July 2020 - Alsa-devel - mailman.alsa-project.org

[RESEND][PATCH v3 7/8] ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio
by Rohit kumar 08 Jul '20

08 Jul '20

From: Ajit Pandey <ajitp(a)codeaurora.org> Add platform driver for configuring sc7180 lpass core I2S and DMA configuration to support playback & capture to external codecs connected over primary & secondary MI2S interfaces. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Resending to update Signed-off mail id. sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-sc7180.c | 216 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 223 insertions(+) create mode 100644 sound/soc/qcom/lpass-sc7180.c diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 0ea4cde..87bec7f 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -24,6 +24,11 @@ config SND_SOC_LPASS_APQ8016 select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM +config SND_SOC_LPASS_SC7180 + tristate + select SND_SOC_LPASS_CPU + select SND_SOC_LPASS_PLATFORM + config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" depends on SND_SOC_QCOM diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 41b2c7a..7972c94 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -4,11 +4,13 @@ snd-soc-lpass-cpu-objs := lpass-cpu.o snd-soc-lpass-platform-objs := lpass-platform.o snd-soc-lpass-ipq806x-objs := lpass-ipq806x.o snd-soc-lpass-apq8016-objs := lpass-apq8016.o +snd-soc-lpass-sc7180-objs := lpass-sc7180.o obj-$(CONFIG_SND_SOC_LPASS_CPU) += snd-soc-lpass-cpu.o obj-$(CONFIG_SND_SOC_LPASS_PLATFORM) += snd-soc-lpass-platform.o obj-$(CONFIG_SND_SOC_LPASS_IPQ806X) += snd-soc-lpass-ipq806x.o obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o +obj-$(CONFIG_SND_SOC_LPASS_SC7180) += snd-soc-lpass-sc7180.o # Machine snd-soc-storm-objs := storm.o diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c new file mode 100644 index 00000000..dd85a97 --- /dev/null +++ b/sound/soc/qcom/lpass-sc7180.c @@ -0,0 +1,216 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Copyright (c) 2020, The Linux Foundation. All rights reserved. + * + * lpass-sc7180.c -- ALSA SoC platform-machine driver for QTi LPASS + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/err.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <dt-bindings/sound/sc7180-lpass.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#include "lpass-lpaif-reg.h" +#include "lpass.h" + +static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { + [MI2S_PRIMARY] = { + .id = MI2S_PRIMARY, + .name = "Primary MI2S", + .playback = { + .stream_name = "Primary Playback", + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .stream_name = "Primary Capture", + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + }, + .probe = &asoc_qcom_lpass_cpu_dai_probe, + .ops = &asoc_qcom_lpass_cpu_dai_ops, + }, + + [MI2S_SECONDARY] = { + .id = MI2S_SECONDARY, + .name = "Secondary MI2S", + .playback = { + .stream_name = "Secondary Playback", + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + }, + .probe = &asoc_qcom_lpass_cpu_dai_probe, + .ops = &asoc_qcom_lpass_cpu_dai_ops, + }, +}; + +static int sc7180_lpass_alloc_dma_channel(struct lpass_data *drvdata, + int direction) +{ + struct lpass_variant *v = drvdata->variant; + int chan = 0; + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + chan = find_first_zero_bit(&drvdata->dma_ch_bit_map, + v->rdma_channels); + + if (chan >= v->rdma_channels) + return -EBUSY; + } else { + chan = find_next_zero_bit(&drvdata->dma_ch_bit_map, + v->wrdma_channel_start + + v->wrdma_channels, + v->wrdma_channel_start); + + if (chan >= v->wrdma_channel_start + v->wrdma_channels) + return -EBUSY; + } + + set_bit(chan, &drvdata->dma_ch_bit_map); + + return chan; +} + +static int sc7180_lpass_free_dma_channel(struct lpass_data *drvdata, int chan) +{ + clear_bit(chan, &drvdata->dma_ch_bit_map); + + return 0; +} + +static int sc7180_lpass_init(struct platform_device *pdev) +{ + struct lpass_data *drvdata = platform_get_drvdata(pdev); + struct lpass_variant *variant = drvdata->variant; + struct device *dev = &pdev->dev; + int ret, i; + + drvdata->clks = devm_kcalloc(dev, variant->num_clks, + sizeof(*drvdata->clks), GFP_KERNEL); + drvdata->num_clks = variant->num_clks; + + for (i = 0; i < drvdata->num_clks; i++) + drvdata->clks[i].id = variant->clk_name[i]; + + ret = devm_clk_bulk_get(dev, drvdata->num_clks, drvdata->clks); + if (ret) { + dev_err(dev, "Failed to get clocks %d\n", ret); + return ret; + } + + ret = clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); + if (ret) { + dev_err(dev, "sc7180 clk_enable failed\n"); + return ret; + } + + return 0; +} + +static int sc7180_lpass_exit(struct platform_device *pdev) +{ + struct lpass_data *drvdata = platform_get_drvdata(pdev); + + clk_bulk_disable_unprepare(drvdata->num_clks, drvdata->clks); + + return 0; +} + +static struct lpass_variant sc7180_data = { + .i2sctrl_reg_base = 0x1000, + .i2sctrl_reg_stride = 0x1000, + .i2s_ports = 3, + .irq_reg_base = 0x9000, + .irq_reg_stride = 0x1000, + .irq_ports = 3, + .rdma_reg_base = 0xC000, + .rdma_reg_stride = 0x1000, + .rdma_channels = 5, + .dmactl_audif_start = 1, + .wrdma_reg_base = 0x18000, + .wrdma_reg_stride = 0x1000, + .wrdma_channel_start = 5, + .wrdma_channels = 4, + + .loopback = REG_FIELD_ID(0x1000, 17, 17, 3, 0x1000), + .spken = REG_FIELD_ID(0x1000, 16, 16, 3, 0x1000), + .spkmode = REG_FIELD_ID(0x1000, 11, 15, 3, 0x1000), + .spkmono = REG_FIELD_ID(0x1000, 10, 10, 3, 0x1000), + .micen = REG_FIELD_ID(0x1000, 9, 9, 3, 0x1000), + .micmode = REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000), + .micmono = REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000), + .wssrc = REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000), + + .rdma_dyncclk = REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000), + .rdma_bursten = REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000), + .rdma_wpscnt = REG_FIELD_ID(0xC000, 16, 19, 5, 0x1000), + .rdma_intf = REG_FIELD_ID(0xC000, 12, 15, 5, 0x1000), + .rdma_fifowm = REG_FIELD_ID(0xC000, 1, 5, 5, 0x1000), + .rdma_enable = REG_FIELD_ID(0xC000, 0, 0, 5, 0x1000), + + .wrdma_dyncclk = REG_FIELD_ID(0x18000, 22, 22, 4, 0x1000), + .wrdma_bursten = REG_FIELD_ID(0x18000, 21, 21, 4, 0x1000), + .wrdma_wpscnt = REG_FIELD_ID(0x18000, 17, 20, 4, 0x1000), + .wrdma_intf = REG_FIELD_ID(0x18000, 12, 16, 4, 0x1000), + .wrdma_fifowm = REG_FIELD_ID(0x18000, 1, 5, 4, 0x1000), + .wrdma_enable = REG_FIELD_ID(0x18000, 0, 0, 4, 0x1000), + + .clk_name = (const char*[]) { + "noc", + "audio-core", + "sysnoc_mport", + }, + .num_clks = 3, + .dai_driver = sc7180_lpass_cpu_dai_driver, + .num_dai = ARRAY_SIZE(sc7180_lpass_cpu_dai_driver), + .dai_osr_clk_names = (const char *[]) { + "mclk0", + "null", + }, + .dai_bit_clk_names = (const char *[]) { + "pri_ibit", + "sec_ibit", + }, + .init = sc7180_lpass_init, + .exit = sc7180_lpass_exit, + .alloc_dma_channel = sc7180_lpass_alloc_dma_channel, + .free_dma_channel = sc7180_lpass_free_dma_channel, +}; + +static const struct of_device_id sc7180_lpass_cpu_device_id[] = { + {.compatible = "qcom,lpass-cpu-sc7180", .data = &sc7180_data}, + {} +}; + +static struct platform_driver sc7180_lpass_cpu_platform_driver = { + .driver = { + .name = "sc7180-lpass-cpu", + .of_match_table = of_match_ptr(sc7180_lpass_cpu_device_id), + }, + .probe = asoc_qcom_lpass_cpu_platform_probe, + .remove = asoc_qcom_lpass_cpu_platform_probe, +}; + +module_platform_driver(sc7180_lpass_cpu_platform_driver); + +MODULE_DESCRIPTION("SC7180 LPASS CPU DRIVER"); +MODULE_LICENSE("GPL v2"); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH v2] ASoC: atmel-classd: remove codec component
by Codrin Ciubotariu 08 Jul '20

