February 2019 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] RME Fireface alsa driver
by Takashi Sakamoto 23 Apr '19

23 Apr '19

Hi, (C.C.ed to alsa-devel) At first, I apologize that ALSA fireface driver since Linux kernel v4.12 includes some bugs for its packet handling for PCM capture substream[1]. This bug can brings disorder of jackd process. Furthermore, even if handling rx packets correctly, Fireface 400 cannot generate sounds because audio data frame is not picked up from isochronous packets to DSP[2]. I've already posted some patches to fix these bugs. The patches were already merged to ALSA upstream and they'll be going to mainline in next merge window for v4.21 kernel. They were also posted for stable kernel and later you can receive patched kernel image if you use any stable kernels. If you'll test these patches in current your environment, please get recent code from my remote repository for backport to Linux v4.17 or later: https://github.com/takaswie/snd-firewire-improve Your test will be great help to me because I've already left Fireface 400. (I rented it just for two weeks and I'll leave Fireface 800 soon as well.) On Fri, 14 Dec 2018, thet wrote: > OK I have the FF400 set up now. In my recent work for Fireface 800[3][4], I ensure that Fireface 800 can generate sounds even if the HOST LED turns on. This is the same experience with Fireface 400. Thus you can ignore whether the LED turn off or not. > What action should I take to create an error that is meaningful? > .. > when I start audacious from the terminal using the alsa output plugin > and the FF400 selected I get: Any GUI tools are not necessarily good for debugging. Please use 'aplay' insted, like: $ aplay -v -D hw:Fireface400 -r 44100 -c 18 -f S32_LE /dev/urandom $ aplay -v -D hw:Fireface400 -r 88200 -c 14 -f S32_LE /dev/urandom $ aplay -v -D hw:Fireface400 -r 192000 -c 10 -f S32_LE /dev/urandom In the above cases, I expect your unit can generate noisy sounds with the latest code of driver. Anyway, I note you not to drop list address (alsa-devel(a)alsa-project.org) from CC list. It's valuable to share experiences of errors/failures because potentially they are good information for developers to seek bugs in software and be going to fix them. [1] [alsa-devel] [PATCH 1/3] ALSA: firewire-lib: fix wrong handling payload_length as payload_quadlet http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143147.h… [2] [alsa-devel] [PATCH] ALSA: fireface: fix for state to fetch PCM frames http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143150.h… [3] [alsa-devel] [PATCH 0/8] ALSA: fireface: add support for Fireface 800 with MIDI functionality only http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/142826.h… [4] [alsa-devel] [PATCH 0/5] ALSA: fireface: add support for Fireface 800 with streaming functionality http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143154.h… Regards Takashi Sakamoto

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[alsa-devel] [PATCH 1/3] sound: Add hikey960 i2s audio driver
by Pengcheng Li 10 Apr '19

