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[PATCH] ASoC: qcom: lpass-cpu: mark IRQ_CLEAR register as volatile and readable
by Srinivas Kandagatla 25 Jun '21

25 Jun '21

Currently IRQ_CLEAR register is marked as write-only, however using regmap_update_bits on this register will have some side effects. so mark IRQ_CLEAR register appropriately as readable and volatile. Fixes: da0363f7bfd3 ("ASoC: qcom: Fix for DMA interrupt clear reg overwriting") Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Tested-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Tested-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 0b9cbf2ce505..8998697cd1e1 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -525,6 +525,8 @@ static bool lpass_cpu_regmap_readable(struct device *dev, unsigned int reg) return true; for (i = 0; i < v->irq_ports; ++i) { + if (reg == LPAIF_IRQCLEAR_REG(v, i)) + return true; if (reg == LPAIF_IRQEN_REG(v, i)) return true; if (reg == LPAIF_IRQSTAT_REG(v, i)) @@ -566,9 +568,12 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->irq_ports; ++i) + for (i = 0; i < v->irq_ports; ++i) { + if (reg == LPAIF_IRQCLEAR_REG(v, i)) + return true; if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; + } for (i = 0; i < v->rdma_channels; ++i) if (reg == LPAIF_RDMACURR_REG(v, i)) -- 2.21.0

2 1

[PATCH alsa-lib 0/5] Add generic exception mechanism for non-standard control-names
by Hans de Goede 25 Jun '21

25 Jun '21

Hi All, This series seems to have fallen through the cracks, so here is a resend of it. Regards, Hans Hans de Goede (5): mixer: simple - Add generic exception mechanism for non-standard control-names mixer: simple - Move handling of 3D Control - Depth controls to the exceptions list mixer: simple - Add exceptions for non " Volume" suffixed capture vol-ctls used in ASoC realtek codec drivers mixer: simple - Add exceptions for some capture-vol-ctls which have a " Volume" suffix mixer: simple - Add exceptions for some Playback Switches with a " Channel Switch" suffix src/mixer/simple_none.c | 74 +++++++++++++++++++++++++---------------- 1 file changed, 46 insertions(+), 28 deletions(-) -- 2.31.1

2 12

[PATCH] ALSA: firewire-lib: Fix 'amdtp_domain_start()' when no AMDTP_OUT_STREAM stream is found
by Christophe JAILLET 25 Jun '21

25 Jun '21

The intent here is to return an error code if we don't find what we are looking for in the 'list_for_each_entry()' loop. 's' is not NULL if the list is empty or if we scan the complete list. Introduce a new 'found' variable to handle such cases. Fixes: 60dd49298ec5 ("ALSA: firewire-lib: handle several AMDTP streams in callback handler of IRQ target") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- We could test with" if (list_entry_is_head(s, &d->streams, list))" instead, but I find it much less readable. --- sound/firewire/amdtp-stream.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index aad9778d1c4d..9be2260e4ca2 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1943,6 +1943,7 @@ int amdtp_domain_start(struct amdtp_domain *d, unsigned int tx_init_skip_cycles, unsigned int events_per_period = d->events_per_period; unsigned int queue_size; struct amdtp_stream *s; + bool found = false; int err; if (replay_seq) { @@ -1955,10 +1956,12 @@ int amdtp_domain_start(struct amdtp_domain *d, unsigned int tx_init_skip_cycles, // Select an IT context as IRQ target. list_for_each_entry(s, &d->streams, list) { - if (s->direction == AMDTP_OUT_STREAM) + if (s->direction == AMDTP_OUT_STREAM) { + found = true; break; + } } - if (!s) + if (!found) return -ENXIO; d->irq_target = s; -- 2.30.2

4 3

[PATCH v4 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by fengzheng923＠gmail.com 25 Jun '21

25 Jun '21

From: Ban Tao <fengzheng923(a)gmail.com> The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. --- v2->v3: None. --- v3->v4: 1.add sig_bits. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e924f9e5df97..8d700baaa3ca 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..720d15e4b8ba --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.17.1

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ALSA: scarlett2: Default on?
by Geoffrey D. Bennett 25 Jun '21

25 Jun '21

On Wed, Jun 23, 2021 at 08:39:24AM +0200, Takashi Iwai wrote: [...] > OK, now all patches have been merged. Thanks! I would next like to consider how we can enable this mixer driver by default. I originally added the device_setup=1 gate because there were reports of the driver making the interface hang. These were all traced back to the problem which was resolved with the commit "Fix device hang with ehci-pci". That commit fixed the issue for those who had reported it and since then there have been no more reports that the mixer driver causes any issues. Simply removing the device_setup=1 check would leave users with no way to disable the driver in case that turns out to be necessary for some reason though, so I don't think that's a good idea. What I think would be the best option would be to have the driver as its own loadable module. Does this sound like a good idea to you? Thanks, Geoffrey.

2 3

[PATCH V3 00/17] Add Scarlett Gen 3 support
by Geoffrey D. Bennett 25 Jun '21

25 Jun '21

This patch series adds a fixed version of Scarlett Gen 3 support on top of the previous "Refactor Scarlett Gen 2 support" patches 1-15/31. Two differences from the previous patches 16-31/31: - Add patch from Takashi fixing scarlett2_add_new_ctl() - Don't increase MAX_ID_ELEMS I tested the above changes on the 18i20 Gen 3 and confirmed no crash with 439 controls and MAX_ID_ELEMS 256. Geoffrey D. Bennett (16): ALSA: usb-audio: scarlett2: Add Gen 3 mixer support ALSA: usb-audio: scarlett2: Add support for "input-other" notify ALSA: usb-audio: scarlett2: Add Gen 3 MSD mode switch ALSA: usb-audio: scarlett2: Move get config above set config ALSA: usb-audio: scarlett2: Allow bit-level access to config ALSA: usb-audio: scarlett2: Add support for Solo and 2i2 Gen 3 ALSA: usb-audio: scarlett2: Add "air" switch support ALSA: usb-audio: scarlett2: Add phantom power switch support ALSA: usb-audio: scarlett2: Add direct monitor support ALSA: usb-audio: scarlett2: Label 18i8 Gen 3 line outputs correctly ALSA: usb-audio: scarlett2: Split up sw_hw_enum_ctl_put() ALSA: usb-audio: scarlett2: Add sw_hw_ctls and mux_ctls ALSA: usb-audio: scarlett2: Update mux controls to allow updates ALSA: usb-audio: scarlett2: Add speaker switching support ALSA: usb-audio: scarlett2: Update get_config to do endian conversion ALSA: usb-audio: scarlett2: Add support for the talkback feature Takashi Iwai (1): ALSA: usb-audio: scarlett2: Fix wrong resume call sound/usb/mixer.c | 3 + sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 6 + sound/usb/mixer_scarlett_gen2.c | 1804 ++++++++++++++++++++++++++++--- 4 files changed, 1637 insertions(+), 177 deletions(-) -- 2.31.1

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[PATCH v3 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 25 Jun '21

25 Jun '21

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some build errors. --- v2->v3: 1.Fix some build errors. --- .../sound/allwinner,sun50i-h6-dmic.yaml | 68 +++++++++++++++++++ 1 file changed, 68 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000000..d9107685b78d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + clocks = <&ccu 101>, <&ccu 100>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu 42>; + }; + +... -- 2.29.0

2 1

[PATCH alsa-lib 1/2] control: fix typo
by Nihal Jere 24 Jun '21

24 Jun '21

Signed-off-by: Nihal Jere <nihal(a)nihaljere.xyz> --- src/control/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control.c b/src/control/control.c index 91415b51..3930c7b1 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -390,7 +390,7 @@ int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info) /** * \brief Get a list of element identifiers * - * Before calling this function, memoru must be allocated using + * Before calling this function, memory must be allocated using * snd_ctl_elem_list_malloc(). * * This function obtains data from the sound card driver and puts it -- 2.32.0

1 1

[GIT PULL] ASoC fixes for v5.13-rc7
by Mark Brown 24 Jun '21

24 Jun '21

The following changes since commit c8a4556d98510ca05bad8d02265a4918b03a8c0b: ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin (2021-06-07 15:54:08 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.13-rc7 for you to fetch changes up to 3b7961a326f8a7e03f54a19f02fedae8d488b80f: ASoC: atmel-i2s: Fix usage of capture and playback at the same time (2021-06-21 13:00:48 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.13 A final batch of fixes for v5.13, this is larger than I'd like due to the fixes for a series of suspend issues that Intel turned up in their testing this week. ---------------------------------------------------------------- Bard Liao (3): ASoC: rt5682-sdw: set regcache_cache_only false before reading RT5682_DEVICE_ID ASoC: rt711-sdca-sdw: add readable for SDW_SDCA_CTL() registers ASoC: rt711-sdca: handle mbq_regmap in rt711_sdca_io_init Codrin Ciubotariu (2): ASoC: atmel-i2s: Set symmetric sample bits ASoC: atmel-i2s: Fix usage of capture and playback at the same time Jack Yu (1): ASoC: rt715-sdca: fix clock stop prepare timeout issue Marek Vasut (1): ASoC: sgtl5000: Add audio-graph-card port Mark Brown (2): Merge tag 'v5.13-rc3' into asoc-5.13 ASoC: rt5645: Avoid upgrading static warnings to errors Oder Chiou (1): ASoC: rt5682: Fix a problem with error handling in the io init function of the soundwire Pierre-Louis Bossart (15): ASoC: max98373-sdw: add missing memory allocation check ASoC: max98373-sdw: use first_hw_init flag on resume ASoC: rt1308-sdw: use first_hw_init flag on resume ASoC: rt1316-sdw: use first_hw_init flag on resume ASoC: rt5682-sdw: use first_hw_init flag on resume ASoC: rt700-sdw: use first_hw_init flag on resume ASoC: rt711-sdca-sdw: use first_hw_init flag on resume ASoC: rt711-sdw: use first_hw_init flag on resume ASoC: rt715-sdca-sdw: use first_hw_init flag on resume ASoC: rt715-sdw: use first_hw_init flag on resume soundwire: export sdw_update() and sdw_update_no_pm() ASoC: rt700-sdw: fix race condition on system suspend ASoC: rt711-sdw: fix race condition on system suspend ASoC: rt5682-sdw: fix race condition on system suspend ASoC: rt711-sdca-sdw: fix race condition on system suspend Richard Fitzgerald (1): ASoC: cs42l42: Correct definition of CS42L42_ADC_PDN_MASK Shengjiu Wang (1): ASoC: fsl_spdif: Fix error handler with pm_runtime_enable Timur Tabi (1): MAINTAINERS: remove Timur Tabi from Freescale SOC sound drivers .gitignore | 10 +- .mailmap | 15 +- CREDITS | 18 +- Documentation/ABI/obsolete/sysfs-class-dax | 2 +- .../ABI/obsolete/sysfs-kernel-fadump_registered | 2 +- .../ABI/obsolete/sysfs-kernel-fadump_release_mem | 2 +- Documentation/ABI/removed/sysfs-bus-nfit | 2 +- Documentation/ABI/stable/procfs-audit_loginuid | 27 + .../ABI/testing/debugfs-driver-habanalabs | 70 +- Documentation/ABI/testing/debugfs-moxtet | 4 +- 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arch/arm/boot/dts/bcm47094-luxul-xwc-2000.dts | 4 +- arch/arm/boot/dts/bcm47094-luxul-xwr-3100.dts | 9 +- arch/arm/boot/dts/bcm47094-luxul-xwr-3150-v1.dts | 9 +- arch/arm/boot/dts/bcm47094-netgear-r8500.dts | 4 +- arch/arm/boot/dts/bcm47094-phicomm-k3.dts | 4 +- arch/arm/boot/dts/dra7-l4.dtsi | 79 +- arch/arm/boot/dts/dra7.dtsi | 240 +- arch/arm/boot/dts/ep7209.dtsi | 18 +- arch/arm/boot/dts/ep7211-edb7211.dts | 2 +- arch/arm/boot/dts/exynos3250-monk.dts | 4 +- arch/arm/boot/dts/exynos3250-rinato.dts | 4 +- arch/arm/boot/dts/exynos4210-i9100.dts | 56 +- arch/arm/boot/dts/exynos4412-midas.dtsi | 6 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 5 +- arch/arm/boot/dts/exynos4412-odroidx.dts | 3 +- arch/arm/boot/dts/exynos4412-p4note.dtsi | 4 +- arch/arm/boot/dts/exynos4412-ppmu-common.dtsi | 48 +- arch/arm/boot/dts/exynos5250-smdk5250.dts | 2 +- arch/arm/boot/dts/exynos5250-snow-common.dtsi | 2 +- arch/arm/boot/dts/exynos5410-pinctrl.dtsi | 28 +- 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arch/arm/tools/syscalltbl.sh | 22 - arch/arm/xen/mm.c | 10 +- arch/arm64/Kconfig | 91 +- arch/arm64/Kconfig.platforms | 26 +- arch/arm64/Makefile | 5 +- arch/arm64/boot/dts/Makefile | 1 + .../dts/allwinner/sun50i-a64-amarula-relic.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-bananapi-m64.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-nanopi-a64.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-olinuxino.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-orangepi-win.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-pine64-lts.dts | 4 + .../arm64/boot/dts/allwinner/sun50i-a64-pine64.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-pinebook.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-pinephone.dtsi | 2 +- .../boot/dts/allwinner/sun50i-a64-pinetab.dts | 2 +- .../boot/dts/allwinner/sun50i-a64-sopine.dtsi | 4 +- .../boot/dts/allwinner/sun50i-a64-teres-i.dts | 2 +- arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 6 +- .../boot/dts/allwinner/sun50i-h6-beelink-gs1.dts | 44 +- .../boot/dts/allwinner/sun50i-h6-orangepi-3.dts | 16 +- .../boot/dts/allwinner/sun50i-h6-orangepi.dtsi | 24 +- .../boot/dts/allwinner/sun50i-h6-pine-h64.dts | 4 +- arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 12 +- arch/arm64/boot/dts/altera/Makefile | 2 +- arch/arm64/boot/dts/amlogic/Makefile | 3 + arch/arm64/boot/dts/amlogic/meson-a1.dtsi | 4 +- arch/arm64/boot/dts/amlogic/meson-axg.dtsi | 1 - arch/arm64/boot/dts/amlogic/meson-g12-common.dtsi | 20 +- .../arm64/boot/dts/amlogic/meson-g12b-gsking-x.dts | 2 +- .../boot/dts/amlogic/meson-g12b-gtking-pro.dts | 4 +- arch/arm64/boot/dts/amlogic/meson-g12b-gtking.dts | 2 +- .../boot/dts/amlogic/meson-g12b-odroid-n2.dtsi | 56 +- arch/arm64/boot/dts/amlogic/meson-gx.dtsi | 6 + .../arm64/boot/dts/amlogic/meson-gxbb-odroidc2.dts | 2 +- .../dts/amlogic/meson-gxl-s905d-mecool-kii-pro.dts | 86 + .../boot/dts/amlogic/meson-gxl-s905d-sml5442tw.dts | 8 +- .../boot/dts/amlogic/meson-gxm-mecool-kiii-pro.dts | 113 + 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.../boot/dts/qcom/msm8998-oneplus-common.dtsi | 514 + .../boot/dts/qcom/msm8998-oneplus-dumpling.dts | 25 + arch/arm64/boot/dts/qcom/msm8998.dtsi | 2 + arch/arm64/boot/dts/qcom/pm8150.dtsi | 4 +- arch/arm64/boot/dts/qcom/pm8350.dtsi | 25 + arch/arm64/boot/dts/qcom/pm8350b.dtsi | 25 + arch/arm64/boot/dts/qcom/pm8350c.dtsi | 25 + arch/arm64/boot/dts/qcom/pmk8350.dtsi | 25 + arch/arm64/boot/dts/qcom/pmr735a.dtsi | 25 + arch/arm64/boot/dts/qcom/pmr735b.dtsi | 25 + arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 18 +- .../boot/dts/qcom/sc7180-trogdor-coachz-r1-lte.dts | 18 + .../boot/dts/qcom/sc7180-trogdor-coachz-r1.dts | 141 + .../boot/dts/qcom/sc7180-trogdor-coachz-r2-lte.dts | 18 + .../boot/dts/qcom/sc7180-trogdor-coachz-r2.dts | 15 + .../arm64/boot/dts/qcom/sc7180-trogdor-coachz.dtsi | 266 + .../qcom/sc7180-trogdor-lazor-limozeen-nots-r4.dts | 34 + .../qcom/sc7180-trogdor-lazor-limozeen-nots.dts | 26 + .../dts/qcom/sc7180-trogdor-lazor-limozeen.dts | 42 + 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| 155 +- arch/arm64/boot/dts/qcom/sm8150.dtsi | 555 +- arch/arm64/boot/dts/qcom/sm8250-mtp.dts | 4 - arch/arm64/boot/dts/qcom/sm8250.dtsi | 676 +- arch/arm64/boot/dts/qcom/sm8350-hdk.dts | 319 + arch/arm64/boot/dts/qcom/sm8350-mtp.dts | 116 + arch/arm64/boot/dts/qcom/sm8350.dtsi | 1637 +- .../boot/dts/renesas/beacon-renesom-baseboard.dtsi | 4 +- arch/arm64/boot/dts/renesas/hihope-common.dtsi | 3 + arch/arm64/boot/dts/renesas/hihope-rev4.dtsi | 2 +- ...hope-rzg2-ex-aistarvision-mipi-adapter-2.1.dtsi | 2 - .../boot/dts/renesas/r8a774a1-beacon-rzg2m-kit.dts | 3 + arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 8 + .../boot/dts/renesas/r8a774b1-beacon-rzg2n-kit.dts | 3 + arch/arm64/boot/dts/renesas/r8a774b1.dtsi | 8 + arch/arm64/boot/dts/renesas/r8a774c0-cat874.dts | 6 +- .../boot/dts/renesas/r8a774c0-ek874-mipi-2.1.dts | 2 +- arch/arm64/boot/dts/renesas/r8a774c0.dtsi | 4 + .../boot/dts/renesas/r8a774e1-beacon-rzg2h-kit.dts | 3 + arch/arm64/boot/dts/renesas/r8a774e1.dtsi | 8 + 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drivers/gpu/drm/amd/display/dmub/inc/dmub_cmd.h | 1498 +- .../drm/amd/display/dmub/inc/dmub_trace_buffer.h | 3 +- drivers/gpu/drm/amd/display/dmub/src/Makefile | 2 +- drivers/gpu/drm/amd/display/dmub/src/dmub_dcn20.c | 57 + drivers/gpu/drm/amd/display/dmub/src/dmub_dcn20.h | 35 +- drivers/gpu/drm/amd/display/dmub/src/dmub_dcn30.c | 7 + drivers/gpu/drm/amd/display/dmub/src/dmub_srv.c | 182 +- .../gpu/drm/amd/display/dmub/src/dmub_srv_stat.c | 105 + drivers/gpu/drm/amd/display/include/logger_types.h | 3 + .../drm/amd/display/modules/color/color_gamma.c | 6 +- .../drm/amd/display/modules/color/color_gamma.h | 4 +- .../drm/amd/display/modules/freesync/freesync.c | 69 +- drivers/gpu/drm/amd/display/modules/hdcp/hdcp.c | 10 +- drivers/gpu/drm/amd/display/modules/hdcp/hdcp.h | 4 +- .../drm/amd/display/modules/hdcp/hdcp1_execution.c | 35 +- .../amd/display/modules/hdcp/hdcp1_transition.c | 6 +- .../drm/amd/display/modules/hdcp/hdcp2_execution.c | 8 +- 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+- drivers/gpu/drm/i915/display/intel_combo_phy.c | 31 +- drivers/gpu/drm/i915/display/intel_crt.c | 18 +- drivers/gpu/drm/i915/display/intel_crt.h | 1 - drivers/gpu/drm/i915/display/intel_crtc.c | 290 +- drivers/gpu/drm/i915/display/intel_csr.c | 16 +- drivers/gpu/drm/i915/display/intel_cursor.c | 29 +- drivers/gpu/drm/i915/display/intel_ddi.c | 2789 +- drivers/gpu/drm/i915/display/intel_ddi.h | 23 +- drivers/gpu/drm/i915/display/intel_ddi_buf_trans.c | 1394 + drivers/gpu/drm/i915/display/intel_ddi_buf_trans.h | 100 + drivers/gpu/drm/i915/display/intel_display.c | 4515 +-- drivers/gpu/drm/i915/display/intel_display.h | 56 +- .../gpu/drm/i915/display/intel_display_debugfs.c | 209 +- drivers/gpu/drm/i915/display/intel_display_power.c | 283 +- drivers/gpu/drm/i915/display/intel_display_power.h | 32 + drivers/gpu/drm/i915/display/intel_display_types.h | 179 +- drivers/gpu/drm/i915/display/intel_dp.c | 1942 +- drivers/gpu/drm/i915/display/intel_dp.h | 11 - 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| 8 +- net/ipv4/netfilter/iptable_filter.c | 17 +- net/ipv4/netfilter/iptable_mangle.c | 23 +- net/ipv4/netfilter/iptable_nat.c | 59 +- net/ipv4/netfilter/iptable_raw.c | 17 +- net/ipv4/netfilter/iptable_security.c | 17 +- net/ipv4/netfilter/nf_defrag_ipv4.c | 46 +- net/ipv4/netfilter/nf_log_arp.c | 172 - net/ipv4/netfilter/nf_log_ipv4.c | 395 - net/ipv4/nexthop.c | 1570 +- net/ipv4/ping.c | 4 +- net/ipv4/route.c | 183 +- net/ipv4/sysctl_net_ipv4.c | 265 +- net/ipv4/tcp.c | 13 + net/ipv4/tcp_bpf.c | 139 +- net/ipv4/tcp_cong.c | 4 + net/ipv4/tcp_cubic.c | 24 +- net/ipv4/tcp_input.c | 10 +- net/ipv4/tcp_ipv4.c | 24 +- net/ipv4/tcp_lp.c | 4 +- net/ipv4/tcp_output.c | 20 +- net/ipv4/udp.c | 41 + net/ipv4/udp_bpf.c | 76 +- net/ipv4/udp_offload.c | 27 +- net/ipv6/addrconf.c | 45 +- net/ipv6/addrconf_core.c | 9 +- net/ipv6/af_inet6.c | 4 +- net/ipv6/ah6.c | 4 +- net/ipv6/esp6.c | 4 +- net/ipv6/esp6_offload.c | 19 +- net/ipv6/exthdrs.c | 5 +- net/ipv6/icmp.c | 20 +- net/ipv6/ip6_gre.c | 7 +- 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| 30 +- net/netfilter/Makefile | 6 +- net/netfilter/ipset/ip_set_core.c | 182 +- net/netfilter/ipvs/ip_vs_core.c | 2 +- net/netfilter/ipvs/ip_vs_ftp.c | 2 - net/netfilter/nf_conntrack_acct.c | 2 +- net/netfilter/nf_conntrack_core.c | 53 +- net/netfilter/nf_conntrack_ecache.c | 31 +- net/netfilter/nf_conntrack_expect.c | 22 +- net/netfilter/nf_conntrack_ftp.c | 5 +- net/netfilter/nf_conntrack_h323_main.c | 3 +- net/netfilter/nf_conntrack_helper.c | 15 +- net/netfilter/nf_conntrack_irc.c | 5 +- net/netfilter/nf_conntrack_netlink.c | 384 +- net/netfilter/nf_conntrack_pptp.c | 4 +- net/netfilter/nf_conntrack_proto.c | 8 +- net/netfilter/nf_conntrack_proto_dccp.c | 1 + net/netfilter/nf_conntrack_proto_tcp.c | 40 +- net/netfilter/nf_conntrack_sane.c | 5 +- net/netfilter/nf_conntrack_standalone.c | 87 +- net/netfilter/nf_flow_table_core.c | 146 +- net/netfilter/nf_flow_table_ip.c | 453 +- net/netfilter/nf_flow_table_offload.c | 281 +- net/netfilter/nf_log.c | 10 - 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(100%) rename drivers/{staging => }/comedi/drivers/s526.c (100%) rename drivers/{staging => }/comedi/drivers/s626.c (100%) rename drivers/{staging => }/comedi/drivers/s626.h (100%) rename drivers/{staging => }/comedi/drivers/ssv_dnp.c (100%) create mode 100644 drivers/comedi/drivers/tests/Makefile create mode 100644 drivers/comedi/drivers/tests/comedi_example_test.c create mode 100644 drivers/comedi/drivers/tests/ni_routes_test.c create mode 100644 drivers/comedi/drivers/tests/unittest.h rename drivers/{staging => }/comedi/drivers/usbdux.c (100%) rename drivers/{staging => }/comedi/drivers/usbduxfast.c (100%) rename drivers/{staging => }/comedi/drivers/usbduxsigma.c (100%) rename drivers/{staging => }/comedi/drivers/vmk80xx.c (100%) rename drivers/{staging => }/comedi/drivers/z8536.h (100%) rename drivers/{staging => }/comedi/kcomedilib/Makefile (100%) rename drivers/{staging => }/comedi/kcomedilib/kcomedilib_main.c (100%) rename drivers/{staging => }/comedi/proc.c (100%) rename drivers/{staging => }/comedi/range.c (100%) create mode 100644 drivers/counter/interrupt-cnt.c create mode 100644 drivers/dma/idxd/perfmon.c create mode 100644 drivers/dma/idxd/perfmon.h create mode 100644 drivers/firmware/smccc/kvm_guest.c create mode 100644 drivers/gpio/gpio-bd71815.c create mode 100644 drivers/gpio/gpio-realtek-otto.c create mode 100644 drivers/gpu/drm/amd/amdgpu/aldebaran.c create mode 100644 drivers/gpu/drm/amd/amdgpu/aldebaran.h create mode 100644 drivers/gpu/drm/amd/amdgpu/aldebaran_reg_init.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_amdkfd_aldebaran.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_amdkfd_arcturus.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_res_cursor.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_reset.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_reset.h mode change 100755 => 100644 drivers/gpu/drm/amd/amdgpu/dimgrey_cavefish_reg_init.c create mode 100644 drivers/gpu/drm/amd/amdgpu/gfx_v9_4_2.c create mode 100644 drivers/gpu/drm/amd/amdgpu/gfx_v9_4_2.h create mode 100644 drivers/gpu/drm/amd/amdgpu/mmhub_v1_7.c create mode 100644 drivers/gpu/drm/amd/amdgpu/mmhub_v1_7.h create mode 100644 drivers/gpu/drm/amd/amdgpu/psp_v13_0.c create mode 100644 drivers/gpu/drm/amd/amdgpu/psp_v13_0.h create mode 100644 drivers/gpu/drm/amd/amdgpu/sdma_v4_4.c create mode 100644 drivers/gpu/drm/amd/amdgpu/sdma_v4_4.h create mode 100644 drivers/gpu/drm/amd/amdgpu/smuio_v13_0.c create mode 100644 drivers/gpu/drm/amd/amdgpu/smuio_v13_0.h create mode 100644 drivers/gpu/drm/amd/amdgpu/umc_v6_7.c create mode 100644 drivers/gpu/drm/amd/amdgpu/umc_v6_7.h create mode 100644 drivers/gpu/drm/amd/display/dc/core/dc_link_enc_cfg.c create mode 100644 drivers/gpu/drm/amd/display/dc/core/dc_stat.c create mode 100644 drivers/gpu/drm/amd/display/dc/dc_stat.h create mode 100644 drivers/gpu/drm/amd/display/dc/dce/dmub_outbox.c create mode 100644 drivers/gpu/drm/amd/display/dc/dce/dmub_outbox.h create mode 100644 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drivers/gpu/drm/amd/include/asic_reg/smuio/smuio_13_0_2_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/thm/thm_13_0_2_offset.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/thm/thm_13_0_2_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/umc/umc_6_7_0_offset.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/umc/umc_6_7_0_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/vcn/vcn_2_6_0_offset.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/vcn/vcn_2_6_0_sh_mask.h create mode 100644 drivers/gpu/drm/amd/pm/inc/aldebaran_ppsmc.h create mode 100644 drivers/gpu/drm/amd/pm/inc/smu13_driver_if_aldebaran.h create mode 100644 drivers/gpu/drm/amd/pm/inc/smu_v13_0.h create mode 100644 drivers/gpu/drm/amd/pm/inc/smu_v13_0_pptable.h create mode 100644 drivers/gpu/drm/amd/pm/swsmu/smu13/Makefile create mode 100644 drivers/gpu/drm/amd/pm/swsmu/smu13/aldebaran_ppt.c create mode 100644 drivers/gpu/drm/amd/pm/swsmu/smu13/aldebaran_ppt.h create mode 100644 drivers/gpu/drm/amd/pm/swsmu/smu13/smu_v13_0.c delete mode 100644 drivers/gpu/drm/arc/Kconfig delete mode 100644 drivers/gpu/drm/arc/Makefile delete mode 100644 drivers/gpu/drm/arc/arcpgu.h delete mode 100644 drivers/gpu/drm/arc/arcpgu_crtc.c delete mode 100644 drivers/gpu/drm/arc/arcpgu_drv.c delete mode 100644 drivers/gpu/drm/arc/arcpgu_hdmi.c delete mode 100644 drivers/gpu/drm/arc/arcpgu_regs.h delete mode 100644 drivers/gpu/drm/arc/arcpgu_sim.c delete mode 100644 drivers/gpu/drm/ast/ast_cursor.c create mode 100644 drivers/gpu/drm/bridge/chipone-icn6211.c create mode 100644 drivers/gpu/drm/bridge/lontium-lt8912b.c create mode 100644 drivers/gpu/drm/drm_displayid.c create mode 100644 drivers/gpu/drm/drm_gem_atomic_helper.c delete mode 100644 drivers/gpu/drm/gma500/accel_2d.c delete mode 100644 drivers/gpu/drm/gma500/blitter.c delete mode 100644 drivers/gpu/drm/gma500/blitter.h create mode 100644 drivers/gpu/drm/gud/Kconfig create mode 100644 drivers/gpu/drm/gud/Makefile create mode 100644 drivers/gpu/drm/gud/gud_connector.c create mode 100644 drivers/gpu/drm/gud/gud_drv.c create mode 100644 drivers/gpu/drm/gud/gud_internal.h create mode 100644 drivers/gpu/drm/gud/gud_pipe.c create mode 100644 drivers/gpu/drm/i915/TODO.txt create mode 100644 drivers/gpu/drm/i915/display/g4x_dp.c create mode 100644 drivers/gpu/drm/i915/display/g4x_dp.h create mode 100644 drivers/gpu/drm/i915/display/g4x_hdmi.c create mode 100644 drivers/gpu/drm/i915/display/g4x_hdmi.h create mode 100644 drivers/gpu/drm/i915/display/intel_ddi_buf_trans.c create mode 100644 drivers/gpu/drm/i915/display/intel_ddi_buf_trans.h create mode 100644 drivers/gpu/drm/i915/display/intel_fb.c create mode 100644 drivers/gpu/drm/i915/display/intel_fb.h create mode 100644 drivers/gpu/drm/i915/display/skl_scaler.c create mode 100644 drivers/gpu/drm/i915/display/skl_scaler.h create mode 100644 drivers/gpu/drm/i915/display/skl_universal_plane.c create mode 100644 drivers/gpu/drm/i915/display/skl_universal_plane.h delete mode 100644 drivers/gpu/drm/i915/gem/i915_gem_fence.c create mode 100644 drivers/gpu/drm/i915/intel_step.c create mode 100644 drivers/gpu/drm/i915/intel_step.h delete mode 100644 drivers/gpu/drm/msm/dsi/pll/dsi_pll.c delete mode 100644 drivers/gpu/drm/msm/dsi/pll/dsi_pll.h delete mode 100644 drivers/gpu/drm/msm/dsi/pll/dsi_pll_10nm.c delete mode 100644 drivers/gpu/drm/msm/dsi/pll/dsi_pll_14nm.c delete mode 100644 drivers/gpu/drm/msm/dsi/pll/dsi_pll_28nm.c delete mode 100644 drivers/gpu/drm/msm/dsi/pll/dsi_pll_28nm_8960.c delete mode 100644 drivers/gpu/drm/msm/dsi/pll/dsi_pll_7nm.c create mode 100644 drivers/gpu/drm/tiny/arcpgu.c create mode 100644 drivers/gpu/drm/ttm/ttm_device.c delete mode 100644 drivers/gpu/drm/ttm/ttm_memory.c create mode 100644 drivers/gpu/drm/vmwgfx/ttm_memory.c create mode 100644 drivers/gpu/drm/vmwgfx/ttm_memory.h create mode 100644 drivers/hid/hid-ft260.c create mode 100644 drivers/hid/hid-thrustmaster.c create mode 100644 drivers/hid/surface-hid/Kconfig create mode 100644 drivers/hid/surface-hid/Makefile create mode 100644 drivers/hid/surface-hid/surface_hid.c create mode 100644 drivers/hid/surface-hid/surface_hid_core.c create mode 100644 drivers/hid/surface-hid/surface_hid_core.h create mode 100644 drivers/hid/surface-hid/surface_kbd.c delete mode 100644 drivers/hwmon/amd_energy.c create mode 100644 drivers/hwmon/nzxt-kraken2.c create mode 100644 drivers/hwmon/pmbus/bpa-rs600.c create mode 100644 drivers/hwmon/pmbus/fsp-3y.c create mode 100644 drivers/hwmon/pmbus/ir36021.c create mode 100644 drivers/hwmon/pmbus/max15301.c create mode 100644 drivers/hwmon/pmbus/stpddc60.c delete mode 100644 drivers/hwspinlock/sirf_hwspinlock.c create mode 100644 drivers/hwtracing/coresight/coresight-trbe.c create mode 100644 drivers/hwtracing/coresight/coresight-trbe.h create mode 100644 drivers/i2c/busses/i2c-cp2615.c create mode 100644 drivers/i2c/busses/i2c-hisi.c create mode 100644 drivers/iio/accel/bmi088-accel-core.c create mode 100644 drivers/iio/accel/bmi088-accel-spi.c create mode 100644 drivers/iio/accel/bmi088-accel.h create mode 100644 drivers/iio/adc/ti-ads131e08.c create mode 100644 drivers/iio/cdc/Kconfig create mode 100644 drivers/iio/cdc/Makefile create mode 100644 drivers/iio/cdc/ad7150.c create mode 100644 drivers/iio/common/scmi_sensors/Kconfig create mode 100644 drivers/iio/common/scmi_sensors/Makefile create mode 100644 drivers/iio/common/scmi_sensors/scmi_iio.c create mode 100644 drivers/iio/proximity/cros_ec_mkbp_proximity.c create mode 100644 drivers/iio/test/Kconfig create mode 100644 drivers/iio/test/Makefile create mode 100644 drivers/iio/test/iio-test-format.c create mode 100644 drivers/infiniband/hw/mlx5/dm.c create mode 100644 drivers/infiniband/hw/mlx5/dm.h create mode 100644 drivers/input/misc/atc260x-onkey.c create mode 100644 drivers/input/misc/iqs626a.c create mode 100644 drivers/input/touchscreen.c create mode 100644 drivers/input/touchscreen/hycon-hy46xx.c create mode 100644 drivers/input/touchscreen/ilitek_ts_i2c.c create mode 100644 drivers/input/touchscreen/msg2638.c delete mode 100644 drivers/input/touchscreen/of_touchscreen.c create mode 100644 drivers/interconnect/qcom/sdm660.c create mode 100644 drivers/interconnect/qcom/sm8350.c create mode 100644 drivers/interconnect/qcom/sm8350.h create mode 100644 drivers/iommu/io-pgfault.c create mode 100644 drivers/iommu/sprd-iommu.c create mode 100644 drivers/irqchip/irq-apple-aic.c create mode 100644 drivers/irqchip/irq-idt3243x.c create mode 100644 drivers/irqchip/irq-wpcm450-aic.c create mode 100644 drivers/leds/flash/leds-rt4505.c delete mode 100644 drivers/media/pci/mantis/mantis_core.c create mode 100644 drivers/media/platform/imx-jpeg/Kconfig create mode 100644 drivers/media/platform/imx-jpeg/Makefile create mode 100644 drivers/media/platform/imx-jpeg/mxc-jpeg-hw.c create mode 100644 drivers/media/platform/imx-jpeg/mxc-jpeg-hw.h create mode 100644 drivers/media/platform/imx-jpeg/mxc-jpeg.c create mode 100644 drivers/media/platform/imx-jpeg/mxc-jpeg.h create mode 100644 drivers/media/platform/qcom/camss/camss-csid-170.c create mode 100644 drivers/media/platform/qcom/camss/camss-csid-4-1.c create mode 100644 drivers/media/platform/qcom/camss/camss-csid-4-7.c create mode 100644 drivers/media/platform/qcom/camss/camss-csid-gen1.h create mode 100644 drivers/media/platform/qcom/camss/camss-csid-gen2.h create mode 100644 drivers/media/platform/qcom/camss/camss-vfe-170.c create mode 100644 drivers/media/platform/qcom/camss/camss-vfe-4-8.c create mode 100644 drivers/media/platform/qcom/camss/camss-vfe-gen1.c create mode 100644 drivers/media/platform/qcom/camss/camss-vfe-gen1.h create mode 100644 drivers/media/rc/keymaps/rc-mecool-kii-pro.c create mode 100644 drivers/media/rc/keymaps/rc-mecool-kiii-pro.c create mode 100644 drivers/media/rc/keymaps/rc-minix-neo.c create mode 100644 drivers/media/rc/keymaps/rc-xbox-360.c delete mode 100644 drivers/media/rc/zx-irdec.c delete mode 100644 drivers/mfd/ab3100-core.c delete mode 100644 drivers/mfd/ab3100-otp.c create mode 100644 drivers/mfd/atc260x-core.c create mode 100644 drivers/mfd/atc260x-i2c.c create mode 100644 drivers/mfd/ntxec.c create mode 100644 drivers/mfd/rohm-bd9576.c create mode 100644 drivers/misc/dw-xdata-pcie.c delete mode 100644 drivers/misc/pvpanic.c create mode 100644 drivers/misc/pvpanic/Kconfig create mode 100644 drivers/misc/pvpanic/Makefile create mode 100644 drivers/misc/pvpanic/pvpanic-mmio.c create mode 100644 drivers/misc/pvpanic/pvpanic-pci.c create mode 100644 drivers/misc/pvpanic/pvpanic.c create mode 100644 drivers/misc/pvpanic/pvpanic.h rename drivers/{staging/most/sound/sound.c => most/most_snd.c} (100%) delete mode 100644 drivers/mtd/parsers/ofpart.c create mode 100644 drivers/mtd/parsers/ofpart_bcm4908.c create mode 100644 drivers/mtd/parsers/ofpart_bcm4908.h create mode 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C-Media 6632A (Xonar U7), 'Input Gain Pad Control'
by Joachim Schwender 24 Jun '21

24 Jun '21

I have a Xonar U7, with a C-Media 6632A. It works somehow, the only issue i have is that the control 'Input Gain Pad Control' is kind of odd: By the control name i would expect it changes the gain on the input path, but in reality it changes the headphones output level. The Line output level is constant, as well as the input level. Alsa settings are: Line capture on Mic capture off Item0: 'Line' Is this Xonar U7 specific or is this a general implementation bug?? best regards -- Dipl.-Ing (FH) Joachim Schwender Hölderlinstrasse 8/1 72636 Frickenhausen ☏+49 7022 8103 Web:schwender-beyer.de <http://www.schwender-beyer.de/> Datenschutzerklärung <http://gedanken-polizei.de/ueberwachung.php>

1 0

[PATCH] ASoC: da7219: Fix an out-of-bound read in an error handling path
by Christophe JAILLET 24 Jun '21

24 Jun '21

If 'of_clk_add_hw_provider()' fails, the previous 'for' loop will have run completely and 'i' is know to be 'DA7219_DAI_NUM_CLKS'. In such a case, there will be an out-of-bounds access when using 'da7219->dai_clks_lookup[i]' and '&da7219->dai_clks_hw[i]'. To avoid that, add a new label, 'err_free_all', which set the expected value of 'i' in such a case. Fixes: 78013a1cf297 ("ASoC: da7219: Fix clock handling around codec level probe") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/da7219.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 13009d08b09a..1e8b491d1fd3 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2204,12 +2204,14 @@ static int da7219_register_dai_clks(struct snd_soc_component *component) da7219->clk_hw_data); if (ret) { dev_err(dev, "Failed to register clock provider\n"); - goto err; + goto err_free_all; } } return 0; +err_free_all: + i = DA7219_DAI_NUM_CLKS - 1; err: do { if (da7219->dai_clks_lookup[i]) -- 2.30.2

2 2

1.2.5 now prefixing devices with "_ucm0001."
by GitHub issues - opened 23 Jun '21

23 Jun '21

alsa-project/alsa-ucm-conf issue #104 was opened from craftyguy: I'm using a ucm2 config that is not yet upstream (see [1]), and 1e6297b650114cb2e043be4c677118f971e31eb7 has caused devices to get renamed; they are now prefixed with `_ucm0001.`: alsaucm json output from 1.2.4: ``` ... "Verbs": { "HiFi": { "Comment": "Default", "Devices": { "Handset": { "Comment": "Handset", "ConflictingDevices": [ "Speaker", "Headphones" ], "Values": { "CaptureCTL": "hw:L5", "PlaybackCTL": "hw:L5", "PlaybackChannels": "2", "PlaybackPCM": "hw:L5,0", "PlaybackPriority": "100", "PlaybackSwitch": "name='Speaker Switch'", "PlaybackVolume": "name='Speaker Volume'" } }, ... ``` alsaucm json output from 1.2.5: ``` ... "Verbs": { "HiFi": { "Comment": "Default", "Devices": { "Handset": { "Comment": "Handset", "ConflictingDevices": [ "Speaker", "Headphones" ], "Values": { "CaptureCTL": "_ucm0001.hw:L5", "PlaybackCTL": "_ucm0001.hw:L5", "PlaybackChannels": "2", "PlaybackPCM": "_ucm0001.hw:L5,0", "PlaybackPriority": "100", "PlaybackSwitch": "name='Speaker Switch'", "PlaybackVolume": "name='Speaker Volume'" } }, ... ``` This breaks things [like pulseaudio config](https://source.puri.sm/Librem5/librem5-base/-/blob/pureos/byzantium/default/audio/pulse/librem5.pa) that expects specific device names. Was this renaming intentional? cc @agx 1. https://source.puri.sm/Librem5/librem5-base/-/tree/pureos/byzantium/default… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/104 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/1] ASoC: fsl: remove unnecessary oom message
by Zhen Lei 23 Jun '21

23 Jun '21

Fixes scripts/checkpatch.pl warning: WARNING: Possible unnecessary 'out of memory' message Remove it can help us save a bit of memory. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/soc/fsl/imx-audmix.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index cbdc0a2c09c5..a364e2415de0 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -209,10 +209,8 @@ static int imx_audmix_probe(struct platform_device *pdev) /* for CPU/Codec/Platform x 2 */ dlc = devm_kcalloc(&pdev->dev, 6, sizeof(*dlc), GFP_KERNEL); - if (!dlc) { - dev_err(&pdev->dev, "failed to allocate dai_link\n"); + if (!dlc) return -ENOMEM; - } ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i, &args); -- 2.25.1

2 1

[PATCH -next 0/4] ASoC: tegra: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 23 Jun '21

23 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Yang Yingliang (4): ASoC: tegra20: i2s: Use devm_platform_get_and_ioremap_resource() ASoC: tegra20: spdif: Use devm_platform_get_and_ioremap_resource() ASoC: tegra: tegra210_admaif: Use devm_platform_get_and_ioremap_resource() ASoC: tegra30: ahub: Use devm_platform_get_and_ioremap_resource() sound/soc/tegra/tegra20_i2s.c | 3 +-- sound/soc/tegra/tegra20_spdif.c | 3 +-- sound/soc/tegra/tegra210_admaif.c | 4 +--- sound/soc/tegra/tegra30_ahub.c | 3 +-- 4 files changed, 4 insertions(+), 9 deletions(-) -- 2.25.1