08 Jul '20

The CPU and the codec both are represented now as components, so for CLASS-D we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v2: - removed no longer used ATMEL_CLASSD_CODEC_DAI_NAME macro; sound/soc/atmel/atmel-classd.c | 134 ++++++++++++--------------------- 1 file changed, 48 insertions(+), 86 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index e98601eccfa3..24bc93c468af 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -5,7 +5,6 @@ * * Author: Songjun Wu <songjun.wu(a)atmel.com> */ - #include <linux/of.h> #include <linux/clk.h> #include <linux/module.h> @@ -120,39 +119,21 @@ static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int err; regmap_write(dd->regmap, CLASSD_THR, 0x0); - return clk_prepare_enable(dd->pclk); -} - -static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - clk_disable_unprepare(dd->pclk); + err = clk_prepare_enable(dd->pclk); + if (err) + return err; + err = clk_prepare_enable(dd->gclk); + if (err) { + clk_disable_unprepare(dd->pclk); + return err; + } + return 0; } -static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { - .startup = atmel_classd_cpu_dai_startup, - .shutdown = atmel_classd_cpu_dai_shutdown, -}; - -static struct snd_soc_dai_driver atmel_classd_cpu_dai = { - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = ATMEL_CLASSD_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &atmel_classd_cpu_dai_ops, -}; - -static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { - .name = "atmel-classd", -}; - /* platform */ static int atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, @@ -306,31 +287,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component) return regcache_sync(dd->regmap); } -static struct snd_soc_component_driver soc_component_dev_classd = { - .probe = atmel_classd_component_probe, - .resume = atmel_classd_component_resume, - .controls = atmel_classd_snd_controls, - .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ -static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - return clk_prepare_enable(dd->gclk); -} - -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, - int mute) +static int atmel_classd_cpu_dai_digital_mute(struct snd_soc_dai *cpu_dai, + int mute) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; @@ -373,13 +333,13 @@ static struct { }; static int -atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; int fs; int i, best, best_val, cur_val, ret; u32 mask, val; @@ -417,8 +377,8 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, } static void -atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); @@ -426,10 +386,10 @@ atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(dd->gclk); } -static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; snd_soc_component_update_bits(component, CLASSD_MR, CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, @@ -439,10 +399,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, return 0; } -static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; @@ -468,19 +428,16 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { - .digital_mute = atmel_classd_codec_dai_digital_mute, - .startup = atmel_classd_codec_dai_startup, - .shutdown = atmel_classd_codec_dai_shutdown, - .hw_params = atmel_classd_codec_dai_hw_params, - .prepare = atmel_classd_codec_dai_prepare, - .trigger = atmel_classd_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, + .digital_mute = atmel_classd_cpu_dai_digital_mute, + .hw_params = atmel_classd_cpu_dai_hw_params, + .prepare = atmel_classd_cpu_dai_prepare, + .trigger = atmel_classd_cpu_dai_trigger, }; -#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" - -static struct snd_soc_dai_driver atmel_classd_codec_dai = { - .name = ATMEL_CLASSD_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, @@ -488,7 +445,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = { .rates = ATMEL_CLASSD_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &atmel_classd_codec_dai_ops, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", + .probe = atmel_classd_component_probe, + .resume = atmel_classd_component_resume, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -517,9 +485,10 @@ static int atmel_classd_asoc_card_init(struct device *dev, dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; - dai_link->codecs->dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; + dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -620,13 +589,6 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd, - &atmel_classd_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