10 Apr '19

From: Youlin Wang <wangyoulin1(a)hisilicon.com> Add i2s driver for hisi3660 soc found on the hikey960 board. Add conpile line in make file. Technical support by Guangke Ji. Signed-off-by: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Tanglei Han <hantanglei(a)huawei.com> Signed-off-by: Guangke Ji <jiguangke(a)huawei.com> Signed-off-by: Feng Chen <puck.chen(a)hisilicon.com> Signed-off-by: Kaihua Zhong <zhongkaihua(a)huawei.com> Signed-off-by: Jun Chen <chenjun14(a)huawei.com> Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Yiping Xu <xuyiping(a)hisilicon.com> Signed-off-by: Pengcheng Li <lipengcheng8(a)huawei.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> --- sound/soc/hisilicon/Kconfig | 8 +- sound/soc/hisilicon/Makefile | 1 + sound/soc/hisilicon/hi3660-i2s.c | 448 +++++++++++++++++++++++++++++++++++++++ sound/soc/hisilicon/hi3660-i2s.h | 99 +++++++++ 4 files changed, 555 insertions(+), 1 deletion(-) create mode 100644 sound/soc/hisilicon/hi3660-i2s.c create mode 100644 sound/soc/hisilicon/hi3660-i2s.h diff --git a/sound/soc/hisilicon/Kconfig b/sound/soc/hisilicon/Kconfig index 4356d5a..b023ef9 100644 --- a/sound/soc/hisilicon/Kconfig +++ b/sound/soc/hisilicon/Kconfig @@ -1,5 +1,11 @@ config SND_I2S_HI6210_I2S - tristate "Hisilicon I2S controller" + tristate "Hisilicon Hi6210 I2S controller" + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Hisilicon I2S + +config SND_I2S_HI3660_I2S + tristate "Hisilicon 960 I2S controller" select SND_SOC_GENERIC_DMAENGINE_PCM help Hisilicon I2S diff --git a/sound/soc/hisilicon/Makefile b/sound/soc/hisilicon/Makefile index e8095e2..6800516 100644 --- a/sound/soc/hisilicon/Makefile +++ b/sound/soc/hisilicon/Makefile @@ -1 +1,2 @@ obj-$(CONFIG_SND_I2S_HI6210_I2S) += hi6210-i2s.o +obj-$(CONFIG_SND_I2S_HI3660_I2S) += hi3660-i2s.o diff --git a/sound/soc/hisilicon/hi3660-i2s.c b/sound/soc/hisilicon/hi3660-i2s.c new file mode 100644 index 0000000..b894007 --- /dev/null +++ b/sound/soc/hisilicon/hi3660-i2s.c @@ -0,0 +1,448 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * linux/sound/soc/hisilicon/hi3660-i2s.c + * + * I2S IP driver for hi3660. + * + * Copyright (c) 2001-2021, Huawei Tech. Co., Ltd. + */ +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/jiffies.h> +#include <linux/io.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <linux/interrupt.h> +#include <linux/reset.h> +#include <linux/of_address.h> +#include <linux/of_irq.h> +#include <linux/reset-controller.h> +#include <linux/clk.h> +#include <linux/regulator/consumer.h> + +#include "hi3660-i2s.h" + +struct hi3660_i2s { + struct device *dev; + struct reset_control *rc; + int clocks; + struct regulator *regu_asp; + struct pinctrl *pctrl; + struct pinctrl_state *pin_default; + struct pinctrl_state *pin_idle; + struct clk *asp_subsys_clk; + struct snd_soc_dai_driver dai; + void __iomem *base; + void __iomem *base_syscon; + phys_addr_t base_phys; + struct snd_dmaengine_dai_dma_data dma_data[2]; + spinlock_t lock; + int rate; + int format; + int bits; + int channels; + u32 master; + u32 status; +}; + +static void update_bits(struct hi3660_i2s *i2s, u32 ofs, u32 reset, u32 set) +{ + u32 val = readl(i2s->base + ofs) & ~reset; + + writel(val | set, i2s->base + ofs); +} + +static void update_bits_syscon(struct hi3660_i2s *i2s, + u32 ofs, u32 reset, u32 set) +{ + u32 val = readl(i2s->base_syscon + ofs) & ~reset; + + writel(val | set, i2s->base_syscon + ofs); +} + +static int enable_format(struct hi3660_i2s *i2s, + struct snd_pcm_substream *substream) +{ + switch (i2s->format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = false; + update_bits_syscon(i2s, HI_ASP_CFG_R_CLK_SEL_REG, + 