2 5

[PATCH v2 0/3] ASoC: tlv320aic32x4: Add support for TAS2505
by Claudius Heine 23 Jun '21

23 Jun '21

Hi, this is v2 from my patchset that add support for the TAS2505 to the tlv320aic32x4 driver. kind regards, Claudius Changes from v1: - clarified commit message of first patch, which add the type value to the struct - removed unnecessary code to put and get speaker volume - removed 'Gain' from 'HP Driver Playback Volume' control - fixed rebase issues Claudius Heine (3): ASoC: tlv320aic32x4: add type to device private data struct ASoC: tlv320aic32x4: add support for TAS2505 ASoC: tlv320aic32x4: dt-bindings: add TAS2505 to compatible .../bindings/sound/tlv320aic32x4.txt | 1 + sound/soc/codecs/tlv320aic32x4-i2c.c | 22 ++- sound/soc/codecs/tlv320aic32x4-spi.c | 23 ++- sound/soc/codecs/tlv320aic32x4.c | 139 +++++++++++++++++- sound/soc/codecs/tlv320aic32x4.h | 10 ++ 5 files changed, 186 insertions(+), 9 deletions(-) base-commit: 70585216fe7730d9fb5453d3e2804e149d0fe201 -- 2.32.0

2 4

[PATCH] ASoC: fsl_spdif: Fix unexpected interrupt after suspend
by Shengjiu Wang 23 Jun '21

23 Jun '21

When system enter suspend, the machine driver suspend callback function will be called, then the cpu driver trigger callback (SNDRV_PCM_TRIGGER_SUSPEND) be called, it would disable the interrupt. But the machine driver suspend and cpu dai driver suspend order maybe changed, the cpu dai driver's suspend callback is called before machine driver's suppend callback, then the interrupt is not cleared successfully in trigger callback. So need to clear interrupts in cpu dai driver's suspend callback to avoid such issue. Fixes: 9cb2b3796e08 ("ASoC: fsl_spdif: Add pm runtime function") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index a9c6b930b04a..8ffb1a6048d6 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1467,6 +1467,9 @@ static int fsl_spdif_runtime_suspend(struct device *dev) struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); int i; + /* Disable all the interrupts */ + regmap_update_bits(spdif_priv->regmap, REG_SPDIF_SIE, 0xffffff, 0); + regmap_read(spdif_priv->regmap, REG_SPDIF_SRPC, &spdif_priv->regcache_srpc); regcache_cache_only(spdif_priv->regmap, true); -- 2.27.0

3 2

UCM2 documentation
by GitHub issues - opened 23 Jun '21

23 Jun '21

alsa-project/alsa-ucm-conf issue #103 was opened from perexg: Add/improve the documentation for - directory tree layout - build-in string substitutions - runtime conditional configuration selection - mixer settings (boot -> use -> verb -> device) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/103 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: snd-soc-dummy: add Device Tree support
by Judy Hsiao 23 Jun '21

23 Jun '21

Support for loading the snd-soc-dummy via DeviceTree. This is useful to create dummy codec devices where we need to have some DAI links without a real Codec. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- .../devicetree/bindings/sound/snd-soc-dummy.txt | 16 ++++++++++++++++ sound/soc/soc-utils.c | 11 +++++++++++ 2 files changed, 27 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/snd-soc-dummy.txt diff --git a/Documentation/devicetree/bindings/sound/snd-soc-dummy.txt b/Documentation/devicetree/bindings/sound/snd-soc-dummy.txt new file mode 100644 index 000000000000..7fa8c5751e62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snd-soc-dummy.txt @@ -0,0 +1,16 @@ +* snd-soc-dummy + +This node models the snd-soc-dummy. +This is useful to create dummy codec devices where we need to have +some DAI links without a real Codec. + +Required properties: +- compatible : "asoc,snd-soc-dummy" + + +Example: + +dummy_codec { + compatible = "asoc,snd-soc-dummy"; + #sound-dai-cells = <0>; +}; diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 299b5d6ebfd1..def2cc687415 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -7,6 +7,8 @@ // Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> // Liam Girdwood <lrg(a)slimlogic.co.uk> +#include <linux/module.h> +#include <linux/of.h> #include <linux/platform_device.h> #include <linux/export.h> #include <sound/core.h> @@ -181,9 +183,18 @@ static int snd_soc_dummy_probe(struct platform_device *pdev) return ret; } +#ifdef CONFIG_OF +static const struct of_device_id soc_dummy_device_id[] = { + { .compatible = "asoc,snd-soc-dummy" }, + {} +}; +MODULE_DEVICE_TABLE(of, soc_dummy_device_id); +#endif + static struct platform_driver soc_dummy_driver = { .driver = { .name = "snd-soc-dummy", + .of_match_table = of_match_ptr(soc_dummy_device_id), }, .probe = snd_soc_dummy_probe, }; -- 2.31.0

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[GIT PULL] HDMI codec improvements, v2
by Maxime Ripard 23 Jun '21

23 Jun '21

Hi, Here's a PR for the changes to hdmi-codec that need to be shared between drm-misc-next and ASoC. This is the second iteration, fixing a bisection issue with compilation Thanks! Maxime The following changes since commit 6efb943b8616ec53a5e444193dccf1af9ad627b5: Linux 5.13-rc1 (2021-05-09 14:17:44 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git tags/asoc-hdmi-codec-improvements-v2 for you to fetch changes up to 2fef64eec23a0840c97977b16dd8919afaffa876: ASoC: hdmi-codec: Add a prepare hook (2021-06-10 11:48:56 +0200) ---------------------------------------------------------------- Improvements to the hdmi-codec driver and ALSA infrastructure around it to support the HDMI Channel Mapping and IEC958 controls ---------------------------------------------------------------- Maxime Ripard (5): ALSA: doc: Clarify IEC958 controls iface ALSA: iec958: Split status creation and fill ASoC: hdmi-codec: Rework to support more controls ASoC: hdmi-codec: Add iec958 controls ASoC: hdmi-codec: Add a prepare hook .../sound/kernel-api/writing-an-alsa-driver.rst | 13 +- include/sound/hdmi-codec.h | 12 +- include/sound/pcm_iec958.h | 8 + sound/core/pcm_iec958.c | 176 +++++++++++++---- sound/soc/codecs/hdmi-codec.c | 219 +++++++++++++++++---- 5 files changed, 337 insertions(+), 91 deletions(-)

2 3

[PATCH v2 00/12] drm/vc4: hdmi: Enable Channel Mapping, IEC958, HBR Passthrough using hdmi-codec
by Maxime Ripard 23 Jun '21

23 Jun '21

Hi, hdmi-codec allows to have a lot of HDMI-audio related infrastructure in place, it's missing a few controls to be able to provide HBR passthrough. This series adds more infrastructure for the drivers, and leverages it in the vc4 HDMI controller driver. Thanks! Maxime Changes from v1: - Added an extra patch to clarify the iec958 controls iface policy - Added kerneldoc for the new iec958 PCM functions - s/EXPORT_SYMBOL/EXPORT_SYMBOL_GPL - Used the ALSA prefix where relevant - Rebased on drm-misc-next-2021-05-17 Dom Cobley (5): drm/vc4: hdmi: Set HD_CTL_WHOLSMP and HD_CTL_CHALIGN_SET drm/vc4: hdmi: Set HDMI_MAI_FMT drm/vc4: hdmi: Set VC4_HDMI_MAI_CONFIG_FORMAT_REVERSE drm/vc4: hdmi: Remove firmware logic for MAI threshold setting ARM: dts: bcm2711: Tune DMA parameters for HDMI audio Maxime Ripard (7): ALSA: doc: Clarify IEC958 controls iface ALSA: iec958: Split status creation and fill ASoC: hdmi-codec: Rework to support more controls ASoC: hdmi-codec: Add iec958 controls ASoC: hdmi-codec: Add a prepare hook drm/vc4: hdmi: Register HDMI codec drm/vc4: hdmi: Remove redundant variables .../kernel-api/writing-an-alsa-driver.rst | 13 +- arch/arm/boot/dts/bcm2711.dtsi | 4 +- drivers/gpu/drm/vc4/Kconfig | 1 + drivers/gpu/drm/vc4/vc4_hdmi.c | 322 ++++++++---------- drivers/gpu/drm/vc4/vc4_hdmi.h | 5 +- drivers/gpu/drm/vc4/vc4_regs.h | 30 ++ include/sound/hdmi-codec.h | 12 +- include/sound/pcm_iec958.h | 8 + sound/core/pcm_iec958.c | 176 +++++++--- sound/soc/codecs/hdmi-codec.c | 219 +++++++++--- 10 files changed, 508 insertions(+), 282 deletions(-) -- 2.31.1

6 32

[PATCH] ALSA: hda/realtek: Apply LED fixup for HP Dragonfly G1, too
by Takashi Iwai 23 Jun '21

23 Jun '21

HP Dragonfly G1 (SSID 103c:861f) also requires the same quirk for the mute and mic-mute LED just as Dragonfly G2 model. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213329 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 49f4cac8b05e..308a58a83f33 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8345,6 +8345,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), + SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.26.2

1 0

[PATCH 0/9] ALSA: firewire-motu: misc fixes and code refactoring
by Takashi Sakamoto 23 Jun '21

23 Jun '21

Hi, This patchset includes misc fixes and code refactoring, mainly for detection of sampling clock source and 828/896 support recently added. Takashi Sakamoto (9): ALSA: firewire-motu: fix detection for S/PDIF source on optical interface in v2 protocol ALSA: firewire-motu: code refactoring for detection of clock source in v2 protocol ALSA: firewire-motu: add support for AES/EBU clock source in v2 protocol ALSA: firewire-motu: use macro instead of magic number for clock source in v2 protocol ALSA: firewire-motu: code refactoring for packet format detection in v2 protocol ALSA: firewire-motu: code refactoring for source detection of sampling clock in v3 protocol ALSA: firewire-motu: use macro instead of magic number for clock source in v3 protocol ALSA: firewire-motu: fix register handling for 828 ALSA: firewire-motu: fix register handling for 896 sound/firewire/motu/motu-protocol-v1.c | 86 ++++++++---- sound/firewire/motu/motu-protocol-v2.c | 176 +++++++++---------------- sound/firewire/motu/motu-protocol-v3.c | 79 ++++------- 3 files changed, 143 insertions(+), 198 deletions(-) -- 2.30.2

2 10

[PATCH 0/8] ASoC: mediatek: Add support for MT8195 SoC
by Trevor Wu 23 Jun '21

23 Jun '21

This series of patches adds support for Mediatek AFE of MT8195 SoC. Patches are based on broonie tree "for-next" branch. Trevor Wu (8): ASoC: mediatek: mt8195: update mediatek common driver ASoC: mediatek: mt8195: support etdm in platform driver ASoC: mediatek: mt8195: support adda in platform driver ASoC: mediatek: mt8195: support pcm in platform driver ASoC: mediatek: mt8195: add platform driver dt-bindings: mediatek: mt8195: add audio afe document ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1019-rt5682 document .../bindings/sound/mt8195-afe-pcm.yaml | 72 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 39 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 20 +- sound/soc/mediatek/common/mtk-base-afe.h | 10 +- sound/soc/mediatek/mt8195/Makefile | 14 + sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 899 +++++ sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 201 + sound/soc/mediatek/mt8195/mt8195-afe-common.h | 200 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3264 +++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 929 +++++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2666 ++++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 393 ++ .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 974 +++++ sound/soc/mediatek/mt8195/mt8195-reg.h | 2793 ++++++++++++++ 16 files changed, 12494 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1019-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/Makefile create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-common.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-adda.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-reg.h -- 2.18.0

3 13

[PATCH] MAINTAINERS: Add Focusrite Scarlett Gen 2/3 Mixer Driver entry
by Geoffrey D. Bennett 23 Jun '21

23 Jun '21

Add Focusrite Scarlett Gen 2/3 Mixer Driver entry. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index bd7aff0c120f..71e327d970d7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7103,6 +7103,13 @@ L: linux-input(a)vger.kernel.org S: Maintained F: drivers/input/joystick/fsia6b.c +FOCUSRITE SCARLETT GEN 2/3 MIXER DRIVER +M: Geoffrey D. Bennett <g(a)b4.vu> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git +F: sound/usb/mixer_scarlett_gen2.c + FORCEDETH GIGABIT ETHERNET DRIVER M: Rain River <rain.1986.08.12(a)gmail.com> M: Zhu Yanjun <zyjzyj2000(a)gmail.com> -- 2.31.1

2 1

Lenovo Legion/Yoga Slim 7 speaker output
by Cameron Berkenpas 23 Jun '21

23 Jun '21

Hello, I'm working on a patch to get audio output working on the speakers of the Lenovo Legion/Yoga Slim 7 laptops: https://bugzilla.kernel.org/attachment.cgi?id=297559&action=diff Please let me know if this isn't the right place for these types of questions. Both of these laptops have ALC287's that both seem to need some codec initialization to get the speakers working. For both laptops, the specs report ALC3306, but Linux on both systems reports that it's an ALC287, and looking at the PCI ID's on Windows agrees that it's an ALC287. This is my first time working on such a patch, and I'm having a headphone related problem: 1. When plugging headphones into the audio jack, I get sound on the headphones as expected. I see that the speakers are muted and the headphones are unmuted as expected in alsamixer. 2. When I unplug the headphones, I can no longer get speaker output. Monitoring alsamixer, I observe that unplugging the headphones results in the speakers being unmuted and the headphones are muted as expected. However, I am unable to get speaker output in this case. 3. If I plug in the headphones, don't play any sound, and then unplug the headphones, I see the same behavior in alsamixer, BUT speaker output still works. 4. If I apply the initialization verbs via hda-verb (the same verbs that I'm using in my patch), speaker output works again. 5. If speaker output is no longer working (due to #2), putting the laptop to sleep and resuming fixes the speakers. The code in the kernel seems to call the sound card's init again so this makes sense. In short, it seems I need to re-initialize the speakers when the headphones are unplugged. Studying sound/pci/hda/patch_realtek.c, there's alc_headset_mode_unplugged(), but as there's currently no case for 0x10ec0287 at all. I'm not sure adding in a case that re-initializes makes sense if the vast majority of ALC287's don't need to when headphones are unplugged. It seems none of the other cards, at least as far as I can tell, do a re-init when headphones are unplugged either. They just seem to setup some pins and coefficients for the most part if anything at all. How can I re-init the speakers when headphones are unplugged specifically for these specific models? The initial-initialization is only done for these laptops, I'm just unclear what, if any, mechanism there is for initialization on headphone unplugs? Thanks! -Cameron

1 2

[PATCH] ALSA: hdsp: fix a test for copy_to_user() failure
by Dan Carpenter 22 Jun '21

22 Jun '21

The copy_to_user() function returns the number of bytes remaining to be copied. It doesn't return negatives. Fixes: 66c8f75919dd ("ALSA: hdsp: Fix assignment in if condition") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/pci/rme9652/hdsp.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index e3e4fabf4abf..8457a4bbc3df 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4877,9 +4877,8 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne memset(&hdsp_version, 0, sizeof(hdsp_version)); hdsp_version.io_type = hdsp->io_type; hdsp_version.firmware_rev = hdsp->firmware_rev; - err = copy_to_user(argp, &hdsp_version, sizeof(hdsp_version)); - if (err < 0) - return -EFAULT; + if (copy_to_user(argp, &hdsp_version, sizeof(hdsp_version))) + return -EFAULT; break; } case SNDRV_HDSP_IOCTL_UPLOAD_FIRMWARE: { -- 2.30.2

2 1

[PATCH -next] ASoC: SOF: imx: Add missing of_node_put() in imx8_probe()
by Yang Yingliang 22 Jun '21

22 Jun '21

This node pointer is returned by of_parse_phandle() with refcount incremented in this function. of_node_put() on it before exiting this function. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/imx/imx8.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 4e7dccadd7d0..12fedf0984bd 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -315,6 +315,7 @@ static int imx8_probe(struct snd_sof_dev *sdev) } ret = of_address_to_resource(res_node, 0, &res); + of_node_put(res_node); if (ret) { dev_err(&pdev->dev, "failed to get reserved region address\n"); goto exit_pdev_unregister; -- 2.25.1

2 1

[PATCH] ASoC: fsl-asoc-card: change dev_err to dev_dbg for defer probe
by Shengjiu Wang 22 Jun '21

22 Jun '21

Don't need to print error message for defer probe Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4f55b316cf0f..b40ba910ae9d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -709,7 +709,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) of_node_put(framemaster); if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { - dev_err(&pdev->dev, "failed to find codec device\n"); + dev_dbg(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; goto asrc_fail; } -- 2.27.0

4 4

[PATCH v2 00/10] ASoC: Intel: machine driver corrections
by Pierre-Louis Bossart 22 Jun '21

22 Jun '21

Some of these patches dealing with Sparse warnings were submitted earlier but not merged. I grouped them with a couple of fixes from Kai. v2: added Mark Brown in CC, was missed in v1 added Richard Fitzgerald Tested-by Kai Vehmanen (3): ASoC: Intel: sof_sdw: use mach data for ADL RVP DMIC count ASoC: Intel: sof_sdw: remove hdac-hdmi support ASoC: Intel: skl_hda_dsp_generic: Update Kconfig documentation Pierre-Louis Bossart (7): ASoC: Intel: sof_rt5682: shrink platform_id names below 20 characters ASoC: Intel: glk_rt5682_max98357a: shrink platform_id below 20 characters ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters ASoC: Intel: sof_sdw: fix signed/unsigned warning ASoC: Intel: soc-acpi: add ull suffix for SoundWire _ADR values ASoC: Intel: use MODULE_DEVICE_TABLE with platform_device_id tables sound/soc/intel/boards/Kconfig | 7 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 3 +- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 5 +-- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 3 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 3 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 3 +- sound/soc/intel/boards/skl_rt286.c | 3 +- sound/soc/intel/boards/sof_cs42l42.c | 3 +- sound/soc/intel/boards/sof_da7219_max98373.c | 6 +-- sound/soc/intel/boards/sof_rt5682.c | 20 +++------- sound/soc/intel/boards/sof_sdw.c | 5 +-- sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/intel/boards/sof_sdw_hdmi.c | 37 +----------------- .../intel/common/soc-acpi-intel-adl-match.c | 28 +++++++------- .../intel/common/soc-acpi-intel-cml-match.c | 20 +++++----- .../intel/common/soc-acpi-intel-cnl-match.c | 2 +- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- .../intel/common/soc-acpi-intel-icl-match.c | 12 +++--- .../intel/common/soc-acpi-intel-jsl-match.c | 6 +-- .../intel/common/soc-acpi-intel-kbl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 38 +++++++++---------- 27 files changed, 83 insertions(+), 144 deletions(-) -- 2.25.1

2 11

[PATCH 1/2] ASoC: codecs: wcd938x: fix unused variable warning
by Srinivas Kandagatla 22 Jun '21

22 Jun '21

This patch fixes below warning: unused variable wcd938x_dt_match by placing device match table under CONFIG_OF Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd938x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 4be61122b0b5..427f51fade37 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3714,12 +3714,14 @@ static int wcd938x_remove(struct platform_device *pdev) return 0; } +#if defined(CONFIG_OF) static const struct of_device_id wcd938x_dt_match[] = { { .compatible = "qcom,wcd9380-codec" }, { .compatible = "qcom,wcd9385-codec" }, {} }; MODULE_DEVICE_TABLE(of, wcd938x_dt_match); +#endif static struct platform_driver wcd938x_codec_driver = { .probe = wcd938x_probe, -- 2.21.0

2 2

[PATCH RFC 00/15] ASoC: add Audio Graph Card2 driver
by Kuninori Morimoto 22 Jun '21

22 Jun '21

Hi Mark, Rob We already have Audio-Graph-Card which is Of-Graph base general sound card driver. Basically it supports basic CPU-Codec connection, and is also supporting DPCM connection. Because it was forcibly expanded to DPCM, DT parsing is very limited and very difficult to add new features on it, for example Multi CPU/Codec support, Codec2Codec support, etc. And some users want to use Audio-Graph-Card as basic connection and expand its own settings on it. This patch adds more flexible new Audio-Graph-Card2 driver for it. Audio-Graph-Card and Audio-Graph-Card2 are similar, but don't have full compatibility. Audio-Graph-Card2 supports Normal Connection, DPCM Connection, Multi CPU/Codec Connection, Codec2Codec Connection, and possible to Customizing. This patch-set adds Audio-Graph-Card2 driver and customized driver sample, and DT settings sample which can be used for testing. To enable testing/debuging, this patch-set also adds Test-Component driver. We already have Dummy Component and/or Dummy DAI on soc-utils, but 1) we can't use it from DT, 2) it do nothing. Added new Test-Component can be used from DT, and it indicates called function name. We can use it to trace callback order / understanding ALSA SoC behavior, etc. Sample DT settings which can be used for testing is using Test-Component. This patch set includes sample DT for custome driver and/or audio-graph-card2. You can easily try to use/test it if you added below line to your DT file. It needs below CONFIGs. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/audio-graph-card2-sample.dtsi" CONFIG_SND_AUDIO_GRAPH_CARD2 CONFIG_SND_SAMPLE_CUSTOM_CARD CONFIG_SND_TEST_COMPONENT There are some notes. Because Audio Graph Card2 is still under experimental stage, it will indicate such warning when probing, and the DT might be updated/exchanged. It can use Codec2Codec, but it will start automatically when probed, and can't stop it so far. It should be updated. 1 : add Of-Graph *port base* loop which is used at Audio-Graph-Card2 2 - 3 : add Test-Component driver 4 - 10 : add Audio-Graph-Card2 driver 11 : add Audio-Graph-Card2 base custome driver sample 12 - 15 : Audio-Graph-Card2 driver / custome driver DT sample Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (15): 1 of: property: add port base loop 2 ASoC: dt-bindings: test-component: add Test Component YAML bindings 3 ASoC: test-component: add Test Component for Sound debug/test 4 ASoC: simple-card-utils: add asoc_graph_is_ports0() 5 ASoC: simple-card-utils: add codec2codec support 6 ASoC: audio-graph-card2: add Audio Graph Card2 driver 7 ASoC: audio-graph-card2: add DPCM support 8 ASoC: audio-graph-card2: add Multi CPU/Codec support 9 ASoC: audio-graph-card2: add Codec2Codec support 10 ASoC: audio-graph-card2: add Yaml Document 11 ASoC: sample-custom-card: add Audio Graph Card2 custome sample 12 ASoC: audio-graph-card2-sample.dtsi: add Sample DT for Audio Graph Card2 13 ASoC: audio-graph-card2-sample.dtsi: add DPCM sample 14 ASoC: audio-graph-card2-sample.dtsi: add Multi CPU/Codec sample 15 ASoC: audio-graph-card2-sample.dtsi: add Codec2Codec sample. .../sound/audio-graph-card2-items.yaml | 36 + .../bindings/sound/audio-graph-card2.yaml | 45 + .../bindings/sound/audio-graph-port2.yaml | 74 + .../bindings/sound/test-component.yaml | 33 + drivers/of/property.c | 69 +- include/linux/of_graph.h | 14 + include/sound/graph_card.h | 24 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + .../soc/generic/audio-graph-card2-sample.dtsi | 140 ++ sound/soc/generic/audio-graph-card2.c | 1253 +++++++++++++++++ sound/soc/generic/sample-custom-card.c | 160 +++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 15 files changed, 2572 insertions(+), 11 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2-items.yaml create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-port2.yaml create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 sound/soc/generic/audio-graph-card2-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/sample-custom-card.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

2 17

[PATCH] pci: hda: Add IRQ check for platform_get_irq()
by Jiajun Cao 22 Jun '21

22 Jun '21

The function hda_tegra_first_init() neglects to check the return value after executing platform_get_irq(). hda_tegra_first_init() should check the return value (if negative error number) for errors so as to not pass a negative value to the devm_request_irq(). Fix it by adding a check for the return value irq_id. Signed-off-by: Jiajun Cao <jjcao20(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> --- sound/pci/hda/hda_tegra.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 6f2b743b9d75..6c6dc3fcde60 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -262,6 +262,9 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) const char *sname, *drv_name = "tegra-hda"; struct device_node *np = pdev->dev.of_node; + if (irq_id < 0) + return irq_id; + err = hda_tegra_init_chip(chip, pdev); if (err) return err; -- 2.17.1