1 0

[PATCH] ASoC: atmel-classd: remove codec component
by Codrin Ciubotariu 08 Jul '20

08 Jul '20

The CPU and the codec both are represented now as components, so for CLASS-D we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/atmel-classd.c | 133 ++++++++++++--------------------- 1 file changed, 48 insertions(+), 85 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index e98601eccfa3..d1ce67d30cd7 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -5,7 +5,6 @@ * * Author: Songjun Wu <songjun.wu(a)atmel.com> */ - #include <linux/of.h> #include <linux/clk.h> #include <linux/module.h> @@ -120,39 +119,21 @@ static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int err; regmap_write(dd->regmap, CLASSD_THR, 0x0); - return clk_prepare_enable(dd->pclk); -} - -static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - clk_disable_unprepare(dd->pclk); + err = clk_prepare_enable(dd->pclk); + if (err) + return err; + err = clk_prepare_enable(dd->gclk); + if (err) { + clk_disable_unprepare(dd->pclk); + return err; + } + return 0; } -static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { - .startup = atmel_classd_cpu_dai_startup, - .shutdown = atmel_classd_cpu_dai_shutdown, -}; - -static struct snd_soc_dai_driver atmel_classd_cpu_dai = { - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = ATMEL_CLASSD_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &atmel_classd_cpu_dai_ops, -}; - -static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { - .name = "atmel-classd", -}; - /* platform */ static int atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, @@ -306,31 +287,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component) return regcache_sync(dd->regmap); } -static struct snd_soc_component_driver soc_component_dev_classd = { - .probe = atmel_classd_component_probe, - .resume = atmel_classd_component_resume, - .controls = atmel_classd_snd_controls, - .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ -static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - return clk_prepare_enable(dd->gclk); -} - -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, - int mute) +static int atmel_classd_cpu_dai_digital_mute(struct snd_soc_dai *cpu_dai, + int mute) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; @@ -373,13 +333,13 @@ static struct { }; static int -atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; int fs; int i, best, best_val, cur_val, ret; u32 mask, val; @@ -417,8 +377,8 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, } static void -atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); @@ -426,10 +386,10 @@ atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(dd->gclk); } -static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; snd_soc_component_update_bits(component, CLASSD_MR, CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, @@ -439,10 +399,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, return 0; } -static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; @@ -468,19 +428,17 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { - .digital_mute = atmel_classd_codec_dai_digital_mute, - .startup = atmel_classd_codec_dai_startup, - .shutdown = atmel_classd_codec_dai_shutdown, - .hw_params = atmel_classd_codec_dai_hw_params, - .prepare = atmel_classd_codec_dai_prepare, - .trigger = atmel_classd_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, + .digital_mute = atmel_classd_cpu_dai_digital_mute, + .hw_params = atmel_classd_cpu_dai_hw_params, + .prepare = atmel_classd_cpu_dai_prepare, + .trigger = atmel_classd_cpu_dai_trigger, }; #define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" - -static struct snd_soc_dai_driver atmel_classd_codec_dai = { - .name = ATMEL_CLASSD_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, @@ -488,7 +446,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = { .rates = ATMEL_CLASSD_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &atmel_classd_codec_dai_ops, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", + .probe = atmel_classd_component_probe, + .resume = atmel_classd_component_resume, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -517,9 +486,10 @@ static int atmel_classd_asoc_card_init(struct device *dev, dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; - dai_link->codecs->dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; + dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -620,13 +590,6 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd, - &atmel_classd_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

1 0

[PATCH v2 00/28] Clean-up ASoC's W=1 build warnings
by Lee Jones 08 Jul '20

08 Jul '20

This set is part of a larger effort attempting to clean-up W=1 kernel builds, which are currently overwhelmingly riddled with niggly little warnings. After these patches are applied, the build system no longer complains about any W=0 nor W=1 level warnings in sound/soc, when building for Arm. Hurrah! Changelog v1 => v2: - Remove whole read from patch 21 - Suggested by Peter Lee Jones (28): ASoC: atmel: atmel-pcm-dma: Demote function header which is clearly not kerneldoc ASoC: atmel: atmel_ssc_dai: Demote function header which is clearly not kerneldoc ASoC: fsl: fsl-asoc-card: Use correct format when providing struct documentation ASoC: fsl: fsl_asrc: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi_dbg: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_spdif: Use correct formatting when denoting struct documentation ASoC: fsl: fsl_spdif: Update 'struct fsl_spdif_priv's descriptions ASoC: fsl: fsl_esai: Fix a bunch of kerneldoc issues ASoC: fsl: fsl_esai: Add descriptions for 'tx_mask' and 'rx_mask' to 'fsl_esai' ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' ASoC: qcom: qdsp6: q6asm: Provide documentation for WMA 'codec_profile' ASoC: samsung: pcm: Demote half-documented kerneldoc header ASoC: samsung: spdif: Fix 3 misspelled struct properties ASoC: sunxi: sun4i-i2s: Demote half-documented kerneldoc header ASoC: uniphier: aio-core: Fix incorrectly named struct property and remove another ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: sti: uniperif: Mark 'uni_tdm_hw' as __maybe_unused ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' ASoC: codecs: da7210: Fix formatting issues in da7210_set_dai_pll() header ASoC: soc-ac97: Demote seemingly unintentional kerneldoc header ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: codecs: rt5631: Demote misuse of kerneldoc to standard comment blocks ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: wm8400: Remove a bunch of unused variables sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/da7210.c | 12 ++++++----- sound/soc/codecs/jz4770.c | 1 - sound/soc/codecs/rt5631.c | 12 +++++------ sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/wm8400.c | 12 ----------- sound/soc/fsl/fsl-asoc-card.c | 16 +++++++------- sound/soc/fsl/fsl_asrc.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_esai.c | 19 ++++++++++------- sound/soc/fsl/fsl_spdif.c | 5 +++-- sound/soc/fsl/fsl_ssi.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 ++-- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 1 + sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/spdif.c | 6 +++--- sound/soc/soc-ac97.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ti/omap-mcbsp-st.c | 3 --- sound/soc/uniphier/aio-core.c | 7 +++---- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 26 files changed, 88 insertions(+), 135 deletions(-) -- 2.25.1