0, HI_ASP_CFG_R_CLK_SEL_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = true; + update_bits_syscon(i2s, HI_ASP_CFG_R_CLK_SEL_REG, + HI_ASP_CFG_R_CLK_SEL_EN, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct hi3660_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + /* deassert reset on sio_bt*/ + update_bits_syscon(i2s, HI_ASP_CFG_R_RST_CTRLDIS_REG, + 0, BIT(2)|BIT(6)|BIT(8)|BIT(16)); + + /* enable clk before frequency division */ + update_bits_syscon(i2s, HI_ASP_CFG_R_GATE_EN_REG, + 0, BIT(5)|BIT(6)); + + /* enable frequency division */ + update_bits_syscon(i2s, HI_ASP_CFG_R_GATE_CLKDIV_EN_REG, + 0, BIT(2)|BIT(5)); + + /* select clk */ + update_bits_syscon(i2s, HI_ASP_CFG_R_CLK_SEL_REG, + HI_ASP_MASK, HI_ASP_CFG_R_CLK_SEL); + + /* select clk_div */ + update_bits_syscon(i2s, HI_ASP_CFG_R_CLK1_DIV_REG, + HI_ASP_MASK, HI_ASP_CFG_R_CLK1_DIV_SEL); + update_bits_syscon(i2s, HI_ASP_CFG_R_CLK4_DIV_REG, + HI_ASP_MASK, HI_ASP_CFG_R_CLK4_DIV_SEL); + update_bits_syscon(i2s, HI_ASP_CFG_R_CLK6_DIV_REG, + HI_ASP_MASK, HI_ASP_CFG_R_CLK6_DIV_SEL); + + /* sio config */ + update_bits(i2s, HI_ASP_SIO_MODE_REG, HI_ASP_MASK, 0x0); + update_bits(i2s, HI_ASP_SIO_DATA_WIDTH_SET_REG, HI_ASP_MASK, 0x9); + update_bits(i2s, HI_ASP_SIO_I2S_POS_MERGE_EN_REG, HI_ASP_MASK, 0x1); + update_bits(i2s, HI_ASP_SIO_I2S_START_POS_REG, HI_ASP_MASK, 0x0); + + return 0; +} + +static void shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct hi3660_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + if (!IS_ERR_OR_NULL(i2s->asp_subsys_clk)) + clk_disable_unprepare(i2s->asp_subsys_clk); +} + +static void txctrl(struct snd_soc_dai *cpu_dai, int on) +{ + struct hi3660_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + spin_lock(&i2s->lock); + + if (on) { + /* enable SIO TX */ + update_bits(i2s, HI_ASP_SIO_CT_SET_REG, 0, + HI_ASP_SIO_TX_ENABLE | + HI_ASP_SIO_TX_DATA_MERGE | + HI_ASP_SIO_TX_FIFO_THRESHOLD | + HI_ASP_SIO_RX_ENABLE | + HI_ASP_SIO_RX_DATA_MERGE | + HI_ASP_SIO_RX_FIFO_THRESHOLD); + } else { + /* disable SIO TX */ + update_bits(i2s, HI_ASP_SIO_CT_CLR_REG, 0, + HI_ASP_SIO_TX_ENABLE | HI_ASP_SIO_RX_ENABLE); + } + spin_unlock(&i2s->lock); +} + +static void rxctrl(struct snd_soc_dai *cpu_dai, int on) +{ + struct hi3660_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + spin_lock(&i2s->lock); + if (on) + /* enable SIO RX */ + update_bits(i2s, HI_ASP_SIO_CT_SET_REG, 0, + HI_ASP_SIO_TX_ENABLE | + HI_ASP_SIO_TX_DATA_MERGE | + HI_ASP_SIO_TX_FIFO_THRESHOLD | + HI_ASP_SIO_RX_ENABLE | + HI_ASP_SIO_RX_DATA_MERGE | + HI_ASP_SIO_RX_FIFO_THRESHOLD); + else + /* disable SIO RX */ + update_bits(i2s, HI_ASP_SIO_CT_CLR_REG, 0, + HI_ASP_SIO_TX_ENABLE | HI_ASP_SIO_RX_ENABLE); + spin_unlock(&i2s->lock); +} + +static int set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + return 0; +} + +static int set_format(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct hi3660_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + i2s->format = fmt; + i2s->master = (i2s->format & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBS_CFS; + + return 0; +} + +static int hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct hi3660_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + + enable_format(i2s, substream); + + dma_data->maxburst = 4; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data->addr = i2s->base_phys + + HI_ASP_SIO_I2S_DUAL_TX_CHN_REG; + else + dma_data->addr = i2s->base_phys + + HI_ASP_SIO_I2S_DUAL_RX_CHN_REG; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + i2s->bits = 16; + dma_data->addr_width = 4; + break; + + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: + i2s->bits = 