3 2

[PATCH] dt-bindings: Drop redundant minItems/maxItems
by Rob Herring 22 Jun '21

22 Jun '21

If a property has an 'items' list, then a 'minItems' or 'maxItems' with the same size as the list is redundant and can be dropped. Note that is DT schema specific behavior and not standard json-schema behavior. The tooling will fixup the final schema adding any unspecified minItems/maxItems. This condition is partially checked with the meta-schema already, but only if both 'minItems' and 'maxItems' are equal to the 'items' length. An improved meta-schema is pending. Cc: Jens Axboe <axboe(a)kernel.dk> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: Herbert Xu <herbert(a)gondor.apana.org.au> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: David Airlie <airlied(a)linux.ie> Cc: Daniel Vetter <daniel(a)ffwll.ch> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Cc: Kamal Dasu <kdasu.kdev(a)gmail.com> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Marc Zyngier <maz(a)kernel.org> Cc: Joerg Roedel <joro(a)8bytes.org> Cc: Jassi Brar <jassisinghbrar(a)gmail.com> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> Cc: Ulf Hansson <ulf.hansson(a)linaro.org> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Wolfgang Grandegger <wg(a)grandegger.com> Cc: Marc Kleine-Budde <mkl(a)pengutronix.de> Cc: Andrew Lunn <andrew(a)lunn.ch> Cc: Vivien Didelot <vivien.didelot(a)gmail.com> Cc: Vladimir Oltean <olteanv(a)gmail.com> Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: Kishon Vijay Abraham I <kishon(a)ti.com> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: "Uwe Kleine-König" <u.kleine-koenig(a)pengutronix.de> Cc: Lee Jones <lee.jones(a)linaro.org> Cc: Ohad Ben-Cohen <ohad(a)wizery.com> Cc: Mathieu Poirier <mathieu.poirier(a)linaro.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Paul Walmsley <paul.walmsley(a)sifive.com> Cc: Palmer Dabbelt <palmer(a)dabbelt.com> Cc: Albert Ou <aou(a)eecs.berkeley.edu> Cc: Alessandro Zummo <a.zummo(a)towertech.it> Cc: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Zhang Rui <rui.zhang(a)intel.com> Cc: Daniel Lezcano <daniel.lezcano(a)linaro.org> Cc: Wim Van Sebroeck <wim(a)linux-watchdog.org> Cc: Guenter Roeck <linux(a)roeck-us.net> Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/ata/nvidia,tegra-ahci.yaml | 1 - .../devicetree/bindings/clock/allwinner,sun4i-a10-ccu.yaml | 2 -- .../devicetree/bindings/clock/qcom,gcc-apq8064.yaml | 1 - Documentation/devicetree/bindings/clock/qcom,gcc-sdx55.yaml | 2 -- .../devicetree/bindings/clock/qcom,gcc-sm8350.yaml | 2 -- .../devicetree/bindings/clock/sprd,sc9863a-clk.yaml | 1 - .../devicetree/bindings/crypto/allwinner,sun8i-ce.yaml | 2 -- Documentation/devicetree/bindings/crypto/fsl-dcp.yaml | 1 - .../display/allwinner,sun4i-a10-display-backend.yaml | 6 ------ .../bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml | 1 - .../bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml | 4 ---- .../bindings/display/allwinner,sun8i-a83t-hdmi-phy.yaml | 2 -- .../bindings/display/allwinner,sun8i-r40-tcon-top.yaml | 2 -- .../devicetree/bindings/display/bridge/cdns,mhdp8546.yaml | 2 -- .../bindings/display/rockchip/rockchip,dw-hdmi.yaml | 2 -- Documentation/devicetree/bindings/display/st,stm32-dsi.yaml | 2 -- .../devicetree/bindings/display/st,stm32-ltdc.yaml | 1 - .../devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml | 4 ---- .../devicetree/bindings/dma/renesas,rcar-dmac.yaml | 1 - .../devicetree/bindings/edac/amazon,al-mc-edac.yaml | 2 -- Documentation/devicetree/bindings/eeprom/at24.yaml | 1 - Documentation/devicetree/bindings/example-schema.yaml | 2 -- Documentation/devicetree/bindings/gpu/brcm,bcm-v3d.yaml | 1 - Documentation/devicetree/bindings/gpu/vivante,gc.yaml | 1 - Documentation/devicetree/bindings/i2c/brcm,brcmstb-i2c.yaml | 1 - .../devicetree/bindings/i2c/marvell,mv64xxx-i2c.yaml | 2 -- .../devicetree/bindings/i2c/mellanox,i2c-mlxbf.yaml | 1 - .../devicetree/bindings/iio/adc/amlogic,meson-saradc.yaml | 1 - .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 2 -- .../bindings/interrupt-controller/fsl,irqsteer.yaml | 1 - .../bindings/interrupt-controller/loongson,liointc.yaml | 1 - Documentation/devicetree/bindings/iommu/arm,smmu-v3.yaml | 1 - .../devicetree/bindings/iommu/renesas,ipmmu-vmsa.yaml | 1 - .../devicetree/bindings/mailbox/st,stm32-ipcc.yaml | 2 -- .../devicetree/bindings/media/amlogic,gx-vdec.yaml | 1 - Documentation/devicetree/bindings/media/i2c/adv7604.yaml | 1 - .../devicetree/bindings/media/marvell,mmp2-ccic.yaml | 1 - .../devicetree/bindings/media/qcom,sc7180-venus.yaml | 1 - .../devicetree/bindings/media/qcom,sdm845-venus-v2.yaml | 1 - .../devicetree/bindings/media/qcom,sm8250-venus.yaml | 1 - Documentation/devicetree/bindings/media/renesas,drif.yaml | 1 - .../bindings/memory-controllers/mediatek,smi-common.yaml | 6 ++---- .../bindings/memory-controllers/mediatek,smi-larb.yaml | 1 - .../devicetree/bindings/mmc/allwinner,sun4i-a10-mmc.yaml | 2 -- Documentation/devicetree/bindings/mmc/fsl-imx-esdhc.yaml | 1 - Documentation/devicetree/bindings/mmc/mtk-sd.yaml | 2 -- Documentation/devicetree/bindings/mmc/renesas,sdhi.yaml | 2 -- Documentation/devicetree/bindings/mmc/sdhci-am654.yaml | 1 - Documentation/devicetree/bindings/mmc/sdhci-pxa.yaml | 1 - .../devicetree/bindings/net/amlogic,meson-dwmac.yaml | 2 -- .../devicetree/bindings/net/brcm,bcm4908-enet.yaml | 2 -- Documentation/devicetree/bindings/net/can/bosch,m_can.yaml | 2 -- Documentation/devicetree/bindings/net/dsa/brcm,sf2.yaml | 2 -- Documentation/devicetree/bindings/net/snps,dwmac.yaml | 2 -- Documentation/devicetree/bindings/net/stm32-dwmac.yaml | 1 - Documentation/devicetree/bindings/pci/brcm,stb-pcie.yaml | 2 -- Documentation/devicetree/bindings/pci/loongson.yaml | 1 - .../devicetree/bindings/pci/mediatek-pcie-gen3.yaml | 1 - .../devicetree/bindings/pci/microchip,pcie-host.yaml | 2 -- Documentation/devicetree/bindings/perf/arm,cmn.yaml | 1 - .../devicetree/bindings/phy/brcm,bcm63xx-usbh-phy.yaml | 1 - .../devicetree/bindings/phy/brcm,brcmstb-usb-phy.yaml | 3 --- Documentation/devicetree/bindings/phy/brcm,sata-phy.yaml | 1 - Documentation/devicetree/bindings/phy/mediatek,tphy.yaml | 2 -- .../devicetree/bindings/phy/phy-cadence-sierra.yaml | 2 -- .../devicetree/bindings/phy/phy-cadence-torrent.yaml | 4 ---- .../devicetree/bindings/phy/qcom,ipq806x-usb-phy-hs.yaml | 1 - .../devicetree/bindings/phy/qcom,ipq806x-usb-phy-ss.yaml | 1 - Documentation/devicetree/bindings/phy/qcom,qmp-phy.yaml | 1 - Documentation/devicetree/bindings/phy/qcom,qusb2-phy.yaml | 2 -- Documentation/devicetree/bindings/phy/renesas,usb2-phy.yaml | 2 -- Documentation/devicetree/bindings/phy/renesas,usb3-phy.yaml | 1 - .../devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml | 1 - .../devicetree/bindings/power/amlogic,meson-ee-pwrc.yaml | 1 - .../devicetree/bindings/pwm/allwinner,sun4i-a10-pwm.yaml | 1 - .../devicetree/bindings/remoteproc/st,stm32-rproc.yaml | 2 -- .../devicetree/bindings/remoteproc/ti,k3-dsp-rproc.yaml | 1 - .../devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml | 1 - Documentation/devicetree/bindings/reset/fsl,imx-src.yaml | 1 - .../devicetree/bindings/riscv/sifive-l2-cache.yaml | 1 - .../devicetree/bindings/rtc/allwinner,sun6i-a31-rtc.yaml | 1 - Documentation/devicetree/bindings/rtc/imxdi-rtc.yaml | 1 - Documentation/devicetree/bindings/serial/fsl-lpuart.yaml | 2 -- Documentation/devicetree/bindings/serial/samsung_uart.yaml | 1 - .../devicetree/bindings/soc/qcom/qcom,geni-se.yaml | 1 - Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml | 2 -- .../bindings/sound/nvidia,tegra-audio-graph-card.yaml | 1 - .../devicetree/bindings/sound/nvidia,tegra210-i2s.yaml | 2 -- Documentation/devicetree/bindings/sound/st,stm32-sai.yaml | 3 --- .../devicetree/bindings/spi/amlogic,meson-gx-spicc.yaml | 1 - .../devicetree/bindings/spi/brcm,spi-bcm-qspi.yaml | 2 -- .../bindings/thermal/allwinner,sun8i-a83t-ths.yaml | 2 -- Documentation/devicetree/bindings/thermal/qcom-tsens.yaml | 1 - .../bindings/timer/allwinner,sun5i-a13-hstimer.yaml | 1 - Documentation/devicetree/bindings/timer/arm,arch_timer.yaml | 1 - .../devicetree/bindings/timer/arm,arch_timer_mmio.yaml | 2 -- .../devicetree/bindings/timer/intel,ixp4xx-timer.yaml | 1 - .../devicetree/bindings/usb/maxim,max3420-udc.yaml | 2 -- .../devicetree/bindings/usb/nvidia,tegra-xudc.yaml | 4 ---- Documentation/devicetree/bindings/usb/renesas,usbhs.yaml | 3 --- .../devicetree/bindings/watchdog/st,stm32-iwdg.yaml | 1 - 101 files changed, 2 insertions(+), 163 deletions(-) diff --git a/Documentation/devicetree/bindings/ata/nvidia,tegra-ahci.yaml b/Documentation/devicetree/bindings/ata/nvidia,tegra-ahci.yaml index a75e9a8f539a..3c7a2425f3e6 100644 --- a/Documentation/devicetree/bindings/ata/nvidia,tegra-ahci.yaml +++ b/Documentation/devicetree/bindings/ata/nvidia,tegra-ahci.yaml @@ -20,7 +20,6 @@ properties: reg: minItems: 2 - maxItems: 3 items: - description: AHCI registers - description: SATA configuration and IPFS registers diff --git a/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-ccu.yaml b/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-ccu.yaml index a27025cd3909..c4b7243ddcf2 100644 --- a/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-ccu.yaml +++ b/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-ccu.yaml @@ -51,7 +51,6 @@ properties: clocks: minItems: 2 - maxItems: 4 items: - description: High Frequency Oscillator (usually at 24MHz) - description: Low Frequency Oscillator (usually at 32kHz) @@ -60,7 +59,6 @@ properties: clock-names: minItems: 2 - maxItems: 4 items: - const: hosc - const: losc diff --git a/Documentation/devicetree/bindings/clock/qcom,gcc-apq8064.yaml b/Documentation/devicetree/bindings/clock/qcom,gcc-apq8064.yaml index eacccc88bbf6..8e2eac6cbfb9 100644 --- a/Documentation/devicetree/bindings/clock/qcom,gcc-apq8064.yaml +++ b/Documentation/devicetree/bindings/clock/qcom,gcc-apq8064.yaml @@ -46,7 +46,6 @@ properties: nvmem-cell-names: minItems: 1 - maxItems: 2 items: - const: calib - const: calib_backup diff --git a/Documentation/devicetree/bindings/clock/qcom,gcc-sdx55.yaml b/Documentation/devicetree/bindings/clock/qcom,gcc-sdx55.yaml index 1121b3934cb9..b0d1c65aa354 100644 --- a/Documentation/devicetree/bindings/clock/qcom,gcc-sdx55.yaml +++ b/Documentation/devicetree/bindings/clock/qcom,gcc-sdx55.yaml @@ -27,7 +27,6 @@ properties: - description: Sleep clock source - description: PLL test clock source (Optional clock) minItems: 2 - maxItems: 3 clock-names: items: @@ -35,7 +34,6 @@ properties: - const: sleep_clk - const: core_bi_pll_test_se # Optional clock minItems: 2 - maxItems: 3 '#clock-cells': const: 1 diff --git a/Documentation/devicetree/bindings/clock/qcom,gcc-sm8350.yaml b/Documentation/devicetree/bindings/clock/qcom,gcc-sm8350.yaml index 78f35832aa41..1122700dcc2b 100644 --- a/Documentation/devicetree/bindings/clock/qcom,gcc-sm8350.yaml +++ b/Documentation/devicetree/bindings/clock/qcom,gcc-sm8350.yaml @@ -36,7 +36,6 @@ properties: - description: USB3 phy wrapper pipe clock source (Optional clock) - description: USB3 phy sec pipe clock source (Optional clock) minItems: 2 - maxItems: 13 clock-names: items: @@ -54,7 +53,6 @@ properties: - const: usb3_phy_wrapper_gcc_usb30_pipe_clk # Optional clock - const: usb3_uni_phy_sec_gcc_usb30_pipe_clk # Optional clock minItems: 2 - maxItems: 13 '#clock-cells': const: 1 diff --git a/Documentation/devicetree/bindings/clock/sprd,sc9863a-clk.yaml b/Documentation/devicetree/bindings/clock/sprd,sc9863a-clk.yaml index 4069e09cb62d..47e1ab08c95d 100644 --- a/Documentation/devicetree/bindings/clock/sprd,sc9863a-clk.yaml +++ b/Documentation/devicetree/bindings/clock/sprd,sc9863a-clk.yaml @@ -40,7 +40,6 @@ properties: clock-names: minItems: 1 - maxItems: 4 items: - const: ext-26m - const: ext-32k diff --git a/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml b/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml index 6ab07eba7778..00648f9d9278 100644 --- a/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml +++ b/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml @@ -30,7 +30,6 @@ properties: - description: Module clock - description: MBus clock minItems: 2 - maxItems: 3 clock-names: items: @@ -38,7 +37,6 @@ properties: - const: mod - const: ram minItems: 2 - maxItems: 3 resets: maxItems: 1 diff --git a/Documentation/devicetree/bindings/crypto/fsl-dcp.yaml b/Documentation/devicetree/bindings/crypto/fsl-dcp.yaml index a30bf38a4a49..99be01539fcd 100644 --- a/Documentation/devicetree/bindings/crypto/fsl-dcp.yaml +++ b/Documentation/devicetree/bindings/crypto/fsl-dcp.yaml @@ -27,7 +27,6 @@ properties: - description: MXS DCP DCP interrupt - description: MXS DCP secure interrupt minItems: 2 - maxItems: 3 clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-display-backend.yaml b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-display-backend.yaml index 12a7df0e38b2..3d8ea3c2d8dd 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-display-backend.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-display-backend.yaml @@ -26,14 +26,12 @@ properties: reg: minItems: 1 - maxItems: 2 items: - description: Display Backend registers - description: SAT registers reg-names: minItems: 1 - maxItems: 2 items: - const: be - const: sat @@ -43,7 +41,6 @@ properties: clocks: minItems: 3 - maxItems: 4 items: - description: The backend interface clock - description: The backend module clock @@ -52,7 +49,6 @@ properties: clock-names: minItems: 3 - maxItems: 4 items: - const: ahb - const: mod @@ -61,14 +57,12 @@ properties: resets: minItems: 1 - maxItems: 2 items: - description: The Backend reset line - description: The SAT reset line reset-names: minItems: 1 - maxItems: 2 items: - const: be - const: sat diff --git a/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml index a738d7c12a97..bf0bdf54e5f9 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml @@ -24,7 +24,6 @@ properties: clocks: minItems: 1 - maxItems: 2 items: - description: Bus Clock - description: Module Clock diff --git a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml index 907fb47cc84a..5d42d36608d9 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml @@ -46,7 +46,6 @@ properties: clocks: minItems: 3 - maxItems: 6 items: - description: Bus Clock - description: Register Clock @@ -57,7 +56,6 @@ properties: clock-names: minItems: 3 - maxItems: 6 items: - const: iahb - const: isfr @@ -68,14 +66,12 @@ properties: resets: minItems: 1 - maxItems: 2 items: - description: HDMI Controller Reset - description: HDCP Reset reset-names: minItems: 1 - maxItems: 2 items: - const: ctrl - const: hdcp diff --git a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-hdmi-phy.yaml b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-hdmi-phy.yaml index 501cec16168c..a97366aaf924 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-hdmi-phy.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-hdmi-phy.yaml @@ -27,7 +27,6 @@ properties: clocks: minItems: 2 - maxItems: 4 items: - description: Bus Clock - description: Module Clock @@ -36,7 +35,6 @@ properties: clock-names: minItems: 2 - maxItems: 4 items: - const: bus - const: mod diff --git a/Documentation/devicetree/bindings/display/allwinner,sun8i-r40-tcon-top.yaml b/Documentation/devicetree/bindings/display/allwinner,sun8i-r40-tcon-top.yaml index ec21e8bf2767..61ef7b337218 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun8i-r40-tcon-top.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun8i-r40-tcon-top.yaml @@ -48,7 +48,6 @@ properties: clocks: minItems: 2 - maxItems: 6 items: - description: The TCON TOP interface clock - description: The TCON TOP TV0 clock @@ -59,7 +58,6 @@ properties: clock-names: minItems: 2 - maxItems: 6 items: - const: bus - const: tcon-tv0 diff --git a/Documentation/devicetree/bindings/display/bridge/cdns,mhdp8546.yaml b/Documentation/devicetree/bindings/display/bridge/cdns,mhdp8546.yaml index 63427878715e..9a3208a15137 100644 --- a/Documentation/devicetree/bindings/display/bridge/cdns,mhdp8546.yaml +++ b/Documentation/devicetree/bindings/display/bridge/cdns,mhdp8546.yaml @@ -18,7 +18,6 @@ properties: reg: minItems: 1 - maxItems: 2 items: - description: Register block of mhdptx apb registers up to PHY mapped area (AUX_CONFIG_P). @@ -29,7 +28,6 @@ properties: reg-names: minItems: 1 - maxItems: 2 items: - const: mhdptx - const: j721e-intg diff --git a/Documentation/devicetree/bindings/display/rockchip/rockchip,dw-hdmi.yaml b/Documentation/devicetree/bindings/display/rockchip/rockchip,dw-hdmi.yaml index 75cd9c686e98..da3b889ad8fc 100644 --- a/Documentation/devicetree/bindings/display/rockchip/rockchip,dw-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/rockchip/rockchip,dw-hdmi.yaml @@ -29,7 +29,6 @@ properties: clocks: minItems: 2 - maxItems: 5 items: - {} - {} @@ -41,7 +40,6 @@ properties: clock-names: minItems: 2 - maxItems: 5 items: - {} - {} diff --git a/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml b/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml index 679daed4124e..ed310bbe3afe 100644 --- a/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml +++ b/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml @@ -29,7 +29,6 @@ properties: - description: DSI bus clock - description: Pixel clock minItems: 2 - maxItems: 3 clock-names: items: @@ -37,7 +36,6 @@ properties: - const: ref - const: px_clk minItems: 2 - maxItems: 3 resets: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml b/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml index d54f9ca207af..4ae3d75492d3 100644 --- a/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml +++ b/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml @@ -22,7 +22,6 @@ properties: - description: events interrupt line. - description: errors interrupt line. minItems: 1 - maxItems: 2 clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml b/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml index 403d57977ee7..d88bd93f4b80 100644 --- a/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml +++ b/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml @@ -65,7 +65,6 @@ properties: The APB clock and at least one video clock are mandatory, the audio clock is optional. minItems: 2 - maxItems: 4 items: - description: dp_apb_clk is the APB clock - description: dp_aud_clk is the Audio clock @@ -78,13 +77,11 @@ properties: clock-names: oneOf: - minItems: 2 - maxItems: 3 items: - const: dp_apb_clk - enum: [ dp_vtc_pixel_clk_in, dp_live_video_in_clk ] - enum: [ dp_vtc_pixel_clk_in, dp_live_video_in_clk ] - minItems: 3 - maxItems: 4 items: - const: dp_apb_clk - const: dp_aud_clk @@ -116,7 +113,6 @@ properties: maxItems: 2 phy-names: minItems: 1 - maxItems: 2 items: - const: dp-phy0 - const: dp-phy1 diff --git a/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml b/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml index 7f2a54bc732d..d8142cbd13d3 100644 --- a/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml +++ b/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml @@ -52,7 +52,6 @@ properties: interrupt-names: minItems: 9 - maxItems: 17 items: - const: error - pattern: "^ch([0-9]|1[0-5])$" diff --git a/Documentation/devicetree/bindings/edac/amazon,al-mc-edac.yaml b/Documentation/devicetree/bindings/edac/amazon,al-mc-edac.yaml index 57e5270a0741..4cfc3a187004 100644 --- a/Documentation/devicetree/bindings/edac/amazon,al-mc-edac.yaml +++ b/Documentation/devicetree/bindings/edac/amazon,al-mc-edac.yaml @@ -30,14 +30,12 @@ properties: interrupts: minItems: 1 - maxItems: 2 items: - description: uncorrectable error interrupt - description: correctable error interrupt interrupt-names: minItems: 1 - maxItems: 2 items: - const: ue - const: ce diff --git a/Documentation/devicetree/bindings/eeprom/at24.yaml b/Documentation/devicetree/bindings/eeprom/at24.yaml index 021d8ae42da3..914a423ec449 100644 --- a/Documentation/devicetree/bindings/eeprom/at24.yaml +++ b/Documentation/devicetree/bindings/eeprom/at24.yaml @@ -32,7 +32,6 @@ properties: oneOf: - allOf: - minItems: 1 - maxItems: 2 items: - pattern: "^(atmel|catalyst|microchip|nxp|ramtron|renesas|rohm|st),(24(c|cs|lc|mac)[0-9]+|spd)$" - pattern: "^atmel,(24(c|cs|mac)[0-9]+|spd)$" diff --git a/Documentation/devicetree/bindings/example-schema.yaml b/Documentation/devicetree/bindings/example-schema.yaml index a97f39109f8d..ff6ec65145cf 100644 --- a/Documentation/devicetree/bindings/example-schema.yaml +++ b/Documentation/devicetree/bindings/example-schema.yaml @@ -91,7 +91,6 @@ properties: interrupts: # Either 1 or 2 interrupts can be present minItems: 1 - maxItems: 2 items: - description: tx or combined interrupt - description: rx interrupt @@ -105,7 +104,6 @@ properties: interrupt-names: # minItems must be specified here because the default would be 2 minItems: 1 - maxItems: 2 items: - const: tx irq - const: rx irq diff --git a/Documentation/devicetree/bindings/gpu/brcm,bcm-v3d.yaml b/Documentation/devicetree/bindings/gpu/brcm,bcm-v3d.yaml index 9d72264fa90a..e6485f7b046f 100644 --- a/Documentation/devicetree/bindings/gpu/brcm,bcm-v3d.yaml +++ b/Documentation/devicetree/bindings/gpu/brcm,bcm-v3d.yaml @@ -34,7 +34,6 @@ properties: - enum: [ bridge, gca ] - enum: [ bridge, gca ] minItems: 2 - maxItems: 4 interrupts: items: diff --git a/Documentation/devicetree/bindings/gpu/vivante,gc.yaml b/Documentation/devicetree/bindings/gpu/vivante,gc.yaml index 3ed172629974..93e7244cdc0e 100644 --- a/Documentation/devicetree/bindings/gpu/vivante,gc.yaml +++ b/Documentation/devicetree/bindings/gpu/vivante,gc.yaml @@ -36,7 +36,6 @@ properties: - description: AHB/slave interface clock (only required if GPU can gate slave interface independently) minItems: 1 - maxItems: 4 clock-names: items: diff --git a/Documentation/devicetree/bindings/i2c/brcm,brcmstb-i2c.yaml b/Documentation/devicetree/bindings/i2c/brcm,brcmstb-i2c.yaml index edbca2476128..7070c04469ed 100644 --- a/Documentation/devicetree/bindings/i2c/brcm,brcmstb-i2c.yaml +++ b/Documentation/devicetree/bindings/i2c/brcm,brcmstb-i2c.yaml @@ -21,7 +21,6 @@ properties: reg: minItems: 1 - maxItems: 2 items: - description: BSC register range - description: Auto-I2C register range diff --git a/Documentation/devicetree/bindings/i2c/marvell,mv64xxx-i2c.yaml b/Documentation/devicetree/bindings/i2c/marvell,mv64xxx-i2c.yaml index eb72dd571def..f771c09aabfc 100644 --- a/Documentation/devicetree/bindings/i2c/marvell,mv64xxx-i2c.yaml +++ b/Documentation/devicetree/bindings/i2c/marvell,mv64xxx-i2c.yaml @@ -43,14 +43,12 @@ properties: clocks: minItems: 1 - maxItems: 2 items: - description: Reference clock for the I2C bus - description: Bus clock (Only for Armada 7K/8K) clock-names: minItems: 1 - maxItems: 2 items: - const: core - const: reg diff --git a/Documentation/devicetree/bindings/i2c/mellanox,i2c-mlxbf.yaml b/Documentation/devicetree/bindings/i2c/mellanox,i2c-mlxbf.yaml index d2b401d062b9..93198d5d43a6 100644 --- a/Documentation/devicetree/bindings/i2c/mellanox,i2c-mlxbf.yaml +++ b/Documentation/devicetree/bindings/i2c/mellanox,i2c-mlxbf.yaml @@ -20,7 +20,6 @@ properties: reg: minItems: 3 - maxItems: 4 items: - description: Smbus block registers - description: Cause master registers diff --git a/Documentation/devicetree/bindings/iio/adc/amlogic,meson-saradc.yaml b/Documentation/devicetree/bindings/iio/adc/amlogic,meson-saradc.yaml index 3be8955587e4..7e8328e9ce13 100644 --- a/Documentation/devicetree/bindings/iio/adc/amlogic,meson-saradc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/amlogic,meson-saradc.yaml @@ -41,7 +41,6 @@ properties: clock-names: minItems: 2 - maxItems: 4 items: - const: clkin - const: core diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index 1e7894e524f9..733351dee252 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -38,14 +38,12 @@ properties: dfsdm clock can also feed CLKOUT, when CLKOUT is used. - description: audio clock can be used as an alternate to feed CLKOUT. minItems: 1 - maxItems: 2 clock-names: items: - const: dfsdm - const: audio minItems: 1 - maxItems: 2 "#address-cells": const: 1 diff --git a/Documentation/devicetree/bindings/interrupt-controller/fsl,irqsteer.yaml b/Documentation/devicetree/bindings/interrupt-controller/fsl,irqsteer.yaml index 3b11a1a15398..bcb5e20fa9ca 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/fsl,irqsteer.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/fsl,irqsteer.yaml @@ -35,7 +35,6 @@ properties: - description: output interrupt 6 - description: output interrupt 7 minItems: 1 - maxItems: 8 clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml b/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml index 067165c4b836..edf26452dc72 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/loongson,liointc.yaml @@ -50,7 +50,6 @@ properties: - const: int2 - const: int3 minItems: 1 - maxItems: 4 '#interrupt-cells': const: 2 diff --git a/Documentation/devicetree/bindings/iommu/arm,smmu-v3.yaml b/Documentation/devicetree/bindings/iommu/arm,smmu-v3.yaml index 5951c6f98c74..e87bfbcc6913 100644 --- a/Documentation/devicetree/bindings/iommu/arm,smmu-v3.yaml +++ b/Documentation/devicetree/bindings/iommu/arm,smmu-v3.yaml @@ -38,7 +38,6 @@ properties: If provided, then the combined interrupt will be used in preference to any others. - minItems: 2 - maxItems: 4 items: - const: eventq # Event Queue not empty - const: gerror # Global Error activated diff --git a/Documentation/devicetree/bindings/iommu/renesas,ipmmu-vmsa.yaml b/Documentation/devicetree/bindings/iommu/renesas,ipmmu-vmsa.yaml index dda44976acc1..02c69a95c332 100644 --- a/Documentation/devicetree/bindings/iommu/renesas,ipmmu-vmsa.yaml +++ b/Documentation/devicetree/bindings/iommu/renesas,ipmmu-vmsa.yaml @@ -49,7 +49,6 @@ properties: interrupts: minItems: 1 - maxItems: 2 description: Specifiers for the MMU fault interrupts. Not required for cache IPMMUs. items: diff --git a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml index 3b7ab61a144f..b15da9ba90b2 100644 --- a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml +++ b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml @@ -32,7 +32,6 @@ properties: - description: tx channel free - description: wakeup source minItems: 2 - maxItems: 3 interrupt-names: items: @@ -40,7 +39,6 @@ properties: - const: tx - const: wakeup minItems: 2 - maxItems: 3 wakeup-source: true diff --git a/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml b/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml index b902495d278b..5044c4bb94e0 100644 --- a/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml +++ b/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml @@ -67,7 +67,6 @@ properties: clock-names: minItems: 4 - maxItems: 5 items: - const: dos_parser - const: dos diff --git a/Documentation/devicetree/bindings/media/i2c/adv7604.yaml b/Documentation/devicetree/bindings/media/i2c/adv7604.yaml index df634b0c1f8c..de15cebe2955 100644 --- a/Documentation/devicetree/bindings/media/i2c/adv7604.yaml +++ b/Documentation/devicetree/bindings/media/i2c/adv7604.yaml @@ -30,7 +30,6 @@ properties: reg-names: minItems: 1 - maxItems: 13 items: - const: main - enum: [ avlink, cec, infoframe, esdp, dpp, afe, rep, edid, hdmi, test, cp, vdp ] diff --git a/Documentation/devicetree/bindings/media/marvell,mmp2-ccic.yaml b/Documentation/devicetree/bindings/media/marvell,mmp2-ccic.yaml index c14c7d827b00..b39b84c5f012 100644 --- a/Documentation/devicetree/bindings/media/marvell,mmp2-ccic.yaml +++ b/Documentation/devicetree/bindings/media/marvell,mmp2-ccic.yaml @@ -43,7 +43,6 @@ properties: clocks: minItems: 1 - maxItems: 3 items: - description: AXI bus interface clock - description: Peripheral clock diff --git a/Documentation/devicetree/bindings/media/qcom,sc7180-venus.yaml b/Documentation/devicetree/bindings/media/qcom,sc7180-venus.yaml index 04013e5dd044..90b4af2c9724 100644 --- a/Documentation/devicetree/bindings/media/qcom,sc7180-venus.yaml +++ b/Documentation/devicetree/bindings/media/qcom,sc7180-venus.yaml @@ -30,7 +30,6 @@ properties: power-domain-names: minItems: 2 - maxItems: 3 items: - const: venus - const: vcodec0 diff --git a/Documentation/devicetree/bindings/media/qcom,sdm845-venus-v2.yaml b/Documentation/devicetree/bindings/media/qcom,sdm845-venus-v2.yaml index 04b9af4db191..177bf81544b1 100644 --- a/Documentation/devicetree/bindings/media/qcom,sdm845-venus-v2.yaml +++ b/Documentation/devicetree/bindings/media/qcom,sdm845-venus-v2.yaml @@ -30,7 +30,6 @@ properties: power-domain-names: minItems: 3 - maxItems: 4 items: - const: venus - const: vcodec0 diff --git a/Documentation/devicetree/bindings/media/qcom,sm8250-venus.yaml b/Documentation/devicetree/bindings/media/qcom,sm8250-venus.yaml index 7b81bd7f2399..ebf8f3d866a5 100644 --- a/Documentation/devicetree/bindings/media/qcom,sm8250-venus.yaml +++ b/Documentation/devicetree/bindings/media/qcom,sm8250-venus.yaml @@ -30,7 +30,6 @@ properties: power-domain-names: minItems: 2 - maxItems: 3 items: - const: venus - const: vcodec0 diff --git a/Documentation/devicetree/bindings/media/renesas,drif.yaml b/Documentation/devicetree/bindings/media/renesas,drif.yaml index 9cd56ff2c316..817a6d566738 100644 --- a/Documentation/devicetree/bindings/media/renesas,drif.yaml +++ b/Documentation/devicetree/bindings/media/renesas,drif.yaml @@ -78,7 +78,6 @@ properties: dma-names: minItems: 1 - maxItems: 2 items: - const: rx - const: rx diff --git a/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-common.yaml b/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-common.yaml index a08a32340987..e87e4382807c 100644 --- a/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-common.yaml +++ b/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-common.yaml @@ -53,14 +53,12 @@ properties: apb and smi are mandatory. the async is only for generation 1 smi HW. gals(global async local sync) also is optional, see below. minItems: 2 - maxItems: 4 items: - description: apb is Advanced Peripheral Bus clock, It's the clock for setting the register. - description: smi is the clock for transfer data and command. - - description: async is asynchronous clock, it help transform the smi - clock into the emi clock domain. - - description: gals0 is the path0 clock of gals. + - description: Either asynchronous clock to help transform the smi clock + into the emi clock domain on Gen1 h/w, or the path0 clock of gals. - description: gals1 is the path1 clock of gals. clock-names: diff --git a/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-larb.yaml b/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-larb.yaml index 7ed7839ff0a7..2353f6cf3c80 100644 --- a/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-larb.yaml +++ b/Documentation/devicetree/bindings/memory-controllers/mediatek,smi-larb.yaml @@ -37,7 +37,6 @@ properties: description: | apb and smi are mandatory. gals(global async local sync) is optional. minItems: 2 - maxItems: 3 items: - description: apb is Advanced Peripheral Bus clock, It's the clock for setting the register. diff --git a/Documentation/devicetree/bindings/mmc/allwinner,sun4i-a10-mmc.yaml b/Documentation/devicetree/bindings/mmc/allwinner,sun4i-a10-mmc.yaml index e75b3a8ba816..4f62ad6ce50c 100644 --- a/Documentation/devicetree/bindings/mmc/allwinner,sun4i-a10-mmc.yaml +++ b/Documentation/devicetree/bindings/mmc/allwinner,sun4i-a10-mmc.yaml @@ -64,7 +64,6 @@ properties: clocks: minItems: 2 - maxItems: 4 items: - description: Bus Clock - description: Module Clock @@ -73,7 +72,6 @@ properties: clock-names: minItems: 2 - maxItems: 4 items: - const: ahb - const: mmc diff --git a/Documentation/devicetree/bindings/mmc/fsl-imx-esdhc.yaml b/Documentation/devicetree/bindings/mmc/fsl-imx-esdhc.yaml index 369471814496..b5baf439fbac 100644 --- a/Documentation/devicetree/bindings/mmc/fsl-imx-esdhc.yaml +++ b/Documentation/devicetree/bindings/mmc/fsl-imx-esdhc.yaml @@ -116,7 +116,6 @@ properties: pinctrl-names: minItems: 1 - maxItems: 4 items: - const: default - const: state_100mhz diff --git a/Documentation/devicetree/bindings/mmc/mtk-sd.yaml b/Documentation/devicetree/bindings/mmc/mtk-sd.yaml index 8648d48dbbfd..4e553fd0349f 100644 --- a/Documentation/devicetree/bindings/mmc/mtk-sd.yaml +++ b/Documentation/devicetree/bindings/mmc/mtk-sd.yaml @@ -38,7 +38,6 @@ properties: description: Should contain phandle for the clock feeding the MMC controller. minItems: 2 - maxItems: 8 items: - description: source clock (required). - description: HCLK which used for host (required). @@ -51,7 +50,6 @@ properties: clock-names: minItems: 2 - maxItems: 8 items: - const: source - const: hclk diff --git a/Documentation/devicetree/bindings/mmc/renesas,sdhi.yaml b/Documentation/devicetree/bindings/mmc/renesas,sdhi.yaml index 1118b6fa93c9..677989bc5924 100644 --- a/Documentation/devicetree/bindings/mmc/renesas,sdhi.yaml +++ b/Documentation/devicetree/bindings/mmc/renesas,sdhi.yaml @@ -75,7 +75,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: core - const: cd @@ -107,7 +106,6 @@ properties: pinctrl-names: minItems: 1 - maxItems: 2 items: - const: default - const: state_uhs diff --git a/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml b/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml index 3a79e39253d2..94a96174f99a 100644 --- a/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml +++ b/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml @@ -45,7 +45,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: clk_ahb - const: clk_xin diff --git a/Documentation/devicetree/bindings/mmc/sdhci-pxa.yaml b/Documentation/devicetree/bindings/mmc/sdhci-pxa.yaml index aa12480648a5..1c87f4218e18 100644 --- a/Documentation/devicetree/bindings/mmc/sdhci-pxa.yaml +++ b/Documentation/devicetree/bindings/mmc/sdhci-pxa.yaml @@ -57,7 +57,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: io - const: core diff --git a/Documentation/devicetree/bindings/net/amlogic,meson-dwmac.yaml b/Documentation/devicetree/bindings/net/amlogic,meson-dwmac.yaml index 0467441d7037..608e1d62bed5 100644 --- a/Documentation/devicetree/bindings/net/amlogic,meson-dwmac.yaml +++ b/Documentation/devicetree/bindings/net/amlogic,meson-dwmac.yaml @@ -43,7 +43,6 @@ allOf: properties: clocks: minItems: 3 - maxItems: 4 items: - description: GMAC main clock - description: First parent clock of the internal mux @@ -52,7 +51,6 @@ allOf: clock-names: minItems: 3 - maxItems: 4 items: - const: stmmaceth - const: clkin0 diff --git a/Documentation/devicetree/bindings/net/brcm,bcm4908-enet.yaml b/Documentation/devicetree/bindings/net/brcm,bcm4908-enet.yaml index 2f46e45dcd60..a93d2f165899 100644 --- a/Documentation/devicetree/bindings/net/brcm,bcm4908-enet.yaml +++ b/Documentation/devicetree/bindings/net/brcm,bcm4908-enet.yaml @@ -23,14 +23,12 @@ properties: interrupts: minItems: 1 - maxItems: 2 items: - description: RX interrupt - description: TX interrupt interrupt-names: minItems: 1 - maxItems: 2 items: - const: rx - const: tx diff --git a/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml b/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml index 798fa5fb7bb2..f84e31348d80 100644 --- a/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml +++ b/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml @@ -30,14 +30,12 @@ properties: - description: interrupt line0 - description: interrupt line1 minItems: 1 - maxItems: 2 interrupt-names: items: - const: int0 - const: int1 minItems: 1 - maxItems: 2 clocks: items: diff --git a/Documentation/devicetree/bindings/net/dsa/brcm,sf2.yaml b/Documentation/devicetree/bindings/net/dsa/brcm,sf2.yaml index d730fe5a4355..d159ac78cec1 100644 --- a/Documentation/devicetree/bindings/net/dsa/brcm,sf2.yaml +++ b/Documentation/devicetree/bindings/net/dsa/brcm,sf2.yaml @@ -48,14 +48,12 @@ properties: clocks: minItems: 1 - maxItems: 2 items: - description: switch's main clock - description: dividing of the switch core clock clock-names: minItems: 1 - maxItems: 2 items: - const: sw_switch - const: sw_switch_mdiv diff --git a/Documentation/devicetree/bindings/net/snps,dwmac.yaml b/Documentation/devicetree/bindings/net/snps,dwmac.yaml index 2edd8bea993e..5d4b028e5620 100644 --- a/Documentation/devicetree/bindings/net/snps,dwmac.yaml +++ b/Documentation/devicetree/bindings/net/snps,dwmac.yaml @@ -82,7 +82,6 @@ properties: interrupts: minItems: 1 - maxItems: 3 items: - description: Combined signal for various interrupt events - description: The interrupt to manage the remote wake-up packet detection @@ -90,7 +89,6 @@ properties: interrupt-names: minItems: 1 - maxItems: 3 items: - const: macirq - const: eth_wake_irq diff --git a/Documentation/devicetree/bindings/net/stm32-dwmac.yaml b/Documentation/devicetree/bindings/net/stm32-dwmac.yaml index 27eb6066793f..19c7bd482a51 100644 --- a/Documentation/devicetree/bindings/net/stm32-dwmac.yaml +++ b/Documentation/devicetree/bindings/net/stm32-dwmac.yaml @@ -46,7 +46,6 @@ properties: clocks: minItems: 3 - maxItems: 5 items: - description: GMAC main clock - description: MAC TX clock diff --git a/Documentation/devicetree/bindings/pci/brcm,stb-pcie.yaml b/Documentation/devicetree/bindings/pci/brcm,stb-pcie.yaml index f90557f6deb8..b9589a0daa5c 100644 --- a/Documentation/devicetree/bindings/pci/brcm,stb-pcie.yaml +++ b/Documentation/devicetree/bindings/pci/brcm,stb-pcie.yaml @@ -25,14 +25,12 @@ properties: interrupts: minItems: 1 - maxItems: 2 items: - description: PCIe host controller - description: builtin MSI controller interrupt-names: minItems: 1 - maxItems: 2 items: - const: pcie - const: msi diff --git a/Documentation/devicetree/bindings/pci/loongson.yaml b/Documentation/devicetree/bindings/pci/loongson.yaml index 81bae060cbde..82bc6c486ca3 100644 --- a/Documentation/devicetree/bindings/pci/loongson.yaml +++ b/Documentation/devicetree/bindings/pci/loongson.yaml @@ -24,7 +24,6 @@ properties: reg: minItems: 1 - maxItems: 2 items: - description: CFG0 standard config space register - description: CFG1 extended config space register diff --git a/Documentation/devicetree/bindings/pci/mediatek-pcie-gen3.yaml b/Documentation/devicetree/bindings/pci/mediatek-pcie-gen3.yaml index e7b1f9892da4..742206dbd965 100644 --- a/Documentation/devicetree/bindings/pci/mediatek-pcie-gen3.yaml +++ b/Documentation/devicetree/bindings/pci/mediatek-pcie-gen3.yaml @@ -70,7 +70,6 @@ properties: reset-names: minItems: 1 - maxItems: 2 items: - const: phy - const: mac diff --git a/Documentation/devicetree/bindings/pci/microchip,pcie-host.yaml b/Documentation/devicetree/bindings/pci/microchip,pcie-host.yaml index 04251d71f56b..fb95c276a986 100644 --- a/Documentation/devicetree/bindings/pci/microchip,pcie-host.yaml +++ b/Documentation/devicetree/bindings/pci/microchip,pcie-host.yaml @@ -26,14 +26,12 @@ properties: interrupts: minItems: 1 - maxItems: 2 items: - description: PCIe host controller - description: builtin MSI controller interrupt-names: minItems: 1 - maxItems: 2 items: - const: pcie - const: msi diff --git a/Documentation/devicetree/bindings/perf/arm,cmn.yaml b/Documentation/devicetree/bindings/perf/arm,cmn.yaml index e4fcc0de25e2..42424ccbdd0c 100644 --- a/Documentation/devicetree/bindings/perf/arm,cmn.yaml +++ b/Documentation/devicetree/bindings/perf/arm,cmn.yaml @@ -21,7 +21,6 @@ properties: interrupts: minItems: 1 - maxItems: 4 items: - description: Overflow interrupt for DTC0 - description: Overflow interrupt for DTC1 diff --git a/Documentation/devicetree/bindings/phy/brcm,bcm63xx-usbh-phy.yaml b/Documentation/devicetree/bindings/phy/brcm,bcm63xx-usbh-phy.yaml index 9a2e779e6d38..0f0bcde9eb88 100644 --- a/Documentation/devicetree/bindings/phy/brcm,bcm63xx-usbh-phy.yaml +++ b/Documentation/devicetree/bindings/phy/brcm,bcm63xx-usbh-phy.yaml @@ -28,7 +28,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: usbh - const: usb_ref diff --git a/Documentation/devicetree/bindings/phy/brcm,brcmstb-usb-phy.yaml b/Documentation/devicetree/bindings/phy/brcm,brcmstb-usb-phy.yaml index 5f9e91bfb5ff..43a4b880534c 100644 --- a/Documentation/devicetree/bindings/phy/brcm,brcmstb-usb-phy.yaml +++ b/Documentation/devicetree/bindings/phy/brcm,brcmstb-usb-phy.yaml @@ -22,7 +22,6 @@ properties: reg: minItems: 1 - maxItems: 6 items: - description: the base CTRL register - description: XHCI EC register @@ -33,7 +32,6 @@ properties: reg-names: minItems: 1 - maxItems: 6 items: - const: ctrl - const: xhci_ec @@ -51,7 +49,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: sw_usb - const: sw_usb3 diff --git a/Documentation/devicetree/bindings/phy/brcm,sata-phy.yaml b/Documentation/devicetree/bindings/phy/brcm,sata-phy.yaml index 04edda504ab6..cb1aa325336f 100644 --- a/Documentation/devicetree/bindings/phy/brcm,sata-phy.yaml +++ b/Documentation/devicetree/bindings/phy/brcm,sata-phy.yaml @@ -35,7 +35,6 @@ properties: reg-names: minItems: 1 - maxItems: 2 items: - const: phy - const: phy-ctrl diff --git a/Documentation/devicetree/bindings/phy/mediatek,tphy.yaml b/Documentation/devicetree/bindings/phy/mediatek,tphy.yaml index b8a7651a3d9a..ef9d9d4e6875 100644 --- a/Documentation/devicetree/bindings/phy/mediatek,tphy.yaml +++ b/Documentation/devicetree/bindings/phy/mediatek,tphy.yaml @@ -131,7 +131,6 @@ patternProperties: clocks: minItems: 1 - maxItems: 2 items: - description: Reference clock, (HS is 48Mhz, SS/P is 24~27Mhz) - description: Reference clock of analog phy @@ -141,7 +140,6 @@ patternProperties: clock-names: minItems: 1 - maxItems: 2 items: - const: ref - const: da_ref diff --git a/Documentation/devicetree/bindings/phy/phy-cadence-sierra.yaml b/Documentation/devicetree/bindings/phy/phy-cadence-sierra.yaml index 84383e2e0b34..e71b32c9c0d1 100644 --- a/Documentation/devicetree/bindings/phy/phy-cadence-sierra.yaml +++ b/Documentation/devicetree/bindings/phy/phy-cadence-sierra.yaml @@ -31,14 +31,12 @@ properties: resets: minItems: 1 - maxItems: 2 items: - description: Sierra PHY reset. - description: Sierra APB reset. This is optional. reset-names: minItems: 1 - maxItems: 2 items: - const: sierra_reset - const: sierra_apb diff --git a/Documentation/devicetree/bindings/phy/phy-cadence-torrent.yaml b/Documentation/devicetree/bindings/phy/phy-cadence-torrent.yaml index 320a232c7208..bd9ae11c9994 100644 --- a/Documentation/devicetree/bindings/phy/phy-cadence-torrent.yaml +++ b/Documentation/devicetree/bindings/phy/phy-cadence-torrent.yaml @@ -52,28 +52,24 @@ properties: reg: minItems: 1 - maxItems: 2 items: - description: Offset of the Torrent PHY configuration registers. - description: Offset of the DPTX PHY configuration registers. reg-names: minItems: 1 - maxItems: 2 items: - const: torrent_phy - const: dptx_phy resets: minItems: 1 - maxItems: 2 items: - description: Torrent PHY reset. - description: Torrent APB reset. This is optional. reset-names: minItems: 1 - maxItems: 2 items: - const: torrent_reset - const: torrent_apb diff --git a/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-hs.yaml b/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-hs.yaml index 17f132ce5516..35296c588e78 100644 --- a/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-hs.yaml +++ b/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-hs.yaml @@ -30,7 +30,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: ref - const: xo diff --git a/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-ss.yaml b/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-ss.yaml index 17fd7f6b83bb..6cf5c6c06072 100644 --- a/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-ss.yaml +++ b/Documentation/devicetree/bindings/phy/qcom,ipq806x-usb-phy-ss.yaml @@ -30,7 +30,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: ref - const: xo diff --git a/Documentation/devicetree/bindings/phy/qcom,qmp-phy.yaml b/Documentation/devicetree/bindings/phy/qcom,qmp-phy.yaml index 7808ec8bc712..a2de5202eb5e 100644 --- a/Documentation/devicetree/bindings/phy/qcom,qmp-phy.yaml +++ b/Documentation/devicetree/bindings/phy/qcom,qmp-phy.yaml @@ -49,7 +49,6 @@ properties: reg: minItems: 1 - maxItems: 2 items: - description: Address and length of PHY's common serdes block. - description: Address and length of PHY's DP_COM control block. diff --git a/Documentation/devicetree/bindings/phy/qcom,qusb2-phy.yaml b/Documentation/devicetree/bindings/phy/qcom,qusb2-phy.yaml index 9f9cf07b7d45..930da598c969 100644 --- a/Documentation/devicetree/bindings/phy/qcom,qusb2-phy.yaml +++ b/Documentation/devicetree/bindings/phy/qcom,qusb2-phy.yaml @@ -36,7 +36,6 @@ properties: clocks: minItems: 2 - maxItems: 3 items: - description: phy config clock - description: 19.2 MHz ref clk @@ -44,7 +43,6 @@ properties: clock-names: minItems: 2 - maxItems: 3 items: - const: cfg_ahb - const: ref diff --git a/Documentation/devicetree/bindings/phy/renesas,usb2-phy.yaml b/Documentation/devicetree/bindings/phy/renesas,usb2-phy.yaml index 0f358d5b84ef..d5dc5a3cdceb 100644 --- a/Documentation/devicetree/bindings/phy/renesas,usb2-phy.yaml +++ b/Documentation/devicetree/bindings/phy/renesas,usb2-phy.yaml @@ -39,7 +39,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: fck - const: usb_x1 @@ -61,7 +60,6 @@ properties: resets: minItems: 1 - maxItems: 2 items: - description: reset of USB 2.0 host side - description: reset of USB 2.0 peripheral side diff --git a/Documentation/devicetree/bindings/phy/renesas,usb3-phy.yaml b/Documentation/devicetree/bindings/phy/renesas,usb3-phy.yaml index f3ef738a3ff6..b8483f9edbfc 100644 --- a/Documentation/devicetree/bindings/phy/renesas,usb3-phy.yaml +++ b/Documentation/devicetree/bindings/phy/renesas,usb3-phy.yaml @@ -33,7 +33,6 @@ properties: # If you want to use the ssc, the clock-frequency of usb_extal # must not be 0. minItems: 2 - maxItems: 3 items: - const: usb3-if # The funcional clock - const: usb3s_clk # The usb3's external clock diff --git a/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml index ccdd9e3820d7..3f94f6944740 100644 --- a/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml @@ -26,7 +26,6 @@ properties: - description: PAD Pull Control + PAD Schmitt Trigger Enable + PAD Control - description: PAD Drive Capacity Select minItems: 1 - maxItems: 4 clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/power/amlogic,meson-ee-pwrc.yaml b/Documentation/devicetree/bindings/power/amlogic,meson-ee-pwrc.yaml index d30f85cc395e..f005abac7079 100644 --- a/Documentation/devicetree/bindings/power/amlogic,meson-ee-pwrc.yaml +++ b/Documentation/devicetree/bindings/power/amlogic,meson-ee-pwrc.yaml @@ -37,7 +37,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: vpu - const: vapb diff --git a/Documentation/devicetree/bindings/pwm/allwinner,sun4i-a10-pwm.yaml b/Documentation/devicetree/bindings/pwm/allwinner,sun4i-a10-pwm.yaml index 7dcab2bf8128..54a7700df08f 100644 --- a/Documentation/devicetree/bindings/pwm/allwinner,sun4i-a10-pwm.yaml +++ b/Documentation/devicetree/bindings/pwm/allwinner,sun4i-a10-pwm.yaml @@ -37,7 +37,6 @@ properties: clocks: minItems: 1 - maxItems: 2 items: - description: Module Clock - description: Bus Clock diff --git a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml index 64afdcfb613d..1e6225677e00 100644 --- a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml @@ -72,7 +72,6 @@ properties: - from local to remote, where ACK from the remote means that communnication as been stopped on the remote side. minItems: 1 - maxItems: 4 mbox-names: items: @@ -81,7 +80,6 @@ properties: - const: shutdown - const: detach minItems: 1 - maxItems: 4 memory-region: description: diff --git a/Documentation/devicetree/bindings/remoteproc/ti,k3-dsp-rproc.yaml b/Documentation/devicetree/bindings/remoteproc/ti,k3-dsp-rproc.yaml index 6070456a7b67..f399743b631b 100644 --- a/Documentation/devicetree/bindings/remoteproc/ti,k3-dsp-rproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/ti,k3-dsp-rproc.yaml @@ -57,7 +57,6 @@ properties: memory-region: minItems: 2 - maxItems: 8 description: | phandle to the reserved memory nodes to be associated with the remoteproc device. There should be at least two reserved memory nodes defined. The diff --git a/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml b/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml index 73400bc6e91d..75161f191ac3 100644 --- a/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/ti,omap-remoteproc.yaml @@ -116,7 +116,6 @@ properties: list, in the specified order, each representing the corresponding internal RAM memory region. minItems: 1 - maxItems: 3 items: - const: l2ram - const: l1pram diff --git a/Documentation/devicetree/bindings/reset/fsl,imx-src.yaml b/Documentation/devicetree/bindings/reset/fsl,imx-src.yaml index 27c5e34a3ac6..b11ac533f914 100644 --- a/Documentation/devicetree/bindings/reset/fsl,imx-src.yaml +++ b/Documentation/devicetree/bindings/reset/fsl,imx-src.yaml @@ -59,7 +59,6 @@ properties: - description: SRC interrupt - description: CPU WDOG interrupts out of SRC minItems: 1 - maxItems: 2 '#reset-cells': const: 1 diff --git a/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml b/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml index 23b227614366..1d38ff76d18f 100644 --- a/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml +++ b/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml @@ -56,7 +56,6 @@ properties: interrupts: minItems: 3 - maxItems: 4 items: - description: DirError interrupt - description: DataError interrupt diff --git a/Documentation/devicetree/bindings/rtc/allwinner,sun6i-a31-rtc.yaml b/Documentation/devicetree/bindings/rtc/allwinner,sun6i-a31-rtc.yaml index b1b0ee769b71..beeb90e55727 100644 --- a/Documentation/devicetree/bindings/rtc/allwinner,sun6i-a31-rtc.yaml +++ b/Documentation/devicetree/bindings/rtc/allwinner,sun6i-a31-rtc.yaml @@ -32,7 +32,6 @@ properties: interrupts: minItems: 1 - maxItems: 2 items: - description: RTC Alarm 0 - description: RTC Alarm 1 diff --git a/Documentation/devicetree/bindings/rtc/imxdi-rtc.yaml b/Documentation/devicetree/bindings/rtc/imxdi-rtc.yaml index 06bd737821c1..4807c95a663c 100644 --- a/Documentation/devicetree/bindings/rtc/imxdi-rtc.yaml +++ b/Documentation/devicetree/bindings/rtc/imxdi-rtc.yaml @@ -21,7 +21,6 @@ properties: - description: rtc alarm interrupt - description: dryice security violation interrupt minItems: 1 - maxItems: 2 clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/serial/fsl-lpuart.yaml b/Documentation/devicetree/bindings/serial/fsl-lpuart.yaml index bd21060d26e0..a90c971b4f1f 100644 --- a/Documentation/devicetree/bindings/serial/fsl-lpuart.yaml +++ b/Documentation/devicetree/bindings/serial/fsl-lpuart.yaml @@ -36,14 +36,12 @@ properties: - description: ipg clock - description: baud clock minItems: 1 - maxItems: 2 clock-names: items: - const: ipg - const: baud minItems: 1 - maxItems: 2 dmas: items: diff --git a/Documentation/devicetree/bindings/serial/samsung_uart.yaml b/Documentation/devicetree/bindings/serial/samsung_uart.yaml index 97ec8a093bf3..3ec3822bd114 100644 --- a/Documentation/devicetree/bindings/serial/samsung_uart.yaml +++ b/Documentation/devicetree/bindings/serial/samsung_uart.yaml @@ -44,7 +44,6 @@ properties: clock-names: description: N = 0 is allowed for SoCs without internal baud clock mux. minItems: 2 - maxItems: 5 items: - const: uart - pattern: '^clk_uart_baud[0-3]$' diff --git a/Documentation/devicetree/bindings/soc/qcom/qcom,geni-se.yaml b/Documentation/devicetree/bindings/soc/qcom/qcom,geni-se.yaml index 84671950ca0d..4663c2bcad50 100644 --- a/Documentation/devicetree/bindings/soc/qcom/qcom,geni-se.yaml +++ b/Documentation/devicetree/bindings/soc/qcom/qcom,geni-se.yaml @@ -164,7 +164,6 @@ patternProperties: interrupts: minItems: 1 - maxItems: 2 items: - description: UART core irq - description: Wakeup irq (RX GPIO) diff --git a/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml b/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml index dbc62821c60b..9790617af1bc 100644 --- a/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml +++ b/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml @@ -100,7 +100,6 @@ patternProperties: properties: reg: minItems: 2 # On AM437x one of two PRUSS units don't contain Shared RAM. - maxItems: 3 items: - description: Address and size of the Data RAM0. - description: Address and size of the Data RAM1. @@ -111,7 +110,6 @@ patternProperties: reg-names: minItems: 2 - maxItems: 3 items: - const: dram0 - const: dram1 diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml index 249970952202..5bdd30a8a404 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml @@ -28,7 +28,6 @@ properties: minItems: 2 clock-names: - minItems: 2 items: - const: pll_a - const: plla_out0 diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml index 38e52e7dbb40..63370709c768 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml @@ -34,7 +34,6 @@ properties: clocks: minItems: 1 - maxItems: 2 items: - description: I2S bit clock - description: @@ -48,7 +47,6 @@ properties: clock-names: minItems: 1 - maxItems: 2 items: - const: i2s - const: sync_input diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index f2443b651282..06e83461705c 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -26,7 +26,6 @@ properties: - description: Base address and size of SAI common register set. - description: Base address and size of SAI identification register set. minItems: 1 - maxItems: 2 ranges: maxItems: 1 @@ -81,14 +80,12 @@ patternProperties: - description: sai_ck clock feeding the internal clock generator. - description: MCLK clock from a SAI set as master clock provider. minItems: 1 - maxItems: 2 clock-names: items: - const: sai_ck - const: MCLK minItems: 1 - maxItems: 2 dmas: maxItems: 1 diff --git a/Documentation/devicetree/bindings/spi/amlogic,meson-gx-spicc.yaml b/Documentation/devicetree/bindings/spi/amlogic,meson-gx-spicc.yaml index e3fb553d9180..4d46c49ec32b 100644 --- a/Documentation/devicetree/bindings/spi/amlogic,meson-gx-spicc.yaml +++ b/Documentation/devicetree/bindings/spi/amlogic,meson-gx-spicc.yaml @@ -35,7 +35,6 @@ properties: clocks: minItems: 1 - maxItems: 2 items: - description: controller register bus clock - description: baud rate generator and delay control clock diff --git a/Documentation/devicetree/bindings/spi/brcm,spi-bcm-qspi.yaml b/Documentation/devicetree/bindings/spi/brcm,spi-bcm-qspi.yaml index 6ee19d49fd3c..ec5873919170 100644 --- a/Documentation/devicetree/bindings/spi/brcm,spi-bcm-qspi.yaml +++ b/Documentation/devicetree/bindings/spi/brcm,spi-bcm-qspi.yaml @@ -56,7 +56,6 @@ properties: reg-names: minItems: 1 - maxItems: 5 items: - const: mspi - const: bspi @@ -71,7 +70,6 @@ properties: interrupt-names: oneOf: - minItems: 1 - maxItems: 7 items: - const: mspi_done - const: mspi_halted diff --git a/Documentation/devicetree/bindings/thermal/allwinner,sun8i-a83t-ths.yaml b/Documentation/devicetree/bindings/thermal/allwinner,sun8i-a83t-ths.yaml index bf97d1fb33e7..6e0b110153b0 100644 --- a/Documentation/devicetree/bindings/thermal/allwinner,sun8i-a83t-ths.yaml +++ b/Documentation/devicetree/bindings/thermal/allwinner,sun8i-a83t-ths.yaml @@ -23,14 +23,12 @@ properties: clocks: minItems: 1 - maxItems: 2 items: - description: Bus Clock - description: Module Clock clock-names: minItems: 1 - maxItems: 2 items: - const: bus - const: mod diff --git a/Documentation/devicetree/bindings/thermal/qcom-tsens.yaml b/Documentation/devicetree/bindings/thermal/qcom-tsens.yaml index 0242fd91b622..0b3b6af7bd5b 100644 --- a/Documentation/devicetree/bindings/thermal/qcom-tsens.yaml +++ b/Documentation/devicetree/bindings/thermal/qcom-tsens.yaml @@ -77,7 +77,6 @@ properties: nvmem-cell-names: minItems: 1 - maxItems: 2 items: - const: calib - enum: diff --git a/Documentation/devicetree/bindings/timer/allwinner,sun5i-a13-hstimer.yaml b/Documentation/devicetree/bindings/timer/allwinner,sun5i-a13-hstimer.yaml index b6a6d03a08b2..2ecac754e1cd 100644 --- a/Documentation/devicetree/bindings/timer/allwinner,sun5i-a13-hstimer.yaml +++ b/Documentation/devicetree/bindings/timer/allwinner,sun5i-a13-hstimer.yaml @@ -24,7 +24,6 @@ properties: interrupts: minItems: 2 - maxItems: 4 items: - description: Timer 0 Interrupt - description: Timer 1 Interrupt diff --git a/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml b/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml index 7f5e3af58255..df8ce87fd54b 100644 --- a/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml +++ b/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml @@ -35,7 +35,6 @@ properties: interrupts: minItems: 1 - maxItems: 5 items: - description: secure timer irq - description: non-secure timer irq diff --git a/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml b/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml index d83a1f97f911..cd2176cad53a 100644 --- a/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml +++ b/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml @@ -71,14 +71,12 @@ patternProperties: interrupts: minItems: 1 - maxItems: 2 items: - description: physical timer irq - description: virtual timer irq reg: minItems: 1 - maxItems: 2 items: - description: 1st view base address - description: 2nd optional view base address diff --git a/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml b/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml index a8de99b0c0f9..f32575d4b5aa 100644 --- a/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml +++ b/Documentation/devicetree/bindings/timer/intel,ixp4xx-timer.yaml @@ -22,7 +22,6 @@ properties: interrupts: minItems: 1 - maxItems: 2 items: - description: Timer 1 interrupt - description: Timer 2 interrupt diff --git a/Documentation/devicetree/bindings/usb/maxim,max3420-udc.yaml b/Documentation/devicetree/bindings/usb/maxim,max3420-udc.yaml index 4241d38d5864..1d893d3d3432 100644 --- a/Documentation/devicetree/bindings/usb/maxim,max3420-udc.yaml +++ b/Documentation/devicetree/bindings/usb/maxim,max3420-udc.yaml @@ -30,14 +30,12 @@ properties: - description: usb irq from max3420 - description: vbus detection irq minItems: 1 - maxItems: 2 interrupt-names: items: - const: udc - const: vbus minItems: 1 - maxItems: 2 spi-max-frequency: maximum: 26000000 diff --git a/Documentation/devicetree/bindings/usb/nvidia,tegra-xudc.yaml b/Documentation/devicetree/bindings/usb/nvidia,tegra-xudc.yaml index e60e590dbe12..8428415896ce 100644 --- a/Documentation/devicetree/bindings/usb/nvidia,tegra-xudc.yaml +++ b/Documentation/devicetree/bindings/usb/nvidia,tegra-xudc.yaml @@ -25,7 +25,6 @@ properties: reg: minItems: 2 - maxItems: 3 items: - description: XUSB device controller registers - description: XUSB device PCI Config registers @@ -33,7 +32,6 @@ properties: reg-names: minItems: 2 - maxItems: 3 items: - const: base - const: fpci @@ -45,7 +43,6 @@ properties: clocks: minItems: 4 - maxItems: 5 items: - description: Clock to enable core XUSB dev clock. - description: Clock to enable XUSB super speed clock. @@ -55,7 +52,6 @@ properties: clock-names: minItems: 4 - maxItems: 5 items: - const: dev - const: ss diff --git a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml index e67223d90bb7..ad73339ffe1d 100644 --- a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml +++ b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml @@ -53,7 +53,6 @@ properties: clocks: minItems: 1 - maxItems: 3 items: - description: USB 2.0 host - description: USB 2.0 peripheral @@ -86,7 +85,6 @@ properties: dma-names: minItems: 2 - maxItems: 4 items: - const: ch0 - const: ch1 @@ -100,7 +98,6 @@ properties: resets: minItems: 1 - maxItems: 2 items: - description: USB 2.0 host - description: USB 2.0 peripheral diff --git a/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml b/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml index 3f1ba1d6c6b5..481bf91f988a 100644 --- a/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml +++ b/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml @@ -27,7 +27,6 @@ properties: - description: Low speed clock - description: Optional peripheral clock minItems: 1 - maxItems: 2 clock-names: items: -- 2.27.0