4 31

[PATCH v2] ALSA: usb/line6: remove 'defined but not used' warning
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

Fix W=1 warning. Variables are declared in a header file included from multiple C files, replace by #defines as suggested by Takashi sound/usb/line6/driver.h:70:18: warning: ‘SYSEX_EXTRA_SIZE’ defined but not used [-Wunused-const-variable=] 70 | static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4; | ^~~~~~~~~~~~~~~~ sound/usb/line6/driver.h:69:18: warning: ‘SYSEX_DATA_OFS’ defined but not used [-Wunused-const-variable=] 69 | static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3; | ^~~~~~~~~~~~~~ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/line6/driver.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 1a4e3700c80c..ec20fbec6475 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -66,8 +66,8 @@ extern const unsigned char line6_midi_id[3]; -static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3; -static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4; +#define SYSEX_DATA_OFS (sizeof(line6_midi_id) + 3) +#define SYSEX_EXTRA_SIZE (sizeof(line6_midi_id) + 4) /* Common properties of Line 6 devices. -- 2.25.1

2 1

[PATCH] ALSA: Use fallthrough pseudo-keyword
by Gustavo A. R. Silva 08 Jul '20

08 Jul '20

Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. Also, remove unnecessary fall-through markings when it is the case. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fa… Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/atmel/ac97c.c | 16 ++++++++-------- sound/core/memalloc.c | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/oss/pcm_plugin.c | 2 +- sound/core/pcm_native.c | 8 ++++---- sound/core/seq/oss/seq_oss_timer.c | 2 +- sound/core/seq/seq_midi_emul.c | 2 +- sound/drivers/opl3/opl3_midi.c | 4 ++-- sound/isa/galaxy/galaxy.c | 6 +++--- sound/isa/msnd/msnd_pinnacle_mixer.c | 3 +-- sound/isa/opti9xx/miro.c | 10 +++++----- sound/isa/opti9xx/opti92x-ad1848.c | 12 +++++++----- sound/isa/sb/sb8_main.c | 10 +++++----- sound/oss/dmasound/dmasound_atari.c | 2 +- sound/oss/dmasound/dmasound_core.c | 4 ++-- sound/pci/ac97/ac97_codec.c | 4 ++-- sound/pci/atiixp.c | 6 +++--- sound/pci/azt3328.c | 2 +- sound/pci/echoaudio/echoaudio.c | 4 ++-- sound/pci/emu10k1/emupcm.c | 5 ++--- sound/pci/hda/hda_beep.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 4 ++-- sound/pci/ice1712/delta.c | 2 +- sound/pci/intel8x0.c | 8 ++++---- sound/pci/mixart/mixart.c | 2 +- sound/pci/mixart/mixart_core.c | 2 +- sound/pci/nm256/nm256.c | 6 +++--- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 4 ++-- sound/sparc/dbri.c | 8 ++++---- sound/usb/card.c | 2 +- sound/usb/clock.c | 2 +- sound/usb/pcm.c | 2 +- sound/usb/stream.c | 3 +-- 35 files changed, 78 insertions(+), 79 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index a1dce9725b98..a62f1cdad68f 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -356,14 +356,14 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) camr = ac97c_readl(chip, CAMR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_TXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_TXTDIS; if (chip->opened <= 1) @@ -388,14 +388,14 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) ptcr = readl(chip->regs + ATMEL_PDC_PTSR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_RXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_RXTDIS; if (chip->opened <= 1) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bea46ed157a6..9f3e14aebe30 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -157,7 +157,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, */ dmab->dev.type = SNDRV_DMA_TYPE_DEV; #endif /* CONFIG_GENERIC_ALLOCATOR */ - /* fall through */ + fallthrough; case SNDRV_DMA_TYPE_DEV: case SNDRV_DMA_TYPE_DEV_UC: snd_malloc_dev_pages(dmab, size); diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 68630244b00f..327ec42a36b0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2851,7 +2851,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream) break; - /* Fall through */ + fallthrough; case VM_READ: substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; break; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 1545f8fdb4db..d5ca161d588c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -357,7 +357,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, if (snd_mask_test(format_mask, (__force int)format1)) return format1; } - /* fall through */ + fallthrough; default: return (__force snd_pcm_format_t)-EINVAL; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9630d2523948..84b140f800fd 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1903,7 +1903,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, switch (substream->runtime->status->state) { case SNDRV_PCM_STATE_PAUSED: snd_pcm_pause(substream, false); - /* fallthru */ + fallthrough; case SNDRV_PCM_STATE_SUSPENDED: snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); break; @@ -2811,7 +2811,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream) case SNDRV_PCM_STATE_DRAINING: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) return -EBADFD; - /* Fall through */ + fallthrough; case SNDRV_PCM_STATE_RUNNING: return snd_pcm_update_hw_ptr(substream); case SNDRV_PCM_STATE_PREPARED: @@ -3816,7 +3816,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_STATUS_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_STATUS_NEW: if (!pcm_status_mmap_allowed(pcm_file)) return -ENXIO; @@ -3824,7 +3824,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW: if (!pcm_control_mmap_allowed(pcm_file)) return -ENXIO; diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index a35d429e4c27..f9f57232a83f 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -79,7 +79,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* fall through */ + fallthrough; case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 198f285594e3..81d2ef5e5811 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -309,7 +309,7 @@ do_control(const struct snd_midi_op *ops, void *drv, break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* fall through */ + fallthrough; case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 2f6e8023e05c..eb23c55323ae 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -354,7 +354,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) instr_4op = 1; break; } - /* fall through */ + fallthrough; default: spin_unlock_irqrestore(&opl3->voice_lock, flags); return; @@ -443,7 +443,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) switch (connection) { case 0x03: snd_opl3_calc_volume(&vol_op[2], vel, chan); - /* fallthru */ + fallthrough; case 0x02: snd_opl3_calc_volume(&vol_op[0], vel, chan); break; diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index ce409e75ae51..65f9f46c9f58 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -247,7 +247,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: irq[n] = 9; - /* Fall through */ + fallthrough; case 9: wss_config[n] |= WSS_CONFIG_IRQ_9; break; @@ -292,7 +292,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) case 1: if (dma1[n] == 0) break; - /* Fall through */ + fallthrough; default: dev_err(dev, "invalid capture DMA %d\n", dma2[n]); return 0; @@ -322,7 +322,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: mpu_irq[n] = 9; - /* Fall through */ + fallthrough; case 9: config[n] |= GALAXY_CONFIG_MPUIRQ_2; break; diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 02c566fca9e5..287ab740dd16 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -219,8 +219,7 @@ static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) case MSND_MIXER_VOLUME: /* master volume */ writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); writew(wRight, dev->SMA + SMA_wCurrMastVolRight); - /* fall through */ - + fallthrough; case MSND_MIXER_AUX: /* aux pot control */ /* scaled by master volume */ /* fall through */ diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b039429e6871..44ed1b65f6ce 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -163,13 +163,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci) switch (timeout-ACI_MINTIME) { case 0 ... 9: out /= 10; - /* fall through */ + fallthrough; case 10 ... 19: out /= 10; - /* fall through */ + fallthrough; case 20 ... 30: out /= 10; - /* fall through */ + fallthrough; default: set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(out); @@ -824,7 +824,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip, retval = inb(chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: retval = inb(chip->mc_base + reg); @@ -854,7 +854,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: outb(value, chip->mc_base + reg); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 0e6d20e49158..881d3b5711d2 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -249,7 +249,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, retval = inb(chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -292,7 +292,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -343,7 +343,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C925: /* enable WSS mode */ @@ -380,7 +380,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: /* disable 3D sound (set GPIO1 as output, low) */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C933: /* * The BTC 1817DW has QS1000 wavetable which is connected @@ -392,7 +393,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C930: snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index e33dfe165276..86d0d2fdf48a 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -116,13 +116,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -261,7 +261,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -361,7 +361,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; if (chip->playback_format == SB_DSP_OUTPUT) @@ -371,7 +371,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_CAPTURE_16: if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_CAPTURE_8: substream = chip->capture_substream; if (chip->capture_format == SB_DSP_INPUT) diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 823ccfa089b2..81c6a9830727 