32; + dma_data->addr_width = 4; + break; + default: + dev_err(cpu_dai->dev, "Bad format\n"); + return -EINVAL; + } + + return 0; +} + +static int trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rxctrl(cpu_dai, 1); + else + txctrl(cpu_dai, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rxctrl(cpu_dai, 0); + else + txctrl(cpu_dai, 0); + break; + default: + dev_err(cpu_dai->dev, "unknown cmd\n"); + return -EINVAL; + } + + return 0; +} + +static int dai_probe(struct snd_soc_dai *dai) +{ + struct hi3660_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, + &i2s->dma_data[SNDRV_PCM_STREAM_PLAYBACK], + &i2s->dma_data[SNDRV_PCM_STREAM_CAPTURE]); + + return 0; +} + + +static struct snd_soc_dai_ops dai_ops = { + .trigger = trigger, + .hw_params = hw_params, + .set_fmt = set_format, + .set_sysclk = set_sysclk, + .startup = startup, + .shutdown = shutdown, +}; + +static struct snd_soc_dai_driver dai_init = { + .name = "hi3660_i2s", + .probe = dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE, + .rates = SNDRV_PCM_RATE_48000, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE, + .rates = SNDRV_PCM_RATE_48000, + }, + .ops = &dai_ops, +}; + +static const struct snd_soc_component_driver component_driver = { + .name = "hi3660_i2s", +}; + +#include <sound/dmaengine_pcm.h> + +static const struct snd_pcm_hardware sound_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX, + .period_bytes_min = 4096, + .period_bytes_max = 4096, + .periods_min = 4, + .periods_max = UINT_MAX, + .buffer_bytes_max = SIZE_MAX, +}; + +static const struct snd_dmaengine_pcm_config dmaengine_pcm_config = { + .pcm_hardware = &sound_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .prealloc_buffer_size = 64 * 1024, +}; + +static int hi3660_i2s_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct hi3660_i2s *i2s; + struct resource *res; + int ret; + + i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + i2s->dev = dev; + spin_lock_init(&i2s->lock); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + return ret; + } + i2s->base_phys = (phys_addr_t)res->start; + + i2s->dai = dai_init; + dev_set_drvdata(&pdev->dev, i2s); + + i2s->base = devm_ioremap_resource(dev, res); + if (IS_ERR(i2s->base)) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = PTR_ERR(i2s->base); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res) { + ret = -ENODEV; + return ret; + } + i2s->base_syscon = devm_ioremap(dev, res->start, resource_size(res)); + if (IS_ERR(i2s->base_syscon)) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = PTR_ERR(i2s->base_syscon); + return ret; + } + + /* i2s iomux config */ + i2s->pctrl = devm_pinctrl_get(dev); + if (IS_ERR(i2s->pctrl)) { + dev_err(dev, "could not get pinctrl\n"); + ret = -EIO; + return ret; + } + + i2s->pin_default = pinctrl_lookup_state(i2s->pctrl, + PINCTRL_STATE_DEFAULT); + if (IS_ERR(i2s->pin_default)) { + dev_err(dev, + "could not get default state (%li)\n", + PTR_ERR(i2s->pin_default)); + ret = -EIO; + return ret; + } + + if (pinctrl_select_state(i2s->pctrl, i2s->pin_default)) { + dev_err(dev, "could not set pins to default state\n"); + ret = -EIO; + return ret; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, + &dmaengine_pcm_config, 0); + if (ret) + return ret; + + ret = snd_soc_register_component(&pdev->dev, &component_driver, + &i2s->dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register dai\n"); + return ret; + } + + return 0; +} + +static int hi3660_i2s_remove(struct platform_device *pdev) +{ + struct hi3660_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + pinctrl_put(i2s->pctrl); + + return 0; +} + +static const struct of_device_id dt_ids[] = { + { .compatible = "hisilicon,hi3660-i2s-1.0" }, + { /* sentinel */ } +}; + +MODULE_DEVICE_TABLE(of, dt_ids); + +static struct platform_driver local_platform_driver = { + .probe = hi3660_i2s_probe, + .remove = hi3660_i2s_remove, + .driver = { + .name = "hi3660_i2s", + .owner = THIS_MODULE, + .of_match_table = dt_ids, + }, +}; + +module_platform_driver(local_platform_driver); + +MODULE_DESCRIPTION("Hisilicon I2S driver"); +MODULE_AUTHOR("Guangke Ji <j00209069(a)notesmail.huawei.