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[alsa-devel] [PATCH 1/6] ASoC: amd: Refactoring of DAI from DMA driver
by Ravulapati Vishnu vardhan rao 28 Dec '19

28 Dec '19

ASoC: PCM DMA driver should only have dma ops. So Removed all DAI related functionality.Refactoring the PCM DMA diver code.Added new file containing only DAI ops Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/Makefile | 2 + sound/soc/amd/raven/acp3x-i2s.c | 262 ++++++++++++++++++++++++++++++++ sound/soc/amd/raven/acp3x-pcm-dma.c | 291 ++++++------------------------------ sound/soc/amd/raven/acp3x.h | 43 ++++++ 4 files changed, 356 insertions(+), 242 deletions(-) create mode 100644 sound/soc/amd/raven/acp3x-i2s.c diff --git a/sound/soc/amd/raven/Makefile b/sound/soc/amd/raven/Makefile index 108d1ac..62c22b6 100644 --- a/sound/soc/amd/raven/Makefile +++ b/sound/soc/amd/raven/Makefile @@ -2,5 +2,7 @@ # Raven Ridge platform Support snd-pci-acp3x-objs := pci-acp3x.o snd-acp3x-pcm-dma-objs := acp3x-pcm-dma.o +snd-acp3x-i2s-objs := acp3x-i2s.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-pci-acp3x.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-pcm-dma.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-i2s.o diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c new file mode 100644 index 0000000..cdc1c61 --- /dev/null +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -0,0 +1,262 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// AMD ALSA SoC PCM Driver +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <linux/dma-mapping.h> + +#include "acp3x.h" + +#define DRV_NAME "acp3x-i2s" + +static int acp3x_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct i2s_dev_data *adata; + int mode; + + adata = snd_soc_dai_get_drvdata(cpu_dai); + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (mode) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = false; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = true; + break; + default: + return -EINVAL; + } + return 0; +} + +static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, + u32 tx_mask, u32 rx_mask, int slots, int slot_width) +{ + struct i2s_dev_data *adata; + u32 val, reg_val, frmt_val, frm_len; + u16 slot_len; + + adata = snd_soc_dai_get_drvdata(cpu_dai); + + /* These values are as per Hardware Spec */ + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + return -EINVAL; + } + + /* Enable I2S/BT channels TDM, respective TX/RX frame lengths.*/ + + frm_len = FRM_LEN | (slots << 15) | (slot_len << 18); + if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) { + reg_val = mmACP_BTTDM_ITER; + frmt_val = mmACP_BTTDM_TXFRMT; + } else { + reg_val = mmACP_BTTDM_IRER; + frmt_val = mmACP_BTTDM_RXFRMT; + } + val = rv_readl(adata->acp3x_base + reg_val); + rv_writel(val | 0x2, adata->acp3x_base + reg_val); + rv_writel(frm_len, adata->acp3x_base + frmt_val); + adata->tdm_fmt = frm_len; + return 0; +} + +static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct i2s_stream_instance *rtd; + u32 val; + u32 reg_val; + + rtd = substream->runtime->private_data; + + /* These values are as per Hardware Spec */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + rtd->xfer_resolution = 0x0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + default: + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg_val = mmACP_BTTDM_ITER; + else + reg_val = mmACP_BTTDM_IRER; + + val = rv_readl(rtd->acp3x_base + reg_val); + val = val | (rtd->xfer_resolution << 3); + rv_writel(val, rtd->acp3x_base + reg_val); + return 0; +} + +static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct i2s_stream_instance *rtd; + u32 val, period_bytes; + int ret, reg_val; + + rtd = substream->runtime->private_data; + period_bytes = frames_to_bytes(substream->runtime, + substream->runtime->period_size); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rtd->bytescount = acp_get_byte_count(rtd, + substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_val = mmACP_BTTDM_ITER; + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_TX_INTR_WATERMARK_SIZE); + } else { + reg_val = mmACP_BTTDM_IRER; + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_RX_INTR_WATERMARK_SIZE); + } + val = rv_readl(rtd->acp3x_base + reg_val); + val = val | BIT(0); + rv_writel(val, rtd->acp3x_base + reg_val); + rv_writel(1, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg_val = mmACP_BTTDM_ITER; + else + reg_val = mmACP_BTTDM_IRER; + + val = rv_readl(rtd->acp3x_base + reg_val); + val = val & ~BIT(0); + rv_writel(val, rtd->acp3x_base + reg_val); + rv_writel(0, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static struct snd_soc_dai_ops acp3x_i2s_dai_ops = { + .hw_params = acp3x_i2s_hwparams, + .trigger = acp3x_i2s_trigger, + .set_fmt = acp3x_i2s_set_fmt, + .set_tdm_slot = acp3x_i2s_set_tdm_slot, +}; + +static const struct snd_soc_component_driver acp3x_dai_component = { + .name = "acp3x-i2s", +}; + +static struct snd_soc_dai_driver acp3x_i2s_dai = { + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp3x_i2s_dai_ops, +}; + +static int acp3x_dai_probe(struct platform_device *pdev) +{ + struct resource *res; + struct i2s_dev_data *adata; + int ret; + + adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), + GFP_KERNEL); + if (!adata) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); + return -ENOMEM; + } + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (IS_ERR(adata->acp3x_base)) + return PTR_ERR(adata->acp3x_base); + + adata->i2s_irq = res->start; + dev_set_drvdata(&pdev->dev, adata); + ret = devm_snd_soc_register_component(&pdev->dev, + &acp3x_dai_component, &acp3x_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); + return -ENODEV; + } + return 0; +} + +static int acp3x_dai_remove(struct platform_device *pdev) +{ + /* As we use devm_ memory alloc there is nothing TBD here */ + + return 0; +} + +static struct platform_driver acp3x_dai_driver = { + .probe = acp3x_dai_probe, + .remove = acp3x_dai_remove, + .driver = { + .name = "acp3x_i2s_playcap", + }, +}; + +module_platform_driver(acp3x_dai_driver); + +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP 3.x PCM Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 9792104..351143b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -18,24 +18,6 @@ #define DRV_NAME "acp3x-i2s-audio" -struct i2s_dev_data { - bool tdm_mode; - unsigned int i2s_irq; - u32 tdm_fmt; - void __iomem *acp3x_base; - struct snd_pcm_substream *play_stream; - struct snd_pcm_substream *capture_stream; -}; - -struct i2s_stream_instance { - u16 num_pages; - u16 channels; - u32 xfer_resolution; - u64 bytescount; - dma_addr_t dma_addr; - void __iomem *acp3x_base; -}; - static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -178,10 +160,11 @@ static int acp3x_deinit(void __iomem *acp3x_base) static irqreturn_t i2s_irq_handler(int irq, void *dev_id) { + struct i2s_dev_data *rv_i2s_data; u16 play_flag, cap_flag; u32 val; - struct i2s_dev_data *rv_i2s_data = dev_id; + rv_i2s_data = dev_id; if (!rv_i2s_data) return IRQ_NONE; @@ -278,11 +261,17 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) static int acp3x_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - struct snd_pcm_runtime *runtime = substream->runtime; - struct i2s_dev_data *adata = dev_get_drvdata(component->dev); - struct i2s_stream_instance *i2s_data = kzalloc(sizeof(struct i2s_stream_instance), - GFP_KERNEL); + struct snd_pcm_runtime *runtime; + struct snd_soc_pcm_runtime *prtd; + struct i2s_dev_data *adata; + struct i2s_stream_instance *i2s_data; + int ret; + + runtime = substream->runtime; + prtd = substream->private_data; + component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + adata = dev_get_drvdata(component->dev); + i2s_data = kzalloc(sizeof(*i2s_data), GFP_KERNEL); if (!i2s_data) return -EINVAL; @@ -312,23 +301,6 @@ static int acp3x_dma_open(struct snd_soc_component *component, return 0; } -static u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) -{ - u64 byte_count; - - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { - byte_count = rv_readl(rtd->acp3x_base + - mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH); - byte_count |= rv_readl(rtd->acp3x_base + - mmACP_BT_TX_LINEARPOSITIONCNTR_LOW); - } else { - byte_count = rv_readl(rtd->acp3x_base + - mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH); - byte_count |= rv_readl(rtd->acp3x_base + - mmACP_BT_RX_LINEARPOSITIONCNTR_LOW); - } - return byte_count; -} static int acp3x_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, @@ -351,12 +323,12 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - u32 pos = 0; - u32 buffersize = 0; - u64 bytescount = 0; - struct i2s_stream_instance *rtd = - substream->runtime->private_data; + struct i2s_stream_instance *rtd; + u32 pos; + u32 buffersize; + u64 bytescount; + rtd = substream->runtime->private_data; buffersize = frames_to_bytes(substream->runtime, substream->runtime->buffer_size); bytescount = acp_get_byte_count(rtd, substream->stream); @@ -385,8 +357,12 @@ static int acp3x_dma_mmap(struct snd_soc_component *component, static int acp3x_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct i2s_stream_instance *rtd = substream->runtime->private_data; - struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + struct snd_soc_pcm_runtime *prtd; + struct i2s_dev_data *adata; + + prtd = substream->private_data; + component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + adata = dev_get_drvdata(component->dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) adata->play_stream = NULL; @@ -398,186 +374,9 @@ static int acp3x_dma_close(struct snd_soc_component *component, */ if (!adata->play_stream && !adata->capture_stream) rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - kfree(rtd); return 0; } -static int acp3x_dai_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) -{ - - struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - adata->tdm_mode = false; - break; - case SND_SOC_DAIFMT_DSP_A: - adata->tdm_mode = true; - break; - default: - return -EINVAL; - } - - return 0; -} - -static int acp3x_dai_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, - u32 rx_mask, int slots, int slot_width) -{ - u32 val = 0; - u16 slot_len; - - struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); - - switch (slot_width) { - case SLOT_WIDTH_8: - slot_len = 8; - break; - case SLOT_WIDTH_16: - slot_len = 16; - break; - case SLOT_WIDTH_24: - slot_len = 24; - break; - case SLOT_WIDTH_32: - slot_len = 0; - break; - default: - return -EINVAL; - } - - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); - rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_ITER); - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); - rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_IRER); - - val = (FRM_LEN | (slots << 15) | (slot_len << 18)); - rv_writel(val, adata->acp3x_base + mmACP_BTTDM_TXFRMT); - rv_writel(val, adata->acp3x_base + mmACP_BTTDM_RXFRMT); - - adata->tdm_fmt = val; - return 0; -} - -static int acp3x_dai_i2s_hwparams(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - u32 val = 0; - struct i2s_stream_instance *rtd = substream->runtime->private_data; - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: - rtd->xfer_resolution = 0x0; - break; - case SNDRV_PCM_FORMAT_S16_LE: - rtd->xfer_resolution = 0x02; - break; - case SNDRV_PCM_FORMAT_S24_LE: - rtd->xfer_resolution = 0x04; - break; - case SNDRV_PCM_FORMAT_S32_LE: - rtd->xfer_resolution = 0x05; - break; - default: - return -EINVAL; - } - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); - val = val | (rtd->xfer_resolution << 3); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); - else - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); - - return 0; -} - -static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - int ret = 0; - struct i2s_stream_instance *rtd = substream->runtime->private_data; - u32 val, period_bytes; - - period_bytes = frames_to_bytes(substream->runtime, - substream->runtime->period_size); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - rtd->bytescount = acp_get_byte_count(rtd, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - rv_writel(period_bytes, rtd->acp3x_base + - mmACP_BT_TX_INTR_WATERMARK_SIZE); - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); - val = val | BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); - } else { - rv_writel(period_bytes, rtd->acp3x_base + - mmACP_BT_RX_INTR_WATERMARK_SIZE); - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); - val = val | BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); - } - rv_writel(1, rtd->acp3x_base + mmACP_BTTDM_IER); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); - val = val & ~BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); - } else { - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); - val = val & ~BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); - } - rv_writel(0, rtd->acp3x_base + mmACP_BTTDM_IER); - break; - default: - ret = -EINVAL; - break; - } - - return ret; -} - -static struct snd_soc_dai_ops acp3x_dai_i2s_ops = { - .hw_params = acp3x_dai_i2s_hwparams, - .trigger = acp3x_dai_i2s_trigger, - .set_fmt = acp3x_dai_i2s_set_fmt, - .set_tdm_slot = acp3x_dai_set_tdm_slot, -}; - -static struct snd_soc_dai_driver acp3x_i2s_dai_driver = { - .playback = { - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .channels_min = 2, - .channels_max = 8, - - .rate_min = 8000, - .rate_max = 96000, - }, - .capture = { - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .channels_min = 2, - .channels_max = 2, - .rate_min = 8000, - .rate_max = 48000, - }, - .ops = &acp3x_dai_i2s_ops, -}; - static const struct snd_soc_component_driver acp3x_i2s_component = { .name = DRV_NAME, .open = acp3x_dma_open, @@ -590,10 +389,10 @@ static const struct snd_soc_component_driver acp3x_i2s_component = { static int acp3x_audio_probe(struct platform_device *pdev) { - int status; struct resource *res; struct i2s_dev_data *adata; unsigned int irqflags; + int status, ret; if (!pdev->dev.platform_data) { dev_err(&pdev->dev, "platform_data not retrieved\n"); @@ -603,7 +402,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); return -ENODEV; } @@ -613,6 +412,8 @@ static int acp3x_audio_probe(struct platform_device *pdev) adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!adata->acp3x_base) + return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (!res) { @@ -621,52 +422,54 @@ static int acp3x_audio_probe(struct platform_device *pdev) } adata->i2s_irq = res->start; - adata->play_stream = NULL; - adata->capture_stream = NULL; dev_set_drvdata(&pdev->dev, adata); /* Initialize ACP */ status = acp3x_init(adata->acp3x_base); if (status) return -ENODEV; + status = devm_snd_soc_register_component(&pdev->dev, &acp3x_i2s_component, - &acp3x_i2s_dai_driver, 1); + NULL, 0); if (status) { - dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); + dev_err(&pdev->dev, "Fail to register acp i2s component\n"); + ret = -ENODEV; goto dev_err; } status = devm_request_irq(&pdev->dev, adata->i2s_irq, i2s_irq_handler, irqflags, "ACP3x_I2S_IRQ", adata); if (status) { dev_err(&pdev->dev, "ACP3x I2S IRQ request failed\n"); + ret = -ENODEV; goto dev_err; } - pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_set_autosuspend_delay(&pdev->dev, 5000); pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_enable(&pdev->dev); return 0; + dev_err: status = acp3x_deinit(adata->acp3x_base); if (status) dev_err(&pdev->dev, "ACP de-init failed\n"); else - dev_info(&pdev->dev, "ACP de-initialized\n"); - /*ignore device status and return driver probe error*/ - return -ENODEV; + dev_dbg(&pdev->dev, "ACP de-initialized\n"); + return ret; } static int acp3x_audio_remove(struct platform_device *pdev) { + struct i2s_dev_data *adata; int ret; - struct i2s_dev_data *adata = dev_get_drvdata(&pdev->dev); + adata = dev_get_drvdata(&pdev->dev); ret = acp3x_deinit(adata->acp3x_base); if (ret) dev_err(&pdev->dev, "ACP de-init failed\n"); else - dev_info(&pdev->dev, "ACP de-initialized\n"); + dev_dbg(&pdev->dev, "ACP de-initialized\n"); pm_runtime_disable(&pdev->dev); return 0; @@ -674,10 +477,11 @@ static int acp3x_audio_remove(struct platform_device *pdev) static int acp3x_resume(struct device *dev) { + struct i2s_dev_data *adata; int status; u32 val; - struct i2s_dev_data *adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_init(adata->acp3x_base); if (status) return -ENODEV; @@ -719,14 +523,15 @@ static int acp3x_resume(struct device *dev) static int acp3x_pcm_runtime_suspend(struct device *dev) { + struct i2s_dev_data *adata; int status; - struct i2s_dev_data *adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_deinit(adata->acp3x_base); if (status) dev_err(dev, "ACP de-init failed\n"); else - dev_info(dev, "ACP de-initialized\n"); + dev_dbg(dev, "ACP de-initialized\n"); rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); @@ -735,8 +540,9 @@ static int acp3x_pcm_runtime_suspend(struct device *dev) static int acp3x_pcm_runtime_resume(struct device *dev) { + struct i2s_dev_data *adata; int status; - struct i2s_dev_data *adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_init(adata->acp3x_base); if (status) @@ -755,13 +561,14 @@ static struct platform_driver acp3x_dma_driver = { .probe = acp3x_audio_probe, .remove = acp3x_audio_remove, .driver = { - .name = "acp3x_rv_i2s", + .name = "acp3x_rv_i2s_dma", .pm = &acp3x_pm_ops, }, }; module_platform_driver(acp3x_dma_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); MODULE_DESCRIPTION("AMD ACP 3.x PCM Driver"); diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 2f15fe1..a6c6d63 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -51,6 +51,30 @@ #define SLOT_WIDTH_24 0x18 #define SLOT_WIDTH_32 0x20 +struct acp3x_platform_info { + u16 play_i2s_instance; + u16 cap_i2s_instance; + u16 capture_channel; +}; + +struct i2s_dev_data { + bool tdm_mode; + unsigned int i2s_irq; + u32 tdm_fmt; + u32 substream_type; + void __iomem *acp3x_base; + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; +}; + +struct i2s_stream_instance { + u16 num_pages; + u16 channels; + u32 xfer_resolution; + u64 bytescount; + dma_addr_t dma_addr; + void __iomem *acp3x_base; +}; static inline u32 rv_readl(void __iomem *base_addr) { @@ -61,3 +85,22 @@ static inline void rv_writel(u32 val, void __iomem *base_addr) { writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); } + +static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, + int direction) +{ + u64 byte_count; + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_LOW); + } else { + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_LOW); + } + return byte_count; +} -- 2.7.4