13 14

[PATCH] ALSA: usb-audio: Fix OOB access at proc output
by Takashi Iwai 22 Jun '21

22 Jun '21

At extending the available mixer values for 32bit types, we forgot to add the corresponding entries for the format dump in the proc output. This may result in OOB access. Here adds the missing entries. Fixes: bc18e31c3042 ("ALSA: usb-audio: Fix parameter block size for UAC2 control requests") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 428d581f988f..4ea4875abdf8 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3294,8 +3294,9 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, struct usb_mixer_elem_list *list) { struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); - static const char * const val_types[] = {"BOOLEAN", "INV_BOOLEAN", - "S8", "U8", "S16", "U16"}; + static const char * const val_types[] = { + "BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16", "S32", "U32", + }; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " "channels=%i, type=\"%s\"\n", cval->head.id, cval->control, cval->cmask, cval->channels, -- 2.26.2

1 0

[PATCH 00/31] Refactor Scarlett Gen 2 support and add Scarlett Gen 3 support
by Geoffrey D. Bennett 22 Jun '21

22 Jun '21

This patch set broadly: - Refactors the Scarlett Gen 2 support to make the Gen 3 mixer support trivial to add - Fixes a couple of minor issues with the Gen 2 support (low priority for stable; the issues were not reported by any user) - Adds support for Gen 3 devices with and without a mixer - Adds support for the major features new with the Gen 3 devices Geoffrey D. Bennett (31): ALSA: usb-audio: scarlett2: Add usb_tx/rx functions ALSA: usb-audio: scarlett2: Update initialisation sequence ALSA: usb-audio: scarlett2: Fix 6i6 Gen 2 line out descriptions ALSA: usb-audio: scarlett2: Always enable interrupt polling ALSA: usb-audio: scarlett2: Add "Sync Status" control ALSA: usb-audio: scarlett2: Merge common line in capture strings ALSA: usb-audio: scarlett2: Reformat scarlett2_config_items[] ALSA: usb-audio: scarlett2: Improve device info lookup ALSA: usb-audio: scarlett2: Move info lookup out of init function ALSA: usb-audio: scarlett2: Remove repeated device info comments ALSA: usb-audio: scarlett2: Add scarlett2_vol_ctl_write() helper ALSA: usb-audio: scarlett2: Add mute support ALSA: usb-audio: scarlett2: Allow arbitrary ordering of mux entries ALSA: usb-audio: scarlett2: Split struct scarlett2_ports ALSA: usb-audio: scarlett2: Fix Level Meter control ALSA: usb-audio: scarlett2: Add Gen 3 mixer support ALSA: usb-audio: scarlett2: Add support for "input-other" notify ALSA: usb-audio: scarlett2: Add Gen 3 MSD mode switch ALSA: usb-audio: scarlett2: Move get config above set config ALSA: usb-audio: scarlett2: Allow bit-level access to config ALSA: usb-audio: scarlett2: Add support for Solo and 2i2 Gen 3 ALSA: usb-audio: scarlett2: Add "air" switch support ALSA: usb-audio: scarlett2: Add phantom power switch support ALSA: usb-audio: scarlett2: Add direct monitor support ALSA: usb-audio: scarlett2: Label 18i8 Gen 3 line outputs correctly ALSA: usb-audio: scarlett2: Split up sw_hw_enum_ctl_put() ALSA: usb-audio: scarlett2: Add sw_hw_ctls and mux_ctls ALSA: usb-audio: scarlett2: Update mux controls to allow updates ALSA: usb-audio: scarlett2: Add speaker switching support ALSA: usb-audio: scarlett2: Update get_config to do endian conversion ALSA: usb-audio: scarlett2: Add support for the talkback feature sound/usb/mixer.c | 2 +- sound/usb/mixer_quirks.c | 6 + sound/usb/mixer_scarlett_gen2.c | 2725 +++++++++++++++++++++++++------ 3 files changed, 2239 insertions(+), 494 deletions(-) base-commit: 6c0a2078134aba6a77291554035304df9e16b85c -- 2.31.1

3 42

[RESEND PATCH v2] ASoC: fsl-asoc-card: change dev_err to dev_err_probe for defer probe
by Shengjiu Wang 22 Jun '21

22 Jun '21

Don't need to print error message for defer probe Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - use dev_err_probe instead of dev_dbg sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 121e08c2af2a..24c890d76da0 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -708,8 +708,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) of_node_put(framemaster); if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { - dev_err(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; + dev_err_probe(&pdev->dev, ret, "failed to find codec device\n"); goto asrc_fail; } -- 2.27.0

3 3

[PATCH -next 0/4] ASoC: samsung: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 21 Jun '21

21 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Yang Yingliang (4): ASoC: samsung: i2s: Use devm_platform_get_and_ioremap_resource() ASoC: samsung: pcm: Use devm_platform_get_and_ioremap_resource() ASoC: samsung: s3c2412-i2s: Use devm_platform_get_and_ioremap_resource() ASoC: samsung: s3c24xx-i2s: Use devm_platform_get_and_ioremap_resource() sound/soc/samsung/i2s.c | 3 +-- sound/soc/samsung/pcm.c | 3 +-- sound/soc/samsung/s3c2412-i2s.c | 3 +-- sound/soc/samsung/s3c24xx-i2s.c | 3 +-- 4 files changed, 4 insertions(+), 8 deletions(-) -- 2.25.1

3 9

[PATCH v2] ASoC: fsl_xcvr: disable all interrupts when suspend happens
by Shengjiu Wang 21 Jun '21

21 Jun '21

There is an unhandled interrupt after suspend, which cause endless interrupt when system resume, so system may hang. Disable all interrupts in runtime suspend callback to avoid above issue. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - Add Fixes tag - Add comments for the change sound/soc/fsl/fsl_xcvr.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index df7c189d97dd..92dd99258edf 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1233,6 +1233,16 @@ static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) struct fsl_xcvr *xcvr = dev_get_drvdata(dev); int ret; + /* + * Clear interrupts, when streams starts or resumes after + * suspend, interrupts are enabled in prepare(), so no need + * to enable interrupts in resume(). + */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, + FSL_XCVR_IRQ_EARC_ALL, 0); + if (ret < 0) + dev_err(dev, "Failed to clear IER0: %d\n", ret); + /* Assert M0+ reset */ ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, FSL_XCVR_EXT_CTRL_CORE_RESET, -- 2.27.0

3 2

[PATCH] MAINTAINERS: remove Timur Tabi from Freescale SOC sound drivers
by Timur Tabi 21 Jun '21

21 Jun '21

I haven't touched these drivers in seven years, and none of the patches sent to me these days affect code that I wrote. The other maintainers are doing a very good job without me. Signed-off-by: Timur Tabi <timur(a)kernel.org> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 8c5ee008301a..b3b9a253316f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7354,7 +7354,6 @@ F: drivers/net/ethernet/freescale/fs_enet/ F: include/linux/fs_enet_pd.h FREESCALE SOC SOUND DRIVERS -M: Timur Tabi <timur(a)kernel.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> R: Fabio Estevam <festevam(a)gmail.com> -- 2.25.1

3 2

[PATCH 0/5] ASoC/SoundWire: fix race condition on system suspend
by Pierre-Louis Bossart 21 Jun '21

21 Jun '21

This patchset fixes a race condition when a SoundWire codec signals an 'Alert' status during a system suspend, which can happen when e.g. a jack detection event is detected and handled in a workqueue scheduled after the link is suspended. When this happens in pm_runtime suspend, we already use pm_runtime_get_resume() to force the link to remain active, or resume immediately, but during a system suspend we don't have this option: we can't prevent the entire system from suspending just because of a jack detection or codec-specific event. This patch suggests instead disabling the implementation-defined or class-defined IRQs at the start of the suspend, along with an additional flag to protect against an interrupt thread/workqueue being scheduled after the interrupts are disabled. One could argue that the SoundWire core could have handled this case, which is a fair objection. I chose to keep the interrupt handling codec-specific since not all codecs can actually generate interrupts, and what they do during these interrupts is also quite custom - see e.g. the difference between the RT711 and RT711-SDCA versions. In the latter case the class-defined interrupts are cleared by the codec driver, not by the core. This patchset implies changes to both ASoC and SoundWire, with a minimal dependency on a sdw_update() helper and its _no_pm version, both of which should have been added a long time ago for read-modify-write operations. This race condition was present in all previous kernel versions but was only identified in late May in automated tests. We added Fixes tag to help linux-stable backports, but there's no real point in back-porting to earlier than 5.10 due to other missing dependencies in the upstream code. Pierre-Louis Bossart (5): soundwire: export sdw_update() and sdw_update_no_pm() ASoC: rt700-sdw: fix race condition on system suspend ASoC: rt711-sdw: fix race condition on system suspend ASoC: rt5682-sdw: fix race condition on system suspend ASoC: rt711-sdca-sdw: fix race condition on system suspend drivers/soundwire/bus.c | 17 +++++++++++- drivers/soundwire/bus.h | 13 --------- include/linux/soundwire/sdw.h | 3 ++ sound/soc/codecs/rt5682-sdw.c | 38 +++++++++++++++++++++++-- sound/soc/codecs/rt5682.h | 2 ++ sound/soc/codecs/rt700-sdw.c | 34 +++++++++++++++++++++-- sound/soc/codecs/rt700.c | 4 +++ sound/soc/codecs/rt700.h | 2 ++ sound/soc/codecs/rt711-sdca-sdw.c | 46 +++++++++++++++++++++++++++++-- sound/soc/codecs/rt711-sdca.c | 4 +++ sound/soc/codecs/rt711-sdca.h | 2 ++ sound/soc/codecs/rt711-sdw.c | 34 +++++++++++++++++++++-- sound/soc/codecs/rt711.c | 4 +++ sound/soc/codecs/rt711.h | 2 ++ 14 files changed, 183 insertions(+), 22 deletions(-) base-commit: bf28c803f2f4b14716df168e98f6ede4ba955866 -- 2.25.1

3 7

[PATCH v3 0/8] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 21 Jun '21

21 Jun '21

Hi Mark These are v3 of parsing for daifmt. I want to add new audio-graph-card2 sound card driver, and this is last part of necessary soc-core cleanup for it. Current some drivers are using DT, and then, snd_soc_of_parse_daifmt() parses daifmt, but bitclock/frame provider parsing part is one of headache, because we are assuming below both cases. A) node { bitclock-master; frame-master; ... }; B) link { bitclock-master = <&xxx>; frame-master = <&xxx>; ... }; The original was style A), and style B) was added later. snd_soc_of_parse_daifmt() parses A) style as original style, and user need to update to B) style for clock_provider part if needed. To handle it more flexibile, this patch-set adds new functions which separates snd_soc_of_parse_daifmt() helper function. snd_soc_daifmt_parse_format() : format part snd_soc_daifmt_parse_clock_provider_as_flag() : clock part for style A) snd_soc_daifmt_parse_clock_provider_as_phandl() : clock part for style B) snd_soc_daifmt_parse_clock_provider_as_bitmap() : clock part use with _from_bitmap v1 -> v2 - tidyup parse_clock_provider functions to _as_flag/phandle/bitmap() - don't exchange code style on each drivers. v2 -> v3 - use daifmt as much as possible (don't use daiclk) on each driver. Link: https://lore.kernel.org/r/875yypdxlm.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87czsvdc4o.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (8): ASoC: soc-core: add snd_soc_daifmt_clock_provider_from_bitmap() ASoC: soc-core: add snd_soc_daifmt_clock_provider_fliped() ASoC: soc-core: add snd_soc_daifmt_parse_format/clock_provider() ASoC: atmel: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: fsl: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: meson: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: simple-card-utils: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: soc-core: remove snd_soc_of_parse_daifmt() include/sound/soc.h | 21 ++++- sound/soc/atmel/mikroe-proto.c | 9 ++- sound/soc/fsl/fsl-asoc-card.c | 7 +- sound/soc/generic/simple-card-utils.c | 16 ++-- sound/soc/meson/meson-card-utils.c | 6 +- sound/soc/soc-core.c | 109 +++++++++++++++++--------- 6 files changed, 106 insertions(+), 62 deletions(-) -- 2.25.1

4 11

[PATCH] ASoC: tegra: Fix a NULL vs IS_ERR() check
by Dan Carpenter 21 Jun '21

21 Jun '21

The tegra_machine_parse_phandle() function doesn't return NULL, it returns error pointers. Fixes: cc8f70f56039 ("ASoC: tegra: Unify ASoC machine drivers") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/tegra/tegra_asoc_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index a53aec361a77..735909310a26 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -409,7 +409,7 @@ int tegra_asoc_machine_probe(struct platform_device *pdev) return PTR_ERR(np_codec); np_i2s = tegra_machine_parse_phandle(dev, "nvidia,i2s-controller"); - if (!np_i2s) + if (IS_ERR(np_i2s)) return PTR_ERR(np_i2s); card->dai_link->cpus->of_node = np_i2s; -- 2.30.2

3 2

[PATCH] ASoC: fsl_xcvr: disable all interrupts when suspend happens
by Shengjiu Wang 21 Jun '21

21 Jun '21

There is an unhandled interrupt after suspend, which cause endless interrupt when system resume, so system may hang. Disable all interrupts in runtime suspend callback to avoid above issue. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index df7c189d97dd..37b4fdb236ee 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1233,6 +1233,11 @@ static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) struct fsl_xcvr *xcvr = dev_get_drvdata(dev); int ret; + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, + FSL_XCVR_IRQ_EARC_ALL, 0); + if (ret < 0) + dev_err(dev, "Failed to clear IER0: %d\n", ret); + /* Assert M0+ reset */ ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, FSL_XCVR_EXT_CTRL_CORE_RESET, -- 2.27.0

4 4

[PATCH -next 1/2] ASoC: qcom: apq8016_sbc: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 21 Jun '21

21 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/qcom/apq8016_sbc.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 270986b2f102..08a05f0ecad7 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -134,7 +134,6 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct snd_soc_card *card; struct apq8016_sbc_data *data; - struct resource *res; int ret; data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); @@ -151,13 +150,11 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) if (ret) return ret; - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mic-iomux"); - data->mic_iomux = devm_ioremap_resource(dev, res); + data->mic_iomux = devm_platform_ioremap_resource_byname(pdev, "mic-iomux"); if (IS_ERR(data->mic_iomux)) return PTR_ERR(data->mic_iomux); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "spkr-iomux"); - data->spkr_iomux = devm_ioremap_resource(dev, res); + data->spkr_iomux = devm_platform_ioremap_resource_byname(pdev, "spkr-iomux"); if (IS_ERR(data->spkr_iomux)) return PTR_ERR(data->spkr_iomux); -- 2.25.1

2 2

[PATCH -next 0/3] ASoC: sunxi: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 21 Jun '21

21 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Yang Yingliang (3): ASoC: sunxi: sun4i-codec: Use devm_platform_get_and_ioremap_resource() ASoC: sun4i-i2s: Use devm_platform_get_and_ioremap_resource() ASoC: sunxi: sun4i-spdif: Use devm_platform_get_and_ioremap_resource() sound/soc/sunxi/sun4i-codec.c | 3 +-- sound/soc/sunxi/sun4i-i2s.c | 3 +-- sound/soc/sunxi/sun4i-spdif.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) -- 2.25.1

2 4

[PATCH 1/2] ASoC: atmel-i2s: Set symmetric sample bits
by Codrin Ciubotariu 21 Jun '21

21 Jun '21

The I2S needs to have the same sample bits for both capture and playback streams. Fixes: b543e467d1a9 ("ASoC: atmel-i2s: add driver for the new Atmel I2S controller") Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/atmel-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index e8fe1a7a4d83..a95002b0cc8e 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -542,6 +542,7 @@ static struct snd_soc_dai_driver atmel_i2s_dai = { }, .ops = &atmel_i2s_dai_ops, .symmetric_rate = 1, + .symmetric_sample_bits = 1, }; static const struct snd_soc_component_driver atmel_i2s_component = { -- 2.30.2

2 2

[PATCH -next v2 0/9] ASoC: fsl: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 21 Jun '21

21 Jun '21

patch #1 ~ #8: Use devm_platform_get_and_ioremap_resource() patch #9 check return value of platform_get_resource_byname() v2: change error message in patch #9 Yang Yingliang (9): ASoC: fsl_asrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_aud2htx: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_easrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_esai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_micfil: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_sai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_spdif: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_ssi: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_xcvr: check return value after calling platform_get_resource_byname() sound/soc/fsl/fsl_asrc.c | 3 +-- sound/soc/fsl/fsl_aud2htx.c | 3 +-- sound/soc/fsl/fsl_easrc.c | 3 +-- sound/soc/fsl/fsl_esai.c | 3 +-- sound/soc/fsl/fsl_micfil.c | 3 +-- sound/soc/fsl/fsl_sai.c | 3 +-- sound/soc/fsl/fsl_spdif.c | 3 +-- sound/soc/fsl/fsl_ssi.c | 3 +-- sound/soc/fsl/fsl_xcvr.c | 4 ++++ 9 files changed, 12 insertions(+), 16 deletions(-) -- 2.25.1

2 10

[PATCH 2/2] ASoC: rt711: add two jack detection modes
by shumingf＠realtek.com 21 Jun '21

21 Jun '21

From: Shuming Fan <shumingf(a)realtek.com> Some boards use different circuits for jack detection. This patch adds two modes as below 1. JD2/2 ports/external resister 100k 2. JD2/1 port/JD voltage 1.8V Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdw.h | 2 ++ sound/soc/codecs/rt711.c | 30 ++++++++++++++++++++++++++++++ sound/soc/codecs/rt711.h | 29 ++++++++++++++++++++++++++++- 3 files changed, 60 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.h b/sound/soc/codecs/rt711-sdw.h index 43b2b984b29c..6acf9858330d 100644 --- a/sound/soc/codecs/rt711-sdw.h +++ b/sound/soc/codecs/rt711-sdw.h @@ -267,7 +267,9 @@ static const struct reg_default rt711_reg_defaults[] = { { 0x8393, 0x00 }, { 0x7319, 0x00 }, { 0x8399, 0x00 }, + { 0x752008, 0xa807 }, { 0x752009, 0x1029 }, + { 0x75200b, 0x7770 }, { 0x752011, 0x007a }, { 0x75201a, 0x8003 }, { 0x752045, 0x5289 }, diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index c3a82b462a93..301c7ab27878 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -389,6 +389,36 @@ static void rt711_jack_init(struct rt711_priv *rt711) RT711_HP_JD_FINAL_RESULT_CTL_JD12, RT711_HP_JD_FINAL_RESULT_CTL_JD12); break; + case RT711_JD2_100K: + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL2, RT711_JD2_2PORT_100K_DECODE | RT711_JD2_1PORT_TYPE_DECODE | + RT711_HP_JD_SEL_JD2 | RT711_JD1_2PORT_TYPE_100K_DECODE, + RT711_JD2_2PORT_100K_DECODE_HP | RT711_JD2_1PORT_JD_HP | + RT711_HP_JD_SEL_JD2 | RT711_JD1_2PORT_JD_RESERVED); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_CC_DET1, + RT711_HP_JD_FINAL_RESULT_CTL_JD12, + RT711_HP_JD_FINAL_RESULT_CTL_JD12); + break; + case RT711_JD2_1P8V_1PORT: + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL1, RT711_JD2_DIGITAL_JD_MODE_SEL, + RT711_JD2_1_JD_MODE); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL2, RT711_JD2_1PORT_TYPE_DECODE | + RT711_HP_JD_SEL_JD2, + RT711_JD2_1PORT_JD_HP | + RT711_HP_JD_SEL_JD2); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_JD_CTL4, RT711_JD2_PAD_PULL_UP_MASK | + RT711_JD2_MODE_SEL_MASK, + RT711_JD2_PAD_PULL_UP | + RT711_JD2_MODE2_1P8V_1PORT); + rt711_index_update_bits(rt711->regmap, RT711_VENDOR_REG, + RT711_CC_DET1, + RT711_HP_JD_FINAL_RESULT_CTL_JD12, + RT711_HP_JD_FINAL_RESULT_CTL_JD12); + break; default: dev_warn(rt711->component->dev, "Wrong JD source\n"); break; diff --git a/sound/soc/codecs/rt711.h b/sound/soc/codecs/rt711.h index ca0f581feec7..5f2ba1341085 100644 --- a/sound/soc/codecs/rt711.h +++ b/sound/soc/codecs/rt711.h @@ -52,7 +52,9 @@ struct sdw_stream_data { /* Index (NID:20h) */ #define RT711_DAC_DC_CALI_CTL1 0x00 +#define RT711_JD_CTL1 0x08 #define RT711_JD_CTL2 0x09 +#define RT711_JD_CTL4 0x0b #define RT711_CC_DET1 0x11 #define RT711_PARA_VERB_CTL 0x1a #define RT711_COMBO_JACK_AUTO_CTL1 0x45 @@ -171,10 +173,33 @@ struct sdw_stream_data { /* DAC DC offset calibration control-1 (0x00)(NID:20h) */ #define RT711_DAC_DC_CALI_TRIGGER (0x1 << 15) +/* jack detect control 1 (0x08)(NID:20h) */ +#define RT711_JD2_DIGITAL_JD_MODE_SEL (0x1 << 1) +#define RT711_JD2_1_JD_MODE (0x0 << 1) +#define RT711_JD2_2_JD_MODE (0x1 << 1) + /* jack detect control 2 (0x09)(NID:20h) */ #define RT711_JD2_2PORT_200K_DECODE_HP (0x1 << 13) +#define RT711_JD2_2PORT_100K_DECODE (0x1 << 12) +#define RT711_JD2_2PORT_100K_DECODE_HP (0x0 << 12) #define RT711_HP_JD_SEL_JD1 (0x0 << 1) #define RT711_HP_JD_SEL_JD2 (0x1 << 1) +#define RT711_JD2_1PORT_TYPE_DECODE (0x3 << 10) +#define RT711_JD2_1PORT_JD_LINE2 (0x0 << 10) +#define RT711_JD2_1PORT_JD_HP (0x1 << 10) +#define RT711_JD2_1PORT_JD_LINE1 (0x2 << 10) +#define RT711_JD1_2PORT_TYPE_100K_DECODE (0x1 << 0) +#define RT711_JD1_2PORT_JD_RESERVED (0x0 << 0) +#define RT711_JD1_2PORT_JD_LINE1 (0x1 << 0) + +/* jack detect control 4 (0x0b)(NID:20h) */ +#define RT711_JD2_PAD_PULL_UP_MASK (0x1 << 3) +#define RT711_JD2_PAD_NOT_PULL_UP (0x0 << 3) +#define RT711_JD2_PAD_PULL_UP (0x1 << 3) +#define RT711_JD2_MODE_SEL_MASK (0x3 << 0) +#define RT711_JD2_MODE0_2PORT (0x0 << 0) +#define RT711_JD2_MODE1_3P3V_1PORT (0x1 << 0) +#define RT711_JD2_MODE2_1P8V_1PORT (0x2 << 0) /* CC DET1 (0x11)(NID:20h) */ #define RT711_HP_JD_FINAL_RESULT_CTL_JD12 (0x1 << 10) @@ -215,7 +240,9 @@ enum { enum rt711_jd_src { RT711_JD_NULL, RT711_JD1, - RT711_JD2 + RT711_JD2, + RT711_JD2_100K, + RT711_JD2_1P8V_1PORT }; int rt711_io_init(struct device *dev, struct sdw_slave *slave); -- 2.31.1

2 1

[PATCH V2 00/14] Cleanup before adding Scarlett Gen 3 support
by Geoffrey D. Bennett 21 Jun '21

21 Jun '21

Hi Takashi, Here is version 2 of a set of patches which is some cleanup of the Scarlett Gen 2 mixer driver in preparation for adding Scarlett Gen 3 support. One review comment I got (from Hin-Tak) was: > 40+ patches is a lot, for modifying just one file. I would collapse > it all into one and break it up again to under 10, maybe, broadly > into "functionally-equivalent re-org", "small isolated bug fixes", > "additional functions, not yet used", "hooking up those new > functions", etc? I'm not sure that I agree with that comment -- I tried to follow the Documentation/process/submitting-patches.rst advice of "Separate each logical change into a separate patch" for easy review of the individual pieces, but perhaps I went too far in that direction? Please let me know if I should combine some of these patches together. After this set of patches I have a second set of patches almost ready. They add a few features to the Gen 2 support and update the code ready to add Gen 3 support: ALSA: usb-audio: scarlett2: Add usb_tx/rx functions ALSA: usb-audio: scarlett2: Update initialisation sequence ALSA: usb-audio: scarlett2: Fix 6i6 Gen 2 line out descriptions ALSA: usb-audio: scarlett2: Always enable interrupt polling ALSA: usb-audio: scarlett2: Add "Sync Status" control ALSA: usb-audio: scarlett2: Merge common line in capture strings ALSA: usb-audio: scarlett2: Reformat scarlett2_config_items[] ALSA: usb-audio: scarlett2: Improve device info lookup ALSA: usb-audio: scarlett2: Move info lookup out of init function ALSA: usb-audio: scarlett2: Remove repeated device info comments ALSA: usb-audio: scarlett2: Add scarlett2_vol_ctl_write() helper ALSA: usb-audio: scarlett2: Add mute support ALSA: usb-audio: scarlett2: Allow arbitrary ordering of mux entries ALSA: usb-audio: scarlett2: Split struct scarlett2_ports ALSA: usb-audio: scarlett2: Fix Level Meter control The final set of patches adds Gen 3 mixer support and then adds support for the new Gen 3 features: ALSA: usb-audio: scarlett2: Add Gen 3 mixer support ALSA: usb-audio: scarlett2: Add support for "input-other" notify ALSA: usb-audio: scarlett2: Add Gen 3 MSD mode switch ALSA: usb-audio: scarlett2: Move get config above set config ALSA: usb-audio: scarlett2: Allow bit-level access to config ALSA: usb-audio: scarlett2: Add support for Solo and 2i2 Gen 3 ALSA: usb-audio: scarlett2: Add "air" switch support ALSA: usb-audio: scarlett2: Add phantom power switch support ALSA: usb-audio: scarlett2: Add direct monitor support ALSA: usb-audio: scarlett2: Label 18i8 Gen 3 line outputs correctly ALSA: usb-audio: scarlett2: Split up sw_hw_enum_ctl_put() ALSA: usb-audio: scarlett2: Add sw_hw_ctls and mux_ctls ALSA: usb-audio: scarlett2: Update mux controls to allow updates ALSA: usb-audio: scarlett2: Add speaker switching support ALSA: usb-audio: scarlett2: Update get_config to do endian conversion ALSA: usb-audio: scarlett2: Add support for the talkback feature If you'd like to see the commits so far all at once, please check this branch: https://github.com/geoffreybennett/scarlett-gen2/commits/scarlett-gen3-for-… Thanks, Geoffrey. Geoffrey D. Bennett (14): ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment ALSA: usb-audio: scarlett2: Fix 18i8 Gen 2 PCM Input count ALSA: usb-audio: scarlett2: Coding style improvements ALSA: usb-audio: scarlett2: Remove unused/useless code ALSA: usb-audio: scarlett2: Remove interrupt debug message ALSA: usb-audio: scarlett2: Remove redundant info->button_count ALSA: usb-audio: scarlett2: Rename buttons/interrupts/vol ALSA: usb-audio: scarlett2: Rename struct scarlett2_mixer_data ALSA: usb-audio: scarlett2: Add temp variable for consistency ALSA: usb-audio: scarlett2: Fix data_mutex lock ALSA: usb-audio: scarlett2: Fix scarlett2_*_ctl_put() return values ALSA: usb-audio: scarlett2: Fix union usage in mixer control callbacks ALSA: usb-audio: scarlett2: Don't copy struct scarlett2_config ALSA: usb-audio: scarlett2: Remove hard-coded USB #defines sound/usb/mixer_scarlett_gen2.c | 425 +++++++++++++++++--------------- 1 file changed, 224 insertions(+), 201 deletions(-) base-commit: f8fbcdfb0665de60997d9746809e1704ed782bbc -- 2.31.1

2 3

[PATCH 1/2] control: Add documentation for snd_ctl_card_* and friends.
by Tanjeff-N. Moos 21 Jun '21

21 Jun '21

I sent this patch before, but it didn't make it into the Git repo. If I could improve something, please tell me. In this patch series, I added a description about control interface handling and how control interfaces are identified. In addition, I added/improved Doxygen documentation for the snd_ctl_card_info_t type and related corresponding functions, e.g. snd_ctl_card_info(). I also documented other card-related like snd_card_next(). Along the way I did minor documentation improvements. Signed-off-by: Tanjeff-N. Moos <tanjeff(a)cccmz.de> --- include/control.h | 40 +++++++++- src/control/cards.c | 56 +++++++++----- src/control/control.c | 175 +++++++++++++++++++++++++++++++----------- 3 files changed, 202 insertions(+), 69 deletions(-) diff --git a/include/control.h b/include/control.h index ccf906e2..9ebb4905 100644 --- a/include/control.h +++ b/include/control.h @@ -50,7 +50,32 @@ typedef struct snd_aes_iec958 { unsigned char dig_subframe[4]; /**< AES/IEC958 subframe bits */ } snd_aes_iec958_t; -/** CTL card info container */ +/** \brief CTL card info container. + * + * This type contains meta information about a sound card, such as the index, + * name, longname, etc. + * + * \par Memory management + * + * Before using a snd_ctl_card_info_t object, it must be allocated using + * snd_ctl_card_info_alloca() or snd_ctl_card_info_malloc(). When using the + * latter, it must be freed again using snd_ctl_card_info_free(). + * + * A card info object can be zeroed out using snd_ctl_card_info_clear(). + * + * A card info object can be copied to another one using + * snd_ctl_card_info_copy(). + * + * \par Obtaining the Information + * + * To obtain the card information, it must first be opened using + * snd_ctl_open(), and a snd_ctl_card_info_t container must be + * allocated. Then, the information can be read using + * snd_ctl_card_info_get_card(). + * + * Thereafter, the card properties can be read using the + * snd_ctl_card_info_get_*() functions. + */ typedef struct _snd_ctl_card_info snd_ctl_card_info_t; /** CTL element identifier container */ @@ -442,11 +467,20 @@ void snd_ctl_elem_id_set_name(snd_ctl_elem_id_t *obj, const char *val); void snd_ctl_elem_id_set_index(snd_ctl_elem_id_t *obj, unsigned int val); size_t snd_ctl_card_info_sizeof(void); + /** \hideinitializer - * \brief allocate an invalid #snd_ctl_card_info_t using standard alloca - * \param ptr returned pointer + * \brief Allocate an invalid #snd_ctl_card_info_t on the stack. + * + * Allocate space for a card info object on the stack. The allocated + * memory need not be freed, because it is on the stack. + * + * See snd_ctl_card_info_t for details. + * + * \param ptr Pointer to a snd_ctl_elem_value_t pointer. The address + * of the allocated space will returned here. */ #define snd_ctl_card_info_alloca(ptr) __snd_alloca(ptr, snd_ctl_card_info) + int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr); void snd_ctl_card_info_free(snd_ctl_card_info_t *obj); void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj); diff --git a/src/control/cards.c b/src/control/cards.c index e57457c2..8226c42d 100644 --- a/src/control/cards.c +++ b/src/control/cards.c @@ -77,8 +77,8 @@ static int snd_card_load1(int card) /** * \brief Try to load the driver for a card. - * \param card Card number. - * \return 1 if driver is present, zero if driver is not present + * \param card Card index. + * \return 1 if driver is present, zero if driver is not present. */ int snd_card_load(int card) { @@ -86,14 +86,24 @@ int snd_card_load(int card) } /** - * \brief Try to determine the next card. - * \param rcard pointer to card number - * \result zero if success, otherwise a negative error code + * \brief Iterate over physical sound cards. + * + * This function takes the index of a physical sound card and sets it to the + * index of the next card. If index is -1, it is set to the index of the first + * card. After the last card, the index is set to -1. + * + * For example, if you have 2 sound cards (with index 0 and 1), the index will + * be modified as follows: + * + * - -1 --> 0 + * - 0 --> 1 + * - 1 --> -1 + * + * This does not work for virtual sound cards. * - * Tries to determine the next card from given card number. - * If card number is -1, then the first available card is - * returned. If the result card number is -1, no more cards - * are available. + * \param rcard Index of current card. The index of the next card is stored + * here. + * \result zero if success, otherwise a negative error code. */ int snd_card_next(int *rcard) { @@ -114,13 +124,18 @@ int snd_card_next(int *rcard) } /** - * \brief Convert card string to an integer value. - * \param string String containing card identifier - * \return zero if success, otherwise a negative error code + * \brief Convert a card string to the card index. * - * The accepted format is an integer value in ASCII representation - * or the card identifier (the id parameter for sound-card drivers). - * The control device name like /dev/snd/controlC0 is accepted, too. + * This works only for physical sound cards, not for virtual cards. + * + * \param string A string identifying the card. + * \return The index of the card. On error, a a negative error code + * is returned. + * + * The accepted formats for "string" are: + * - The index of the card (as listed in /proc/asound/cards), given as string + * - The ID of the card (as listed in /proc/asound/cards) + * - The control device name (like /dev/snd/controlC0) */ int snd_card_get_index(const char *string) { @@ -163,8 +178,9 @@ int snd_card_get_index(const char *string) /** * \brief Obtain the card name. - * \param card Card number - * \param name Result - card name corresponding to card number + * + * \param card The index of the card. + * \param name Result - card name corresponding to card index. * \result zero if success, otherwise a negative error code * * The value returned in name is allocated with strdup and should be @@ -193,9 +209,9 @@ int snd_card_get_name(int card, char **name) /** * \brief Obtain the card long name. - * \param card Card number - * \param name Result - card long name corresponding to card number - * \result zero if success, otherwise a negative error code + * \param card Index of the card. + * \param name Result - card long name corresponding to card index. + * \result Zero if success, otherwise a negative error code. * * The value returned in name is allocated with strdup and should be * freed when no longer used. diff --git a/src/control/control.c b/src/control/control.c index ed986e54..7cf4decb 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -33,8 +33,50 @@ <P>Control interface is designed to access primitive controls. There is also an interface for notifying about control and structure changes. + \section control_general_overview General overview +In Alsa, there are physical sound cards, such as USB headsets, and +virtual sound cards, such as "pulse", which represents the PulseAudio +Sound system. Each sound card offers a control interface, making its +settings (e.g. volume knobs) available. The complete list of available +control interfaces can be obtained using snd_device_name_hint(), +giving -1 as card index and "ctl" as interface type. Each returned +NAME hint identifies a control interface. + +Sound cards have an ID (a string), an index (an int, sometimes called +the "card number"), a name, a longname, a mixername and a "components" +property. The file /proc/asound/cards lists most of these properties +for physical sound cards. Virtual sound cards are not listed in that +file. The format is: + +\verbatim +index [ID ] Driver - name + longname +\endverbatim + +Note that the mixername and components are not listed. + + +\subsection control_cards_id Identifying and Opening Control Interfaces + +To work with a control interface, is must be opened first, using +snd_ctl_open(). This function takes the interface name. + +For physical sound cards, the control interface can be identified +using the string "hw:<index>" (e.g. `hw:2`). The NAME hint - which is +"hw:CARD=<ID>" - can also be used. Further, its device file (something +like `/dev/snd/controlC0`) is also acceptable. Either of them can be +given to snd_ctl_open(). + +For virtual sound cards, the NAME hint is given to snd_ctl_open(). + +The functions snd_card_get_index(), snd_card_get_name() and +snd_card_get_longname() can be used to find an identifying property +when another one is already known. + +\section control_elements Elements + In ALSA control feature, each sound card can have control elements. The elements are managed according to below model. @@ -65,7 +107,7 @@ are managed according to below model. of userspace applications and drivers in kernel. -\section identifying_elements Identifying Elements +\subsection identifying_elements Identifying Elements Each element has the following identifying properties: @@ -84,7 +126,7 @@ but in practice this is rare). The numid can change on each boot. In case of an USB sound card, the numid can also change when it is reconnected. The short numid is used to reduce the lookup time. -\section element_lists Element Lists +\subsection element_lists Element Lists An element list can be used to obtain a list of all elements of the sound card. The list contains generic information (e.g. how many @@ -93,7 +135,7 @@ elements the card has), and the identifying properties of the elements element lists. -\section working_with_elements Working with Elements +\subsection working_with_elements Working with Elements It is possible to obtain information about an element using the snd_ctl_elem_info_*() functions. For enums, the allowed values can be @@ -108,7 +150,7 @@ actual values or settings. It is also possible to get and set the ID values (such as the numid or the name). -\section element_sets Element Sets +\subsection element_sets Element Sets The type of element set is one of integer, integer64, boolean, enumerators, bytes and IEC958 structure. This indicates the type of value for each member in @@ -329,10 +371,15 @@ int snd_ctl_subscribe_events(snd_ctl_t *ctl, int subscribe) /** - * \brief Get card related information - * \param ctl CTL handle - * \param info Card info pointer - * \return 0 on success otherwise a negative error code + * \brief Get information about the sound card. + * + * Obtain information about the sound card previously opened using + * snd_ctl_open(). The object "info" must be allocated prior to calling this + * function. See snd_ctl_card_info_t for details. + * + * \param ctl The CTL handle. + * \param info The card information is stored here. + * \return 0 on success, otherwise a negative error code. */ int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info) { @@ -1508,11 +1555,13 @@ int _snd_ctl_open_named_child(snd_ctl_t **pctl, const char *name, #endif /** - * \brief Opens a CTL - * \param ctlp Returned CTL handle - * \param name ASCII identifier of the CTL handle - * \param mode Open mode (see #SND_CTL_NONBLOCK, #SND_CTL_ASYNC) - * \return 0 on success otherwise a negative error code + * \brief Opens a sound card. + * + * \param ctlp Returned CTL handle. + * \param name A string identifying the card (See \ref control_cards_id). + * \param mode Open mode (see #SND_CTL_NONBLOCK, #SND_CTL_ASYNC). + * + * \return 0 on success otherwise a negative error code. */ int snd_ctl_open(snd_ctl_t **ctlp, const char *name, int mode) { @@ -2027,8 +2076,8 @@ void snd_ctl_elem_id_set_index(snd_ctl_elem_id_t *obj, unsigned int val) } /** - * \brief get size of #snd_ctl_card_info_t - * \return size in bytes + * \brief get size of #snd_ctl_card_info_t. + * \return Size in bytes. */ size_t snd_ctl_card_info_sizeof() { @@ -2036,9 +2085,16 @@ size_t snd_ctl_card_info_sizeof() } /** - * \brief allocate an invalid #snd_ctl_card_info_t using standard malloc - * \param ptr returned pointer - * \return 0 on success otherwise negative error code + * \brief Allocate an invalid #snd_ctl_card_info_t on the heap. + * + * Allocate space for a card info object on the heap. The allocated memory + * must be freed using snd_ctl_card_info_free(). + * + * See snd_ctl_card_info_t for details. + * + * \param ptr Pointer to a snd_ctl_card_info_t pointer. The address + * of the allocated space will be returned here. + * \return 0 on success, otherwise a negative error code. */ int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr) { @@ -2050,8 +2106,10 @@ int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr) } /** - * \brief frees a previously allocated #snd_ctl_card_info_t - * \param obj pointer to object to free + * \brief Free an #snd_ctl_card_info_t previously allocated using + * snd_ctl_card_info_malloc(). + * + * \param obj Pointer to the snd_ctl_card_info_t. */ void snd_ctl_card_info_free(snd_ctl_card_info_t *obj) { @@ -2059,8 +2117,11 @@ void snd_ctl_card_info_free(snd_ctl_card_info_t *obj) } /** - * \brief clear given #snd_ctl_card_info_t object - * \param obj pointer to object to clear + * \brief Clear given card info object. + * + * See snd_ctl_elem_value_t for details. + * + * \param obj Card info object. */ void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj) { @@ -2068,9 +2129,10 @@ void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj) } /** - * \brief copy one #snd_ctl_card_info_t to another - * \param dst pointer to destination - * \param src pointer to source + * \brief Bitwise copy of a #snd_ctl_card_info_t object. + * + * \param dst Pointer to destination. + * \param src Pointer to source. */ void snd_ctl_card_info_copy(snd_ctl_card_info_t *dst, const snd_ctl_card_info_t *src) { @@ -2079,9 +2141,12 @@ void snd_ctl_card_info_copy(snd_ctl_card_info_t *dst, const snd_ctl_card_info_t } /** - * \brief Get card number from a CTL card info - * \param obj CTL card info - * \return card number + * \brief Get the sound card index from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card index. */ int snd_ctl_card_info_get_card(const snd_ctl_card_info_t *obj) { @@ -2090,9 +2155,12 @@ int snd_ctl_card_info_get_card(const snd_ctl_card_info_t *obj) } /** - * \brief Get card identifier from a CTL card info - * \param obj CTL card info - * \return card identifier + * \brief Get the sound card ID from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card ID. */ const char *snd_ctl_card_info_get_id(const snd_ctl_card_info_t *obj) { @@ -2101,9 +2169,12 @@ const char *snd_ctl_card_info_get_id(const snd_ctl_card_info_t *obj) } /** - * \brief Get card driver name from a CTL card info - * \param obj CTL card info - * \return card driver name + * \brief Get the sound card driver from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return The sound card driver. */ const char *snd_ctl_card_info_get_driver(const snd_ctl_card_info_t *obj) { @@ -2112,9 +2183,12 @@ const char *snd_ctl_card_info_get_driver(const snd_ctl_card_info_t *obj) } /** - * \brief Get card name from a CTL card info - * \param obj CTL card info - * \return card name + * \brief Get the sound card name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card name. */ const char *snd_ctl_card_info_get_name(const snd_ctl_card_info_t *obj) { @@ -2123,9 +2197,12 @@ const char *snd_ctl_card_info_get_name(const snd_ctl_card_info_t *obj) } /** - * \brief Get card long name from a CTL card info - * \param obj CTL card info - * \return card long name + * \brief Get the sound cards long name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound cards long name. */ const char *snd_ctl_card_info_get_longname(const snd_ctl_card_info_t *obj) { @@ -2134,9 +2211,12 @@ const char *snd_ctl_card_info_get_longname(const snd_ctl_card_info_t *obj) } /** - * \brief Get card mixer name from a CTL card info - * \param obj CTL card info - * \return card mixer name + * \brief Get the sound card mixer name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card mixer name. */ const char *snd_ctl_card_info_get_mixername(const snd_ctl_card_info_t *obj) { @@ -2145,9 +2225,12 @@ const char *snd_ctl_card_info_get_mixername(const snd_ctl_card_info_t *obj) } /** - * \brief Get card component list from a CTL card info - * \param obj CTL card info - * \return card mixer identifier + * \brief Get the sound cards "components" property from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound cards "components" property. */ const char *snd_ctl_card_info_get_components(const snd_ctl_card_info_t *obj) { -- 2.25.1