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/hisilicon/hi3660-i2s.h b/sound/soc/hisilicon/hi3660-i2s.h new file mode 100644 index 0000000..1874855 --- /dev/null +++ b/sound/soc/hisilicon/hi3660-i2s.h @@ -0,0 +1,99 @@ +/* SPDX-License-Identifier: GPL-2.0+ + * + * linux/sound/soc/hisilicon/hi3660-i2s.c + * + * I2S IP driver for hi3660. + * + * Copyright (c) 2001-2021, Huawei Tech. Co., Ltd. + * + */ + +#ifndef _HI3660_I2S_H +#define _HI3660_I2S_H + +enum hisi_bits { + HII2S_BITS_16, + HII2S_BITS_18, + HII2S_BITS_20, + HII2S_BITS_24, +}; + +enum hisi_i2s_rates { + HII2S_FS_RATE_8KHZ = 0, + HII2S_FS_RATE_16KHZ = 1, + HII2S_FS_RATE_32KHZ = 2, + HII2S_FS_RATE_48KHZ = 4, + HII2S_FS_RATE_96KHZ = 5, + HII2S_FS_RATE_192KHZ = 6, +}; + +#define HI_ASP_CFG_R_RST_CTRLEN_REG 0x0 +#define HI_ASP_CFG_R_RST_CTRLDIS_REG 0x4 +#define HI_ASP_CFG_R_GATE_EN_REG 0xC +#define HI_ASP_CFG_R_GATE_DIS_REG 0x10 +#define HI_ASP_CFG_R_GATE_CLKEN_REG 0x14 +#define HI_ASP_CFG_R_GATE_CLKSTAT_REG 0x18 +#define HI_ASP_CFG_R_GATE_CLKDIV_EN_REG 0x1C +#define HI_ASP_CFG_R_CLK1_DIV_REG 0x20 +#define HI_ASP_CFG_R_CLK2_DIV_REG 0x24 +#define HI_ASP_CFG_R_CLK3_DIV_REG 0x28 +#define HI_ASP_CFG_R_CLK4_DIV_REG 0x2C +#define HI_ASP_CFG_R_CLK5_DIV_REG 0x30 +#define HI_ASP_CFG_R_CLK6_DIV_REG 0x34 +#define HI_ASP_CFG_R_CLK_SEL_REG 0x38 +#define HI_ASP_CFG_R_SEC_REG 0x100 + + +#define HI_ASP_SIO_VERSION_REG (0x3C) +#define HI_ASP_SIO_MODE_REG (0x40) +#define HI_ASP_SIO_INTSTATUS_REG (0x44) +#define HI_ASP_SIO_INTCLR_REG (0x48) +#define HI_ASP_SIO_I2S_LEFT_XD_REG (0x4C) +#define HI_ASP_SIO_I2S_RIGHT_XD_REG (0x50) +#define HI_ASP_SIO_I2S_LEFT_RD_REG (0x54) +#define HI_ASP_SIO_I2S_RIGHT_RD_REG (0x58) +#define HI_ASP_SIO_CT_SET_REG (0x5C) +#define HI_ASP_SIO_CT_CLR_REG (0x60) +#define HI_ASP_SIO_RX_STA_REG (0x68) +#define HI_ASP_SIO_TX_STA_REG (0x6C) +#define HI_ASP_SIO_DATA_WIDTH_SET_REG (0x78) +#define HI_ASP_SIO_I2S_START_POS_REG (0x7C) +#define HI_ASP_SIO_I2S_POS_FLAG_REG (0x80) +#define HI_ASP_SIO_SIGNED_EXT_REG (0x84) +#define HI_ASP_SIO_I2S_POS_MERGE_EN_REG (0x88) +#define HI_ASP_SIO_INTMASK_REG (0x8C) +#define HI_ASP_SIO_I2S_DUAL_RX_CHN_REG (0xA0) +#define HI_ASP_SIO_I2S_DUAL_TX_CHN_REG (0xC0) + + +#define HI_ASP_CFG_R_CLK_SEL_EN BIT(2) +#define HI_ASP_CFG_R_CLK_SEL 0x140010 +#define HI_ASP_CFG_R_CLK1_DIV_SEL 0xbcdc9a +#define HI_ASP_CFG_R_CLK4_DIV_SEL 0x00ff000f +#define HI_ASP_CFG_R_CLK6_DIV_SEL 0x00ff003f +#define HI_ASP_CFG_SIO_MODE 0 +#define HI_ASP_SIO_MODE_SEL_EN BIT(0) +#define HI_ASP_MASK 0xffffffff + +#define HI_ASP_SIO_RX_ENABLE BIT(13) +#define HI_ASP_SIO_TX_ENABLE BIT(12) +#define HI_ASP_SIO_RX_FIFO_DISABLE BIT(11) +#define HI_ASP_SIO_TX_FIFO_DISABLE BIT(10) +#define HI_ASP_SIO_RX_DATA_MERGE BIT(9) +#define HI_ASP_SIO_TX_DATA_MERGE BIT(8) +#define HI_ASP_SIO_RX_FIFO_THRESHOLD (0x5 << 4) +#define HI_ASP_SIO_TX_FIFO_THRESHOLD (0xB << 0) +#define HI_ASP_SIO_RX_FIFO_THRESHOLD_CLR (0xF << 4) +#define HI_ASP_SIO_TX_FIFO_THRESHOLD_CLR (0xF << 0) +#define HI_ASP_SIO_BURST (0x4) + + +enum hisi_i2s_formats { + HII2S_FORMAT_I2S, + HII2S_FORMAT_PCM_STD, + HII2S_FORMAT_PCM_USER, + HII2S_FORMAT_LEFT_JUST, + HII2S_FORMAT_RIGHT_JUST, +}; + +#endif/* _HI3660_I2S_H */ -- 2.8.0