3 2

[alsa-devel] [PATCH AUTOSEL 5.4 001/187] ASoC: rt5682: fix i2c arbitration lost issue
by Sasha Levin 28 Dec '19

28 Dec '19

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit bc094709de0192a756c6946a7c89c543243ae609 ] This patch modified the HW initial setting to fix i2c arbitration lost issue. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20191125091940.11953-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index c50b75ce82e0..05e883a65d7a 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -72,6 +72,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, + {RT5682_I2C_CTRL, 0x000f}, }; static const struct reg_default rt5682_reg[] = { @@ -2481,6 +2482,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); -- 2.20.1

2 1

[alsa-devel] [PATCH AUTOSEL 4.4 17/25] ASoC: wm8962: fix lambda value
by Sasha Levin 27 Dec '19

27 Dec '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 556672d75ff486e0b6786056da624131679e0576 ] According to user manual, it is required that FLL_LAMBDA > 0 in all cases (Integer and Franctional modes). Fixes: 9a76f1ff6e29 ("ASoC: Add initial WM8962 CODEC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/1576065442-19763-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a7e79784fc16..4a3ce9b85253 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2792,7 +2792,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2862,7 +2862,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 26/38] ASoC: wm8962: fix lambda value
by Sasha Levin 27 Dec '19

27 Dec '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 556672d75ff486e0b6786056da624131679e0576 ] According to user manual, it is required that FLL_LAMBDA > 0 in all cases (Integer and Franctional modes). Fixes: 9a76f1ff6e29 ("ASoC: Add initial WM8962 CODEC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/1576065442-19763-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index fd2731d171dd..0e8008d38161 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2791,7 +2791,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2861,7 +2861,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 39/57] ASoC: wm8962: fix lambda value
by Sasha Levin 27 Dec '19

27 Dec '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 556672d75ff486e0b6786056da624131679e0576 ] According to user manual, it is required that FLL_LAMBDA > 0 in all cases (Integer and Franctional modes). Fixes: 9a76f1ff6e29 ("ASoC: Add initial WM8962 CODEC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/1576065442-19763-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index fd2731d171dd..0e8008d38161 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2791,7 +2791,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2861,7 +2861,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 19/57] ASoC: topology: Check return value for soc_tplg_pcm_create()
by Sasha Levin 27 Dec '19

27 Dec '19

From: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> [ Upstream commit a3039aef52d9ffeb67e9211899cd3e8a2953a01f ] The return value of soc_tplg_pcm_create() is currently not checked in soc_tplg_pcm_elems_load(). If an error is to occur there, the topology ignores it and continues loading. Fix that by checking the status and rejecting the topology on error. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191210003939.15752-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-topology.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 2d5cf263515b..72301bcad3bd 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1921,6 +1921,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, int count = hdr->count; int i; bool abi_match; + int ret; if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; @@ -1957,7 +1958,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ - soc_tplg_pcm_create(tplg, _pcm); + ret = soc_tplg_pcm_create(tplg, _pcm); + if (ret < 0) { + if (!abi_match) + kfree(_pcm); + return ret; + } /* offset by version-specific struct size and * real priv data size -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 59/84] ASoC: wm8962: fix lambda value
by Sasha Levin 27 Dec '19