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -1449,7 +1449,7 @@ static int FalconMixerIoctl(u_int cmd, u_long arg) tt_dmasnd.input_gain = RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 | RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff); - /* fall through - return set value */ + fallthrough; /* return set value */ case SOUND_MIXER_READ_MIC: return IOCTL_OUT(arg, RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) | diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index f802ea331e24..38f25e97538f 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1478,13 +1478,13 @@ static int dmasound_setup(char *str) printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius); else catchRadius = ints[3]; - /* fall through */ + fallthrough; case 2: if (ints[1] < MIN_BUFFERS) printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs); else numWriteBufs = ints[1]; - /* fall through */ + fallthrough; case 1: if ((size = ints[2]) < 256) /* check for small buffer specs */ size <<= 10 ; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6758c072000e..012a7ee849e8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -218,11 +218,11 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) case AC97_ID_ST_AC97_ID4: if (reg == 0x08) return 0; - /* fall through */ + fallthrough; case AC97_ID_ST7597: if (reg == 0x22 || reg == 0x7a) return 1; - /* fall through */ + fallthrough; case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 85d3b4e95489..a25d75455802 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -896,15 +896,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) case 8: data |= ATI_REG_OUT_DMA_SLOT_BIT(10) | ATI_REG_OUT_DMA_SLOT_BIT(11); - /* fall through */ + fallthrough; case 6: data |= ATI_REG_OUT_DMA_SLOT_BIT(7) | ATI_REG_OUT_DMA_SLOT_BIT(8); - /* fall through */ + fallthrough; case 4: data |= ATI_REG_OUT_DMA_SLOT_BIT(6) | ATI_REG_OUT_DMA_SLOT_BIT(9); - /* fall through */ + fallthrough; default: data |= ATI_REG_OUT_DMA_SLOT_BIT(3) | ATI_REG_OUT_DMA_SLOT_BIT(4); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 58167d8469e1..77c7030ebbfa 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1232,7 +1232,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - /* fall-through */ + fallthrough; case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break; case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break; case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 0941a7a17623..6c285380b150 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -690,7 +690,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) break; case SNDRV_PCM_FORMAT_S32_BE: format.data_are_bigendian = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_FORMAT_S32_LE: format.bits_per_sample = 32; break; @@ -742,7 +742,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) pipe->last_counter = 0; pipe->position = 0; *pipe->dma_counter = 0; - /* fall through */ + fallthrough; case PIPE_STATE_PAUSED: pipe->state = PIPE_STATE_STARTED; break; diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b934c6ac52dd..b2ddabb99438 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -753,7 +753,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]); - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) @@ -902,8 +902,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]); } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); - - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 841523f6b88d..53a2b89f8983 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -102,7 +102,7 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, case SND_BELL: if (hz) hz = 1000; - /* fallthru */ + fallthrough; case SND_TONE: if (beep->linear_tone) beep->tone = beep_linear_tone(beep, hz); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 396b5503038a..9002888670f9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1014,7 +1014,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f150f2: codec->power_save_node = 1; - /* Fall through */ + fallthrough; default: codec->pin_amp_workaround = 1; snd_hda_pick_fixup(codec, cxt5066_fixup_models, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..fcd303b696e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -374,7 +374,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0299: alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); - /* fallthrough */ + fallthrough; case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: @@ -4709,7 +4709,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, break; case 0x10ec0867: alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14); - /* fallthru */ + fallthrough; case 0x10ec0221: case 0x10ec0662: snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 81929063b2fa..1d2a0287284b 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -691,7 +691,7 @@ static int snd_ice1712_delta_init(struct snd_ice1712 *ice) break; case ICE1712_SUBDEVICE_DELTADIO2496: ice->gpio.set_pro_rate = delta_1010_set_rate_val; - /* fall thru */ + fallthrough; case ICE1712_SUBDEVICE_DELTA66: ice->spdif.ops.open = delta_open_spdif; ice->spdif.ops.setup_rate = delta_setup_spdif; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 1781a1c081c3..675bb0da00a0 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -810,7 +810,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ICH_IOCE | ICH_STARTBM; @@ -818,7 +818,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: val = 0; break; @@ -852,7 +852,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -869,7 +869,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* pause */ diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7ba487443c7f..efff220b26ea 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -169,7 +169,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_RUNNING: if(rate != 0) break; - /* fall through */ + fallthrough; default: if(rate == 0) return 0; /* nothing to do */ diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 048a2660d18d..0bdd33b0af65 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -527,7 +527,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) dev_err(&mgr->pci->dev, "canceled notification %x !\n", msg); } - /* fall through */ + fallthrough; case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ mutex_lock(&mgr->msg_lock); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index ebce4d259e06..240195e85e3d 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -560,7 +560,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: s->suspended = 0; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_START: if (! s->running) { snd_nm256_playback_start(chip, s, substream); @@ -569,7 +569,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: s->suspended = 1; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: if (s->running) { snd_nm256_playback_stop(chip); @@ -1664,7 +1664,7 @@ static int snd_nm256_probe(struct pci_dev *pci, return -ENODEV; case NM_RESET_WORKAROUND_2: reset_workaround_2 = 1; - /* Fall-through */ + fallthrough; case NM_RESET_WORKAROUND: reset_workaround = 1; break; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index db7d76a3cfeb..cacc6a9d14c8 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -400,7 +400,7 @@ static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream, kctl = chip->pcm_mixer[substream->number].ctl; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; } - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: chip->ctrl_playback[ypcm->voices[0]->number + 1] = 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 6cbe5cb34358..dfc4295b69c4 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -45,7 +45,7 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) case SNDRV_PCM_TRIGGER_START: chip->pcm_hwptr = 0; chip->pcm_tdone = 0; - /* fall thru */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: mask = 0; @@ -132,7 +132,7 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs) case SNDRV_PCM_FORMAT_S24_3LE: case SNDRV_PCM_FORMAT_S24_3BE: chip->pcm_sample = 3; - /* fall through */ + fallthrough; default: /* 24-bit */ aval = AK4117_DIF_24R; chip->pcm_frame = 3; diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index cf7049999261..17b79d45d17f 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -580,16 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len) switch (len) { case 32: b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16); - /* fall through */ + fallthrough; case 16: b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8); - /* fall through */ + fallthrough; case 8: b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4); - /* fall through */ + fallthrough; case 4: b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2); - /* fall through */ + fallthrough; case 2: b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1); case 1: diff --git a/sound/usb/card.c b/sound/usb/card.c index 162bdd6eb4d4..696e788c5d31 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -230,7 +230,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) dev_warn(&dev->dev, "unknown interface protocol %#02x, assuming v1\n", protocol); - /* fall through */ + fallthrough; case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f..f3ca59005d91 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -670,7 +670,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, else return 0; } - /* fall through */ + fallthrough; case UAC_VERSION_2: return set_sample_rate_v2v3(chip, iface, alts, fmt, rate); } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 8a05dcb1344f..6282c539bbb4 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1741,7 +1741,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea switch (cmd) { case SNDRV_PCM_TRIGGER_START: subs->trigger_tstamp_pending_update = true; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 15296f2c902c..4d1e6579e54d 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1146,9 +1146,8 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n", iface_no, altno, protocol); protocol = UAC_VERSION_1; - /* fall through */ + fallthrough; case UAC_VERSION_1: - /* fall through */ case UAC_VERSION_2: { int bm_quirk = 0;