2 3

A52 documentation in need of updates.
by GitHub issues - opened 21 Jun '21

21 Jun '21

alsa-project/alsa-plugins issue #29 was opened from quequotion: [a52.txt](https://github.com/alsa-project/alsa-plugins/blob/4219f4b3fded74d7… hasn't been changed in fifteen years. In that time, [a preconfiguration has been supplied that ends up getting merged into a dysfunctional mess if users add a manual configuration to their asound.conf without prepending an exclamation point to override it](https://github.com/alsa-project/alsa-lib/issues/154#issuecomment-859799…, and more recently [more sample formats have been added](https://github.com/alsa-project/alsa-plugins/commit/f11e7a8994111f97…. This file needs some attention. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/29 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v3] ASoC: qcom: Fix for DMA interrupt clear reg overwriting
by Srinivasa Rao Mandadapu 21 Jun '21

21 Jun '21

The DMA interrupt clear register overwritten during simultaneous playback and capture in lpass platform interrupt handler. It's causing playback or capture stuck in similtaneous plaback on speaker and capture on dmic test. Update appropriate reg fields of corresponding channel instead of entire register write. Fixes: commit c5c8635a04711 ("ASoC: qcom: Add LPASS platform driver") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v2: -- Removed redundant variables. Changes since v1: -- Subject lines changed. sound/soc/qcom/lpass-platform.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 0df9481ea4c6..f9df76d37858 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -526,7 +526,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_write(map, reg_irqclr, val_irqclr); + ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -650,10 +650,11 @@ static irqreturn_t lpass_dma_interrupt_handler( struct lpass_variant *v = drvdata->variant; irqreturn_t ret = IRQ_NONE; int rv; - unsigned int reg = 0, val = 0; + unsigned int reg, val, mask; struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; + mask = LPAIF_IRQ_ALL(chan); switch (dai_id) { case LPASS_DP_RX: map = drvdata->hdmiif_map; @@ -676,8 +677,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - - rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); + rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_PER(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +688,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_XRUN(chan) | val); + rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_XRUN(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -700,7 +700,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_ERR(chan) | val); + rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_ERR(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 6

Lenovo Legion ALC287/ALC3306 issue
by GitHub issues - opened 21 Jun '21

21 Jun '21

alsa-project/alsa-lib issue #160 was opened from camberkenpas: Hello, I was able to do a little digging into this issue by running a Windows 10 virtual machine and passing the sound card device to the virtual machine I was able to get audio to play through the laptop's virtual machine using the virtual machine. More importantly, I was able to capture traces that might help us determine what this sound card needs. I've attached traces at the ticket for this bug: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Perhaps this is enough to deduce at least some of what verbs the codec wants in order to work. Is this the right place to share this? Or is the kernel bug tracker the only appropriate place for this? Issue URL : https://github.com/alsa-project/alsa-lib/issues/160 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 14/14] ALSA: usb-audio: scarlett2: Remove hard-coded USB #defines
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Remove the hard-coded interface number and related constants for the vendor-specific interface and look them up from the USB endpoint descriptor. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 74 ++++++++++++++++++++++++--------- 1 file changed, 55 insertions(+), 19 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 45fd540920b9..2e1937b072ee 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -230,6 +230,10 @@ struct scarlett2_data { struct mutex data_mutex; /* lock access to this data */ struct delayed_work work; const struct scarlett2_device_info *info; + __u8 bInterfaceNumber; + __u8 bEndpointAddress; + __u16 wMaxPacketSize; + __u8 bInterval; int num_mux_srcs; int num_mux_dsts; u16 scarlett2_seq; @@ -444,12 +448,6 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, /*** USB Interactions ***/ -/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ -#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 -#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 -#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 -#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 - /* Interrupt flags for dim/mute button and monitor changes */ #define SCARLETT2_USB_NOTIFY_DIM_MUTE 0x00200000 #define SCARLETT2_USB_NOTIFY_MONITOR 0x00400000 @@ -615,7 +613,7 @@ static int scarlett2_usb( SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, 0, - SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + private->bInterfaceNumber, req, req_buf_size); @@ -635,7 +633,7 @@ static int scarlett2_usb( SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 0, - SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + private->bInterfaceNumber, resp, resp_buf_size); @@ -1886,12 +1884,45 @@ static void scarlett2_count_mux_io(struct scarlett2_data *private) private->num_mux_dsts = dsts; } -/* Initialise private data and sequence number */ +/* Look through the interface descriptors for the Focusrite Control + * interface (bInterfaceClass = 255 Vendor Specific Class) and set + * bInterfaceNumber, bEndpointAddress, wMaxPacketSize, and bInterval + * in private + */ +static int scarlett2_find_fc_interface(struct usb_device *dev, + struct scarlett2_data *private) +{ + struct usb_host_config *config = dev->actconfig; + int i; + + for (i = 0; i < config->desc.bNumInterfaces; i++) { + struct usb_interface *intf = config->interface[i]; + struct usb_interface_descriptor *desc = + &intf->altsetting[0].desc; + struct usb_endpoint_descriptor *epd; + + if (desc->bInterfaceClass != 255) + continue; + + epd = get_endpoint(intf->altsetting, 0); + private->bInterfaceNumber = desc->bInterfaceNumber; + private->bEndpointAddress = epd->bEndpointAddress & + USB_ENDPOINT_NUMBER_MASK; + private->wMaxPacketSize = le16_to_cpu(epd->wMaxPacketSize); + private->bInterval = epd->bInterval; + return 0; + } + + return -EINVAL; +} + +/* Initialise private data, sequence number, and get the USB data */ static int scarlett2_init_private(struct usb_mixer_interface *mixer, const struct scarlett2_device_info *info) { struct scarlett2_data *private = kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL); + int err; if (!private) return -ENOMEM; @@ -1899,13 +1930,19 @@ static int scarlett2_init_private(struct usb_mixer_interface *mixer, mutex_init(&private->usb_mutex); mutex_init(&private->data_mutex); INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work); + + mixer->private_data = private; + mixer->private_free = scarlett2_private_free; + mixer->private_suspend = scarlett2_private_suspend; + private->info = info; scarlett2_count_mux_io(private); private->scarlett2_seq = 0; private->mixer = mixer; - mixer->private_data = private; - mixer->private_free = scarlett2_private_free; - mixer->private_suspend = scarlett2_private_suspend; + + err = scarlett2_find_fc_interface(mixer->chip->dev, private); + if (err < 0) + return err; /* Initialise the sequence number used for the proprietary commands */ return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, NULL, 0, NULL, 0); @@ -2050,8 +2087,8 @@ static void scarlett2_notify(struct urb *urb) static int scarlett2_init_notify(struct usb_mixer_interface *mixer) { struct usb_device *dev = mixer->chip->dev; - unsigned int pipe = usb_rcvintpipe(dev, - SCARLETT2_USB_INTERRUPT_ENDPOINT); + struct scarlett2_data *private = mixer->private_data; + unsigned int pipe = usb_rcvintpipe(dev, private->bEndpointAddress); void *transfer_buffer; if (mixer->urb) { @@ -2067,14 +2104,13 @@ static int scarlett2_init_notify(struct usb_mixer_interface *mixer) if (!mixer->urb) return -ENOMEM; - transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + transfer_buffer = kmalloc(private->wMaxPacketSize, GFP_KERNEL); if (!transfer_buffer) return -ENOMEM; usb_fill_int_urb(mixer->urb, dev, pipe, - transfer_buffer, SCARLETT2_USB_INTERRUPT_MAX_DATA, - scarlett2_notify, mixer, - SCARLETT2_USB_INTERRUPT_INTERVAL); + transfer_buffer, private->wMaxPacketSize, + scarlett2_notify, mixer, private->bInterval); return usb_submit_urb(mixer->urb, GFP_KERNEL); } @@ -2084,7 +2120,7 @@ static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, { int err; - /* Initialise private data and sequence number */ + /* Initialise private data, sequence number, and get the USB data */ err = scarlett2_init_private(mixer, info); if (err < 0) return err; -- 2.31.1

1 0

[PATCH 13/14] ALSA: usb-audio: scarlett2: Don't copy struct scarlett2_config
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

scarlett2_usb_set_config() and scarlett2_usb_get_config() were copying struct scarlett2_config. Use a pointer instead. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 49c4662531a3..45fd540920b9 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -706,8 +706,8 @@ static int scarlett2_usb_set_config( struct usb_mixer_interface *mixer, int config_item_num, int index, int value) { - const struct scarlett2_config config_item = - scarlett2_config_items[config_item_num]; + const struct scarlett2_config *config_item = + &scarlett2_config_items[config_item_num]; struct { __le32 offset; __le32 bytes; @@ -721,17 +721,17 @@ static int scarlett2_usb_set_config( cancel_delayed_work_sync(&private->work); /* Send the configuration parameter data */ - req.offset = cpu_to_le32(config_item.offset + index * config_item.size); - req.bytes = cpu_to_le32(config_item.size); + req.offset = cpu_to_le32(config_item->offset + index * config_item->size); + req.bytes = cpu_to_le32(config_item->size); req.value = cpu_to_le32(value); err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, - &req, sizeof(u32) * 2 + config_item.size, + &req, sizeof(u32) * 2 + config_item->size, NULL, 0); if (err < 0) return err; /* Activate the change */ - req2 = cpu_to_le32(config_item.activate); + req2 = cpu_to_le32(config_item->activate); err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, &req2, sizeof(req2), NULL, 0); if (err < 0) @@ -764,11 +764,11 @@ static int scarlett2_usb_get_config( struct usb_mixer_interface *mixer, int config_item_num, int count, void *buf) { - const struct scarlett2_config config_item = - scarlett2_config_items[config_item_num]; - int size = config_item.size * count; + const struct scarlett2_config *config_item = + &scarlett2_config_items[config_item_num]; + int size = config_item->size * count; - return scarlett2_usb_get(mixer, config_item.offset, buf, size); + return scarlett2_usb_get(mixer, config_item->offset, buf, size); } /* Send a USB message to get volume status; result placed in *buf */ -- 2.31.1

1 0

[PATCH 12/14] ALSA: usb-audio: scarlett2: Fix union usage in mixer control callbacks
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Fix mixer control callbacks to use the correct members of the struct snd_ctl_elem_value. The use of value.integer and value.enumerated were swapped in a few places. Update scarlett2_mux_src_enum_ctl_put() to use min() instead of clamp() as value.enumerated.item is unsigned. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index fe142b7ddb32..49c4662531a3 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1270,7 +1270,7 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); oval = private->vol_sw_hw_switch[index]; - val = !!ucontrol->value.integer.value[0]; + val = !!ucontrol->value.enumerated.item[0]; if (oval == val) goto unlock; @@ -1356,7 +1356,7 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); oval = private->level_switch[index]; - val = !!ucontrol->value.integer.value[0]; + val = !!ucontrol->value.enumerated.item[0]; if (oval == val) goto unlock; @@ -1390,7 +1390,7 @@ static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct scarlett2_data *private = elem->head.mixer->private_data; - ucontrol->value.enumerated.item[0] = + ucontrol->value.integer.value[0] = private->pad_switch[elem->control]; return 0; } @@ -1448,7 +1448,7 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, scarlett2_update_volumes(mixer); mutex_unlock(&private->data_mutex); - ucontrol->value.enumerated.item[0] = private->dim_mute[elem->control]; + ucontrol->value.integer.value[0] = private->dim_mute[elem->control]; return 0; } @@ -1748,8 +1748,8 @@ static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); oval = private->mux[index]; - val = clamp(ucontrol->value.integer.value[0], - 0L, private->num_mux_srcs - 1L); + val = min(ucontrol->value.enumerated.item[0], + private->num_mux_srcs - 1U); if (oval == val) goto unlock; -- 2.31.1

1 0

[PATCH 11/14] ALSA: usb-audio: scarlett2: Fix scarlett2_*_ctl_put() return values
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Mixer control put callbacks should return 1 if the value is changed. Fix the sw_hw, level, pad, and button controls accordingly. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index d6298ab1de56..fe142b7ddb32 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1304,6 +1304,8 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, /* Send SW/HW switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1364,6 +1366,8 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1414,6 +1418,8 @@ static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1469,6 +1475,8 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_DIM_MUTE, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); -- 2.31.1

1 0

[PATCH 10/14] ALSA: usb-audio: scarlett2: Fix data_mutex lock
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The private->vol_updated flag was being checked outside of the mutex_lock/unlock() of private->data_mutex leading to the volume data being fetched twice from the device unnecessarily or old volume data being returned. Update scarlett2_*_ctl_get() and include the private->vol_updated flag check inside the critical region. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index cb8483d3c58a..d6298ab1de56 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1153,11 +1153,10 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->master_vol; return 0; @@ -1171,11 +1170,10 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, struct scarlett2_data *private = mixer->private_data; int index = elem->control; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->vol[index]; return 0; @@ -1439,11 +1437,10 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.enumerated.item[0] = private->dim_mute[elem->control]; return 0; -- 2.31.1

1 0

[PATCH 09/14] ALSA: usb-audio: scarlett2: Add temp variable for consistency
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Add index temporary variable to scarlett2_mixer_ctl_put() for consistency with the other *_ctl_put() functions. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index c1e74918425a..cb8483d3c58a 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1622,18 +1622,19 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int oval, val, num_mixer_in, mix_num, err = 0; + int index = elem->control; mutex_lock(&private->data_mutex); - oval = private->mix[elem->control]; + oval = private->mix[index]; val = ucontrol->value.integer.value[0]; num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; - mix_num = elem->control / num_mixer_in; + mix_num = index / num_mixer_in; if (oval == val) goto unlock; - private->mix[elem->control] = val; + private->mix[index] = val; err = scarlett2_usb_set_mix(mixer, mix_num); if (err == 0) err = 1; -- 2.31.1

1 0

[PATCH 08/14] ALSA: usb-audio: scarlett2: Rename struct scarlett2_mixer_data
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Rename struct scarlett2_mixer_data to struct scarlett2_data. A less-wordy name is better because it is used everywhere, and although this is a mixer driver, it also controls other vendor-specific features. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 80 ++++++++++++++++----------------- 1 file changed, 40 insertions(+), 40 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 4e2ee979b9bd..c1e74918425a 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -224,7 +224,7 @@ struct scarlett2_device_info { struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; }; -struct scarlett2_mixer_data { +struct scarlett2_data { struct usb_mixer_interface *mixer; struct mutex usb_mutex; /* prevent sending concurrent USB requests */ struct mutex data_mutex; /* lock access to this data */ @@ -564,7 +564,7 @@ struct scarlett2_usb_packet { u8 data[]; }; -static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, +static void scarlett2_fill_request_header(struct scarlett2_data *private, struct scarlett2_usb_packet *req, u32 cmd, u16 req_size) { @@ -583,7 +583,7 @@ static int scarlett2_usb( struct usb_mixer_interface *mixer, u32 cmd, void *req_data, u16 req_size, void *resp_data, u16 resp_size) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; struct scarlett2_usb_packet *req, *resp = NULL; @@ -693,8 +693,8 @@ static void scarlett2_config_save(struct usb_mixer_interface *mixer) /* Delayed work to save config */ static void scarlett2_config_save_work(struct work_struct *work) { - struct scarlett2_mixer_data *private = - container_of(work, struct scarlett2_mixer_data, work.work); + struct scarlett2_data *private = + container_of(work, struct scarlett2_data, work.work); scarlett2_config_save(private->mixer); } @@ -715,7 +715,7 @@ static int scarlett2_usb_set_config( } __packed req; __le32 req2; int err; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; /* Cancel any pending NVRAM save */ cancel_delayed_work_sync(&private->work); @@ -786,7 +786,7 @@ static int scarlett2_usb_get_volume_status( static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, int mix_num) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; int num_mixer_in = @@ -829,7 +829,7 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, int mix_num) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; struct { @@ -895,7 +895,7 @@ static u32 scarlett2_mux_id_to_num(const struct scarlett2_ports *ports, } /* Convert one mux entry from the interface and load into private->mux[] */ -static void scarlett2_usb_populate_mux(struct scarlett2_mixer_data *private, +static void scarlett2_usb_populate_mux(struct scarlett2_data *private, u32 mux_entry) { const struct scarlett2_device_info *info = private->info; @@ -933,7 +933,7 @@ static void scarlett2_usb_populate_mux(struct scarlett2_mixer_data *private, /* Send USB message to get mux inputs and then populate private->mux[] */ static int scarlett2_usb_get_mux(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int count = private->num_mux_dsts; int err, i; @@ -962,7 +962,7 @@ static int scarlett2_usb_get_mux(struct usb_mixer_interface *mixer) /* Send USB messages to set mux inputs */ static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int rate, port_dir_rate; @@ -1103,7 +1103,7 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, */ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; struct scarlett2_usb_volume_status volume_status; @@ -1151,7 +1151,7 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; if (private->vol_updated) { mutex_lock(&private->data_mutex); @@ -1168,7 +1168,7 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; if (private->vol_updated) { @@ -1186,7 +1186,7 @@ static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1252,7 +1252,7 @@ static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->vol_sw_hw_switch[elem->control]; @@ -1264,7 +1264,7 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1336,7 +1336,7 @@ static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->level_switch[elem->control]; @@ -1348,7 +1348,7 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1386,7 +1386,7 @@ static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->pad_switch[elem->control]; @@ -1398,7 +1398,7 @@ static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1437,7 +1437,7 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; if (private->vol_updated) { mutex_lock(&private->data_mutex); @@ -1454,7 +1454,7 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1490,7 +1490,7 @@ static const struct snd_kcontrol_new scarlett2_dim_mute_ctl = { static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int num_line_out = @@ -1562,7 +1562,7 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; int err, i; char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -1607,7 +1607,7 @@ static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.integer.value[0] = private->mix[elem->control]; return 0; @@ -1618,7 +1618,7 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int oval, val, num_mixer_in, mix_num, err = 0; @@ -1663,7 +1663,7 @@ static const struct snd_kcontrol_new scarlett2_mixer_ctl = { static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; int err, i, j; int index; @@ -1692,7 +1692,7 @@ static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; unsigned int item = uinfo->value.enumerated.item; int items = private->num_mux_srcs; @@ -1724,7 +1724,7 @@ static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; ucontrol->value.enumerated.item[0] = private->mux[elem->control]; return 0; @@ -1735,7 +1735,7 @@ static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int index = elem->control; int oval, val, err = 0; @@ -1768,7 +1768,7 @@ static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; int port_type, channel, i; @@ -1847,7 +1847,7 @@ static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) static void scarlett2_private_free(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; cancel_delayed_work_sync(&private->work); kfree(private); @@ -1856,7 +1856,7 @@ static void scarlett2_private_free(struct usb_mixer_interface *mixer) static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; if (cancel_delayed_work_sync(&private->work)) scarlett2_config_save(private->mixer); @@ -1864,7 +1864,7 @@ static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) /*** Initialisation ***/ -static void scarlett2_count_mux_io(struct scarlett2_mixer_data *private) +static void scarlett2_count_mux_io(struct scarlett2_data *private) { const struct scarlett2_ports *ports = private->info->ports; int port_type, srcs = 0, dsts = 0; @@ -1884,8 +1884,8 @@ static void scarlett2_count_mux_io(struct scarlett2_mixer_data *private) static int scarlett2_init_private(struct usb_mixer_interface *mixer, const struct scarlett2_device_info *info) { - struct scarlett2_mixer_data *private = - kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + struct scarlett2_data *private = + kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL); if (!private) return -ENOMEM; @@ -1908,7 +1908,7 @@ static int scarlett2_init_private(struct usb_mixer_interface *mixer, /* Read configuration from the interface on start */ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_ports *ports = info->ports; int num_line_out = @@ -1978,7 +1978,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) static void scarlett2_notify_monitor( struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_ports *ports = private->info->ports; int num_line_out = ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; @@ -2001,7 +2001,7 @@ static void scarlett2_notify_monitor( static void scarlett2_notify_dim_mute( struct usb_mixer_interface *mixer) { - struct scarlett2_mixer_data *private = mixer->private_data; + struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; int i; -- 2.31.1

1 0

[PATCH 07/14] ALSA: usb-audio: scarlett2: Rename buttons/interrupts/vol
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

To match the vendor's terminology, change #defines, identifiers, and comments: - mute/dim/hardware buttons are now called dim/mute - mixer status/interrupt is now notify - vol is now monitor Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 94 ++++++++++++++++----------------- 1 file changed, 47 insertions(+), 47 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 043b0929ad9d..4e2ee979b9bd 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -194,10 +194,10 @@ enum { SCARLETT2_PORT_DIRNS = 5, }; -/* Hardware buttons on the 18i20 */ -#define SCARLETT2_BUTTON_MAX 2 +/* Dim/Mute buttons on the 18i20 */ +#define SCARLETT2_DIM_MUTE_COUNT 2 -static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { +static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { "Mute", "Dim" }; @@ -239,10 +239,10 @@ struct scarlett2_mixer_data { u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; - u8 buttons[SCARLETT2_BUTTON_MAX]; + u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT]; struct snd_kcontrol *master_vol_ctl; struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; - struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *dim_mute_ctls[SCARLETT2_DIM_MUTE_COUNT]; u8 mux[SCARLETT2_MUX_MAX]; u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; }; @@ -450,9 +450,9 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, #define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 #define SCARLETT2_USB_INTERRUPT_INTERVAL 3 -/* Interrupt flags for volume and mute/dim button changes */ -#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x00400000 -#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x00200000 +/* Interrupt flags for dim/mute button and monitor changes */ +#define SCARLETT2_USB_NOTIFY_DIM_MUTE 0x00200000 +#define SCARLETT2_USB_NOTIFY_MONITOR 0x00400000 /* Commands for sending/receiving requests/responses */ #define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 @@ -475,8 +475,8 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, /* volume status is read together (matches scarlett2_config_items[]) */ struct scarlett2_usb_volume_status { - /* mute & dim buttons */ - u8 buttons[SCARLETT2_BUTTON_MAX]; + /* dim/mute buttons */ + u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT]; u8 pad1; @@ -499,7 +499,7 @@ struct scarlett2_usb_volume_status { /* Configuration parameters that can be read and written */ enum { - SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_DIM_MUTE = 0, SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, SCARLETT2_CONFIG_SW_HW_SWITCH = 2, SCARLETT2_CONFIG_LEVEL_SWITCH = 3, @@ -518,7 +518,7 @@ struct scarlett2_config { static const struct scarlett2_config scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { - /* Mute/Dim Buttons */ + /* Dim/Mute Buttons */ { .offset = 0x31, .size = 1, @@ -1127,8 +1127,8 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) } if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) - private->buttons[i] = !!volume_status.buttons[i]; + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) + private->dim_mute[i] = !!volume_status.dim_mute[i]; return 0; } @@ -1430,10 +1430,10 @@ static const struct snd_kcontrol_new scarlett2_pad_ctl = { .put = scarlett2_pad_ctl_put, }; -/*** Mute/Dim Controls ***/ +/*** Dim/Mute Controls ***/ -static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; @@ -1445,12 +1445,12 @@ static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, mutex_unlock(&private->data_mutex); } - ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + ucontrol->value.enumerated.item[0] = private->dim_mute[elem->control]; return 0; } -static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; @@ -1461,16 +1461,16 @@ static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); - oval = private->buttons[index]; + oval = private->dim_mute[index]; val = !!ucontrol->value.integer.value[0]; if (oval == val) goto unlock; - private->buttons[index] = val; + private->dim_mute[index] = val; /* Send switch change to the device */ - err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_DIM_MUTE, index, val); unlock: @@ -1478,12 +1478,12 @@ static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, return err; } -static const struct snd_kcontrol_new scarlett2_button_ctl = { +static const struct snd_kcontrol_new scarlett2_dim_mute_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", .info = snd_ctl_boolean_mono_info, - .get = scarlett2_button_ctl_get, - .put = scarlett2_button_ctl_put + .get = scarlett2_dim_mute_ctl_get, + .put = scarlett2_dim_mute_ctl_put }; /*** Create the analogue output controls ***/ @@ -1544,13 +1544,13 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) } } - /* Add HW button controls */ + /* Add dim/mute controls */ if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) { + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) { err = scarlett2_add_new_ctl( - mixer, &scarlett2_button_ctl, - i, 1, scarlett2_button_names[i], - &private->button_ctls[i]); + mixer, &scarlett2_dim_mute_ctl, + i, 1, scarlett2_dim_mute_names[i], + &private->dim_mute_ctls[i]); if (err < 0) return err; } @@ -1962,8 +1962,8 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) } if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) - private->buttons[i] = !!volume_status.buttons[i]; + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) + private->dim_mute[i] = !!volume_status.dim_mute[i]; for (i = 0; i < num_mixer_out; i++) { err = scarlett2_usb_get_mix(mixer, i); @@ -1974,8 +1974,8 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) return scarlett2_usb_get_mux(mixer); } -/* Notify on volume change */ -static void scarlett2_mixer_interrupt_vol_change( +/* Notify on monitor change */ +static void scarlett2_notify_monitor( struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; @@ -1997,8 +1997,8 @@ static void scarlett2_mixer_interrupt_vol_change( } } -/* Notify on button change */ -static void scarlett2_mixer_interrupt_button_change( +/* Notify on dim/mute change */ +static void scarlett2_notify_dim_mute( struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; @@ -2010,13 +2010,13 @@ static void scarlett2_mixer_interrupt_button_change( if (!info->line_out_hw_vol) return; - for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->button_ctls[i]->id); + &private->dim_mute_ctls[i]->id); } /* Interrupt callback */ -static void scarlett2_mixer_interrupt(struct urb *urb) +static void scarlett2_notify(struct urb *urb) { struct usb_mixer_interface *mixer = urb->context; int len = urb->actual_length; @@ -2027,10 +2027,10 @@ static void scarlett2_mixer_interrupt(struct urb *urb) goto requeue; data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); - if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) - scarlett2_mixer_interrupt_vol_change(mixer); - if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) - scarlett2_mixer_interrupt_button_change(mixer); + if (data & SCARLETT2_USB_NOTIFY_MONITOR) + scarlett2_notify_monitor(mixer); + if (data & SCARLETT2_USB_NOTIFY_DIM_MUTE) + scarlett2_notify_dim_mute(mixer); requeue: if (ustatus != -ENOENT && @@ -2041,7 +2041,7 @@ static void scarlett2_mixer_interrupt(struct urb *urb) } } -static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +static int scarlett2_init_notify(struct usb_mixer_interface *mixer) { struct usb_device *dev = mixer->chip->dev; unsigned int pipe = usb_rcvintpipe(dev, @@ -2067,7 +2067,7 @@ static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) usb_fill_int_urb(mixer->urb, dev, pipe, transfer_buffer, SCARLETT2_USB_INTERRUPT_MAX_DATA, - scarlett2_mixer_interrupt, mixer, + scarlett2_notify, mixer, SCARLETT2_USB_INTERRUPT_INTERVAL); return usb_submit_urb(mixer->urb, GFP_KERNEL); @@ -2115,7 +2115,7 @@ static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, /* Set up the interrupt polling if there are hardware buttons */ if (info->line_out_hw_vol) { - err = scarlett2_mixer_status_create(mixer); + err = scarlett2_init_notify(mixer); if (err < 0) return err; } -- 2.31.1

1 0

[PATCH 06/14] ALSA: usb-audio: scarlett2: Remove redundant info->button_count
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The per-model button_count value was used to determine whether dim/mute controls should be added, but these are present iff line_out_hw_vol is true. Remove button_count and replace with SCARLETT2_BUTTON_MAX and a check for line_out_hw_vol true. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 41 ++++++++++++++++++--------------- 1 file changed, 23 insertions(+), 18 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 3225ec709d98..043b0929ad9d 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -218,7 +218,6 @@ struct scarlett2_ports { struct scarlett2_device_info { u8 line_out_hw_vol; /* line out hw volume is sw controlled */ - u8 button_count; /* number of buttons */ u8 level_input_count; /* inputs with level selectable */ u8 pad_input_count; /* inputs with pad selectable */ const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; @@ -370,9 +369,6 @@ static const struct scarlett2_device_info s18i20_gen2_info = { */ .line_out_hw_vol = 1, - /* Mute and dim buttons */ - .button_count = 2, - .line_out_descrs = { "Monitor L", "Monitor R", @@ -1108,7 +1104,8 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; - const struct scarlett2_ports *ports = private->info->ports; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; struct scarlett2_usb_volume_status volume_status; int num_line_out = ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; @@ -1129,8 +1126,9 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) private->vol[i] = private->master_vol; } - for (i = 0; i < private->info->button_count; i++) - private->buttons[i] = !!volume_status.buttons[i]; + if (info->line_out_hw_vol) + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) + private->buttons[i] = !!volume_status.buttons[i]; return 0; } @@ -1547,13 +1545,15 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) } /* Add HW button controls */ - for (i = 0; i < private->info->button_count; i++) { - err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, - i, 1, scarlett2_button_names[i], - &private->button_ctls[i]); - if (err < 0) - return err; - } + if (info->line_out_hw_vol) + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) { + err = scarlett2_add_new_ctl( + mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } return 0; } @@ -1961,8 +1961,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) private->vol[i] = volume; } - for (i = 0; i < info->button_count; i++) - private->buttons[i] = !!volume_status.buttons[i]; + if (info->line_out_hw_vol) + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) + private->buttons[i] = !!volume_status.buttons[i]; for (i = 0; i < num_mixer_out; i++) { err = scarlett2_usb_get_mix(mixer, i); @@ -2001,11 +2002,15 @@ static void scarlett2_mixer_interrupt_button_change( struct usb_mixer_interface *mixer) { struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; int i; private->vol_updated = 1; - for (i = 0; i < private->info->button_count; i++) + if (!info->line_out_hw_vol) + return; + + for (i = 0; i < SCARLETT2_BUTTON_MAX; i++) snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &private->button_ctls[i]->id); } @@ -2109,7 +2114,7 @@ static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, return err; /* Set up the interrupt polling if there are hardware buttons */ - if (info->button_count) { + if (info->line_out_hw_vol) { err = scarlett2_mixer_status_create(mixer); if (err < 0) return err; -- 2.31.1

1 0

[PATCH 05/14] ALSA: usb-audio: scarlett2: Remove interrupt debug message
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Just ignore instead of printing an error if the interrupt data is not the expected length. This check was for development and the condition has not been observed. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index a5f5c537e344..3225ec709d98 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -2018,19 +2018,14 @@ static void scarlett2_mixer_interrupt(struct urb *urb) int ustatus = urb->status; u32 data; - if (ustatus != 0) + if (ustatus != 0 || len != 8) goto requeue; - if (len == 8) { - data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); - if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) - scarlett2_mixer_interrupt_vol_change(mixer); - if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) - scarlett2_mixer_interrupt_button_change(mixer); - } else { - usb_audio_err(mixer->chip, - "scarlett mixer interrupt length %d\n", len); - } + data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); requeue: if (ustatus != -ENOENT && -- 2.31.1

1 0

[PATCH 04/14] ALSA: usb-audio: scarlett2: Remove unused/useless code
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Remove #define not used. Remove useless assignments and copies. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 8026847d27b6..a5f5c537e344 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -271,7 +271,6 @@ static const struct scarlett2_device_info s6i6_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 8 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -327,7 +326,6 @@ static const struct scarlett2_device_info s18i8_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 4 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -393,7 +391,6 @@ static const struct scarlett2_device_info s18i20_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 6 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -571,8 +568,6 @@ struct scarlett2_usb_packet { u8 data[]; }; -#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) - static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, struct scarlett2_usb_packet *req, u32 cmd, u16 req_size) @@ -595,8 +590,8 @@ static int scarlett2_usb( struct scarlett2_mixer_data *private = mixer->private_data; u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; - struct scarlett2_usb_packet *req = NULL, *resp = NULL; - int err = 0; + struct scarlett2_usb_packet *req, *resp = NULL; + int err; req = kmalloc(req_buf_size, GFP_KERNEL); if (!req) { @@ -1026,10 +1021,6 @@ static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) ports, private->mux[j] ) << 12 ); - - /* skip private->mux[j] entries not output */ - j += ports[port_type].num[SCARLETT2_PORT_OUT] - - ports[port_type].num[port_dir_rate]; } err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, @@ -1681,7 +1672,7 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; - for (i = 0, index = 0; i < num_outputs; i++) { + for (i = 0, index = 0; i < num_outputs; i++) for (j = 0; j < num_inputs; j++, index++) { snprintf(s, sizeof(s), "Mix %c Input %02d Playback Volume", @@ -1691,7 +1682,6 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) if (err < 0) return err; } - } return 0; } @@ -1925,8 +1915,6 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; int num_mixer_out = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; - u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; - u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; struct scarlett2_usb_volume_status volume_status; int err, i; @@ -1935,11 +1923,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, info->level_input_count, - level_switches); + private->level_switch); if (err < 0) return err; - for (i = 0; i < info->level_input_count; i++) - private->level_switch[i] = level_switches[i]; } if (info->pad_input_count) { @@ -1947,11 +1933,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) mixer, SCARLETT2_CONFIG_PAD_SWITCH, info->pad_input_count, - pad_switches); + private->pad_switch); if (err < 0) return err; - for (i = 0; i < info->pad_input_count; i++) - private->pad_switch[i] = pad_switches[i]; } err = scarlett2_usb_get_volume_status(mixer, &volume_status); -- 2.31.1

1 0

[PATCH 03/14] ALSA: usb-audio: scarlett2: Coding style improvements
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Improve alignment and readability with: - Whitespace fixes - Add leading zeros to 32-bit flag values - Rename SCARLETT2_USB_GET_METER_LEVELS to SCARLETT2_USB_GET_METER - Rename SCARLETT2_PORT_DIRECTIONS to SCARLETT2_PORT_DIRNS Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 53 +++++++++++++++++---------------- 1 file changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index e327f15b89ce..8026847d27b6 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -175,23 +175,23 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { * - PCM I/O */ enum { - SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_NONE = 0, SCARLETT2_PORT_TYPE_ANALOGUE = 1, - SCARLETT2_PORT_TYPE_SPDIF = 2, - SCARLETT2_PORT_TYPE_ADAT = 3, - SCARLETT2_PORT_TYPE_MIX = 4, - SCARLETT2_PORT_TYPE_PCM = 5, - SCARLETT2_PORT_TYPE_COUNT = 6, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, }; /* Count of total I/O and number available at each sample rate */ enum { - SCARLETT2_PORT_IN = 0, - SCARLETT2_PORT_OUT = 1, - SCARLETT2_PORT_OUT_44 = 2, - SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, SCARLETT2_PORT_OUT_176 = 4, - SCARLETT2_PORT_DIRECTIONS = 5, + SCARLETT2_PORT_DIRNS = 5, }; /* Hardware buttons on the 18i20 */ @@ -210,7 +210,7 @@ static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { */ struct scarlett2_ports { u16 id; - int num[SCARLETT2_PORT_DIRECTIONS]; + int num[SCARLETT2_PORT_DIRNS]; const char * const src_descr; int src_num_offset; const char * const dst_descr; @@ -458,22 +458,23 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, #define SCARLETT2_USB_INTERRUPT_INTERVAL 3 /* Interrupt flags for volume and mute/dim button changes */ -#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 -#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x00400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x00200000 /* Commands for sending/receiving requests/responses */ -#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 #define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 -#define SCARLETT2_USB_INIT_SEQ 0x00000000 -#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 -#define SCARLETT2_USB_GET_MIX 0x00002001 -#define SCARLETT2_USB_SET_MIX 0x00002002 -#define SCARLETT2_USB_GET_MUX 0x00003001 -#define SCARLETT2_USB_SET_MUX 0x00003002 -#define SCARLETT2_USB_GET_DATA 0x00800000 -#define SCARLETT2_USB_SET_DATA 0x00800001 -#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER 0x00001001 +#define SCARLETT2_USB_GET_MIX 0x00002001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_GET_MUX 0x00003001 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 + #define SCARLETT2_USB_CONFIG_SAVE 6 #define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 @@ -835,7 +836,7 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, * (values obtained from private->mix[]) */ static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, - int mix_num) + int mix_num) { struct scarlett2_mixer_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; @@ -1056,7 +1057,7 @@ static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, req.pad = 0; req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); - err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER, &req, sizeof(req), resp, sizeof(resp)); if (err < 0) return err; -- 2.31.1

1 0

[PATCH 02/14] ALSA: usb-audio: scarlett2: Fix 18i8 Gen 2 PCM Input count
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The 18i8 Gen 2 has 8 PCM Inputs, not 20. Fix the ports entry in s18i8_gen2_info. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 962050436a9f..e327f15b89ce 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -358,7 +358,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, - .num = { 20, 18, 18, 14, 10 }, + .num = { 8, 18, 18, 14, 10 }, .src_descr = "PCM %d", .src_num_offset = 1, .dst_descr = "PCM %02d Capture" -- 2.31.1

1 0

[PATCH 01/14] ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The 18i8 Gen 2 S/PDIF outputs are available at 192kHz, unlike the 18i20 Gen 2. Remove the comment that says otherwise. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index a461317dc8c6..962050436a9f 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -338,10 +338,6 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_SPDIF] = { .id = 0x180, - /* S/PDIF outputs aren't available at 192kHz - * but are included in the USB mux I/O - * assignment message anyway - */ .num = { 2, 2, 2, 2, 2 }, .src_descr = "S/PDIF %d", .src_num_offset = 1, -- 2.31.1

1 0

[PATCH AUTOSEL 5.12 11/33] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Sasha Levin 20 Jun '21

20 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 19a0aa9b04c5ab9a063b6ceaf7211ee7d9a9d24d ] More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210531145502.6079-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 050a61fe9693..47a4dfd81a46 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.30.2

2 2

[PATCH] soundwire: stream: Fix test for DP prepare complete
by Richard Fitzgerald 20 Jun '21

20 Jun '21

In sdw_prep_deprep_slave_ports(), after the wait_for_completion() the DP prepare status register is read. If this indicates that the port is now prepared, the code should continue with the port setup. It is irrelevant whether the wait_for_completion() timed out if the port is now ready. The previous implementation would always fail if the wait_for_completion() timed out, even if the port was reporting successful prepare. This patch also fixes a minor bug where the return from sdw_read() was not checked for error - any error code with LSBits clear could be misinterpreted as a successful port prepare. Fixes: 79df15b7d37c ("soundwire: Add helpers for ports operations") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/stream.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1eaedaaba094..1a18308f4ef4 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -422,7 +422,6 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, struct completion *port_ready; struct sdw_dpn_prop *dpn_prop; struct sdw_prepare_ch prep_ch; - unsigned int time_left; bool intr = false; int ret = 0, val; u32 addr; @@ -479,15 +478,15 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, /* Wait for completion on port ready */ port_ready = &s_rt->slave->port_ready[prep_ch.num]; - time_left = wait_for_completion_timeout(port_ready, - msecs_to_jiffies(dpn_prop->ch_prep_timeout)); + wait_for_completion_timeout(port_ready, + msecs_to_jiffies(dpn_prop->ch_prep_timeout)); val = sdw_read(s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); - val &= p_rt->ch_mask; - if (!time_left || val) { + if ((val < 0) || (val & p_rt->ch_mask)) { + ret = (val < 0) ? val : -ETIMEDOUT; dev_err(&s_rt->slave->dev, - "Chn prep failed for port:%d\n", prep_ch.num); - return -ETIMEDOUT; + "Chn prep failed for port %d: %d\n", prep_ch.num, ret); + return ret; } } -- 2.20.1

3 2

[PATCH v2] soundwire: bus: Make sdw_nwrite() data pointer argument const
by Richard Fitzgerald 20 Jun '21

20 Jun '21

Idiomatically, write functions should take const pointers to the data buffer, as they don't change the data. They are also likely to be called from functions that receive a const data pointer. Internally the pointer is passed to function/structs shared with the read functions, requiring a cast, but this is an implementation detail that should be hidden by the public API. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 8 ++++---- include/linux/soundwire/sdw.h | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a9e0aa72654d..49c66deba106 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -394,13 +394,13 @@ sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) } static int -sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { struct sdw_msg msg; int ret; ret = sdw_fill_msg(&msg, slave, addr, count, - slave->dev_num, SDW_MSG_FLAG_WRITE, val); + slave->dev_num, SDW_MSG_FLAG_WRITE, (u8 *)val); if (ret < 0) return ret; @@ -535,9 +535,9 @@ EXPORT_SYMBOL(sdw_nread); * @slave: SDW Slave * @addr: Register address * @count: length - * @val: Buffer for values to be read + * @val: Buffer for values to be written */ -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ced07f8fde87..ee45d2984334 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1040,7 +1040,7 @@ int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value); int sdw_read_no_pm(struct sdw_slave *slave, u32 addr); int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); int sdw_compare_devid(struct sdw_slave *slave, struct sdw_slave_id id); void sdw_extract_slave_id(struct sdw_bus *bus, u64 addr, struct sdw_slave_id *id); -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: Fix bass speaker DAC mapping for Asus UM431D
by Takashi Iwai 20 Jun '21

20 Jun '21

Asus Zenbook 14 UM431D has two speaker pins and a headphone pin, and the auto-parser ends up assigning the bass to the third DAC 0x06. Although the tone comes out, it's inconvenient because this DAC has no volume control unlike two other DACs. For obtaining the volume control for the bass speaker, this patch enforces the mapping to let both front and bass speaker pins sharing the same DAC. It's not ideal but a little bit of improvement. Since we've already applied the same workaround for another ASUS machine, we just need to hook the chain to the existing quirk. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212547 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 07eabcf22b5f..49f4cac8b05e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7839,6 +7839,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b }, { } }, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA401, }, [ALC285_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, -- 2.26.2

1 0

[PATCH] ALSA: hda/realtek: Improve fixup for HP Spectre x360 15-df0xxx
by Elia Devito 20 Jun '21

20 Jun '21

On HP Spectre x360 15-df0xxx, after system boot with plugged headset, the headset mic are not detected. Moving pincfg and DAC's config to single fixup function fix this. Signed-off-by: Elia Devito <eliadevito(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d0936377eb..07eabcf22b5f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6354,6 +6354,24 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* rear speaker */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + /* force front speaker to DAC1 */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -8138,13 +8156,8 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED, }, [ALC285_FIXUP_HP_SPECTRE_X360] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x90170110 }, /* enable top speaker */ - {} - }, - .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360, }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, -- 2.31.1