27 Dec '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 556672d75ff486e0b6786056da624131679e0576 ] According to user manual, it is required that FLL_LAMBDA > 0 in all cases (Integer and Franctional modes). Fixes: 9a76f1ff6e29 ("ASoC: Add initial WM8962 CODEC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/1576065442-19763-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index efd8910b1ff7..dde015fd70a4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2792,7 +2792,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2862,7 +2862,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 30/84] ASoC: topology: Check return value for soc_tplg_pcm_create()
by Sasha Levin 27 Dec '19

27 Dec '19

From: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> [ Upstream commit a3039aef52d9ffeb67e9211899cd3e8a2953a01f ] The return value of soc_tplg_pcm_create() is currently not checked in soc_tplg_pcm_elems_load(). If an error is to occur there, the topology ignores it and continues loading. Fix that by checking the status and rejecting the topology on error. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191210003939.15752-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-topology.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 88a7e860b175..069f38fbf07b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1890,6 +1890,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, int count = hdr->count; int i; bool abi_match; + int ret; if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; @@ -1926,7 +1927,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ - soc_tplg_pcm_create(tplg, _pcm); + ret = soc_tplg_pcm_create(tplg, _pcm); + if (ret < 0) { + if (!abi_match) + kfree(_pcm); + return ret; + } /* offset by version-specific struct size and * real priv data size -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 11/84] ASoC: Intel: bytcr_rt5640: Update quirk for Teclast X89
by Sasha Levin 27 Dec '19

27 Dec '19

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 7eccc05c7101f34cc36afe9405d15de6d4099fb4 ] When the Teclast X89 quirk was added we did not have jack-detection support yet. Note the over-current detection limit is set to 2mA instead of the usual 1.5mA because this tablet tends to give false-positive button-presses when it is set to 1.5mA. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191203221442.2657-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b6dc524830b2..35983eae584a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -675,13 +675,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Teclast X89 */ .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), DMI_MATCH(DMI_BOARD_NAME, "tPAD"), }, .driver_data = (void *)(BYT_RT5640_IN3_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_1P0 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { /* Toshiba Satellite Click Mini L9W-B */ .matches = { -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 04/84] ASoC: max98090: fix possible race conditions
by Sasha Levin 27 Dec '19

27 Dec '19

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit 45dfbf56975994822cce00b7475732a49f8aefed ] max98090_interrupt() and max98090_pll_work() run in 2 different threads. There are 2 possible races: Note: M98090_REG_DEVICE_STATUS = 0x01. Note: ULK == 0, PLL is locked; ULK == 1, PLL is unlocked. max98090_interrupt max98090_pll_work ---------------------------------------------- schedule max98090_pll_work restart max98090 codec receive ULK INT assert ULK == 0 schedule max98090_pll_work (1). In the case (1), the PLL is locked but max98090_interrupt unnecessarily schedules another max98090_pll_work. max98090_interrupt max98090_pll_work max98090 codec ---------------------------------------------------------------------- ULK = 1 receive ULK INT read 0x01 ULK = 0 (clear on read) schedule max98090_pll_work restart max98090 codec ULK = 1 receive ULK INT read 0x01 ULK = 0 (clear on read) read 0x01 assert ULK == 0 (2). In the case (2), both max98090_interrupt and max98090_pll_work read the same clear-on-read register. max98090_pll_work would falsely thought PLL is locked. Note: the case (2) race is introduced by the previous commit ("ASoC: max98090: exit workaround earlier if PLL is locked") to check the status and exit the loop earlier in max98090_pll_work. There are 2 possible solution options: A. turn off ULK interrupt before scheduling max98090_pll_work; and turn on again before exiting max98090_pll_work. B. remove the second thread of execution. Option A cannot fix the case (2) race because it still has 2 threads access the same clear-on-read register simultaneously. Although we could suppose the register is volatile and read the status via I2C could be much slower than the hardware raises the bits. Option B introduces a maximum 10~12 msec penalty delay in the interrupt handler. However, it could only punish the jack detection by extra 10~12 msec. Adopts option B which is the better solution overall. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20191122073114.219945-4-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 8 ++------ sound/soc/codecs/max98090.h | 1 - 2 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 172a48d42479..f8cbed6cd0bf 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2121,10 +2121,8 @@ static void max98090_pll_det_disable_work(struct work_struct *work) M98090_IULK_MASK, 0); } -static void max98090_pll_work(struct work_struct *work) +static void max98090_pll_work(struct max98090_priv *max98090) { - struct max98090_priv *max98090 = - container_of(work, struct max98090_priv, pll_work); struct snd_soc_component *component = max98090->component; unsigned int pll; int i; @@ -2293,7 +2291,7 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (active & M98090_ULK_MASK) { dev_dbg(component->dev, "M98090_ULK_MASK\n"); - schedule_work(&max98090->pll_work); + max98090_pll_work(max98090); } if (active & M98090_JDET_MASK) { @@ -2456,7 +2454,6 @@ static int max98090_probe(struct snd_soc_component *component) max98090_pll_det_enable_work); INIT_WORK(&max98090->pll_det_disable_work, max98090_pll_det_disable_work); - INIT_WORK(&max98090->pll_work, max98090_pll_work); /* Enable jack detection */ snd_soc_component_write(component, M98090_REG_JACK_DETECT, @@ -2509,7 +2506,6 @@ static void max98090_remove(struct snd_soc_component *component) cancel_delayed_work_sync(&max98090->jack_work); cancel_delayed_work_sync(&max98090->pll_det_enable_work); cancel_work_sync(&max98090->pll_det_disable_work); - cancel_work_sync(&max98090->pll_work); max98090->component = NULL; } diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index b1572a2d19da..388d2f74674b 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1533,7 +1533,6 @@ struct max98090_priv { struct delayed_work jack_work; struct delayed_work pll_det_enable_work; struct work_struct pll_det_disable_work; - struct work_struct pll_work; struct snd_soc_jack *jack; unsigned int dai_fmt; int tdm_slots; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 03/84] ASoC: max98090: exit workaround earlier if PLL is locked
by Sasha Levin 27 Dec '19