2 1

[PATCH 00/28] Clean-up ASoC's W=1 build warnings
by Lee Jones 07 Jul '20

07 Jul '20

This set is part of a larger effort attempting to clean-up W=1 kernel builds, which are currently overwhelmingly riddled with niggly little warnings. After these patches are applied, the build system no longer complains about any W=0 nor W=1 level warnings in sound/soc. Hurrah! Lee Jones (28): ASoC: atmel: atmel-pcm-dma: Demote function header which is clearly not kerneldoc ASoC: atmel: atmel_ssc_dai: Demote function header which is clearly not kerneldoc ASoC: fsl: fsl-asoc-card: Use correct format when providing struct documentation ASoC: fsl: fsl_asrc: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi_dbg: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_spdif: Use correct formatting when denoting struct documentation ASoC: fsl: fsl_spdif: Update 'struct fsl_spdif_priv's descriptions ASoC: fsl: fsl_esai: Fix a bunch of kerneldoc issues ASoC: fsl: fsl_esai: Add descriptions for 'tx_mask' and 'rx_mask' to 'fsl_esai' ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' ASoC: qcom: qdsp6: q6asm: Provide documentation for WMA 'codec_profile' ASoC: samsung: pcm: Demote half-documented kerneldoc header ASoC: samsung: spdif: Fix 3 misspelled struct properties ASoC: sunxi: sun4i-i2s: Demote half-documented kerneldoc header ASoC: uniphier: aio-core: Fix incorrectly named struct property and remove another ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: sti: uniperif: Mark 'uni_tdm_hw' as __maybe_unused ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' ASoC: codecs: da7210: Fix formatting issues in da7210_set_dai_pll() header ASoC: soc-ac97: Demote seemingly unintentional kerneldoc header ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: codecs: rt5631: Demote misuse of kerneldoc to standard comment blocks ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: wm8400: Remove a bunch of unused variables sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/da7210.c | 12 ++++++----- sound/soc/codecs/jz4770.c | 1 - sound/soc/codecs/rt5631.c | 12 +++++------ sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/wm8400.c | 12 ----------- sound/soc/fsl/fsl-asoc-card.c | 16 +++++++------- sound/soc/fsl/fsl_asrc.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_esai.c | 19 ++++++++++------- sound/soc/fsl/fsl_spdif.c | 5 +++-- sound/soc/fsl/fsl_ssi.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 ++-- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 1 + sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/spdif.c | 6 +++--- sound/soc/soc-ac97.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ti/omap-mcbsp-st.c | 3 +-- sound/soc/uniphier/aio-core.c | 7 +++---- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 26 files changed, 89 insertions(+), 134 deletions(-) -- 2.25.1