2 1

[PATCH] ALSA: bebob: add support for ToneWeal FW66
by Takashi Sakamoto 20 Jun '21

20 Jun '21

A user of FFADO project reported the issue of ToneWeal FW66. As a result, the device is identified as one of applications of BeBoB solution. I note that in the report the device returns contradictory result in plug discovery process for audio subunit. Fortunately ALSA BeBoB driver doesn't perform it thus it's likely to handle the device without issues. I receive no reaction to test request for this patch yet, however it would be worth to add support for it. daniel@gibbonmoon:/sys/bus/firewire/devices/fw1$ grep -r . * Binary file config_rom matches dev:244:1 guid:0x0023270002000000 hardware_version:0x000002 is_local:0 model:0x020002 model_name:FW66 power/runtime_active_time:0 power/runtime_active_kids:0 power/runtime_usage:0 power/runtime_status:unsupported power/async:disabled power/runtime_suspended_time:0 power/runtime_enabled:disabled power/control:auto subsystem/drivers_autoprobe:1 uevent:MAJOR=244 uevent:MINOR=1 uevent:DEVNAME=fw1 units:0x00a02d:0x010001 vendor:0x002327 vendor_name:ToneWeal fw1.0/uevent:MODALIAS=ieee1394:ven00002327mo00020002sp0000A02Dver00010001 fw1.0/power/runtime_active_time:0 fw1.0/power/runtime_active_kids:0 fw1.0/power/runtime_usage:0 fw1.0/power/runtime_status:unsupported fw1.0/power/async:disabled fw1.0/power/runtime_suspended_time:0 fw1.0/power/runtime_enabled:disabled fw1.0/power/control:auto fw1.0/model:0x020002 fw1.0/rom_index:15 fw1.0/specifier_id:0x00a02d fw1.0/model_name:FW66 fw1.0/version:0x010001 fw1.0/modalias:ieee1394:ven00002327mo00020002sp0000A02Dver00010001 Cc: Daniel Jozsef <daniel.jozsef(a)gmail.com> Reference: https://lore.kernel.org/alsa-devel/20200119164335.GA11974@workstation/ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 62658e2b111a..fd109bea4c53 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -127,6 +127,7 @@ config SND_BEBOB * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro + * ToneWeal FW66 To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 7ca452efb133..42980da45fbf 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -56,6 +56,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_FOCUSRITE 0x0000130e #define VEN_MAUDIO 0x00000d6c #define VEN_DIGIDESIGN 0x00a07e +#define OUI_SHOUYO 0x002327 #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -469,6 +470,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { &maudio_special_spec), /* Digidesign Mbox 2 Pro */ SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), + // Toneweal FW66. + SND_BEBOB_DEV_ENTRY(OUI_SHOUYO, 0x020002, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ -- 2.30.2

2 1

[PATCH 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 19 Jun '21

19 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 408 ++++++++++++++++++++++++++++++++++ 4 files changed, 424 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..78d512d93974 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,408 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * ALSA SoC DMIC Audio Layer + * + * Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + * + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct platform_device *pdev; + struct clk *dmic_clk; + struct clk *apb_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static void sun50i_snd_rxctrl_enable(struct snd_pcm_substream *substream, + struct sun50i_dmic_dev *host, bool enable) +{ + if (enable) { + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + } else { + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + } +} + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_write(host->regmap, SUN50I_DMIC_INT_STA, + SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING | + SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct platform_device *pdev = host->pdev; + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(&pdev->dev, "Invalid format!\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + if (i == ARRAY_SIZE(dmic_rate_s)) { + dev_err(&pdev->dev, "Invalid rate!\n"); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sun50i_snd_rxctrl_enable(substream, host, true); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sun50i_snd_rxctrl_enable(substream, host, false); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (clk_set_rate(host->dmic_clk, freq)) { + dev_err(dai->dev, "Freq : %u not support\n", freq); + return -EINVAL; + } + + return 0; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, + .set_sysclk = sun50i_dmic_set_sysclk, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->apb_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->apb_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + host->pdev = pdev; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->apb_clk = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(host->apb_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->apb_clk), + "failed to get apb clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "dmic"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst) && PTR_ERR(host->rst) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + dev_err(&pdev->dev, "Failed to get reset: %d\n", ret); + return ret; + } + if (!IS_ERR(host->rst)) + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

5 10

[PATCH] ALSA: hda/realtek: Add another ALC236 variant support
by Takashi Iwai 18 Jun '21

18 Jun '21

The codec chip 10ec:0230 is another variant of ALC236, combined with a card reader. Apply the equivalent setup as 10ec:0236. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1184869 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bbbe37f3e724..fac2f15d9e26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -385,6 +385,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); fallthrough; case 0x10ec0215: + case 0x10ec0230: case 0x10ec0233: case 0x10ec0235: case 0x10ec0236: @@ -3127,6 +3128,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0x0); @@ -3154,6 +3156,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0xd011); @@ -4689,6 +4692,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -4803,6 +4807,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0255); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x45, 0xc489); @@ -4952,6 +4957,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -5050,6 +5056,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5163,6 +5170,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5263,6 +5271,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x0070) == 0x0070; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -5556,6 +5565,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, alc255fw); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, alc256fw); @@ -6156,6 +6166,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); break; + case 0x10ec0230: case 0x10ec0235: case 0x10ec0236: case 0x10ec0255: @@ -9130,6 +9141,7 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc256_shutup; spec->init_hook = alc256_init; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: spec->codec_variant = ALC269_TYPE_ALC256; @@ -10421,6 +10433,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0230, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), -- 2.26.2

1 0

[PATCH v3 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH v3 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- v2 -> v3: no changes. v1 -> v2: fix typo from HP ProBook 440 to HP ProBook 450 G8 (https://lkml.org/lkml/2021/6/18/325) 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 ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

2 1

[PATCH 1/1] ASoC: Intel: bdw-rt5677: remove unnecessary oom message
by Zhen Lei 18 Jun '21

18 Jun '21

Fixes scripts/checkpatch.pl warning: WARNING: Possible unnecessary 'out of memory' message Remove it can help us save a bit of memory. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/soc/intel/boards/bdw-rt5677.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 021bc59aac80..e01b7a90ca6c 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -423,10 +423,8 @@ static int bdw_rt5677_probe(struct platform_device *pdev) /* Allocate driver private struct */ bdw_rt5677 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5677_priv), GFP_KERNEL); - if (!bdw_rt5677) { - dev_err(&pdev->dev, "Can't allocate bdw_rt5677\n"); + if (!bdw_rt5677) return -ENOMEM; - } /* override plaform name, if required */ mach = pdev->dev.platform_data; -- 2.25.1

3 2

[PATCH] ASoC: fsi: fix spelling mistake
by Flavio Suligoi 18 Jun '21

18 Jun '21

Fix "thse" --> "these" in struct fsi_stream declaration. Signed-off-by: Flavio Suligoi <f.suligoi(a)asem.it> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c934f87c242..cdf3b7f69ba7 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -219,7 +219,7 @@ struct fsi_stream { u32 bus_option; /* - * thse are initialized by fsi_handler_init() + * these are initialized by fsi_handler_init() */ struct fsi_stream_handler *handler; struct fsi_priv *priv; -- 2.25.1

2 1

[PATCH -next] ASoC: rk817: Constify static struct snd_soc_dai_ops
by Wei Yongjun 18 Jun '21

18 Jun '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..9a896e4326c3 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -382,7 +382,7 @@ static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops rk817_dai_ops = { +static const struct snd_soc_dai_ops rk817_dai_ops = { .hw_params = rk817_hw_params, .set_fmt = rk817_set_dai_fmt, .set_sysclk = rk817_set_dai_sysclk,

3 2

[PATCH] soundwire: stream: Use polling for DP Prepare completion
by Richard Fitzgerald 18 Jun '21

18 Jun '21

sdw_prep_deprep_slave_ports() cannot use the port interrupt to signal CP_SM completion because it is called while holding the bus lock, which blocks the alert handler from running. Change to polling the PREPARESTATUS register and remove the infrastructure that was implemented for the previous interrupt waiting. A new configuration field 'ch_prep_poll_us' is added to struct sdw_dpn_prop so that the peripheral driver may select a polling interval. If this is left at zero a default interval of 500 usec is used. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 2 -- drivers/soundwire/slave.c | 4 ---- drivers/soundwire/stream.c | 24 ++++++++++++++---------- include/linux/soundwire/sdw.h | 8 ++++++-- 4 files changed, 20 insertions(+), 18 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index adcbf3969110..606fc26d407f 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1351,7 +1351,6 @@ static int sdw_handle_dp0_interrupt(struct sdw_slave *slave, u8 *slave_status) */ if (status & SDW_DP0_INT_PORT_READY) { - complete(&slave->port_ready[0]); clear |= SDW_DP0_INT_PORT_READY; } @@ -1429,7 +1428,6 @@ static int sdw_handle_port_interrupt(struct sdw_slave *slave, * for ports implementing CP_SM. */ if (status & SDW_DPN_INT_PORT_READY) { - complete(&slave->port_ready[port]); clear |= SDW_DPN_INT_PORT_READY; } diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 669d7573320b..55ca884ea951 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -26,7 +26,6 @@ int sdw_slave_add(struct sdw_bus *bus, { struct sdw_slave *slave; int ret; - int i; slave = kzalloc(sizeof(*slave), GFP_KERNEL); if (!slave) @@ -62,9 +61,6 @@ int sdw_slave_add(struct sdw_bus *bus, slave->probed = false; slave->first_interrupt_done = false; - for (i = 0; i < SDW_MAX_PORTS; i++) - init_completion(&slave->port_ready[i]); - mutex_lock(&bus->bus_lock); list_add_tail(&slave->node, &bus->slaves); mutex_unlock(&bus->bus_lock); diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1eaedaaba094..bd6b3b64de90 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -8,11 +8,13 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/init.h> +#include <linux/iopoll.h> #include <linux/module.h> #include <linux/mod_devicetable.h> #include <linux/slab.h> #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw.h> +#include <linux/time.h> #include <sound/soc.h> #include "bus.h" @@ -419,11 +421,10 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, struct sdw_port_runtime *p_rt, bool prep) { - struct completion *port_ready; struct sdw_dpn_prop *dpn_prop; struct sdw_prepare_ch prep_ch; - unsigned int time_left; bool intr = false; + unsigned long prep_poll_us, prep_timeout_us; int ret = 0, val; u32 addr; @@ -478,16 +479,19 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, } /* Wait for completion on port ready */ - port_ready = &s_rt->slave->port_ready[prep_ch.num]; - time_left = wait_for_completion_timeout(port_ready, - msecs_to_jiffies(dpn_prop->ch_prep_timeout)); + prep_timeout_us = dpn_prop->ch_prep_timeout * USEC_PER_MSEC; + if (dpn_prop->ch_prep_poll_us) + prep_poll_us = dpn_prop->ch_prep_poll_us; + else + prep_poll_us = SDW_DEFAULT_DP_PREP_POLL_US; - val = sdw_read(s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); - val &= p_rt->ch_mask; - if (!time_left || val) { + ret = read_poll_timeout(sdw_read, val, ((val & p_rt->ch_mask) == 0), + prep_poll_us, prep_timeout_us, false, + s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); + if (ret < 0) { dev_err(&s_rt->slave->dev, - "Chn prep failed for port:%d\n", prep_ch.num); - return -ETIMEDOUT; + "Chn prep failed for port %d: %d\n", prep_ch.num, ret); + return ret; } } diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ddbeb00799e4..4e5290b083bf 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -67,6 +67,9 @@ enum { #define SDW_BLOCK_PACKG_PER_PORT BIT(0) #define SDW_BLOCK_PACKG_PER_CH BIT(1) +/* Default interval to poll for DP prepare completion */ +#define SDW_DEFAULT_DP_PREP_POLL_US 500 + /** * enum sdw_slave_status - Slave status * @SDW_SLAVE_UNATTACHED: Slave is not attached with the bus. @@ -295,6 +298,8 @@ struct sdw_dpn_audio_mode { * @simple_ch_prep_sm: If the port supports simplified channel prepare state * machine * @ch_prep_timeout: Port-specific timeout value, in milliseconds + * @ch_prep_poll_us: Interval to poll for CP_SM prepare completion. Zero means + * poll at SDW_DEFAULT_DP_PREP_POLL_US intervals. * @imp_def_interrupts: If set, each bit corresponds to support for * implementation-defined interrupts * @max_ch: Maximum channels supported @@ -321,6 +326,7 @@ struct sdw_dpn_prop { u32 max_grouping; bool simple_ch_prep_sm; u32 ch_prep_timeout; + u32 ch_prep_poll_us; u32 imp_def_interrupts; u32 max_ch; u32 min_ch; @@ -641,7 +647,6 @@ struct sdw_slave_ops { * @prop: Slave properties * @debugfs: Slave debugfs * @node: node for bus list - * @port_ready: Port ready completion flag for each Slave port * @m_port_map: static Master port map for each Slave port * @dev_num: Current Device Number, values can be 0 or dev_num_sticky * @dev_num_sticky: one-time static Device Number assigned by Bus @@ -673,7 +678,6 @@ struct sdw_slave { struct dentry *debugfs; #endif struct list_head node; - struct completion port_ready[SDW_MAX_PORTS]; unsigned int m_port_map[SDW_MAX_PORTS]; u16 dev_num; u16 dev_num_sticky; -- 2.20.1

2 3

[PATCH v3 2/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From f6d2556f8f6bcdd2261b48604b8086fb67ae9a03 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:23:06 +0800 Subject: [PATCH v3 2/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G8 The HP ProBook 445 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- v2 -> v3: no changes. v1 -> v2: no changes. 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 70bc5b11acfd..a1e62b2de167 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8344,6 +8344,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.25.1

1 0

[PATCH v2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH v2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- 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 ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

1 0

[PATCH 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- 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 ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

1 1

[bug report] ASoC: codecs: wcd938x: add capture dapm widgets
by Dan Carpenter 18 Jun '21

18 Jun '21

Hello Srinivas Kandagatla, The patch d5add08fcbce: "ASoC: codecs: wcd938x: add capture dapm widgets" from Jun 9, 2021, leads to the following static checker warning: sound/soc/codecs/wcd938x.c:2189 wcd938x_tx_channel_config() error: uninitialized symbol 'reg'. sound/soc/codecs/wcd938x.c 2165 static void wcd938x_tx_channel_config(struct snd_soc_component *component, 2166 int channel, int mode) 2167 { 2168 int reg, mask; 2169 2170 switch (channel) { 2171 case 0: 2172 reg = WCD938X_ANA_TX_CH2; 2173 mask = WCD938X_HPF1_INIT_MASK; 2174 break; 2175 case 1: 2176 reg = WCD938X_ANA_TX_CH2; 2177 mask = WCD938X_HPF2_INIT_MASK; 2178 break; 2179 case 2: 2180 reg = WCD938X_ANA_TX_CH4; 2181 mask = WCD938X_HPF3_INIT_MASK; 2182 break; 2183 case 3: 2184 reg = WCD938X_ANA_TX_CH4; 2185 mask = WCD938X_HPF4_INIT_MASK; 2186 break; We could add a default case? default: return; 2187 } 2188 2189 snd_soc_component_write_field(component, reg, mask, mode); 2190 } regards, dan carpenter

1 0

[PATCH] ALSA: firewire-motu: fix rx packet format at higher rate for MOTU 828 mk3 Hybrid
by Takashi Sakamoto 18 Jun '21

18 Jun '21

I assumed that the combination of packet formats for MOTU 828 mk3 Hybrid is the same as MOTU 828 mk3 FireWire. However at higher sampling rate, it is different. MOTU 828 mk3 Hybrid has additional 4 dummy data chunks for rx packet. This commit fixes the issue to which I address at a commit f2ac3b839540 ("ALSA: firewire-motu: sequence replay for source packet header"). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 15 ++++++++++++--- sound/firewire/motu/motu.c | 4 ++-- sound/firewire/motu/motu.h | 3 ++- 3 files changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 4e6b0e449ee4..77e61e89770b 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -185,7 +185,7 @@ int snd_motu_protocol_v3_get_clock_source(struct snd_motu *motu, return err; data = be32_to_cpu(reg) & V3_CLOCK_SOURCE_MASK; - if (motu->spec == &snd_motu_spec_828mk3) + if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) return detect_clock_source_828mk3(motu, data, src); else return v3_detect_clock_source(motu, data, src); @@ -284,14 +284,14 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) motu->spec->rx_fixed_pcm_chunks, sizeof(motu->rx_packet_formats.pcm_chunks)); - if (motu->spec == &snd_motu_spec_828mk3) + if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) return detect_packet_formats_828mk3(motu, data); else return 0; } -const struct snd_motu_spec snd_motu_spec_828mk3 = { +const struct snd_motu_spec snd_motu_spec_828mk3_fw = { .name = "828mk3", .protocol_version = SND_MOTU_PROTOCOL_V3, .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | @@ -300,6 +300,15 @@ const struct snd_motu_spec snd_motu_spec_828mk3 = { .rx_fixed_pcm_chunks = {14, 14, 10}, }; +const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { + .name = "828mk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | + SND_MOTU_SPEC_TX_MIDI_3RD_Q, + .tx_fixed_pcm_chunks = {18, 18, 14}, + .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. +}; + const struct snd_motu_spec snd_motu_spec_ultralite_mk3 = { .name = "UltraLiteMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 0cae670d711c..543136578c70 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -156,10 +156,10 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), - SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. - SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3), // Hybrid. + SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), { } diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index a3deabdf9e34..73f36d1be515 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -130,7 +130,8 @@ extern const struct snd_motu_spec snd_motu_spec_traveler; extern const struct snd_motu_spec snd_motu_spec_ultralite; extern const struct snd_motu_spec snd_motu_spec_8pre; -extern const struct snd_motu_spec snd_motu_spec_828mk3; +extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; +extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; extern const struct snd_motu_spec snd_motu_spec_4pre; -- 2.30.2

2 1

[PATCH] ALSA: bebob: fix rx packet format for Yamaha GO44/GO46, Terratec Phase 24/x24
by Takashi Sakamoto 18 Jun '21

18 Jun '21

Below devices reports zero as the number of channels for external output plug with MIDI type: * Yamaha GO44/GO46 * Terratec Phase 24/X24 As a result, rx packet format is invalid and they generate silent sound. This is a regression added in v5.13. This commit fixes the bug, addressed at a commit 1bd1b3be8655 ("ALSA: bebob: perform sequence replay for media clock recovery"). Cc: <stable(a)vger.kernel.org> Fixes: 5c6ea94f2b7c ("ALSA: bebob: detect the number of available MIDI ports") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index e3e23e42add3..8629b14ded76 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -856,6 +856,11 @@ static int detect_midi_ports(struct snd_bebob *bebob, err = avc_bridgeco_get_plug_ch_count(bebob->unit, addr, &ch_count); if (err < 0) break; + // Yamaha GO44, GO46, Terratec Phase 24, Phase x24 reports 0 for the number of + // channels in external output plug 3 (MIDI type) even if it has a pair of physical + // MIDI jacks. As a workaround, assume it as one. + if (ch_count == 0) + ch_count = 1; *midi_ports += ch_count; } @@ -934,12 +939,12 @@ int snd_bebob_stream_discover(struct snd_bebob *bebob) if (err < 0) goto end; - err = detect_midi_ports(bebob, bebob->rx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_IN, + err = detect_midi_ports(bebob, bebob->tx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_IN, plugs[2], &bebob->midi_input_ports); if (err < 0) goto end; - err = detect_midi_ports(bebob, bebob->tx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_OUT, + err = detect_midi_ports(bebob, bebob->rx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_OUT, plugs[3], &bebob->midi_output_ports); if (err < 0) goto end; -- 2.30.2

2 1

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook x360 830 G8
by Jeremy Szu 18 Jun '21

18 Jun '21

The HP EliteBook x360 830 G8 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- 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 dcd9bceace68..ae5963345093 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8330,6 +8330,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.31.1

2 1

[bug report] ASoC: codecs: wcd938x: add capture dapm widgets
by Dan Carpenter 18 Jun '21

18 Jun '21

Hello Srinivas Kandagatla, The patch d5add08fcbce: "ASoC: codecs: wcd938x: add capture dapm widgets" from Jun 9, 2021, leads to the following static checker warning: sound/soc/codecs/wcd938x.c:2092 wcd938x_tx_swr_ctrl() error: uninitialized symbol 'rate'. sound/soc/codecs/wcd938x.c 2064 switch (event) { 2065 case SND_SOC_DAPM_PRE_PMU: 2066 if (strnstr(w->name, "ADC", sizeof("ADC"))) { Use strncmp() here instead of strnstr(). The sizeof() will include the NUL terminator so it's size 4 and you wanted 3. The off by one means that strnstr() will do two memcmp()s. The equivalent of: if (memcmp(w->name, "ADC", 3) == 0) return w->name; if (memcmp(w->name + 1, "ADC", 3) == 0) return w->name + 1; return NULL; 2067 int i = 0, mode = 0; 2068 2069 if (test_bit(WCD_ADC1, &wcd938x->status_mask)) 2070 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC1]]; 2071 if (test_bit(WCD_ADC2, &wcd938x->status_mask)) 2072 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC2]]; 2073 if (test_bit(WCD_ADC3, &wcd938x->status_mask)) 2074 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC3]]; 2075 if (test_bit(WCD_ADC4, &wcd938x->status_mask)) 2076 mode |= tx_mode_bit[wcd938x->tx_mode[WCD_ADC4]]; 2077 2078 if (mode != 0) { 2079 for (i = 0; i < ADC_MODE_ULP2; i++) { 2080 if (mode & (1 << i)) { 2081 i++; 2082 break; 2083 } 2084 } 2085 } 2086 rate = wcd938x_get_clk_rate(i); 2087 wcd938x_set_swr_clk_rate(component, rate, bank); 2088 } 2089 2090 if (strnstr(w->name, "ADC", sizeof("ADC"))) 2091 /* Copy clk settings to active bank */ 2092 wcd938x_set_swr_clk_rate(component, rate, !bank); This a false positive, but you could silence it by combining it with the previous if block. 2093 break; 2094 case SND_SOC_DAPM_POST_PMD: 2095 if (strnstr(w->name, "ADC", sizeof("ADC"))) { 2096 rate = wcd938x_get_clk_rate(ADC_MODE_INVALID); 2097 wcd938x_set_swr_clk_rate(component, rate, !bank); 2098 wcd938x_set_swr_clk_rate(component, rate, bank); 2099 } 2100 break; 2101 } regards, dan carpenter

1 0

[PATCH v3 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 18 Jun '21

18 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. --- v2->v3: None. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 404 ++++++++++++++++++++++++++++++++++ 4 files changed, 420 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d0b576528552 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,404 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH v2 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 18 Jun '21

18 Jun '21

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- .../sound/allwinner,sun50i-h6-dmic.yaml | 68 +++++++++++++++++++ 1 file changed, 68 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000000..d7af66775e86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,68 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + dmic: dmic@5095000 { + #sound-dai-cells = <1>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + clocks = <&ccu 101>, <&ccu 100>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu 42>; + }; + +... -- 2.29.0

2 1

[PATCH 1/1] ALSA: aoa: remove unnecessary oom message
by Zhen Lei 17 Jun '21

17 Jun '21

Fixes scripts/checkpatch.pl warning: WARNING: Possible unnecessary 'out of memory' message Remove it can help us save a bit of memory. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/aoa/soundbus/i2sbus/pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 1c8e8131a716..a9e502a6cdeb 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -918,10 +918,8 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, } cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); - if (!cii) { - printk(KERN_DEBUG "i2sbus: failed to allocate cii\n"); + if (!cii) return -ENOMEM; - } /* use the private data to point to the codec info */ cii->sdev = soundbus_dev_get(dev); -- 2.25.1

2 1

[PATCH 1/1] ASoC: soc-core: Fix the error return code in snd_soc_of_parse_audio_routing()
by Zhen Lei 17 Jun '21

17 Jun '21

When devm_kcalloc() fails, the error code -ENOMEM should be returned instead of -EINVAL. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9e7f4ceffb65..dbf7de925f8d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2957,7 +2957,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, if (!routes) { dev_err(card->dev, "ASoC: Could not allocate DAPM route table\n"); - return -EINVAL; + return -ENOMEM; } for (i = 0; i < num_routes; i++) { -- 2.25.1

2 1

[PATCH -next] ASoC: spear: spdif_out: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/spear/spdif_out.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 38f9fff5be6b..549295a6ed50 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -287,8 +287,7 @@ static int spdif_out_probe(struct platform_device *pdev) if (!host) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - host->io_base = devm_ioremap_resource(&pdev->dev, res); + host->io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(host->io_base)) return PTR_ERR(host->io_base); -- 2.25.1

2 1

[PATCH -next 0/3] ASoC: stm32: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Yang Yingliang (3): ASoC: stm32: i2s: Use devm_platform_get_and_ioremap_resource() ASoC: stm32: sai: Use devm_platform_get_and_ioremap_resource() ASoC: stm32: spdifrx: Use devm_platform_get_and_ioremap_resource() sound/soc/stm/stm32_i2s.c | 3 +-- sound/soc/stm/stm32_sai_sub.c | 3 +-- sound/soc/stm/stm32_spdifrx.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) -- 2.25.1

2 4

[PATCH -next] ASoC: sti: sti_uniperif: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sti/sti_uniperif.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index e3561f00ed40..34668fe3909d 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -410,16 +410,8 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, *dai = sti_uniperiph_dai_template; dai->name = dev_data->dai_names; - /* Get resources */ - uni->mem_region = platform_get_resource(priv->pdev, IORESOURCE_MEM, 0); - - if (!uni->mem_region) { - dev_err(dev, "Failed to get memory resource\n"); - return -ENODEV; - } - - uni->base = devm_ioremap_resource(dev, uni->mem_region); - + /* Get resources and base address */ + uni->base = devm_platform_get_and_ioremap_resource(priv->pdev, 0, &uni->mem_region); if (IS_ERR(uni->base)) return PTR_ERR(uni->base); -- 2.25.1

2 1

[PATCH -next] ASoC: sprd: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 17 Jun '21

17 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 34b2ce733b54..f6a55fa60c1b 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -949,8 +949,7 @@ static int sprd_mcdt_probe(struct platform_device *pdev) if (!mcdt) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - mcdt->base = devm_ioremap_resource(&pdev->dev, res); + mcdt->base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(mcdt->base)) return PTR_ERR(mcdt->base); -- 2.25.1

2 1

[PATCH] ALSA: seq: oss: Fix error check at system port creation
by Takashi Iwai 17 Jun '21

17 Jun '21

The system port creation in ALSA OSS sequencer was wrongly checked against to the port number that can be never negative. The error code should be checked rather against the ioctl call. This patch corrects the error check. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_init.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index a53d81a86af2..0ee4a5081fd6 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -94,11 +94,10 @@ snd_seq_oss_create_client(void) port_callback.event_input = receive_announce; port->kernel = &port_callback; - call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, port); - system_port = port->addr.port; - if (system_port >= 0) { + if (call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, port) >= 0) { struct snd_seq_port_subscribe subs; + system_port = port->addr.port; memset(&subs, 0, sizeof(subs)); subs.sender.client = SNDRV_SEQ_CLIENT_SYSTEM; subs.sender.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; -- 2.26.2

1 0

[bug report] ALSA: seq: Fix assignment in if condition
by Dan Carpenter 17 Jun '21

17 Jun '21

Hello Takashi Iwai, The patch f9a6bb841f73: "ALSA: seq: Fix assignment in if condition" from Jun 8, 2021, leads to the following static checker warning: sound/core/seq/oss/seq_oss_init.c:99 snd_seq_oss_create_client() warn: 'port->addr.port' is unsigned sound/core/seq/oss/seq_oss_init.c 82 83 /* create annoucement receiver port */ 84 memset(port, 0, sizeof(*port)); 85 strcpy(port->name, "Receiver"); 86 port->addr.client = system_client; 87 port->capability = SNDRV_SEQ_PORT_CAP_WRITE; /* receive only */ 88 port->type = 0; 89 90 memset(&port_callback, 0, sizeof(port_callback)); 91 /* don't set port_callback.owner here. otherwise the module counter 92 * is incremented and we can no longer release the module.. 93 */ 94 port_callback.event_input = receive_announce; 95 port->kernel = &port_callback; 96 97 call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, port); 98 system_port = port->addr.port; 99 if (system_port >= 0) { ^^^^^^^^^^^^^^^^ This was from the old code. It's not clear what is going on. I think the condition can be deleted. 100 struct snd_seq_port_subscribe subs; 101 102 memset(&subs, 0, sizeof(subs)); 103 subs.sender.client = SNDRV_SEQ_CLIENT_SYSTEM; 104 subs.sender.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; 105 subs.dest.client = system_client; 106 subs.dest.port = system_port; 107 call_ctl(SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &subs); 108 } 109 rc = 0; 110 111 /* look up midi devices */ 112 schedule_work(&async_lookup_work); 113 114 __error: 115 kfree(port); 116 return rc; 117 } regards, dan carpenter

2 1

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.fix some compilation errors. 2.Modify some code styles. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.fix some compilation errors. 2.Modify some code styles. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

a52 plugin crashing pulseaudio
by GitHub issues - edited 17 Jun '21

17 Jun '21

alsa-project/alsa-plugins issue #28 was edited from quequotion: Ever since f11e7a8994111f971bc881d4069f768472452f4f, the recent commit that brings the a52 plugin up to speed with ffmpeg's requirement for s32le format sampling, sending any audio to plug:a52 from any pulseaudio client crashes pulseaudio. I [posted this log](https://gist.github.com/quequotion/78972bac339f5249cf6b383519cdcedf) earlier in #23, in which audio from mpv crashes pulseaudio. I have tried other clients since and they all crash the same way. This is with a default pulseaudio configuration, as shipped in Archlinux, and the following `/etc/asound.conf`: ``` # Alsa Resampler defaults.pcm.rate_converter "speexrate_best" # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } defaults.namehint.showall on defaults.namehint.extended on ``` To enable the use of plug:a52 in pulseaudio without configuring it manually, I select the "Digital Surround 5.1 (IEC958/AC3) Output" profile in `pavucontrol`. Pulseaudio, as far as I know, is agnostic of how ALSA plugins work internally. This leads me to think there is something not getting done right in the new patchset, even if it is tolerable in an ALSA-only setup. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/28 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

a52 plugin crashing pulseaudio
by GitHub issues - edited 17 Jun '21

17 Jun '21

alsa-project/alsa-plugins issue #28 was edited from quequotion: Ever since f11e7a8994111f971bc881d4069f768472452f4f, the recent commit that brings the a52 plugin up to speed with ffmpeg's requirement for s32le format sampling, sending any audio via plug:a52 from any pulseaudio client crashes pulseaudio. I [posted this log](https://gist.github.com/quequotion/78972bac339f5249cf6b383519cdcedf) earlier in #23, in which audio from mpv crashes pulseaudio. I have tried other clients since and they all crash the same way. This is with a default pulseaudio configuration, as shipped in Archlinux, and the following `/etc/asound.conf`: ``` # Alsa Resampler defaults.pcm.rate_converter "speexrate_best" # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } defaults.namehint.showall on defaults.namehint.extended on ``` To enable the use of plug:a52 in pulseaudio without configuring it manually, I select the "Digital Surround 5.1 (IEC958/AC3) Output" profile in `pavucontrol`. Pulseaudio, as far as I know, is agnostic of how ALSA plugins work internally. This leads me to think there is something not getting done right in the new patchset, even if it is tolerable in an ALSA-only setup. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/28 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH v2 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 17 Jun '21

17 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..d495ee0f6a2d --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

1 0

[PATCH -next] ASoC: tegra: Fix missing clk_disable_unprepare() on error path
by Zou Wei 17 Jun '21

17 Jun '21

Fix the missing clk_disable_unprepare() before return from tegra_machine_hw_params() in the error handling case. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/tegra/tegra_asoc_machine.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index a53aec3..397f326 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -306,6 +306,7 @@ static int tegra_machine_hw_params(struct snd_pcm_substream *substream, err = snd_soc_dai_set_sysclk(codec_dai, clk_id, mclk, SND_SOC_CLOCK_IN); if (err < 0) { + clk_disable_unprepare(machine->clk_cdev1); dev_err(card->dev, "codec_dai clock not set: %d\n", err); return err; } @@ -523,8 +524,10 @@ int tegra_asoc_machine_probe(struct platform_device *pdev) } err = devm_snd_soc_register_card(dev, card); - if (err) + if (err) { + clk_disable_unprepare(machine->clk_cdev1); return err; + } return 0; } -- 2.6.2

2 1

[PATCH 1/2] ASoC: rt711: remap buttons
by shumingf＠realtek.com 17 Jun '21

17 Jun '21

From: Shuming Fan <shumingf(a)realtek.com> This patch uses the same mapping as the machine driver: BTN_0 : KEY_PLAYPAUSE BTN_1 : KEY_VOICECOMMAND BTN_2 : KEY_VOLUMEUP BTN_3 : KEY_VOLUMEDOWN Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 9f5b2dc16c54..c3a82b462a93 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -154,26 +154,26 @@ static unsigned int rt711_button_detect(struct rt711_priv *rt711) case 0x0200: case 0x0100: case 0x0080: - btn_type |= SND_JACK_BTN_0; + btn_type |= SND_JACK_BTN_2; break; case 0x0001: - btn_type |= SND_JACK_BTN_3; + btn_type |= SND_JACK_BTN_1; break; } switch (val81) { case 0x8000: case 0x4000: case 0x2000: - btn_type |= SND_JACK_BTN_1; + btn_type |= SND_JACK_BTN_0; break; case 0x1000: case 0x0800: case 0x0400: - btn_type |= SND_JACK_BTN_2; + btn_type |= SND_JACK_BTN_3; break; case 0x0200: case 0x0100: - btn_type |= SND_JACK_BTN_3; + btn_type |= SND_JACK_BTN_1; break; } read_error: -- 2.31.1

2 1

a52 plugin crashing pulseaudio
by GitHub issues - opened 17 Jun '21

17 Jun '21

alsa-project/alsa-plugins issue #28 was opened from quequotion: Ever since f11e7a8994111f971bc881d4069f768472452f4f, the recent commit that brings the a52 plugin up to speed with ffmpeg's requirement for s32le format sampling, sending any audio via plug:a52 from any pulseaudio client crashes pulseaudio. I [posted this log](https://gist.github.com/quequotion/78972bac339f5249cf6b383519cdcedf) earlier in #23, in which audio from mpv crashes pulseaudio. I have tried other clients since and they all crash the same way. This is with a default pulseaudio configuration, as shipped in Archlinux, and the following `/etc/asound.conf`: ``` # Alsa Resampler defaults.pcm.rate_converter "speexrate_best" # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } defaults.namehint.showall on defaults.namehint.extended on ``` To enable the use of plug:52 in pulseaudio without configuring it manually, I select the "Digital Surround 5.1 (IEC958/AC3) Output" profile in `pavucontrol`. Pulseaudio, as far as I know, is agnostic of how ALSA plugins work internally. This leads me to think there is something not getting done right in the new patchset, even if it is tolerable in an ALSA-only setup. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/28 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH -next] ASoC: tegra: Add missing of_node_put() in tegra_machine_parse_phandle()
by Zou Wei 17 Jun '21

17 Jun '21

The function is missing a of_node_put on node, fix this by adding the call before returning. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/tegra/tegra_asoc_machine.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index 397f326..cba55ca 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -336,9 +336,12 @@ tegra_machine_parse_phandle(struct device *dev, const char *name) } err = devm_add_action_or_reset(dev, tegra_machine_node_release, np); - if (err) + if (err) { + of_node_put(np); return ERR_PTR(err); + } + of_node_put(np); return np; } -- 2.6.2

2 1

[PATCH] ALSA: firewire-motu: fix stream format for MOTU 8pre FireWire
by Takashi Sakamoto 17 Jun '21

17 Jun '21

My previous refactoring for ALSA firewire-motu driver brought regression to handle MOTU 8pre FireWire. The packet format is not operated correctly. Cc: <stable(a)vger.kernel.org> Fixes: dfbaa4dc11eb ("ALSA: firewire-motu: add model-specific table of chunk count") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v2.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index e59e69ab1538..784073aa1026 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -353,6 +353,7 @@ const struct snd_motu_spec snd_motu_spec_8pre = { .protocol_version = SND_MOTU_PROTOCOL_V2, .flags = SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_2ND_Q, - .tx_fixed_pcm_chunks = {10, 6, 0}, - .rx_fixed_pcm_chunks = {10, 6, 0}, + // Two dummy chunks always in the end of data block. + .tx_fixed_pcm_chunks = {10, 10, 0}, + .rx_fixed_pcm_chunks = {6, 6, 0}, }; -- 2.30.2

2 1

[PATCH alsa-plugins 0/2] Support multiple formats
by Takashi Iwai 17 Jun '21

17 Jun '21

Hi, this is a patch set to add the multi-format support for the existing rate plugins. Takashi === Takashi Iwai (2): rate-lav: Support multiple formats samplerate: Support multiple formats rate-lav/rate_lavrate.c | 115 ++++++++++++++++++++++++++++++++++------ rate/rate_samplerate.c | 99 ++++++++++++++++++++++++++++------ 2 files changed, 183 insertions(+), 31 deletions(-) -- 2.26.2

1 2

[PATCH 0/2] ALSA: firewire-motu: add support for MOTU 828 and 896
by Takashi Sakamoto 17 Jun '21

17 Jun '21

Hi, This patchset is to add support for Mark of the unicorn (MOTU) 828 and 896, which were shipped two decades ago (2001) and already discontinued. (yes, it's 2021). It's reasonable to count them as first generation of MOTU FireWire series since they are based on QuickLogic QuickRAM FPGA apart from Altera or Xilinx FPGA in the latter generation. Unlike the latter generation, they doesn't allow software to configure internal multiplexer. I note that the sequence replay for media clock recovery[1] allows to support them. The device doesn't generate better sound with nominal (ideal) sequence of packets, thus the patches should be applied to development tree for v5.14 kernel. [1] [PATCH 0/3] ALSA: firewire-motu: media clock recovery for sph-aware devices https://lore.kernel.org/alsa-devel/20210602013406.26442-1-o-takashi@sakamoc… Takashi Sakamoto (2): ALSA: firewire-motu: add support for MOTU 828 ALSA: firewire-motu: add support for MOTU 896 sound/firewire/Kconfig | 2 + sound/firewire/motu/Makefile | 3 +- sound/firewire/motu/motu-protocol-v1.c | 440 +++++++++++++++++++++++++ sound/firewire/motu/motu.c | 2 + sound/firewire/motu/motu.h | 24 ++ 5 files changed, 470 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/motu/motu-protocol-v1.c -- 2.30.2

2 4

[PATCH alsa-lib 0/2] Allow multiple formats for PCM rate plugin
by Takashi Iwai 17 Jun '21

17 Jun '21

Hi, this is a patch set for alsa-lib to extend the PCM rate plugin for allowing multiple formats for its converter plugins. This optimizes and improves the quality (skips the unnecessary buffer and conversion) when the converter supports the source and/or the target format natively. The patches for libswresample and libsamplerate will follow. Takashi === Takashi Iwai (2): pcm: rate: Refactoring temporary buffer allocations pcm: rate: Improve the support multiple formats include/pcm_rate.h | 36 +++- src/pcm/pcm_rate.c | 424 +++++++++++++++++++++++++++------------------ 2 files changed, 293 insertions(+), 167 deletions(-) -- 2.26.2

1 2

ALSA: usb-audio: fix rate on Ozone Z90 USB headset
by Daehwan Jung 17 Jun '21

17 Jun '21

It mislabels its 96 kHz altsetting and that's why it causes some noise Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Daehwan Jung <dh10.jung(a)samsung.com> --- sound/usb/format.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 2287f8c..eb216fe 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -223,9 +223,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ /* Terratec Aureon 7.1 USB C-Media 6206, too */ + /* Ozone Z90 USB C-Media, too */ if (rate == 48000 && nr_rates == 1 && (chip->usb_id == USB_ID(0x0d8c, 0x0201) || chip->usb_id == USB_ID(0x0d8c, 0x0102) || + chip->usb_id == USB_ID(0x0d8c, 0x0078) || chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; -- 2.7.4

3 2

HDA-Intel: add Boost volume control for Headset Mic
by GitHub pull_request - opened 17 Jun '21

17 Jun '21

alsa-project/alsa-ucm-conf pull request #101 was opened from jason77-wang: Ubuntu Linux users complain even they set the input volume to maximum, they still can't record sound with a pretty satisfied volume, they need to run commandline commands to adjust "Mic Boost/Headset Mic Boost/Headphone Mic boost" value to increase the input volume. That is because the current ucm only defines "Capture Volume" to control the input volume. Here we add Boost control, then users could adjust both "Capture Volume" and "Boost Volume" through Pulseaudio and Gnome. BugLink: https://bugs.launchpad.net/bugs/1930188 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/101 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/101.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: max98357a: set channels_max to 4
by Judy Hsiao 16 Jun '21

16 Jun '21

Sets channels_max to 4 to support QUAD channel. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/codecs/max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 918812763884..c3cf1743caad 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -117,7 +117,7 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { .rate_min = 8000, .rate_max = 96000, .channels_min = 1, - .channels_max = 2, + .channels_max = 4, }, .ops = &max98357a_dai_ops, }; -- 2.31.0

5 6

[PATCH -next 1/3] ASoC: rockchip: i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 16 Jun '21

16 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/rockchip/rockchip_i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0740764e7f71..c7dc3509bceb 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -618,8 +618,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) return PTR_ERR(i2s->mclk); } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - regs = devm_ioremap_resource(&pdev->dev, res); + regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(regs)) return PTR_ERR(regs); -- 2.25.1

2 3

[PATCH -next] SoC: codecs: wcd938x: fix boolreturn.cocci warning
by Yang Li 16 Jun '21

16 Jun '21

Return statements in functions returning bool should use true/false instead of 1/0. Fix the following coccicheck warning: ./sound/soc/codecs/wcd938x.c:1190:9-10: WARNING: return of 0/1 in function 'wcd938x_volatile_register' with return type bool. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- 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 2cf6145..cfe00536 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1187,7 +1187,7 @@ static bool wcd938x_writeable_register(struct device *dev, unsigned int reg) static bool wcd938x_volatile_register(struct device *dev, unsigned int reg) { if (reg <= WCD938X_BASE_ADDRESS) - return 0; + return false; if (reg == WCD938X_DIGITAL_SWR_TX_CLK_RATE) return true; -- 1.8.3.1

2 1

[PATCH] ASoC: codecs: Fix duplicate included sound/soc.h
by Jiapeng Chong 16 Jun '21

16 Jun '21

Clean up the following includecheck warnings: ./sound/soc/codecs/wcd938x.c: sound/soc.h is included more than once. ./sound/soc/codecs/wcd938x-sdw.c: sound/soc.h is included more than once. No functional change. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/wcd938x-sdw.c | 1 - sound/soc/codecs/wcd938x.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index d82c40e..88334d3 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -7,7 +7,6 @@ #include <linux/device.h> #include <linux/kernel.h> #include <linux/component.h> -#include <sound/soc.h> #include <linux/pm_runtime.h> #include <linux/irqdomain.h> #include <linux/of.h> diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 2cf6145..c0d7452 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -9,7 +9,6 @@ #include <linux/kernel.h> #include <linux/pm_runtime.h> #include <linux/component.h> -#include <sound/soc.h> #include <sound/tlv.h> #include <linux/of_gpio.h> #include <linux/of.h> -- 1.8.3.1

2 1

[PATCH] ASoC: hdmi-codec: make hdmi_codec_controls static
by Jiapeng Chong 16 Jun '21