27 Dec '19

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit 6f49919d11690a9b5614445ba30fde18083fdd63 ] According to the datasheet, PLL lock time typically takes 2 msec and at most takes 7 msec. Check the lock status every 1 msec and exit the workaround if PLL is locked. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20191122073114.219945-3-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f8c8b9decab6..172a48d42479 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2126,6 +2126,8 @@ static void max98090_pll_work(struct work_struct *work) struct max98090_priv *max98090 = container_of(work, struct max98090_priv, pll_work); struct snd_soc_component *component = max98090->component; + unsigned int pll; + int i; if (!snd_soc_component_is_active(component)) return; @@ -2145,8 +2147,16 @@ static void max98090_pll_work(struct work_struct *work) snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); - /* Give PLL time to lock */ - msleep(10); + for (i = 0; i < 10; ++i) { + /* Give PLL time to lock */ + usleep_range(1000, 1200); + + /* Check lock status */ + pll = snd_soc_component_read32( + component, M98090_REG_DEVICE_STATUS); + if (!(pll & M98090_ULK_MASK)) + break; + } } static void max98090_jack_work(struct work_struct *work) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 02/84] ASoC: max98090: remove msleep in PLL unlocked workaround
by Sasha Levin 27 Dec '19

27 Dec '19

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit acb874a7c049ec49d8fc66c893170fb42c01bdf7 ] It was observed Baytrail-based chromebooks could cause continuous PLL unlocked when using playback stream and capture stream simultaneously. Specifically, starting a capture stream after started a playback stream. As a result, the audio data could corrupt or turn completely silent. As the datasheet suggested, the maximum PLL lock time should be 7 msec. The workaround resets the codec softly by toggling SHDN off and on if PLL failed to lock for 10 msec. Notably, there is no suggested hold time for SHDN off. On Baytrail-based chromebooks, it would easily happen continuous PLL unlocked if there is a 10 msec delay between SHDN off and on. Removes the msleep(). Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20191122073114.219945-2-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c3b28b2f4b10..f8c8b9decab6 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2132,10 +2132,16 @@ static void max98090_pll_work(struct work_struct *work) dev_info_ratelimited(component->dev, "PLL unlocked\n"); + /* + * As the datasheet suggested, the maximum PLL lock time should be + * 7 msec. The workaround resets the codec softly by toggling SHDN + * off and on if PLL failed to lock for 10 msec. Notably, there is + * no suggested hold time for SHDN off. + */ + /* Toggle shutdown OFF then ON */ snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, 0); - msleep(10); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); -- 2.20.1

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[alsa-devel] [PATCH 6/6] ASoC: amd MMAP_INTERLEAVED Support
by Ravulapati Vishnu vardhan rao 27 Dec '19

27 Dec '19

ACP-I2S device support MMAP_INTERLEAVED. Added support for the same. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 7045a34..aae7ec6 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -22,6 +22,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | @@ -42,7 +43,8 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, -- 2.7.4

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[alsa-devel] [PATCH 5/6] ASoC: amd: Added ACP3x system resume and runtime pm
by Ravulapati Vishnu vardhan rao 27 Dec '19