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[alsa-devel] [PATCH 2/3] ASoC: add hikey960-i2s DT bindings
by Pengcheng Li 10 Apr '19

10 Apr '19

From: Youlin Wang <wangyoulin1(a)hisilicon.com> Adds DT bindings documentation for the hikey960-i2s driver and audio card device. Signed-off-by: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Tanglei Han <hantanglei(a)huawei.com> Signed-off-by: Guangke Ji <jiguangke(a)huawei.com> Signed-off-by: Feng Chen <puck.chen(a)hisilicon.com> Signed-off-by: Kaihua Zhong <zhongkaihua(a)huawei.com> Signed-off-by: Jun Chen <chenjun14(a)huawei.com> Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Yiping Xu <xuyiping(a)hisilicon.com> Signed-off-by: Pengcheng Li <lipengcheng8(a)huawei.com> --- .../bindings/sound/hisilicon,hi3660-i2s.txt | 49 ++++++++++++++++++++++ 1 file changed, 49 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/hisilicon,hi3660-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/hisilicon,hi3660-i2s.txt b/Documentation/devicetree/bindings/sound/hisilicon,hi3660-i2s.txt new file mode 100644 index 0000000..654c234 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/hisilicon,hi3660-i2s.txt @@ -0,0 +1,49 @@ +* Hisilicon 6210 i2s controller + +Required properties: + +- compatible: should be one of the following: + - "hisilicon,hi3660-i2s-1.0" +- reg: physical base address of the i2s controller unit and length of + memory mapped region. +- interrupts: should contain the i2s interrupt. +- clocks: a list of phandle + clock-specifier pairs, one for each entry + in clock-names. +- clock-names: should contain following: + - "dacodec" + - "i2s-base" +- dmas: DMA specifiers for tx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should be "tx" and "rx" +- hisilicon,sysctrl-syscon: phandle to sysctrl syscon +- #sound-dai-cells: Should be set to 1 (for multi-dai) + +Example for the hi3660 i2s controller: + + i2s2: hisi_i2s { + compatible = "hisilicon,hi3660-i2s-1.0"; + reg = <0x0 0xe804f800 0x0 0x400>, + <0x0 0xe804e000 0x0 0x400>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pmx_func &i2s2_cfg_func>; + dmas = <&asp_dmac 18 &asp_dmac 19>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "hikey-hdmi"; + simple-audio-card,format = "i2s"; + + simple-audio-card,bitclock-master = <&sound_master>; + simple-audio-card,frame-master = <&sound_master>; + + sound_master: simple-audio-card,cpu { + sound-dai = <&i2s2>; + }; + + simple-audio-card,codec { + sound-dai = <&adv7533>; + }; + }; -- 2.8.0