16 Jun '21

This symbol is not used outside of hdmi-codec.c, so marks it static. Fix the following sparse warning: sound/soc/codecs/hdmi-codec.c:750:25: warning: symbol 'hdmi_codec_controls' was not declared. Should it be static? Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e0460a9..b61f980 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -747,7 +747,7 @@ static int hdmi_codec_mute(struct snd_soc_dai *dai, int mute, int direction) SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |\ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -struct snd_kcontrol_new hdmi_codec_controls[] = { +static struct snd_kcontrol_new hdmi_codec_controls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, -- 1.8.3.1

2 1

[PATCH] ASoC: cs42l42: Correct definition of CS42L42_ADC_PDN_MASK
by Richard Fitzgerald 16 Jun '21

16 Jun '21

The definition of CS42L42_ADC_PDN_MASK was incorrectly defined as the HP_PDN bit. Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 7bf05ff05f74..206b3c81d3e0 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -79,7 +79,7 @@ #define CS42L42_HP_PDN_SHIFT 3 #define CS42L42_HP_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) #define CS42L42_ADC_PDN_SHIFT 2 -#define CS42L42_ADC_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) +#define CS42L42_ADC_PDN_MASK (1 << CS42L42_ADC_PDN_SHIFT) #define CS42L42_PDN_ALL_SHIFT 0 #define CS42L42_PDN_ALL_MASK (1 << CS42L42_PDN_ALL_SHIFT) -- 2.20.1

2 1

[PATCH v2] media: Fix Media Controller API config checks
by Shuah Khan 16 Jun '21

16 Jun '21

Smatch static checker warns that "mdev" can be null: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL If CONFIG_MEDIA_CONTROLLER is disabled, this file should not be included in the build. The below conditions in the sound/usb/Makefile are in place to ensure that media.c isn't included in the build. sound/usb/Makefile: snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) The following config check in include/media/media-dev-allocator.h is in place to enable the API only when CONFIG_MEDIA_CONTROLLER and CONFIG_USB are enabled. #if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) This check doesn't work as intended when CONFIG_USB=m. When CONFIG_USB=m, CONFIG_USB_MODULE is defined and CONFIG_USB is not. The above config check doesn't catch that CONFIG_USB is defined as a module and disables the API. This results in sound/usb enabling Media Controller specific ALSA driver code, while Media disables the Media Controller API. Fix the problem requires two changes: 1. Change the check to use IS_ENABLED to detect when CONFIG_USB is enabled as a module or static. Since CONFIG_MEDIA_CONTROLLER is a bool, leave the check unchanged to be consistent with drivers/media/Makefile. 2. Change the drivers/media/mc/Makefile to include mc-dev-allocator.o in mc-objs when CONFIG_USB is enabled. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/alsa-devel/YLeAvT+R22FQ%2FEyw@mwanda/ Signed-off-by: Shuah Khan <skhan(a)linuxfoundation.org> --- Changes since v1: - Fixed the drivers/media/mc/Makefile incorrect logic to check CONFIG_USB value. Test with CONFIG_USB_SUPPORT enabled/disabled CONFIG_USB y/m and disabled. drivers/media/mc/Makefile | 2 +- include/media/media-dev-allocator.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/media/mc/Makefile b/drivers/media/mc/Makefile index 119037f0e686..2b7af42ba59c 100644 --- a/drivers/media/mc/Makefile +++ b/drivers/media/mc/Makefile @@ -3,7 +3,7 @@ mc-objs := mc-device.o mc-devnode.o mc-entity.o \ mc-request.o -ifeq ($(CONFIG_USB),y) +ifneq ($(CONFIG_USB),) mc-objs += mc-dev-allocator.o endif diff --git a/include/media/media-dev-allocator.h b/include/media/media-dev-allocator.h index b35ea6062596..2ab54d426c64 100644 --- a/include/media/media-dev-allocator.h +++ b/include/media/media-dev-allocator.h @@ -19,7 +19,7 @@ struct usb_device; -#if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) +#if defined(CONFIG_MEDIA_CONTROLLER) && IS_ENABLED(CONFIG_USB) /** * media_device_usb_allocate() - Allocate and return struct &media device * -- 2.30.2

1 0

HDSPconf don't recognize RME RayDat
by GitHub issues - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-tools issue #6 was opened from profejuanca: Hi, I'm using a RME Raydat (PCI express) many years ago, but HDSPconf don't recognize my card never, hdspmixer and the rest of jack/alsa tools see my card without problems, but hdspconf in various distros never sees my card. The output is: `HDSPConf 1.4 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org> This program comes WITH ABSOLUTELY NO WARRANTY HDSPConf is free software, see the file copying for details Looking for HDSP cards : Card 0 : RME RayDAT S/N 0x3250cb at 0xfe300000, irq 57 Card 1 : HDA ATI SB at 0xfe800000 irq 16 Card 2 : HDA NVidia at 0xfe080000 irq 60 No Hammerfall DSP card found. ` Card 0 es my RME card... I suposse that raydat is not in the list of the tool. Issue URL : https://github.com/alsa-project/alsa-tools/issues/6 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH 00/10] ASoC: Intel: machine driver corrections
by Pierre-Louis Bossart 16 Jun '21

16 Jun '21

Some of these patches dealing with Sparse warnings were submitted earlier but not merged. I grouped them with a couple of fixes from Kai. Kai Vehmanen (3): ASoC: Intel: sof_sdw: use mach data for ADL RVP DMIC count ASoC: Intel: sof_sdw: remove hdac-hdmi support ASoC: Intel: skl_hda_dsp_generic: Update Kconfig documentation Pierre-Louis Bossart (7): ASoC: Intel: sof_rt5682: shrink platform_id names below 20 characters ASoC: Intel: glk_rt5682_max98357a: shrink platform_id below 20 characters ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters ASoC: Intel: sof_sdw: fix signed/unsigned warning ASoC: Intel: soc-acpi: add ull suffix for SoundWire _ADR values ASoC: Intel: use MODULE_DEVICE_TABLE with platform_device_id tables sound/soc/intel/boards/Kconfig | 7 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 3 +- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 5 +-- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 3 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 3 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 3 +- sound/soc/intel/boards/skl_rt286.c | 3 +- sound/soc/intel/boards/sof_cs42l42.c | 3 +- sound/soc/intel/boards/sof_da7219_max98373.c | 6 +-- sound/soc/intel/boards/sof_rt5682.c | 20 +++------- sound/soc/intel/boards/sof_sdw.c | 5 +-- sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/intel/boards/sof_sdw_hdmi.c | 37 +----------------- .../intel/common/soc-acpi-intel-adl-match.c | 28 +++++++------- .../intel/common/soc-acpi-intel-cml-match.c | 20 +++++----- .../intel/common/soc-acpi-intel-cnl-match.c | 2 +- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- .../intel/common/soc-acpi-intel-icl-match.c | 12 +++--- .../intel/common/soc-acpi-intel-jsl-match.c | 6 +-- .../intel/common/soc-acpi-intel-kbl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 38 +++++++++---------- 27 files changed, 83 insertions(+), 144 deletions(-) -- 2.25.1

2 11

alsa-lib errors on boot
by GitHub issues - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-lib issue #159 was opened from miquecg: ``` alsactl[769]: alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HD-Audio Generic at 0xfd3c8000 irq 108) alsactl[769]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -6 alsactl[769]: alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HD-Audio Generic at 0xfd3c0000 irq 109) alsactl[769]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:1 use case configuration -6 ``` This is happening on a ThinkPad T14 AMD running Arch Linux and `alsa-lib` 1.2.5.1. Previous version also had this issue. Issue URL : https://github.com/alsa-project/alsa-lib/issues/159 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 16 Jun '21

16 Jun '21

The Allwinner SoCs feature an I2S controller across multiple SoC generations. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- .../sound/allwinner,sun50i-h6-dmic.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000000..81d40c374e44 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Controller Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: dmic + + dmas: + maxItems: 1 + + dma-names: + const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + spdif: spdif@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "apb", "dmic"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... -- 2.29.0

4 4

[PATCH] ALSA: oss: Remove redundant assignment to src_access
by Jiapeng Chong 16 Jun '21

16 Jun '21

Variable src_access is set to 'dst_access', but this value is never read as it is not used later on, hence it is a redundant assignment and can be removed. Clean up the following clang-analyzer warning: sound/core/oss/pcm_plugin.c:472:4: warning: Value stored to 'src_access' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/core/oss/pcm_plugin.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 061ba06..a7c0e50 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -469,7 +469,6 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, return err; } srcformat = tmpformat; - src_access = dst_access; } tmpformat.rate = dstformat.rate; err = snd_pcm_plugin_build_rate(plug, -- 1.8.3.1

2 1

ALSA: usb-audio: Fix rate on Ozone Z90 USB headset
by Daehwan Jung 16 Jun '21

16 Jun '21

It mislabels its 96 kHz altsetting and that's why it causes some noise Signed-off-by: Daehwan Jung <dh10.jung(a)samsung.com> --- sound/usb/format.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 2287f8c..eb216fe 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -223,9 +223,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ /* Terratec Aureon 7.1 USB C-Media 6206, too */ + /* Ozone Z90 USB C-Media, too */ if (rate == 48000 && nr_rates == 1 && (chip->usb_id == USB_ID(0x0d8c, 0x0201) || chip->usb_id == USB_ID(0x0d8c, 0x0102) || + chip->usb_id == USB_ID(0x0d8c, 0x0078) || chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; -- 2.7.4

2 1

ThinkPad T14 AMD microphone LED always ON
by GitHub issues - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-ucm-conf issue #100 was opened from miquecg: I'm running Arch Linux with `alsa-ucm-conf` 1.2.5.1. This problem was present in previous versions and it's being discussed on a [forum thread](https://bbs.archlinux.org/viewtopic.php?pid=1977273) among other things. So the problem is that microphone LED is always ON but mute/unmute behaviour still works. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/100 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

rate-lav: Convert to libswresample
by GitHub pull_request - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-plugins pull request #27 was opened from tiwai: The libavresample has been deprecated. Convert to the new API for libswresample. The phase shift and cutoff seem to be either redundant or not-working properly, so those are dropped. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Request URL : https://github.com/alsa-project/alsa-plugins/pull/27 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/27.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH -next] ASoC: hdmi-codec: Make symbol 'hdmi_codec_controls' static
by Wei Yongjun 16 Jun '21

16 Jun '21

The sparse tool complains as follows: sound/soc/codecs/hdmi-codec.c:750:25: warning: symbol 'hdmi_codec_controls' was not declared. Should it be static? This symbol is not used outside of hdmi-codec.c, so marks it static. Fixes: 366b45b97448 ("ASoC: hdmi-codec: Rework to support more controls") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e0460a9bd17a..b61f980cabdc 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -747,7 +747,7 @@ static const struct snd_soc_dai_ops hdmi_codec_spdif_dai_ops = { SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |\ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -struct snd_kcontrol_new hdmi_codec_controls[] = { +static struct snd_kcontrol_new hdmi_codec_controls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM,

2 1

[PATCH -next] ALSA: fm801: Fix missing pci_release_regions() on error in snd_fm801_create()
by Yang Yingliang 16 Jun '21

16 Jun '21

Fix the missing pci_release_regions() before return from snd_fm801_create() in the error handling case. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/pci/fm801.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index ed9dae87145b..adac9c6a67f0 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1254,6 +1254,7 @@ static int snd_fm801_create(struct snd_card *card, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); snd_fm801_free(chip); + pci_release_regions(pci); return -EBUSY; } chip->irq = pci->irq; @@ -1266,6 +1267,7 @@ static int snd_fm801_create(struct snd_card *card, err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_fm801_free(chip); + pci_release_regions(pci); return err; } @@ -1273,6 +1275,7 @@ static int snd_fm801_create(struct snd_card *card, err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_fm801_free(chip); + pci_release_regions(pci); return err; } chip->tea.v4l2_dev = &chip->v4l2_dev; @@ -1285,6 +1288,7 @@ static int snd_fm801_create(struct snd_card *card, if (snd_tea575x_init(&chip->tea, THIS_MODULE)) { dev_err(card->dev, "TEA575x radio not found\n"); snd_fm801_free(chip); + pci_release_regions(pci); return -ENODEV; } } else if ((chip->tea575x_tuner & TUNER_TYPE_MASK) == 0) { -- 2.25.1

2 2

[PATCH][next] ALSA: bebob: Fix bit flag quirk constants
by Colin King 16 Jun '21

16 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The quirking bit-flags are currently set as contiguous integer enum values and so currently SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC is zero and so he quirking never getting set or tested correctly for this quirk. Fix this by setting the quirking constants as shifted bit values. Addresses-Coverity: ("Bitwise-and with zero") Fixes: 93cd12d6e88a ("ALSA: bebob: code refactoring for model-dependent quirks") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/firewire/bebob/bebob.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index c06579d9380e..4d73ecb30d79 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -76,8 +76,8 @@ struct snd_bebob_spec { }; enum snd_bebob_quirk { - SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC, - SND_BEBOB_QUIRK_WRONG_DBC, + SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC = (1 << 0), + SND_BEBOB_QUIRK_WRONG_DBC = (1 << 1), }; struct snd_bebob { -- 2.31.1

3 2

[PATCH -next] ALSA: ppc: fix error return code in snd_pmac_probe()
by Yang Yingliang 16 Jun '21

16 Jun '21

If snd_pmac_tumbler_init() or snd_pmac_tumbler_post_init() fails, snd_pmac_probe() need return error code. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/ppc/powermac.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 1d2865c43d4b..db414b61157e 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -79,7 +79,11 @@ static int snd_pmac_probe(struct platform_device *devptr) sprintf(card->shortname, "PowerMac %s", name_ext); sprintf(card->longname, "%s (Dev %d) Sub-frame %d", card->shortname, chip->device_id, chip->subframe); - if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0) + err = snd_pmac_tumbler_init(chip); + if (err < 0) + goto __error; + err = snd_pmac_tumbler_post_init(); + if (err < 0) goto __error; break; case PMAC_AWACS: -- 2.25.1

2 1

Fix a52 for the recent avcodec
by GitHub pull_request - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-plugins pull request #26 was opened from tiwai: Request URL : https://github.com/alsa-project/alsa-plugins/pull/26 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/26.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] media: Fix Media Controller API config checks
by Shuah Khan 16 Jun '21

16 Jun '21

Smatch static checker warns that "mdev" can be null: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL If CONFIG_MEDIA_CONTROLLER is disabled, this file should not be included in the build. The below conditions in the sound/usb/Makefile are in place to ensure that media.c isn't included in the build. sound/usb/Makefile: snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) The following config check in include/media/media-dev-allocator.h is in place to enable the API only when CONFIG_MEDIA_CONTROLLER and CONFIG_USB are enabled. #if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) This check doesn't work as intended when CONFIG_USB=m. When CONFIG_USB=m, CONFIG_USB_MODULE is defined and CONFIG_USB is not. The above config check doesn't catch that CONFIG_USB is defined as a module and disables the API. This results in sound/usb enabling Media Controller specific ALSA driver code, while Media disables the Media Controller API. Fix the problem requires two changes: 1. Change the check to use IS_ENABLED to detect when CONFIG_USB is enabled as a module or static. Since CONFIG_MEDIA_CONTROLLER is a bool, leave the check unchanged to be consistent with drivers/media/Makefile. 2. Change the drivers/media/mc/Makefile to include mc-dev-allocator.o in mc-objs when CONFIG_USB is y or m. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/alsa-devel/YLeAvT+R22FQ%2FEyw@mwanda/ Signed-off-by: Shuah Khan <skhan(a)linuxfoundation.org> --- drivers/media/mc/Makefile | 2 +- include/media/media-dev-allocator.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/media/mc/Makefile b/drivers/media/mc/Makefile index 119037f0e686..140f0a78540e 100644 --- a/drivers/media/mc/Makefile +++ b/drivers/media/mc/Makefile @@ -3,7 +3,7 @@ mc-objs := mc-device.o mc-devnode.o mc-entity.o \ mc-request.o -ifeq ($(CONFIG_USB),y) +ifeq ($(CONFIG_USB),$(filter $(CONFIG_USB),y m)) mc-objs += mc-dev-allocator.o endif diff --git a/include/media/media-dev-allocator.h b/include/media/media-dev-allocator.h index b35ea6062596..2ab54d426c64 100644 --- a/include/media/media-dev-allocator.h +++ b/include/media/media-dev-allocator.h @@ -19,7 +19,7 @@ struct usb_device; -#if defined(CONFIG_MEDIA_CONTROLLER) && defined(CONFIG_USB) +#if defined(CONFIG_MEDIA_CONTROLLER) && IS_ENABLED(CONFIG_USB) /** * media_device_usb_allocate() - Allocate and return struct &media device * -- 2.30.2

2 4

conf: refine check for snd_config_is_array
by GitHub pull_request - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-lib pull request #158 was opened from ranj063: Apart from checking the index, also check if the node is not compound to qualify a config as an array. For ex: mixer [ "mixer1" ] is an array but mixer.0 { name "mixer 1" } is not an array even though the id is 0. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/158 Patch URL : https://github.com/alsa-project/alsa-lib/pull/158.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Build fails with "undefined macro AM_GNU_GETTEXT"
by GitHub issues - edited 16 Jun '21

16 Jun '21

alsa-project/alsa-utils issue #99 was edited from edward-a: **./gitcompile** fails with the following output: configure.ac:10: warning: macro 'AM_GNU_GETTEXT' not found in library configure.ac:11: warning: macro 'AM_GNU_GETTEXT_VERSION' not found in library ./gitcompile: line 10: gettextize: command not found configure.ac:15: installing './compile' configure.ac:6: installing './install-sh' configure.ac:6: installing './missing' alsactl/Makefile.am: installing './depcomp' axfer/test/Makefile.am:9: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled automake: warning: possible forward-incompatibility. automake: At least a source file is in a subdirectory, but the 'subdir-objects' automake: automake option hasn't been enabled. For now, the corresponding output automake: object file(s) will be placed in the top-level directory. However, automake: this behaviour will change in future Automake versions: they will automake: unconditionally cause object files to be placed in the same subdirectory automake: of the corresponding sources. automake: You are advised to start using 'subdir-objects' option throughout your automake: project, to avoid future incompatibilities. axfer/test/Makefile.am:9: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-single.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-multiple.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled parallel-tests: installing './test-driver' **configure.ac:10: error: possibly undefined macro: AM_GNU_GETTEXT If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:11: error: possibly undefined macro: AM_GNU_GETTEXT_VERSION** CFLAGS=-O2 -Wall -pipe -g ./configure checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... /usr/bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether make supports nested variables... yes checking whether to enable maintainer-specific portions of Makefiles... yes ./configure: line 2937: syntax error near unexpected token `external' ./configure: line 2937: `AM_GNU_GETTEXT(external)' ### System info: DISTRIB_ID=Ubuntu DISTRIB_RELEASE=20.04 DISTRIB_CODENAME=focal DISTRIB_DESCRIPTION="Ubuntu 20.04.2 LTS" NAME="Ubuntu" VERSION="20.04.2 LTS (Focal Fossa)" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 20.04.2 LTS" VERSION_ID="20.04" HOME_URL="https://www.ubuntu.com/" SUPPORT_URL="https://help.ubuntu.com/" BUG_REPORT_URL="https://bugs.launchpad.net/ubuntu/" PRIVACY_POLICY_URL="https://www.ubuntu.com/legal/terms-and-policies/privacy-policy" VERSION_CODENAME=focal UBUNTU_CODENAME=focal Issue URL : https://github.com/alsa-project/alsa-utils/issues/99 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build failes with "undefined macro AM_GNU_GETTEXT"
by GitHub issues - opened 16 Jun '21

16 Jun '21

alsa-project/alsa-utils issue #99 was opened from edward-a: **./gitcompile** fails with the following output: configure.ac:10: warning: macro 'AM_GNU_GETTEXT' not found in library configure.ac:11: warning: macro 'AM_GNU_GETTEXT_VERSION' not found in library ./gitcompile: line 10: gettextize: command not found configure.ac:15: installing './compile' configure.ac:6: installing './install-sh' configure.ac:6: installing './missing' alsactl/Makefile.am: installing './depcomp' axfer/test/Makefile.am:9: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled automake: warning: possible forward-incompatibility. automake: At least a source file is in a subdirectory, but the 'subdir-objects' automake: automake option hasn't been enabled. For now, the corresponding output automake: object file(s) will be placed in the top-level directory. However, automake: this behaviour will change in future Automake versions: they will automake: unconditionally cause object files to be placed in the same subdirectory automake: of the corresponding sources. automake: You are advised to start using 'subdir-objects' option throughout your automake: project, to avoid future incompatibilities. axfer/test/Makefile.am:9: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:9: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:9: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-riff-wave.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-au.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-voc.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../container-raw.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-single.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled axfer/test/Makefile.am:20: warning: source file '../mapper-multiple.c' is in a subdirectory, axfer/test/Makefile.am:20: but option 'subdir-objects' is disabled parallel-tests: installing './test-driver' **configure.ac:10: error: possibly undefined macro: AM_GNU_GETTEXT If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:11: error: possibly undefined macro: AM_GNU_GETTEXT_VERSION** CFLAGS=-O2 -Wall -pipe -g ./configure checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... /usr/bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether make supports nested variables... yes checking whether to enable maintainer-specific portions of Makefiles... yes ./configure: line 2937: syntax error near unexpected token `external' ./configure: line 2937: `AM_GNU_GETTEXT(external)' ### System info: DISTRIB_ID=Ubuntu DISTRIB_RELEASE=20.04 DISTRIB_CODENAME=focal DISTRIB_DESCRIPTION="Ubuntu 20.04.2 LTS" NAME="Ubuntu" VERSION="20.04.2 LTS (Focal Fossa)" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 20.04.2 LTS" VERSION_ID="20.04" HOME_URL="https://www.ubuntu.com/" SUPPORT_URL="https://help.ubuntu.com/" BUG_REPORT_URL="https://bugs.launchpad.net/ubuntu/" PRIVACY_POLICY_URL="https://www.ubuntu.com/legal/terms-and-policies/privacy-policy" VERSION_CODENAME=focal UBUNTU_CODENAME=focal Issue URL : https://github.com/alsa-project/alsa-utils/issues/99 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH -next] ASoC: dwc: dwc-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/dwc/dwc-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index fd4160289fac..8ebf76e04702 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -636,8 +636,7 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai->ops = &dw_i2s_dai_ops; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); + dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(dev->i2s_base)) return PTR_ERR(dev->i2s_base); -- 2.25.1

2 1

[PATCH] ASoC: codecs: wcd938x: remove incorrect module interdependency
by Srinivas Kandagatla 15 Jun '21

15 Jun '21

For some reason we ended up with cyclic dependency between snd_soc_wcd938x and snd_soc_wcd938x_sdw modules. Remove this cyclic dependency by handling them in respective modules. Without this below error is reported during make modules_install depmod: ERROR: Cycle detected: snd_soc_wcd938x -> snd_soc_wcd938x_sdw -> snd_soc_wcd938x depmod: ERROR: Found 2 modules in dependency cycles! Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd938x-sdw.c | 19 ++++++++++++------- sound/soc/codecs/wcd938x.c | 27 +++++---------------------- sound/soc/codecs/wcd938x.h | 4 +--- 3 files changed, 18 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index d82c40ec6898..d1cabf300c36 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -176,8 +176,19 @@ static int wcd9380_interrupt_callback(struct sdw_slave *slave, struct sdw_slave_intr_status *status) { struct wcd938x_sdw_priv *wcd = dev_get_drvdata(&slave->dev); + struct irq_domain *slave_irq = wcd->slave_irq; + struct regmap *regmap = dev_get_regmap(&slave->dev, NULL); + u32 sts1, sts2, sts3; - return wcd938x_handle_sdw_irq(wcd); + do { + handle_nested_irq(irq_find_mapping(slave_irq, 0)); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); + + } while (sts1 || sts2 || sts3); + + return IRQ_HANDLED; } static struct sdw_slave_ops wcd9380_slave_ops = { @@ -239,16 +250,10 @@ static int wcd9380_probe(struct sdw_slave *pdev, SDW_SCP_INT1_PARITY; pdev->prop.lane_control_support = true; if (wcd->is_tx) { - struct regmap *rm; - pdev->prop.source_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.src_dpn_prop = wcd938x_dpn_prop; wcd->ch_info = &wcd938x_sdw_tx_ch_info[0]; pdev->prop.wake_capable = true; - - rm = devm_regmap_init_sdw(pdev, &wcd938x_regmap_config); - if (IS_ERR(rm)) - return PTR_ERR(rm); } else { pdev->prop.sink_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wcd938x_dpn_prop; diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index a2c76dc8fd89..24afe93830ce 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1198,7 +1198,7 @@ static bool wcd938x_volatile_register(struct device *dev, unsigned int reg) return false; } -struct regmap_config wcd938x_regmap_config = { +static struct regmap_config wcd938x_regmap_config = { .name = "wcd938x_csr", .reg_bits = 32, .val_bits = 8, @@ -1211,7 +1211,6 @@ struct regmap_config wcd938x_regmap_config = { .volatile_reg = wcd938x_volatile_register, .can_multi_write = true, }; -EXPORT_SYMBOL_GPL(wcd938x_regmap_config); static const struct regmap_irq wcd938x_irqs[WCD938X_NUM_IRQS] = { REGMAP_IRQ_REG(WCD938X_IRQ_MBHC_BUTTON_PRESS_DET, 0, 0x01), @@ -3472,24 +3471,6 @@ static int wcd938x_reset(struct wcd938x_priv *wcd938x) return 0; } -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *wcd) -{ - struct wcd938x_priv *wcd938x = wcd->wcd938x; - struct irq_domain *slave_irq = wcd938x->virq; - u32 sts1, sts2, sts3; - - do { - handle_nested_irq(irq_find_mapping(slave_irq, 0)); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); - - } while (sts1 || sts2 || sts3); - - return IRQ_HANDLED; -} -EXPORT_SYMBOL_GPL(wcd938x_handle_sdw_irq); - static int wcd938x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -3573,6 +3554,7 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -3581,6 +3563,7 @@ 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->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; 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"); @@ -3607,8 +3590,8 @@ static int wcd938x_bind(struct device *dev) return -EINVAL; } - wcd938x->regmap = dev_get_regmap(wcd938x->txdev, NULL); - if (!wcd938x->regmap) { + wcd938x->regmap = devm_regmap_init_sdw(wcd938x->tx_sdw_dev, &wcd938x_regmap_config); + if (IS_ERR(wcd938x->regmap)) { dev_err(dev, "%s: tx csr regmap not found\n", __func__); return PTR_ERR(wcd938x->regmap); } diff --git a/sound/soc/codecs/wcd938x.h b/sound/soc/codecs/wcd938x.h index 9db3ab6e47a6..07b08de4cebf 100644 --- a/sound/soc/codecs/wcd938x.h +++ b/sound/soc/codecs/wcd938x.h @@ -663,11 +663,9 @@ struct wcd938x_sdw_priv { int num_ports; bool is_tx; struct wcd938x_priv *wcd938x; + struct irq_domain *slave_irq; }; -extern struct regmap_config wcd938x_regmap_config; -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *priv); - #if IS_ENABLED(CONFIG_SND_SOC_WCD938X_SDW) int wcd938x_sdw_free(struct wcd938x_sdw_priv *wcd, struct snd_pcm_substream *substream, -- 2.21.0

2 2

[PATCH -next] ASoC: jz4740-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/jz4740/jz4740-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 65d0bf939134..7ad5d9a924d8 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -525,8 +525,7 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) i2s->soc_info = device_get_match_data(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - i2s->base = devm_ioremap_resource(dev, mem); + i2s->base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(i2s->base)) return PTR_ERR(i2s->base); -- 2.25.1

2 1

[PATCH 0/4] ASoC: qdsp6: Add Quinary MI2S ports support
by Gabriel David 15 Jun '21

15 Jun '21

Various devices(such as the Motorola Moto G7 Power, codename ocean) use the Quinary MI2S ports for reproducing audio. Add support to them in kernel. Signed-off-by: Gabriel David <ultracoolguy(a)disroot.org> Gabriel David (4): ASoC: q6afe: dt-bindings: Add QUIN_MI2S_RX/TX ASoC: qdsp6: q6afe: Add Quinary MI2S ports ASoC: qdsp6: q6afe-dai: Add Quinary MI2S ports ASoC: qdsp6: q6routing: Add Quinary MI2S ports include/dt-bindings/sound/qcom,q6afe.h | 2 ++ sound/soc/qcom/qdsp6/q6afe-dai.c | 41 ++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 8 +++++ sound/soc/qcom/qdsp6/q6afe.h | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 11 +++++++ 5 files changed, 63 insertions(+), 1 deletion(-) --- As a warning, I'm currently the only tester of these patches. If that's gonna be a problem then I understand. -- 2.31.1

3 6

[PATCH -next 1/5] ASoC: img-i2s-in: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/img/img-i2s-in.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index fd3432a1d6ab..1bf5d6edbd32 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -434,8 +434,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) i2s->dev = dev; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 5

[PATCH v2] ASoC: codecs: wcd938x: remove incorrect module interdependency
by Srinivas Kandagatla 15 Jun '21

15 Jun '21

For some reason we ended up with cyclic dependency between snd_soc_wcd938x and snd_soc_wcd938x_sdw modules. Remove this cyclic dependency by handling them in respective modules. Without this below error is reported during make modules_install depmod: ERROR: Cycle detected: snd_soc_wcd938x -> snd_soc_wcd938x_sdw -> snd_soc_wcd938x depmod: ERROR: Found 2 modules in dependency cycles! Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - included irq.h as x86 allmodconfig build reported implicit declaration of function handle_nested_irq sound/soc/codecs/wcd938x-sdw.c | 20 +++++++++++++------- sound/soc/codecs/wcd938x.c | 27 +++++---------------------- sound/soc/codecs/wcd938x.h | 4 +--- 3 files changed, 19 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index d82c40ec6898..d8d4caf48dc7 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -9,6 +9,7 @@ #include <linux/component.h> #include <sound/soc.h> #include <linux/pm_runtime.h> +#include <linux/irq.h> #include <linux/irqdomain.h> #include <linux/of.h> #include <linux/soundwire/sdw.h> @@ -176,8 +177,19 @@ static int wcd9380_interrupt_callback(struct sdw_slave *slave, struct sdw_slave_intr_status *status) { struct wcd938x_sdw_priv *wcd = dev_get_drvdata(&slave->dev); + struct irq_domain *slave_irq = wcd->slave_irq; + struct regmap *regmap = dev_get_regmap(&slave->dev, NULL); + u32 sts1, sts2, sts3; - return wcd938x_handle_sdw_irq(wcd); + do { + handle_nested_irq(irq_find_mapping(slave_irq, 0)); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); + + } while (sts1 || sts2 || sts3); + + return IRQ_HANDLED; } static struct sdw_slave_ops wcd9380_slave_ops = { @@ -239,16 +251,10 @@ static int wcd9380_probe(struct sdw_slave *pdev, SDW_SCP_INT1_PARITY; pdev->prop.lane_control_support = true; if (wcd->is_tx) { - struct regmap *rm; - pdev->prop.source_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.src_dpn_prop = wcd938x_dpn_prop; wcd->ch_info = &wcd938x_sdw_tx_ch_info[0]; pdev->prop.wake_capable = true; - - rm = devm_regmap_init_sdw(pdev, &wcd938x_regmap_config); - if (IS_ERR(rm)) - return PTR_ERR(rm); } else { pdev->prop.sink_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wcd938x_dpn_prop; diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index a2c76dc8fd89..24afe93830ce 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1198,7 +1198,7 @@ static bool wcd938x_volatile_register(struct device *dev, unsigned int reg) return false; } -struct regmap_config wcd938x_regmap_config = { +static struct regmap_config wcd938x_regmap_config = { .name = "wcd938x_csr", .reg_bits = 32, .val_bits = 8, @@ -1211,7 +1211,6 @@ struct regmap_config wcd938x_regmap_config = { .volatile_reg = wcd938x_volatile_register, .can_multi_write = true, }; -EXPORT_SYMBOL_GPL(wcd938x_regmap_config); static const struct regmap_irq wcd938x_irqs[WCD938X_NUM_IRQS] = { REGMAP_IRQ_REG(WCD938X_IRQ_MBHC_BUTTON_PRESS_DET, 0, 0x01), @@ -3472,24 +3471,6 @@ static int wcd938x_reset(struct wcd938x_priv *wcd938x) return 0; } -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *wcd) -{ - struct wcd938x_priv *wcd938x = wcd->wcd938x; - struct irq_domain *slave_irq = wcd938x->virq; - u32 sts1, sts2, sts3; - - do { - handle_nested_irq(irq_find_mapping(slave_irq, 0)); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); - - } while (sts1 || sts2 || sts3); - - return IRQ_HANDLED; -} -EXPORT_SYMBOL_GPL(wcd938x_handle_sdw_irq); - static int wcd938x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -3573,6 +3554,7 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -3581,6 +3563,7 @@ 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->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; 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"); @@ -3607,8 +3590,8 @@ static int wcd938x_bind(struct device *dev) return -EINVAL; } - wcd938x->regmap = dev_get_regmap(wcd938x->txdev, NULL); - if (!wcd938x->regmap) { + wcd938x->regmap = devm_regmap_init_sdw(wcd938x->tx_sdw_dev, &wcd938x_regmap_config); + if (IS_ERR(wcd938x->regmap)) { dev_err(dev, "%s: tx csr regmap not found\n", __func__); return PTR_ERR(wcd938x->regmap); } diff --git a/sound/soc/codecs/wcd938x.h b/sound/soc/codecs/wcd938x.h index 9db3ab6e47a6..07b08de4cebf 100644 --- a/sound/soc/codecs/wcd938x.h +++ b/sound/soc/codecs/wcd938x.h @@ -663,11 +663,9 @@ struct wcd938x_sdw_priv { int num_ports; bool is_tx; struct wcd938x_priv *wcd938x; + struct irq_domain *slave_irq; }; -extern struct regmap_config wcd938x_regmap_config; -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *priv); - #if IS_ENABLED(CONFIG_SND_SOC_WCD938X_SDW) int wcd938x_sdw_free(struct wcd938x_sdw_priv *wcd, struct snd_pcm_substream *substream, -- 2.21.0

2 1

[PATCH -next] ASoC: hisilicon: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/hisilicon/hi6210-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index ff05b9779e4b..a297d4af5099 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -556,8 +556,7 @@ static int hi6210_i2s_probe(struct platform_device *pdev) i2s->dev = dev; spin_lock_init(&i2s->lock); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - i2s->base = devm_ioremap_resource(dev, res); + i2s->base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(i2s->base)) return PTR_ERR(i2s->base); -- 2.25.1

2 1

[PATCH AUTOSEL 4.19 01/12] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index b331b3ba61a9..ab73f84b5970 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2473,13 +2473,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2504,8 +2509,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3700,10 +3703,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 02/15] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index afd61599d94c..a28afb480060 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2470,13 +2470,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2501,8 +2506,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3697,10 +3700,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 21/30] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Sasha Levin 15 Jun '21

15 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> [ Upstream commit c8a4556d98510ca05bad8d02265a4918b03a8c0b ] This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20210604154545.1198337-1-judyhsiao@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 7a30a12519a7..e620a62ef534 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 1d926dd5f590..0484ad39b3dc 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 18/30] ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
by Sasha Levin 15 Jun '21

15 Jun '21

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 49783c6f4a4f49836b5a109ae0daf2f90b0d7713 ] Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also should be set in soundwire mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Link: https://lore.kernel.org/r/20210604063150.29925-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 58fb13132602..aa6c325faeab 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -455,7 +455,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600); regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, -- 2.30.2

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[PATCH AUTOSEL 5.10 14/30] ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
by Sasha Levin 15 Jun '21

15 Jun '21

From: Richard Weinberger <richard(a)nod.at> [ Upstream commit 8bef925e37bdc9b6554b85eda16ced9a8e3c135f ] TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger <richard(a)nod.at> Link: https://lore.kernel.org/r/20210530203446.19022-1-richard@nod.at Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- 2.30.2

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[PATCH AUTOSEL 5.10 13/30] ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
by Sasha Levin 15 Jun '21

15 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 320232caf1d8febea17312dab4b2dfe02e033520 ] Unfortunately the previous patch to fix issues using the AMD ACP bridge has the side effect of breaking the dmic in other cases and needs to be reverted. Removing the changes while we revisit the fix and find something better. Apologies for the churn. Suggested-by: Gabriel Craciunescu <unix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210602171251.3243-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 ----------------------------- 1 file changed, 35 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 6de536b02b57..917536def5f2 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -192,41 +192,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, - { - /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20Y1"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UH"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 Campus */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UJ"), - } - }, - { - /* Lenovo ThinkPad T14 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UD"), - } - }, - { - /* Lenovo ThinkPad X13 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UF"), - } - }, {} }; -- 2.30.2

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[PATCH AUTOSEL 5.10 10/30] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Sasha Levin 15 Jun '21

15 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 19a0aa9b04c5ab9a063b6ceaf7211ee7d9a9d24d ] More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210531145502.6079-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 917536def5f2..6de536b02b57 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -192,6 +192,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.30.2

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[PATCH AUTOSEL 5.10 07/30] ASoC: fsl-asoc-card: Set .owner attribute when registering card.
by Sasha Levin 15 Jun '21

15 Jun '21

From: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> [ Upstream commit a8437f05384cb472518ec21bf4fffbe8f0a47378 ] Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14) [<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104) [<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114) [<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4) [<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd]) [<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Link: https://lore.kernel.org/r/20210527163409.22049-1-nicolas.cavallari@green-co… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index a2dd3b6b7fec..7cd14d6b9436 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -720,6 +720,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- 2.30.2

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[PATCH AUTOSEL 5.10 05/30] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 91a4ef7f620c..a9b079d56fd6 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.12 23/33] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Sasha Levin 15 Jun '21

15 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> [ Upstream commit c8a4556d98510ca05bad8d02265a4918b03a8c0b ] This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20210604154545.1198337-1-judyhsiao@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 936384a94f25..74d3d8c58608 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.30.2

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[PATCH AUTOSEL 5.12 20/33] ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
by Sasha Levin 15 Jun '21

15 Jun '21

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 49783c6f4a4f49836b5a109ae0daf2f90b0d7713 ] Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also should be set in soundwire mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Link: https://lore.kernel.org/r/20210604063150.29925-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index b49f1e16125d..d1dd7f720ba4 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600); regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 15/33] ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
by Sasha Levin 15 Jun '21

15 Jun '21

From: Richard Weinberger <richard(a)nod.at> [ Upstream commit 8bef925e37bdc9b6554b85eda16ced9a8e3c135f ] TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger <richard(a)nod.at> Link: https://lore.kernel.org/r/20210530203446.19022-1-richard@nod.at Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 14/33] ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
by Sasha Levin 15 Jun '21

15 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 320232caf1d8febea17312dab4b2dfe02e033520 ] Unfortunately the previous patch to fix issues using the AMD ACP bridge has the side effect of breaking the dmic in other cases and needs to be reverted. Removing the changes while we revisit the fix and find something better. Apologies for the churn. Suggested-by: Gabriel Craciunescu <unix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210602171251.3243-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 ----------------------------- 1 file changed, 35 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 47a4dfd81a46..050a61fe9693 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,41 +199,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, - { - /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20Y1"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UH"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 Campus */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UJ"), - } - }, - { - /* Lenovo ThinkPad T14 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UD"), - } - }, - { - /* Lenovo ThinkPad X13 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UF"), - } - }, {} }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 07/33] ASoC: fsl-asoc-card: Set .owner attribute when registering card.
by Sasha Levin 15 Jun '21

15 Jun '21

From: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> [ Upstream commit a8437f05384cb472518ec21bf4fffbe8f0a47378 ] Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14) [<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104) [<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114) [<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4) [<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd]) [<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Link: https://lore.kernel.org/r/20210527163409.22049-1-nicolas.cavallari@green-co… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index f62f81ceab0d..9dcbe5d5a428 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -732,6 +732,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 05/33] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 91a4ef7f620c..a9b079d56fd6 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

1 0

Undefined symbol in static library
by GitHub issues - opened 15 Jun '21

15 Jun '21

alsa-project/alsa-lib issue #157 was opened from jpalus: There is a mismatch between declaration: https://github.com/alsa-project/alsa-lib/blob/f4c061f349188c548497607efd462… and actual definition: https://github.com/alsa-project/alsa-lib/blob/f4c061f349188c548497607efd462… resulting in undefined symbol in static library. Judging by convention used in other files I suppose the latter should be renamed to match declaration. Issue URL : https://github.com/alsa-project/alsa-lib/issues/157 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH -next] ASoC: codecs: wcd938x: constify static struct snd_soc_dai_ops
by Pu Lehui 15 Jun '21

15 Jun '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Signed-off-by: Pu Lehui <pulehui(a)huawei.com> --- 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 2cf6145b7219..8ec9357fd855 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3425,7 +3425,7 @@ static int wcd938x_codec_set_sdw_stream(struct snd_soc_dai *dai, } -static struct snd_soc_dai_ops wcd938x_sdw_dai_ops = { +static const struct snd_soc_dai_ops wcd938x_sdw_dai_ops = { .hw_params = wcd938x_codec_hw_params, .hw_free = wcd938x_codec_free, .set_sdw_stream = wcd938x_codec_set_sdw_stream, -- 2.17.1

2 1

[PATCH 0/3] ASoC: tlv320aic32x4: Add support for TAS2505
by Claudius Heine 15 Jun '21

15 Jun '21

Hi, this is an early patchset that provides support for the TAS2505 by the tlv320aic32x4 driver. Playback with aplay and the right mixer settings seems to be working, but some mixer settings either cause distortions or popping. Any input on how to prevent that is welcome. Also with this change TAS2505 is compiled in unconditionally whenever SND_SOC_TLV320AIC32X4_I2C is enabled. Maybe an additional option for the TAS2505 should be added, but then we probably also need a setting to provide only support for the TAS2505 and not the TLV320AIC32X4. Not sure how to set that up in the best way. This is my first time working with the sound subsystem, so expect stupidity and obvious mistakes. regards, Claudius Claudius Heine (3): ASoC: tlv320aic32x4: prepare driver for different device variants ASoC: tlv320aic32x4: add support for TAS2505 ASoC: tlv320aic32x4: dt-bindings: add TAS2505 to compatible .../bindings/sound/tlv320aic32x4.txt | 1 + sound/soc/codecs/tlv320aic32x4-i2c.c | 22 ++- sound/soc/codecs/tlv320aic32x4-spi.c | 23 ++- sound/soc/codecs/tlv320aic32x4.c | 173 +++++++++++++++++- sound/soc/codecs/tlv320aic32x4.h | 10 + 5 files changed, 219 insertions(+), 10 deletions(-) base-commit: 009c9aa5be652675a06d5211e1640e02bbb1c33d -- 2.32.0

2 9

[PATCH] ASoC: dt-bindings: fsl-sai: Add compatible string for imx8mm/8mn/8mp/8ulp
by Shengjiu Wang 15 Jun '21

15 Jun '21

Add compatible string for imx8mm/8mn/8mp/8ulp, these platforms all support SAI IP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 0dc83cc4a236..c71c5861d787 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -9,8 +9,10 @@ Required properties: - compatible : Compatible list, contains "fsl,vf610-sai", "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai" or - "fsl,imx8qm-sai". + "fsl,imx7ulp-sai", "fsl,imx8mq-sai", + "fsl,imx8qm-sai", "fsl,imx8mm-sai", + "fsl,imx8mn-sai", "fsl,imx8mp-sai", or + "fsl,imx8ulp-sai". - reg : Offset and length of the register set for the device. -- 2.27.0