27 Dec '19

When system wide suspend happens, ACP will be powered off and when system resumes,for audio usecase to continue,all the runtime configuration data needs to be programmed again. Added resume pm call back to ACP pm ops and also added runtime PM operations for ACP3x PCM platform device. Device will enter into D3 state when there is no activity on audio I2S lines Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 149 ++----------------------------- sound/soc/amd/raven/acp3x.h | 7 ++ sound/soc/amd/raven/pci-acp3x.c | 173 ++++++++++++++++++++++++++++++++++-- 3 files changed, 180 insertions(+), 149 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 1bd4f76..7045a34 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -58,106 +58,6 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { .periods_max = CAPTURE_MAX_NUM_PERIODS, }; -static int acp3x_power_on(void __iomem *acp3x_base, bool on) -{ - u16 val, mask; - u32 timeout; - - if (on == true) { - val = 1; - mask = ACP3x_POWER_ON; - } else { - val = 0; - mask = ACP3x_POWER_OFF; - } - - rv_writel(val, acp3x_base + mmACP_PGFSM_CONTROL); - timeout = 0; - while (true) { - val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); - if ((val & ACP3x_POWER_OFF_IN_PROGRESS) == mask) - break; - if (timeout > 100) { - pr_err("ACP3x power state change failure\n"); - return -ENODEV; - } - timeout++; - cpu_relax(); - } - return 0; -} - -static int acp3x_reset(void __iomem *acp3x_base) -{ - u32 val, timeout; - - rv_writel(1, acp3x_base + mmACP_SOFT_RESET); - timeout = 0; - while (true) { - val = rv_readl(acp3x_base + mmACP_SOFT_RESET); - if ((val & ACP3x_SOFT_RESET__SoftResetAudDone_MASK) || - timeout > 100) { - if (val & ACP3x_SOFT_RESET__SoftResetAudDone_MASK) - break; - return -ENODEV; - } - timeout++; - cpu_relax(); - } - - rv_writel(0, acp3x_base + mmACP_SOFT_RESET); - timeout = 0; - while (true) { - val = rv_readl(acp3x_base + mmACP_SOFT_RESET); - if (!val || timeout > 100) { - if (!val) - break; - return -ENODEV; - } - timeout++; - cpu_relax(); - } - return 0; -} - -static int acp3x_init(void __iomem *acp3x_base) -{ - int ret; - - /* power on */ - ret = acp3x_power_on(acp3x_base, true); - if (ret) { - pr_err("ACP3x power on failed\n"); - return ret; - } - /* Reset */ - ret = acp3x_reset(acp3x_base); - if (ret) { - pr_err("ACP3x reset failed\n"); - return ret; - } - return 0; -} - -static int acp3x_deinit(void __iomem *acp3x_base) -{ - int ret; - - /* Reset */ - ret = acp3x_reset(acp3x_base); - if (ret) { - pr_err("ACP3x reset failed\n"); - return ret; - } - /* power off */ - ret = acp3x_power_on(acp3x_base, false); - if (ret) { - pr_err("ACP3x power off failed\n"); - return ret; - } - return 0; -} - static irqreturn_t i2s_irq_handler(int irq, void *dev_id) { struct i2s_dev_data *rv_i2s_data; @@ -480,7 +380,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) struct resource *res; struct i2s_dev_data *adata; unsigned int irqflags; - int status, ret; + int status; if (!pdev->dev.platform_data) { dev_err(&pdev->dev, "platform_data not retrieved\n"); @@ -512,53 +412,29 @@ static int acp3x_audio_probe(struct platform_device *pdev) adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata); - /* Initialize ACP */ - status = acp3x_init(adata->acp3x_base); - if (status) - return -ENODEV; - status = devm_snd_soc_register_component(&pdev->dev, &acp3x_i2s_component, NULL, 0); if (status) { dev_err(&pdev->dev, "Fail to register acp i2s component\n"); - ret = -ENODEV; - goto dev_err; + return -ENODEV; } status = devm_request_irq(&pdev->dev, adata->i2s_irq, i2s_irq_handler, irqflags, "ACP3x_I2S_IRQ", adata); if (status) { dev_err(&pdev->dev, "ACP3x I2S IRQ request failed\n"); - ret = -ENODEV; - goto dev_err; + return -ENODEV; } - pm_runtime_set_autosuspend_delay(&pdev->dev, 5000); + pm_runtime_set_autosuspend_delay(&pdev->dev, 2000); pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_enable(&pdev->dev); + pm_runtime_allow(&pdev->dev); return 0; - -dev_err: - status = acp3x_deinit(adata->acp3x_base); - if (status) - dev_err(&pdev->dev, "ACP de-init failed\n"); - else - dev_dbg(&pdev->dev, "ACP de-initialized\n"); - return ret; } static int acp3x_audio_remove(struct platform_device *pdev) { - struct i2s_dev_data *adata; - int ret; - - adata = dev_get_drvdata(&pdev->dev); - ret = acp3x_deinit(adata->acp3x_base); - if (ret) - dev_err(&pdev->dev, "ACP de-init failed\n"); - else - dev_dbg(&pdev->dev, "ACP de-initialized\n"); - pm_runtime_disable(&pdev->dev); return 0; } @@ -566,15 +442,11 @@ static int acp3x_audio_remove(struct platform_device *pdev) static int acp3x_resume(struct device *dev) { struct i2s_dev_data *adata; - int status; u32 val, reg_val, frmt_val; reg_val = 0; frmt_val = 0; adata = dev_get_drvdata(dev); - status = acp3x_init(adata->acp3x_base); - if (status) - return -ENODEV; if (adata->play_stream && adata->play_stream->runtime) { struct i2s_stream_instance *rtd = @@ -621,14 +493,8 @@ static int acp3x_resume(struct device *dev) static int acp3x_pcm_runtime_suspend(struct