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[PATCH] ASoC: amd: add ACPI dependency check
by Vijendar Mukunda 07 Jul '20

07 Jul '20

Add ACPI dependency for evaluating DMIC hardware runtime. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index d1faea5..165c93f 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -218,6 +218,7 @@ static int snd_rn_acp_probe(struct pci_dev *pci, ret = -ENODEV; goto de_init; } else if (dmic_acpi_check == ACP_DMIC_AUTO) { +#if defined(CONFIG_ACPI) handle = ACPI_HANDLE(&pci->dev); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); if (ACPI_FAILURE(ret)) { @@ -228,6 +229,7 @@ static int snd_rn_acp_probe(struct pci_dev *pci, ret = -ENODEV; goto de_init; } +#endif } adata->res = devm_kzalloc(&pci->dev, -- 2.7.4

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Ignored USB-audio implicit feedback in kernel 5.8rc3
by Pavel Hofman 07 Jul '20

07 Jul '20

Hi, Audio analyzer RTX6001 (XMOS-based) is using implicit feedback, yet the feedback is not used by the latest kernel 5.8 rc3 (i.e. already with the latest implicit-feedback patches). Two analyzers on two different PCs are getting clicks in duplex-mode loobpack, one every 10 secs, the other one every 50 secs. While running aplay/arecord loopback I captured USB packets with tcpdump/tshark. The dump contained 931484 IN packets with 48bytes payload, and 684 IN packets with 40 bytes payload. Packet interval 125us (bInterval = 1), 32bits, 2ch => precise samplerate 47,994.1298Hz. Clearly the incoming stream works OK. Yet stream0 file states for Capture: Momentary freq = 48000 Hz (0x6.0000) But ONLY 48bytes packets were sent by the host in OUT packets, i.e. exactly 48kHz output rate. 6 frames OUT/IN difference per second, the incoming FIFO in the device overflows after several seconds. Please may I ask for help with making the driver respect the implicit feedback stream? Best regards, Pavel. 5.8.0-050800-generic #202006282330 SMP Sun Jun 28 23:35:41 UTC 2020 x86_64 GNU/Linux cat /proc/asound/card4/stream0 RTX A/S RTX6001 USB Audio 2.0 at usb-0000:00:13.2-4, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 72 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 32 Channel map: FL FR Interface 1 Altset 2 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 32 Channel map: FL FR Interface 2 Altset 2 Format: S32_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR Interface 2 Altset 3 Format: S16_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR lsusb: Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 007 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 006 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 002 Device 002: ID 058f:6362 Alcor Micro Corp. Flash Card Reader/Writer Bus 002 Device 003: ID 0d9a:00df RTX AS RTX6001 USB Audio 2.0 <---- Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 005 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 008: ID 2047:0301 Texas Instruments Arduino Micro Bus 004 Device 007: ID 18f8:0f97 [Maxxter] Elo TouchSystems 2216 AccuTouch® USB Touchmonitor Interface Bus 004 Device 006: ID 413c:2001 Dell Computer Corp. Keyboard HID Support Bus 004 Device 005: ID 413c:1001 Dell Computer Corp. Keyboard Hub Bus 004 Device 004: ID 0bd9:0001 Liberty Instruments, Inc. Liberty Praxis AudPod 1.1 Bus 004 Device 003: ID 2341:8037 Arduino SA Arduino Micro Bus 004 Device 002: ID 04e7:0050 Elo TouchSystems 2216 AccuTouch® Touchmonitor Interface Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 009 Device 002: ID 18a5:0422 Verbatim, Ltd Portable SSD Bus 009 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 008 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub lsusb -v see the attachment

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[PATCH v2] ASoC: Intel: bxt-da7219-max98357a: support MAX98390 speaker amp
by Brent Lu 07 Jul '20