3 2

[PATCH v6 0/4] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 15 Jun '21

15 Jun '21

This series squashes all the ASoC machine drivers into a single one, this change was suggested by Jon Hunter. It also sets driver_name and components string of each card, allowing userspace alsa-lib to find UCMs at predictable path. Changelog: v6: - Fixed missed configuration of AC97 clock rate for the WM9712 codec in the unified driver. - Added new patch that removes now obsolete "utils" helpers and moves code into the unified driver. ASoC: tegra: Squash utils into common machine driver v5: - The v4 removed the customization of components string for Nexus 7, but I missed to remove the "components" hook which is unused now, it's removed in v5 for consistency. - Slightly improved naming of the common 12MHz MCLK rate function to make it more consistent with the rest of the driver functions. v4: - Moved out mclk_rate callback that is currently used only by WM8903 machine driver from the common driver. This was suggested by Jon Hunter. - Dropped patch which was setting custom components string for Nexus 7. Jaroslav Kysela wants it to be specified in a device-tree, but the components string doesn't have a firm specification for today. It's better to drop this change for now since it's optional anyways. - Fixed compilation error that was reported by kernel robot for v3. - Jaroslav Kysela merged alsa-ucm-conf PR [1] which added UCMs for Nexus 7 and Acer A500. The UCMs are fully working using a combination of updated kernel + alsa-ucm-conf master + alsa-lib master, meaning that they will work with the next releases of kernel and ALSA userspace upstream packages. - Added ack from Jaroslav Kysela to the "Specify components string for each card" patch that he gave to v3. v3: - Added components string as was suggested by Jaroslav Kysela to v2. - Renamed MCLK rate function that is used by max98090 and other codecs to make it look more generic. Added option for specifying CLK ID per device. This all was suggested by Jon Hunter to v2. v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (4): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers ASoC: tegra: Specify components string for each card ASoC: tegra: Squash utils into common machine driver sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 19 +- sound/soc/tegra/tegra_alc5632.c | 259 -------- sound/soc/tegra/tegra_asoc_machine.c | 854 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 49 ++ sound/soc/tegra/tegra_asoc_utils.c | 225 ------- sound/soc/tegra/tegra_asoc_utils.h | 38 -- sound/soc/tegra/tegra_max98090.c | 276 --------- sound/soc/tegra/tegra_rt5640.c | 222 ------- sound/soc/tegra/tegra_rt5677.c | 324 ---------- sound/soc/tegra/tegra_sgtl5000.c | 211 ------- sound/soc/tegra/tegra_wm8753.c | 185 ------ sound/soc/tegra/tegra_wm8903.c | 351 +++-------- sound/soc/tegra/tegra_wm9712.c | 166 ------ sound/soc/tegra/trimslice.c | 172 ------ 15 files changed, 996 insertions(+), 2367 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_asoc_utils.c delete mode 100644 sound/soc/tegra/tegra_asoc_utils.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

2 6

[PATCH] ASoC: dt-bindings: fsl, spdif: Add compatible string for imx8ulp
by Shengjiu Wang 15 Jun '21

15 Jun '21

Add compatible string for imx8ulp, which supports spdif module Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 4454aca34d56..f226ec13167a 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -25,6 +25,7 @@ properties: - fsl,imx8mq-spdif - fsl,imx8mm-spdif - fsl,imx8mn-spdif + - fsl,imx8ulp-spdif reg: maxItems: 1 -- 2.27.0

3 2

Xone43C: USB communication
by Roman Muller 15 Jun '21

15 Jun '21

Hello Gerardo, hello all of you, > Is your capture complete from a software standpoint? i.e. are you > starting Serato or Traktor on the VM while doing the usbmon capture? Not tested yet, I need to get a few items for that and investigate the matter! It is complete up to loading the Win10 driver (which has been supplied by the manufacturer). Maybe I am lucky and this is already sufficient. Have a nice day, Roman

2 1

[PATCH -next] ASoC: axi-spdif: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/adi/axi-spdif.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 9b3d81c41c8c..8d4a6cb4e5c5 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -189,8 +189,7 @@ static int axi_spdif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, spdif); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH -next] ASoC: codecs: wcd: Remove unneeded semicolon
by Yang Li 14 Jun '21

14 Jun '21

Eliminate the following coccicheck warning: ./sound/soc/codecs/wcd-mbhc-v2.c:990:2-3: Unneeded semicolon. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index dee9410..405128c 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -987,7 +987,7 @@ static void wcd_mbhc_adc_update_fsm_source(struct wcd_mbhc *mbhc, wcd_mbhc_write_field(mbhc, WCD_MBHC_BTN_ISRC_CTL, 0); break; - }; + } } static void wcd_mbhc_bcs_enable(struct wcd_mbhc *mbhc, int plug_type, bool enable) -- 1.8.3.1

3 2

[PATCH -next] ASoC: mchp-i2s-mcc: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/mchp-i2s-mcc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 673bc16cb46a..8988f024a732 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -1008,8 +1008,7 @@ static int mchp_i2s_mcc_probe(struct platform_device *pdev) if (!dev) return -ENOMEM; - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[RESEND PATCH] ASoC: rt5640: Make codec selectable
by Martin Blumenstingl 14 Jun '21

14 Jun '21

The Realtek rt5640 codec driver can be used with the generic sound card drivers, so it should be selectable. For example, with the addition of #sound-dai-cells = <0> property in DT, it can be used with simple and graph card drivers. Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> --- Re-sent because my last mail did not make it to the list This will be used in arch/arm/boot/dts/meson8b-ec100.dts sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2a7b3e363069..96fc0144f9fc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1180,7 +1180,7 @@ config SND_SOC_RT5631 depends on I2C config SND_SOC_RT5640 - tristate + tristate "Realtek RT5640/RT5639 Codec" depends on I2C config SND_SOC_RT5645 -- 2.31.1

2 1

[PATCH] ASoC: soc-pcm: fix the return value in dpcm_apply_symmetry()
by Jaroslav Kysela 14 Jun '21

14 Jun '21

In case, where the loops are not executed for a reason, the uninitialized variable 'err' is returned to the caller. Make code fully predictible and assign zero in the declaration. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8659089a87a0..46513bb97904 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1700,7 +1700,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_soc_dpcm *dpcm; struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(fe_substream); struct snd_soc_dai *fe_cpu_dai; - int err; + int err = 0; int i; /* apply symmetry for FE */ -- 2.30.2

2 1

Re: [PATCH] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq()
by Mark Brown 14 Jun '21

14 Jun '21

On Thu, 10 Jun 2021 20:50:52 +0800, Zhongjun Tan wrote: > Clean up the check for irq.dev_err is superfluous as platform_get_irq() > already prints an error.Remove curly braces to confirm to styling > requirements. Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq() commit: 4d5f3a096f3d9e7067c7c2e730d989668e06d552 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH v8 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 14 Jun '21

14 Jun '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v7: - Fixed errors reported by kernel test robot by moving SoundWire specific bits to sdw driver. Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 146 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 315 ++ sound/soc/codecs/wcd938x.c | 3743 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 720 ++++ 9 files changed, 5365 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 14

[PATCH -next] ASoC: atmel-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 584656cc7d3c..e8fe1a7a4d83 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -610,8 +610,7 @@ static int atmel_i2s_probe(struct platform_device *pdev) dev->caps = match->data; /* Map I/O registers. */ - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH] dt-bindings: ASoC: sgtl5000: Add audio-graph-card port
by Marek Vasut 14 Jun '21

14 Jun '21

The SGTL5000 codec can be connected via audio-graph-card, add the missing port: entry into the bindings. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: kernel(a)dh-electronics.com Cc: alsa-devel(a)alsa-project.org To: devicetree(a)vger.kernel.org --- Documentation/devicetree/bindings/sound/sgtl5000.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml index 70b4a8831073..e762c320b574 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.yaml +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -75,6 +75,10 @@ properties: $ref: "/schemas/types.yaml#/definitions/uint32" enum: [ 0, 1, 2, 3 ] + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - reg -- 2.30.2

2 1

[PATCH v9 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 14 Jun '21

14 Jun '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v8: - moved Kconfig and Makefile changes to last patch as suggested by Mark - removed array of enums and used static entries instead. Suggested by Mark - return true if put succeeds, Suggested by Mark - removed some unneeded semi-colons in switch Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing and Kconfig .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 146 + sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 315 ++ sound/soc/codecs/wcd938x.c | 3753 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 720 ++++ 9 files changed, 5376 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 10

[PATCH] ASoC: rt5645: Avoid upgrading static warnings to errors
by Mark Brown 14 Jun '21

14 Jun '21

One of the fixes reverted as part of the UMN fallout was actually fine, however rather than undoing the revert the process that handled all this stuff resulted in a patch which attempted to add extra error checks instead. Unfortunately this new change wasn't really based on a good understanding of the subsystem APIs and bypassed the usual patch flow without ensuring it was reviewed by people with subsystem knowledge and was merged as a fix rather than during the merge window. The effect of the new fix is to upgrade what were previously warnings on static data in the code to hard errors on that data. If this actually happens then it would break existing systems, if it doesn't happen then the change has no effect so this was not a safe change to apply as a fix to the release candidates. Since the new code has not been tested and doesn't in practice improve error handling revert it instead, and also drop the original revert since the original fix was fine. This takes the driver back to what it was in -rc1. Fixes: 5e70b8e22b64e ("ASoC: rt5645: add error checking to rt5645_probe function") Fixes: 1e0ce84215dbf ("Revert "ASoC: rt5645: fix a NULL pointer dereference") Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Phillip Potter <phil(a)philpotter.co.uk> --- sound/soc/codecs/rt5645.c | 49 +++++++++------------------------------ 1 file changed, 11 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 438fa18bcb55..9408ee63cb26 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3388,44 +3388,30 @@ static int rt5645_probe(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); - int ret = 0; rt5645->component = component; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - ret = snd_soc_dapm_new_controls(dapm, + snd_soc_dapm_new_controls(dapm, rt5645_specific_dapm_widgets, ARRAY_SIZE(rt5645_specific_dapm_widgets)); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); - if (ret < 0) - goto exit; - if (rt5645->v_id < 3) { - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5645_old_dapm_routes, ARRAY_SIZE(rt5645_old_dapm_routes)); - if (ret < 0) - goto exit; } break; case CODEC_TYPE_RT5650: - ret = snd_soc_dapm_new_controls(dapm, + snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); - if (ret < 0) - goto exit; break; } @@ -3433,17 +3419,9 @@ static int rt5645_probe(struct snd_soc_component *component) /* for JD function */ if (rt5645->pdata.jd_mode) { - ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_sync(dapm); - if (ret < 0) - goto exit; + snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_sync(dapm); } if (rt5645->pdata.long_name) @@ -3454,14 +3432,9 @@ static int rt5645_probe(struct snd_soc_component *component) GFP_KERNEL); if (!rt5645->eq_param) - ret = -ENOMEM; -exit: - /* - * If there was an error above, everything will be cleaned up by the - * caller if we return an error here. This will be done with a later - * call to rt5645_remove(). - */ - return ret; + return -ENOMEM; + + return 0; } static void rt5645_remove(struct snd_soc_component *component) -- 2.20.1

2 2

[PATCH -next] ASoC: bcm: cygnus_ssp: Use devm_platform_ioremap_resource_byname()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use the devm_platform_ioremap_resource_byname() helper instead of calling platform_get_resource_byname() and devm_ioremap_resource() separately. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/bcm/cygnus-ssp.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index ba03bb62ba96..fca5a3f2eec5 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1308,7 +1308,6 @@ static int cygnus_ssp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct device_node *child_node; - struct resource *res; struct cygnus_audio *cygaud; int err; int node_count; @@ -1320,13 +1319,11 @@ static int cygnus_ssp_probe(struct platform_device *pdev) dev_set_drvdata(dev, cygaud); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "aud"); - cygaud->audio = devm_ioremap_resource(dev, res); + cygaud->audio = devm_platform_ioremap_resource_byname(pdev, "aud"); if (IS_ERR(cygaud->audio)) return PTR_ERR(cygaud->audio); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "i2s_in"); - cygaud->i2s_in = devm_ioremap_resource(dev, res); + cygaud->i2s_in = devm_platform_ioremap_resource_byname(pdev, "i2s_in"); if (IS_ERR(cygaud->i2s_in)) return PTR_ERR(cygaud->i2s_in); -- 2.25.1

2 1

[PATCH] ASoC: fsl_spdif: Fix error handler with pm_runtime_enable
by Shengjiu Wang 14 Jun '21

14 Jun '21

There is error message when defer probe happens: fsl-spdif-dai 2dab0000.spdif: Unbalanced pm_runtime_enable! Fix the error handler with pm_runtime_enable and add fsl_spdif_remove() for pm_runtime_disable. Fixes: 9cb2b3796e08 ("ASoC: fsl_spdif: Add pm runtime function") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 2a76714eb8e6..76a7c9581333 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1441,16 +1441,29 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - return ret; + goto err_pm_disable; } ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); - if (ret && ret != -EPROBE_DEFER) - dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); + if (ret) { + dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); + goto err_pm_disable; + } + + return ret; +err_pm_disable: + pm_runtime_disable(&pdev->dev); return ret; } +static int fsl_spdif_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + #ifdef CONFIG_PM static int fsl_spdif_runtime_suspend(struct device *dev) { @@ -1554,6 +1567,7 @@ static struct platform_driver fsl_spdif_driver = { .pm = &fsl_spdif_pm, }, .probe = fsl_spdif_probe, + .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); -- 2.27.0

2 1

[PATCH -next] ASoC: atmel-pdmic: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-pdmic.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 8e1d8230b180..42117de299e7 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -620,8 +620,7 @@ static int atmel_pdmic_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - io_base = devm_ioremap_resource(dev, res); + io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(io_base)) return PTR_ERR(io_base); -- 2.25.1

2 1

[PATCH v3] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 14 Jun '21

14 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V2: -- Added empty entry at the end of channel maps array. Changes Since V1: -- removed set_channel_map/get_channel_map implementation as default kcontrols added in pcm_new API. sound/soc/qcom/lpass-cpu.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 2 ++ 3 files changed, 36 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..aff39c9f6326 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -29,6 +29,15 @@ #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +/* + * Channel maps for Quad channel playbacks on MI2S Secondary + */ +static struct snd_pcm_chmap_elem lpass_quad_chmaps[] = { + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_RL, + SNDRV_CHMAP_FR, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { @@ -324,6 +333,25 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + if (drvdata->mi2s_playback_sd_mode[dai->id] == LPAIF_I2SCTL_MODE_QUAD01) { + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + lpass_quad_chmaps, drv->playback.channels_max, 0, + NULL); + if (ret < 0) + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +884,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..623ddccdafff 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -259,5 +259,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 3

[PATCH -next] ASoC: axi-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/adi/axi-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index aa082131fb90..1289cb4e2988 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -198,8 +198,7 @@ static int axi_i2s_probe(struct platform_device *pdev) axi_i2s_parse_of(i2s, pdev->dev.of_node); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH -next] ASoC: atmel-classd: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-classd.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 6023369e0f1a..a9f9f449c48c 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -558,8 +558,7 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - io_base = devm_ioremap_resource(dev, res); + io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(io_base)) return PTR_ERR(io_base); -- 2.25.1

2 1

Re: [PATCH] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq()
by Mark Brown 14 Jun '21

14 Jun '21

On Thu, 10 Jun 2021 12:00:37 +0800, Zhongjun Tan wrote: > The platform_get_irq() prints error message telling that interrupt is > missing, hence there is no need to duplicated that message. Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq() commit: 2e8a8adb96a335a04f1697dd4314f5569521328f All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

ALSA 1.2.5.1 bugfix release
by Jaroslav Kysela 14 Jun '21

14 Jun '21

Hello all, the ALSA userspace alsa-lib, alsa-utils and alsa-ucm-conf packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.5_v1.2.5.1 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

USB sound card issues
by Muni Sekhar 14 Jun '21

14 Jun '21

Hi all, I am using a USB sound card(snd-usb-audio), and it fails while doing the audio recording "arecord: pcm_read:2032: read error: Input/output error". What could be the problem? and is there a workaround for this? $ arecord -l **** List of CAPTURE Hardware Devices **** card 1: BT_AG [BT_002], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord --device hw:1,0 --channels 1 --format S16_LE --rate 32000_Hz x.wav Recording WAVE 'x.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono arecord: pcm_read:2032: read error: Input/output error -- Thanks, Sekhar

3 10

added an option to aseqnet to optionally set the midi process name
by GitHub pull_request - reopened 14 Jun '21

14 Jun '21

alsa-project/alsa-utils pull request #95 was reopened from BrainDamage: This option allows to run multiple instances of aseqnet without having to double check the assigned port number, since each one can get spawned with a unique name. The patch is pretty minimal and clean, thopefully here shouldn't be troubles merging it. Request URL : https://github.com/alsa-project/alsa-utils/pull/95 Patch URL : https://github.com/alsa-project/alsa-utils/pull/95.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] alsa: control_led - fix initialization in the mode show callback
by Jaroslav Kysela 14 Jun '21

14 Jun '21

The str variable should be always initialized before use even if the switch covers all cases. This is a minimalistic fix: Assign NULL, the sprintf() may print '(null)' if something is corrupted. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/control_led.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 861f71f2fab5..764058cc345d 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -397,7 +397,7 @@ static ssize_t mode_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); - const char *str; + const char *str = NULL; switch (led->mode) { case MODE_FOLLOW_MUTE: str = "follow-mute"; break; -- 2.30.2

2 1

[PATCH 0/7] Prepare EP93xx drivers for Common Clock Framework
by Alexander Sverdlin 14 Jun '21

14 Jun '21

Nikita posted a patch converting EP93xx to use Common Clock Framework. It turns out some cleanup is necessary in the EP93xx drivers to avoid "Enabling unprepared" clock warnings. Patches with stack traces in the commit messages are tested on EP9302. Link: https://lore.kernel.org/patchwork/patch/1435884/ Alexander Sverdlin (7): iio: ep93xx: Prepare clock before using it spi: spi-ep93xx: Prepare clock before using it Input: ep93xx_keypad: Prepare clock before using it video: ep93xx: Prepare clock before using it dmaengine: ep93xx: Prepare clock before using it ASoC: cirrus: i2s: Prepare clock before using it pwm: ep93xx: Prepare clock before using it drivers/dma/ep93xx_dma.c | 6 +++--- drivers/iio/adc/ep93xx_adc.c | 6 +++--- drivers/input/keyboard/ep93xx_keypad.c | 4 ++-- drivers/pwm/pwm-ep93xx.c | 12 ++++++------ drivers/spi/spi-ep93xx.c | 4 ++-- drivers/video/fbdev/ep93xx-fb.c | 4 ++-- sound/soc/cirrus/ep93xx-i2s.c | 12 ++++++------ 7 files changed, 24 insertions(+), 24 deletions(-) -- 2.32.0

3 3

[PATCH] ALSA: firewire: fill Kconfig with entries for devices based on DICE ASICs
by Takashi Sakamoto 14 Jun '21

14 Jun '21

Nowadays a lot of devices are supported by ALSA dice driver. This commit fills Kconfig with entries for them. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 21 +++++++++++++++++++-- 1 file changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index fc6858131b59..a08a0dbfc951 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -18,8 +18,25 @@ config SND_DICE select SND_HWDEP select SND_FIREWIRE_LIB help - Say Y here to include support for many DACs based on the DICE - chip family (DICE-II/Jr/Mini) which TC Applied Technologies produces. + Say Y here to include support for devices based on the DICE chip family + (DICE-II/TCD2210(Mini)/TCD2220(Jr.)) which TC Applied Technologies (TCAT) produced. + * Allen and Heath Zed R16 + * Alesis iO 14/26 FireWire, MasterControl, MultiMix 8/12/16 FireWire + * Avid Mbox 3 Pro + * FlexRadio Systems FLEX-3000, FLEX-5000 + * Focusrite Liquid Saffire 56 + * Focusrite Saffire Pro 14, Pro 24, Pro 24 DSP, Pro 26, Pro 40(TCD2220) + * Harman Music Group Lexicon I-ONIX FW810S + * Loud Technologies Mackie Onyx Blackbird, Onyx 820i/1220i/1620i/1640i (latter models) + * M-Audio ProFire 610/2626 + * Mytek Stereo192-DSD DAC + * Midas Klark Teknik VeniceF series + * PreSonus FireStudio, FireStudio Mobile, FireStudio Project, FireStudio Tube + * PreSonus StudioLive 16.4.2, 16.0.2, 24.4.2, 32.4.2 + * Solid State Logic Duende Classic, Duende Mini + * TC Electronic Studio Konnekt 48, Konnekt 24D, Konnekt Live, Impact Twin + * TC Electronic Digital Konnekt x32, Desktop Konnekt 6 + * Weiss Engineering ADC2, Vesta, Minerva, AFI1, DAC1, INT202, DAC202 To compile this driver as a module, choose M here: the module will be called snd-dice. -- 2.30.2

2 1

Android porting patches
by GitHub pull_request - opened 14 Jun '21

14 Jun '21

alsa-project/alsa-lib pull request #156 was opened from cwhuang: This is a series of patches to make alsa-lib buildable with Android source tree. Test on with Android 8.1 and later. Request URL : https://github.com/alsa-project/alsa-lib/pull/156 Patch URL : https://github.com/alsa-project/alsa-lib/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4] soundwire: intel: move to auxiliary bus
by Bard Liao 14 Jun '21

14 Jun '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Now that the auxiliary_bus exists, there's no reason to use platform devices as children of a PCI device any longer. This patch refactors the code by extending a basic auxiliary device with Intel link-specific structures that need to be passed between controller and link levels. This refactoring is much cleaner with no need for cross-pointers between device and link structures. Note that the auxiliary bus API has separate init and add steps, which requires more attention in the error unwinding paths. The main loop needs to deal with kfree() and auxiliary_device_uninit() for the current iteration before jumping to the common label which releases everything allocated in prior iterations. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - add link_dev_register for all kzalloc, device_init, and device_add. v3: - Modify the function description of sdw_intel_probe() which was missing in previous version. v4: - Move intel_link_dev_unregister(ldev) before pm_runtime_put_noidle( ldev->link_res.dev) to fix use-after-free reported by KASAN. --- drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 56 ++++--- drivers/soundwire/intel.h | 14 +- drivers/soundwire/intel_init.c | 232 +++++++++++++++++++--------- include/linux/soundwire/sdw_intel.h | 6 +- 5 files changed, 202 insertions(+), 107 deletions(-) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 016e74230bb7..2b7795233282 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -25,6 +25,7 @@ config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE select SOUNDWIRE_GENERIC_ALLOCATION + select AUXILIARY_BUS depends on ACPI && SND_SOC help SoundWire Intel Master driver. diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fd95f94630b1..c11e3d8cd308 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -11,7 +11,7 @@ #include <linux/module.h> #include <linux/interrupt.h> #include <linux/io.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <sound/pcm_params.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -1327,11 +1327,14 @@ static int intel_init(struct sdw_intel *sdw) } /* - * probe and init + * probe and init (aux_dev_id argument is required by function prototype but not used) */ -static int intel_master_probe(struct platform_device *pdev) +static int intel_link_probe(struct auxiliary_device *auxdev, + const struct auxiliary_device_id *aux_dev_id) + { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); struct sdw_intel *sdw; struct sdw_cdns *cdns; struct sdw_bus *bus; @@ -1344,14 +1347,14 @@ static int intel_master_probe(struct platform_device *pdev) cdns = &sdw->cdns; bus = &cdns->bus; - sdw->instance = pdev->id; - sdw->link_res = dev_get_platdata(dev); + sdw->instance = auxdev->id; + sdw->link_res = &ldev->link_res; cdns->dev = dev; cdns->registers = sdw->link_res->registers; cdns->instance = sdw->instance; cdns->msg_count = 0; - bus->link_id = pdev->id; + bus->link_id = auxdev->id; sdw_cdns_probe(cdns); @@ -1384,10 +1387,10 @@ static int intel_master_probe(struct platform_device *pdev) return 0; } -int intel_master_startup(struct platform_device *pdev) +int intel_link_startup(struct auxiliary_device *auxdev) { struct sdw_cdns_stream_config config; - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1524,9 +1527,9 @@ int intel_master_startup(struct platform_device *pdev) return ret; } -static int intel_master_remove(struct platform_device *pdev) +static void intel_link_remove(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1542,19 +1545,17 @@ static int intel_master_remove(struct platform_device *pdev) snd_soc_unregister_component(dev); } sdw_bus_master_delete(bus); - - return 0; } -int intel_master_process_wakeen_event(struct platform_device *pdev) +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_intel *sdw; struct sdw_bus *bus; void __iomem *shim; u16 wake_sts; - sdw = platform_get_drvdata(pdev); + sdw = dev_get_drvdata(dev); bus = &sdw->cdns.bus; if (bus->prop.hw_disabled) { @@ -1976,17 +1977,22 @@ static const struct dev_pm_ops intel_pm = { SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) }; -static struct platform_driver sdw_intel_drv = { - .probe = intel_master_probe, - .remove = intel_master_remove, +static const struct auxiliary_device_id intel_link_id_table[] = { + { .name = "soundwire_intel.link" }, + {}, +}; +MODULE_DEVICE_TABLE(auxiliary, intel_link_id_table); + +static struct auxiliary_driver sdw_intel_drv = { + .probe = intel_link_probe, + .remove = intel_link_remove, .driver = { - .name = "intel-sdw", + /* auxiliary_driver_register() sets .name to be the modname */ .pm = &intel_pm, - } + }, + .id_table = intel_link_id_table }; - -module_platform_driver(sdw_intel_drv); +module_auxiliary_driver(sdw_intel_drv); MODULE_LICENSE("Dual BSD/GPL"); -MODULE_ALIAS("platform:intel-sdw"); -MODULE_DESCRIPTION("Intel Soundwire Master Driver"); +MODULE_DESCRIPTION("Intel Soundwire Link Driver"); diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 06bac8ba14e9..0b47b148da3f 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -7,7 +7,6 @@ /** * struct sdw_intel_link_res - Soundwire Intel link resource structure, * typically populated by the controller driver. - * @pdev: platform_device * @mmio_base: mmio base of SoundWire registers * @registers: Link IO registers base * @shim: Audio shim pointer @@ -23,7 +22,6 @@ * @list: used to walk-through all masters exposed by the same controller */ struct sdw_intel_link_res { - struct platform_device *pdev; void __iomem *mmio_base; /* not strictly needed, useful for debug */ void __iomem *registers; void __iomem *shim; @@ -48,7 +46,15 @@ struct sdw_intel { #endif }; -int intel_master_startup(struct platform_device *pdev); -int intel_master_process_wakeen_event(struct platform_device *pdev); +int intel_link_startup(struct auxiliary_device *auxdev); +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev); + +struct sdw_intel_link_dev { + struct auxiliary_device auxdev; + struct sdw_intel_link_res link_res; +}; + +#define auxiliary_dev_to_sdw_intel_link_dev(auxiliary_dev) \ + container_of(auxiliary_dev, struct sdw_intel_link_dev, auxdev) #endif /* __SDW_INTEL_LOCAL_H */ diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 30ce95ec2d70..9e283bef53d2 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -12,7 +12,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/module.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <linux/pm_runtime.h> #include <linux/soundwire/sdw_intel.h> #include "cadence_master.h" @@ -24,28 +24,108 @@ #define SDW_LINK_BASE 0x30000 #define SDW_LINK_SIZE 0x10000 +static void intel_link_dev_release(struct device *dev) +{ + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); + + kfree(ldev); +} + +/* alloc, init and add link devices */ +static struct sdw_intel_link_dev *intel_link_dev_register(struct sdw_intel_res *res, + struct sdw_intel_ctx *ctx, + struct fwnode_handle *fwnode, + const char *name, + int link_id) +{ + struct sdw_intel_link_dev *ldev; + struct sdw_intel_link_res *link; + struct auxiliary_device *auxdev; + int ret; + + ldev = kzalloc(sizeof(*ldev), GFP_KERNEL); + if (!ldev) + return ERR_PTR(-ENOMEM); + + auxdev = &ldev->auxdev; + auxdev->name = name; + auxdev->dev.parent = res->parent; + auxdev->dev.fwnode = fwnode; + auxdev->dev.release = intel_link_dev_release; + + /* we don't use an IDA since we already have a link ID */ + auxdev->id = link_id; + + /* + * keep a handle on the allocated memory, to be used in all other functions. + * Since the same pattern is used to skip links that are not enabled, there is + * no need to check if ctx->ldev[i] is NULL later on. + */ + ctx->ldev[link_id] = ldev; + + /* Add link information used in the driver probe */ + link = &ldev->link_res; + link->mmio_base = res->mmio_base; + link->registers = res->mmio_base + SDW_LINK_BASE + + (SDW_LINK_SIZE * link_id); + link->shim = res->mmio_base + SDW_SHIM_BASE; + link->alh = res->mmio_base + SDW_ALH_BASE; + + link->ops = res->ops; + link->dev = res->dev; + + link->clock_stop_quirks = res->clock_stop_quirks; + link->shim_lock = &ctx->shim_lock; + link->shim_mask = &ctx->shim_mask; + link->link_mask = ctx->link_mask; + + /* now follow the two-step init/add sequence */ + ret = auxiliary_device_init(auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to initialize link dev %s link_id %d\n", + name, link_id); + kfree(ldev); + return ERR_PTR(ret); + } + + ret = auxiliary_device_add(&ldev->auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to add link dev %s link_id %d\n", + ldev->auxdev.name, link_id); + /* ldev will be freed with the put_device() and .release sequence */ + auxiliary_device_uninit(&ldev->auxdev); + return ERR_PTR(ret); + } + + return ldev; +} + +static void intel_link_dev_unregister(struct sdw_intel_link_dev *ldev) +{ + auxiliary_device_delete(&ldev->auxdev); + auxiliary_device_uninit(&ldev->auxdev); +} + static int sdw_intel_cleanup(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link = ctx->links; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!link) - return 0; - link_mask = ctx->link_mask; - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - if (link->pdev) { - pm_runtime_disable(&link->pdev->dev); - platform_device_unregister(link->pdev); - } + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) - pm_runtime_put_noidle(link->dev); + pm_runtime_disable(&ldev->auxdev.dev); + if (!ldev->link_res.clock_stop_quirks) + pm_runtime_put_noidle(ldev->link_res.dev); + + intel_link_dev_unregister(ldev); } return 0; @@ -91,9 +171,8 @@ EXPORT_SYMBOL_NS(sdw_intel_thread, SOUNDWIRE_INTEL_INIT); static struct sdw_intel_ctx *sdw_intel_probe_controller(struct sdw_intel_res *res) { - struct platform_device_info pdevinfo; - struct platform_device *pdev; struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; struct sdw_intel_ctx *ctx; struct acpi_device *adev; struct sdw_slave *slave; @@ -116,67 +195,60 @@ static struct sdw_intel_ctx count = res->count; dev_dbg(&adev->dev, "Creating %d SDW Link devices\n", count); - ctx = devm_kzalloc(&adev->dev, sizeof(*ctx), GFP_KERNEL); + /* + * we need to alloc/free memory manually and can't use devm: + * this routine may be called from a workqueue, and not from + * the parent .probe. + * If devm_ was used, the memory might never be freed on errors. + */ + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return NULL; ctx->count = count; - ctx->links = devm_kcalloc(&adev->dev, ctx->count, - sizeof(*ctx->links), GFP_KERNEL); - if (!ctx->links) + + /* + * allocate the array of pointers. The link-specific data is allocated + * as part of the first loop below and released with the auxiliary_device_uninit(). + * If some links are disabled, the link pointer will remain NULL. Given that the + * number of links is small, this is simpler than using a list to keep track of links. + */ + ctx->ldev = kcalloc(ctx->count, sizeof(*ctx->ldev), GFP_KERNEL); + if (!ctx->ldev) { + kfree(ctx); return NULL; + } - ctx->count = count; ctx->mmio_base = res->mmio_base; ctx->link_mask = res->link_mask; ctx->handle = res->handle; mutex_init(&ctx->shim_lock); - link = ctx->links; link_mask = ctx->link_mask; INIT_LIST_HEAD(&ctx->link_list); - /* Create SDW Master devices */ - for (i = 0; i < count; i++, link++) { - if (!(link_mask & BIT(i))) { - dev_dbg(&adev->dev, - "Link %d masked, will not be enabled\n", i); + for (i = 0; i < count; i++) { + if (!(link_mask & BIT(i))) continue; - } - link->mmio_base = res->mmio_base; - link->registers = res->mmio_base + SDW_LINK_BASE - + (SDW_LINK_SIZE * i); - link->shim = res->mmio_base + SDW_SHIM_BASE; - link->alh = res->mmio_base + SDW_ALH_BASE; - - link->ops = res->ops; - link->dev = res->dev; - - link->clock_stop_quirks = res->clock_stop_quirks; - link->shim_lock = &ctx->shim_lock; - link->shim_mask = &ctx->shim_mask; - link->link_mask = link_mask; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - - pdevinfo.parent = res->parent; - pdevinfo.name = "intel-sdw"; - pdevinfo.id = i; - pdevinfo.fwnode = acpi_fwnode_handle(adev); - pdevinfo.data = link; - pdevinfo.size_data = sizeof(*link); - - pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(pdev)) { - dev_err(&adev->dev, - "platform device creation failed: %ld\n", - PTR_ERR(pdev)); + /* + * init and add a device for each link + * + * The name of the device will be soundwire_intel.link.[i], + * with the "soundwire_intel" module prefix automatically added + * by the auxiliary bus core. + */ + ldev = intel_link_dev_register(res, + ctx, + acpi_fwnode_handle(adev), + "link", + i); + if (IS_ERR(ldev)) goto err; - } - link->pdev = pdev; - link->cdns = platform_get_drvdata(pdev); + + link = &ldev->link_res; + link->cdns = dev_get_drvdata(&ldev->auxdev.dev); if (!link->cdns) { dev_err(&adev->dev, "failed to get link->cdns\n"); @@ -194,8 +266,7 @@ static struct sdw_intel_ctx num_slaves++; } - ctx->ids = devm_kcalloc(&adev->dev, num_slaves, - sizeof(*ctx->ids), GFP_KERNEL); + ctx->ids = kcalloc(num_slaves, sizeof(*ctx->ids), GFP_KERNEL); if (!ctx->ids) goto err; @@ -213,8 +284,14 @@ static struct sdw_intel_ctx return ctx; err: - ctx->count = i; - sdw_intel_cleanup(ctx); + while (i--) { + if (!(link_mask & BIT(i))) + continue; + ldev = ctx->ldev[i]; + intel_link_dev_unregister(ldev); + } + kfree(ctx->ldev); + kfree(ctx); return NULL; } @@ -222,7 +299,7 @@ static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { struct acpi_device *adev; - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; @@ -241,27 +318,28 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) return -EINVAL; } - if (!ctx->links) + if (!ctx->ldev) return -EINVAL; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_startup(link->pdev); + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) { + intel_link_startup(&ldev->auxdev); + + if (!ldev->link_res.clock_stop_quirks) { /* * we need to prevent the parent PCI device * from entering pm_runtime suspend, so that * power rails to the SoundWire IP are not * turned off. */ - pm_runtime_get_noresume(link->dev); + pm_runtime_get_noresume(ldev->link_res.dev); } } @@ -272,8 +350,8 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) * sdw_intel_probe() - SoundWire Intel probe routine * @res: resource data * - * This registers a platform device for each Master handled by the controller, - * and SoundWire Master and Slave devices will be created by the platform + * This registers an auxiliary device for each Master handled by the controller, + * and SoundWire Master and Slave devices will be created by the auxiliary * device probe. All the information necessary is stored in the context, and * the res argument pointer can be freed after this step. * This function will be called after sdw_intel_acpi_scan() by SOF probe. @@ -306,27 +384,31 @@ EXPORT_SYMBOL_NS(sdw_intel_startup, SOUNDWIRE_INTEL_INIT); void sdw_intel_exit(struct sdw_intel_ctx *ctx) { sdw_intel_cleanup(ctx); + kfree(ctx->ids); + kfree(ctx->ldev); + kfree(ctx); } EXPORT_SYMBOL_NS(sdw_intel_exit, SOUNDWIRE_INTEL_INIT); void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!ctx->links) + if (!ctx->ldev) return; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_process_wakeen_event(link->pdev); + ldev = ctx->ldev[i]; + + intel_link_process_wakeen_event(&ldev->auxdev); } } EXPORT_SYMBOL_NS(sdw_intel_process_wakeen_event, SOUNDWIRE_INTEL_INIT); diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 3a5446ac014a..1ebea7764011 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -58,7 +58,7 @@ struct sdw_intel_acpi_info { u32 link_mask; }; -struct sdw_intel_link_res; +struct sdw_intel_link_dev; /* Intel clock-stop/pm_runtime quirk definitions */ @@ -109,7 +109,7 @@ struct sdw_intel_slave_id { * Controller * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle - * @links: information for each link (controller-specific and kept + * @ldev: information for each link (controller-specific and kept * opaque here) * @ids: array of slave_id, representing Slaves exposed across all enabled * links @@ -123,7 +123,7 @@ struct sdw_intel_ctx { u32 link_mask; int num_slaves; acpi_handle handle; - struct sdw_intel_link_res *links; + struct sdw_intel_link_dev **ldev; struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ -- 2.17.1

5 14

[PATCH] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Kuninori Morimoto 14 Jun '21

14 Jun '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This driver is assuming that all adg->clk[i] is not NULL. Because of this prerequisites, for_each_rsnd_clk() is possible to work for all clk without checking NULL. In other words, all adg->clk[i] should not NULL. Some SoC might doesn't have clk_a/b/c/i. devm_clk_get() returns error in such case. This driver calls rsnd_adg_null_clk_get() and use null_clk instead of NULL in such cases. But devm_clk_get() might returns NULL even though such clocks exist, but it doesn't mean error (user deliberately chose to disable the feature). NULL clk itself is not error from clk point of view, but is error from this driver point of view because it is not assuming such case. But current code is using IS_ERR() which doesn't care NULL. This driver uses IS_ERR_OR_NULL() instead of IS_ERR() for clk check. And it uses ERR_CAST() to clarify null_clk error. One concern here is that it unconditionally uses null_clk if clk_a/b/c/i was error. It is correct if it doesn't exist, but is not correct if it returns error even though it exist. It needs to check "clock-names" from DT before calling devm_clk_get() to handling such case. But let's assume it is overkill so far. Link: https://lore.kernel.org/r/YMCmhfQUimHCSH/n@mwanda Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0ebee1ed06a9..abe9d539709b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -393,7 +393,7 @@ static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, clk = clk_register_fixed_rate(dev, name, parent, 0, 0); if (IS_ERR(clk)) { dev_err(dev, "create null clk error\n"); - return NULL; + return ERR_CAST(clk); } return clk; @@ -430,9 +430,9 @@ static int rsnd_adg_get_clkin(struct rsnd_priv *priv) for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) clk = rsnd_adg_null_clk_get(priv); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clk[i] = clk; -- 2.25.1

2 2

Some minor fixes for Topology2.0
by GitHub pull_request - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-utils pull request #98 was opened from ranj063: Request URL : https://github.com/alsa-project/alsa-utils/pull/98 Patch URL : https://github.com/alsa-project/alsa-utils/pull/98.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH -next 0/9] ASoC: fsl: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 13 Jun '21

13 Jun '21

patch #1 ~ #8: Use devm_platform_get_and_ioremap_resource() patch #9 check return value of platform_get_resource_byname() Yang Yingliang (9): ASoC: fsl_asrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_aud2htx: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_easrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_esai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_micfil: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_sai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_spdif: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_ssi: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_xcvr: check return value after calling platform_get_resource_byname() sound/soc/fsl/fsl_asrc.c | 3 +-- sound/soc/fsl/fsl_aud2htx.c | 3 +-- sound/soc/fsl/fsl_easrc.c | 3 +-- sound/soc/fsl/fsl_esai.c | 3 +-- sound/soc/fsl/fsl_micfil.c | 3 +-- sound/soc/fsl/fsl_sai.c | 3 +-- sound/soc/fsl/fsl_spdif.c | 3 +-- sound/soc/fsl/fsl_ssi.c | 3 +-- sound/soc/fsl/fsl_xcvr.c | 4 ++++ 9 files changed, 12 insertions(+), 16 deletions(-) -- 2.25.1

2 10

alsa-lib 1.2.5 breaks espeak
by GitHub issues - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-lib issue #155 was opened from MrBitBucket: In archlinux amd64 upgrading alsa-lib from 1.2.4 - 1.2.5 breaks the espeak application https://sourceforge.net/projects/espeak/. I see extra errors in espeak output eg ```Expression 'framesAvail' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3629 Expression 'PaAlsaStreamComponent_GetAvailableFrames( self, &framesAvail, xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3989 Expression 'PaAlsaStreamComponent_RegisterChannels( &self->playback, &self->bufferProcessor, &playbackFrames, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4118 Expression 'PaAlsaStream_SetUpBuffers( stream, &framesGot, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4359``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/155 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] ALSA: bebob/fireworks: device entry corrections
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Hi, After working for hardware database of systemd[1], I realize that current device entries of ALSA bebob/fireworks drivers includes some issues. This patchset corrects them. [1] https://github.com/systemd/systemd/pull/19124 Takashi Sakamoto (6): ALSA: bebob: fulfil device entries ALSA: fireworks: fulfil device entries ALSA: bebob: correct device entry for Mackie D.2 FireWire option card ALSA: bebob: correct device entry for Acoustic Reality eAR Master One, Eroica, Figaro, and Ciaccona ALSA: bebob: code refactoring for M-Audio models ALSA: bebob: correct device entries for Phonic Helix Board and FireFly series sound/firewire/bebob/bebob.c | 92 ++++++++++++++-------------- sound/firewire/bebob/bebob.h | 9 --- sound/firewire/fireworks/fireworks.c | 15 +++++ sound/firewire/fireworks/fireworks.h | 8 --- 4 files changed, 61 insertions(+), 63 deletions(-) -- 2.30.2

2 7

[PATCH][next] ALSA: i2c: tea6330t: Remove redundant initialization of variable err
by Colin King 12 Jun '21

12 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The variable err is being initialized with a value that is never read, it is being updated later on. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/i2c/tea6330t.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 742d0f724375..037d6293f728 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -284,7 +284,7 @@ int snd_tea6330t_update_mixer(struct snd_card *card, struct tea6330t *tea; const struct snd_kcontrol_new *knew; unsigned int idx; - int err = -ENOMEM; + int err; u8 default_treble, default_bass; unsigned char bytes[7]; -- 2.31.1

2 1

[PATCH] ALSA: firewire-lib: replace in_interrupt() with in_softirq()
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Tasklet is used to handle isochronous context of 1394 OHCI. The explicit usage of in_softirq() may be preferable than in_interrupt(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 2 +- sound/firewire/amdtp-stream.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index aa53c13b89d3..5fd2aeccdfc2 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -49,7 +49,7 @@ TRACE_EVENT(amdtp_packet, __entry->data_blocks = data_blocks; __entry->data_block_counter = data_block_counter, __entry->packet_index = packet_index; - __entry->irq = !!in_interrupt(); + __entry->irq = !!in_softirq(); __entry->index = index; ), TP_printk( diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 1d9bc7b07df1..aad9778d1c4d 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -610,7 +610,7 @@ static void update_pcm_pointers(struct amdtp_stream *s, // buffer associated to PCM substream to process PCM frames in the buffer, instead // of receiving notification of period elapsed by poll wait. if (!pcm->runtime->no_period_wakeup) { - if (in_interrupt()) { + if (in_softirq()) { // In software IRQ context for 1394 OHCI. snd_pcm_period_elapsed(pcm); } else { @@ -1026,7 +1026,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, const __be32 *ctx_header, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (in_interrupt()) + if (in_softirq()) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } @@ -1738,7 +1738,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. - if (!in_interrupt()) + if (!in_softirq()) fw_iso_context_flush_completions(irq_target->context); } -- 2.30.2

2 1