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[alsa-devel] [PATCH 1/3] ASoC: wcd9335: fix a leaked reference by adding missing of_node_put
by Wen Yang 05 Apr '19

05 Apr '19

The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> (commit_signer:1/11=9%,authored:1/11=9%) Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/codecs/wcd9335.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 981f88a..a04a7ce 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -5188,6 +5188,7 @@ static int wcd9335_slim_status(struct slim_device *sdev, wcd->slim = sdev; wcd->slim_ifc_dev = of_slim_get_device(sdev->ctrl, ifc_dev_np); + of_node_put(ifc_dev_np); if (!wcd->slim_ifc_dev) { dev_err(dev, "Unable to get SLIM Interface device\n"); return -EINVAL; -- 2.9.5

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[alsa-devel] [RFC 0/2] Add MICFIL DAI support
by Cosmin Samoila 27 Mar '19

27 Mar '19

PDM is a popular way to deliver audio from microphones to the processor in several applications, such as mobile telephones. However, current digital-audio systems use multibit audio signal (also known as multibit PCM) to represent the signal. For this purpose a set of FIR, CIC or/and Half Band filters are usually implemented on DSPs or software. This block implements the required digital interface to provide a 16-bit audio signal from a PDM microphone bitstream in a configurable output sampling rate. The implementation of this digital interface is based on the application of digital signal processing techniques in hardware. The PDM Microphone Interface architecture was designed to gate saving and minimal power consumption. It implements a bunch of filters to transform a 1-bit PDM bitstream to a 16-bit PCM signal in the audio band. To avoid aliasing frequencies in passband, the overall filter has 80 dB stopband attenuation and passband ripple less than 0.2dB. The whole module is implemented to work in a multichannel mode. All channels have the same configuration but each input channel could be turned on/off independently. Cosmin-Gabriel Samoila (2): ASoC: micfil: Add bindings for MICFIL DAI ASoC: Add MICFIL SoC Digital Audio Interface driver. .../devicetree/bindings/sound/fsl,micfil.txt | 38 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_micfil.c | 2451 ++++++++++++++++++++ sound/soc/fsl/fsl_micfil.h | 317 +++ 5 files changed, 2817 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 sound/soc/fsl/fsl_micfil.c create mode 100644 sound/soc/fsl/fsl_micfil.h -- 2.7.4

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[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH v4 0/3] Add NXP AUDMIX device and machine drivers
by Viorel Suman 26 Mar '19