07 Jul '20

Support MAX98390 speaker amplifier on cometlake platform. Driver now detects amplifier type in the probe function and installs corresponding controls and DAPM widgets/routes in the late_probe function. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 117 ++++++++++++++++++++-- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 13 +++ 3 files changed, 127 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3d820e1..f176df2 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -288,6 +288,7 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC tristate select SND_SOC_DA7219 select SND_SOC_MAX98357A + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -298,14 +299,14 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH - tristate "Broxton with DA7219 and MAX98357A in I2S Mode" + tristate "Broxton with DA7219 and MAX98357A/MAX98390 in I2S Mode" depends on I2C && ACPI && GPIOLIB depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Broxton-P platforms - with DA7219 + MAX98357A I2S audio codec. + with DA7219 + MAX98357A/MAX98390 I2S audio codec. Say Y or m if you have such a device. This is a recommended option. If unsure select "N". diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 44016c1..4d39253 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -25,9 +25,14 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" +#define MAX98390_DEV0_NAME "i2c-MX98390:00" +#define MAX98390_DEV1_NAME "i2c-MX98390:01" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 +#define SPKAMP_MAX98357A 1 +#define SPKAMP_MAX98390 2 + static struct snd_soc_jack broxton_headset; static struct snd_soc_jack broxton_hdmi[3]; @@ -40,6 +45,7 @@ struct bxt_hdmi_pcm { struct bxt_card_private { struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int spkamp; }; enum { @@ -85,13 +91,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_kcontrol_new max98357a_controls[] = { SOC_DAPM_PIN_SWITCH("Spk"), }; +static const struct snd_kcontrol_new max98390_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + static const struct snd_soc_dapm_widget broxton_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("SoC DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), @@ -100,14 +113,20 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = { platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; +static const struct snd_soc_dapm_widget max98357a_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget max98390_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route audio_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone Jack", NULL, "HPL"}, {"Headphone Jack", NULL, "HPR"}, - /* speaker */ - {"Spk", NULL, "Speaker"}, - /* other jacks */ {"MIC", NULL, "Headset Mic"}, @@ -134,6 +153,17 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headset Mic", NULL, "Platform Clock" }, }; +static const struct snd_soc_dapm_route max98357a_routes[] = { + /* speaker */ + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_soc_dapm_route max98390_routes[] = { + /* Speaker */ + {"Left Spk", NULL, "Left BE_OUT"}, + {"Right Spk", NULL, "Right BE_OUT"}, +}; + static const struct snd_soc_dapm_route broxton_map[] = { {"HiFi Playback", NULL, "ssp5 Tx"}, {"ssp5 Tx", NULL, "codec0_out"}, @@ -404,6 +434,10 @@ SND_SOC_DAILINK_DEF(ssp5_pin, SND_SOC_DAILINK_DEF(ssp5_codec, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", BXT_MAXIM_CODEC_DAI))); +SND_SOC_DAILINK_DEF(max98390_codec, + DAILINK_COMP_ARRAY( + /* Left */ COMP_CODEC(MAX98390_DEV0_NAME, "max98390-aif1"), + /* Right */ COMP_CODEC(MAX98390_DEV1_NAME, "max98390-aif1"))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); @@ -601,15 +635,69 @@ static struct snd_soc_dai_link broxton_dais[] = { }, }; +static struct snd_soc_codec_conf max98390_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV1_NAME), + .name_prefix = "Right", + }, +}; + #define NAME_SIZE 32 static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; - int err, i = 0; + const struct snd_kcontrol_new *controls; + const struct snd_soc_dapm_widget *widgets; + const struct snd_soc_dapm_route *routes; + int num_controls, num_widgets, num_routes, err, i = 0; char jack_name[NAME_SIZE]; + switch (ctx->spkamp) { + case SPKAMP_MAX98357A: + controls = max98357a_controls; + num_controls = ARRAY_SIZE(max98357a_controls); + widgets = max98357a_widgets; + num_widgets = ARRAY_SIZE(max98357a_widgets); + routes = max98357a_routes; + num_routes = ARRAY_SIZE(max98357a_routes); + break; + case SPKAMP_MAX98390: + controls = max98390_controls; + num_controls = ARRAY_SIZE(max98390_controls); + widgets = max98390_widgets; + num_widgets = ARRAY_SIZE(max98390_widgets); + routes = max98390_routes; + num_routes = ARRAY_SIZE(max98390_routes); + break; + default: + dev_err(card->dev, "Invalid speaker amplifier %d\n", ctx->spkamp); + break; + } + + err = snd_soc_dapm_new_controls(&card->dapm, widgets, num_widgets); + if (err) { + dev_err(card->dev, "Fail to new widgets\n"); + return err; + } + + err = snd_soc_add_card_controls(card, controls, num_controls); + if (err) { + dev_err(card->dev, "Fail to add controls\n"); + return err; + } + + err = snd_soc_dapm_add_routes(&card->dapm, routes, num_routes); + if (err) { + dev_err(card->dev, "Fail to add routes\n"); + return err; + } + if (soc_intel_is_glk()) snd_soc_dapm_add_routes(&card->dapm, gemini_map, ARRAY_SIZE(gemini_map)); @@ -678,6 +766,11 @@ static int broxton_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + if (acpi_dev_present("MX98390", NULL, -1)) + ctx->spkamp = SPKAMP_MAX98390; + else + ctx->spkamp = SPKAMP_MAX98357A; + broxton_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&broxton_audio_card, ctx); if (soc_intel_is_glk()) { @@ -702,7 +795,13 @@ static int broxton_audio_probe(struct platform_device *pdev) } else if (soc_intel_is_cml()) { unsigned int i; - broxton_audio_card.name = "cmlda7219max"; + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_audio_card.name = "cml_max98390_da7219"; + + broxton_audio_card.codec_conf = max98390_codec_confs; + broxton_audio_card.num_configs = ARRAY_SIZE(max98390_codec_confs); + } else + broxton_audio_card.name = "cmlda7219max"; for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { /* MAXIM_CODEC is connected to SSP1. */ @@ -710,6 +809,11 @@ static int broxton_audio_probe(struct platform_device *pdev) BXT_MAXIM_CODEC_DAI)) { broxton_dais[i].name = "SSP1-Codec"; broxton_dais[i].cpus->dai_name = "SSP1 Pin"; + + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_dais[i].codecs = max98390_codec; + broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec); + } } /* DIALOG_CODEC is connected to SSP0 */ else if (!strcmp(broxton_dais[i].codecs->dai_name, @@ -759,6 +863,7 @@ MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bxt_da7219_max98357a"); MODULE_ALIAS("platform:glk_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index cdea0c0..dee1f0f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs max98357a_spk_codecs = { .codecs = {"MX98357A"} }; +static struct snd_soc_acpi_codecs max98390_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + /* * The order of the three entries with .id = "10EC5682" matters * here, because DSDT tables expose an ACPI HID for the MAX98357A @@ -55,6 +60,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", }, + { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &max98390_spk_codecs, + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -- 2.7.4

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