26 Mar '19

The patchset adds NXP Audio Mixer (AUDMIX) device and machine drivers and related DT bindings documentation. Changes since V3: 1. Removed machine driver DT bindings documentation. 2. Trigger machine driver probe from device driver as suggested by Nicolin. Changes since V2: 1. Moved "dais" node from machine driver DTS node to device driver DTS node as suggested by Rob. Changes since V1: 1. Original patch split into distinct patches for the device driver and DT binding documentation. 2. Replaced AMIX with AUDMIX in both code and file names as it looks more RM-compliant. 3. Removed polarity control from CPU DAI driver as suggested by Nicolin. 4. Added machine driver and related DT binding documentation. Viorel Suman (3): ASoC: fsl: Add Audio Mixer CPU DAI driver ASoC: add fsl_audmix DT binding documentation ASoC: fsl: Add Audio Mixer machine driver .../devicetree/bindings/sound/fsl,audmix.txt | 54 ++ sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++ sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ 6 files changed, 1080 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h create mode 100644 sound/soc/fsl/imx-audmix.c -- 2.7.4

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[alsa-devel] [PATCH v4] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 26 Mar '19

26 Mar '19

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. v3:- o Remove driver supplied time validation. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 17 ++++++++++++----- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..eecbfdc7a2c9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { -- 2.17.1

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[alsa-devel] [PATCH V4] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 21 Mar '19

21 Mar '19

There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the beginning with the normal enable flow. This is hardware issue, but there is no errata, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting. Fixes commit 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 47 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..3623aa9a6f2e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -54,6 +54,8 @@ struct fsl_esai { u32 fifo_depth; u32 slot_width; u32 slots; + u32 tx_mask; + u32 rx_mask; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -361,21 +363,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); - esai_priv->slot_width = slot_width; esai_priv->slots = slots; + esai_priv->tx_mask = tx_mask; + esai_priv->rx_mask = rx_mask; return 0; } @@ -596,6 +590,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 mask; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -608,15 +603,38 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, for (i = 0; tx && i < channels; i++) regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + /* + * When set the TE/RE in the end of enablement flow, there + * will be channel swap issue for multi data line case. + * In order to workaround this issue, we switch the bit + * enablement sequence to below sequence + * 1) clear the xSMB & xSMA: which is done in probe and + * stop state. + * 2) set TE/RE + * 3) set xSMB + * 4) set xSMA: xSMA is the last one in this flow, which + * will trigger esai to start. + */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); + mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, 0); /* Disable and reset FIFO */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), @@ -906,6 +924,15 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + esai_priv->tx_mask = 0xFFFFFFFF; + esai_priv->rx_mask = 0xFFFFFFFF; + + /* Clear the TSMA, TSMB, RSMA, RSMB */ + regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { -- 1.9.1

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[alsa-devel] [PATCH 0/2] arm64: allwinner: a64: Rename hpvcc-supply to cpvdd-supply
by Chen-Yu Tsai 21 Mar '19

21 Mar '19

Hi everyone, This is a small series to fix the CPVDD regulator supply name for the A64 codec. The datasheet specifies the name "CPVDD" for the supply rail for the audio codec's internal headphone amp charge pump. Somehow I managed to write "CPVDD" in the commit message and the device tree binding, but used "HPVCC" (which is the name used by the A33) in the device tree files and the device driver. (facepalm) This feature was just introduced in v5.0-rc1, so if we could merge this as fixes for v5.0 that would be great. Thanks, and sorry for the blunder on my part. Regards ChenYu Chen-Yu Tsai (2): ASoC: sunxi: sun50i-codec-analog: Rename hpvcc regulator supply to cpvdd arm64: dts: allwinner: a64: Rename hpvcc-supply to cpvdd-supply arch/arm64/boot/dts/allwinner/sun50i-a64-bananapi-m64.dts | 2 +- arch/arm64/boot/dts/allwinner/sun50i-a64-pine64.dts | 2 +- arch/arm64/boot/dts/allwinner/sun50i-a64-pinebook.dts | 2 +- arch/arm64/boot/dts/allwinner/sun50i-a64-sopine.dtsi | 2 +- sound/soc/sunxi/sun50i-codec-analog.c | 4 ++-- 5 files changed, 6 insertions(+), 6 deletions(-) -- 2.20.1

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