device *dev) { struct i2s_dev_data *adata; - int status; adata = dev_get_drvdata(dev); - status = acp3x_deinit(adata->acp3x_base); - if (status) - dev_err(dev, "ACP de-init failed\n"); - else - dev_dbg(dev, "ACP de-initialized\n"); rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); @@ -638,12 +504,9 @@ static int acp3x_pcm_runtime_suspend(struct device *dev) static int acp3x_pcm_runtime_resume(struct device *dev) { struct i2s_dev_data *adata; - int status; adata = dev_get_drvdata(dev); - status = acp3x_init(adata->acp3x_base); - if (status) - return -ENODEV; + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); return 0; } diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 43213ae..b6a80dc 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -68,6 +68,13 @@ #define SLOT_WIDTH_16 0x10 #define SLOT_WIDTH_24 0x18 #define SLOT_WIDTH_32 0x20 +#define ACP_PGFSM_CNTL_POWER_ON_MASK 0x01 +#define ACP_PGFSM_CNTL_POWER_OFF_MASK 0x00 +#define ACP_PGFSM_STATUS_MASK 0x03 +#define ACP_POWERED_ON 0x00 +#define ACP_POWER_ON_IN_PROGRESS 0x01 +#define ACP_POWERED_OFF 0x02 +#define ACP_POWER_OFF_IN_PROGRESS 0x03 struct acp3x_platform_info { u16 play_i2s_instance; diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 94f5f21..2f9f5290 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -9,6 +9,8 @@ #include <linux/io.h> #include <linux/platform_device.h> #include <linux/interrupt.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> #include "acp3x.h" @@ -19,6 +21,109 @@ struct acp3x_dev_data { struct platform_device *pdev[ACP3x_DEVS]; }; +static int acp3x_power_on(void __iomem *acp3x_base) +{ + u32 val; + int timeout; + + val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); + + if (val == 0) + return val; + + if (!((val & ACP_PGFSM_STATUS_MASK) == + ACP_POWER_ON_IN_PROGRESS)) + rv_writel(ACP_PGFSM_CNTL_POWER_ON_MASK, + acp3x_base + mmACP_PGFSM_CONTROL); + timeout = 0; + while (++timeout < 500) { + val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); + if (!val) + return 0; + udelay(1); + } + return -ETIMEDOUT; +} + +static int acp3x_power_off(void __iomem *acp3x_base) +{ + u32 val; + int timeout; + + rv_writel(ACP_PGFSM_CNTL_POWER_OFF_MASK, + acp3x_base + mmACP_PGFSM_CONTROL); + timeout = 0; + while (++timeout < 500) { + val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); + if ((val & ACP_PGFSM_STATUS_MASK) == ACP_POWERED_OFF) + return 0; + udelay(1); + } + return -ETIMEDOUT; +} + +static int acp3x_reset(void __iomem *acp3x_base) +{ + u32 val; + int timeout; + + rv_writel(1, acp3x_base + mmACP_SOFT_RESET); + timeout = 0; + while (++timeout < 500) { + val = rv_readl(acp3x_base + mmACP_SOFT_RESET); + if (val & ACP3x_SOFT_RESET__SoftResetAudDone_MASK) + break; + cpu_relax(); + } + rv_writel(0, acp3x_base + mmACP_SOFT_RESET); + timeout = 0; + while (++timeout < 500) { + val = rv_readl(acp3x_base + mmACP_SOFT_RESET); + if (!val) + return 0; + cpu_relax(); + } + return -ETIMEDOUT; +} + +static int acp3x_init(void __iomem *acp3x_base) +{ + int ret; + + /* power on */ + ret = acp3x_power_on(acp3x_base); + if (ret) { + pr_err("ACP3x power on failed\n"); + return ret; + } + /* Reset */ + ret = acp3x_reset(acp3x_base); + if (ret) { + pr_err("ACP3x reset failed\n"); + return ret; + } + return 0; +} + +static int acp3x_deinit(void __iomem *acp3x_base) +{ + int ret; + + /* Reset */ + ret = acp3x_reset(acp3x_base); + if (ret) { + pr_err("ACP3x reset failed\n"); + return ret; + } + /* power off */ + ret = acp3x_power_off(acp3x_base); + if (ret) { + pr_err("ACP3x power off failed\n"); + return ret; + } + return 0; +} + static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -64,6 +169,9 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); + ret = acp3x_init(adata->acp3x_base); + if (ret) + goto disable_msi; val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); switch (val) { @@ -73,7 +181,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, GFP_KERNEL); if (!adata->res) { ret = -ENOMEM; - goto disable_msi; + goto de_init; } adata->res[0].name = "acp3x_i2s_iomem"; @@ -118,7 +226,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, pdevinfo[2].parent = &pci->dev; pdevinfo[2].num_res = 1; pdevinfo[2].res = &adata->res[2]; - for (i = 0; i < ACP3x_DEVS ; i++) { + for (i = 0; i < ACP3x_DEVS; i++) { adata->pdev[i] = platform_device_register_full(&pdevinfo[i]); if (IS_ERR(adata->pdev[i])) { @@ -134,12 +242,21 @@ static int snd_acp3x_probe(struct pci_dev *pci, ret = -ENODEV; goto disable_msi; } + pm_runtime_set_autosuspend_delay(&pci->dev, 2000); + pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_set_active(&pci->dev); + pm_runtime_put_noidle(&pci->dev); + pm_runtime_enable(&pci->dev); + pm_runtime_allow(&pci->dev); return 0; unregister_devs: if (val == I2S_MODE) - for (i = 0 ; i < ACP3x_DEVS ; i++) + for (i = 0; i < ACP3x_DEVS; i++) platform_device_unregister(adata->pdev[i]); +de_init: + if (acp3x_deinit(adata->acp3x_base)) + dev_err(&pci->dev, "ACP de-init failed\n"); disable_msi: pci_disable_msi(pci); release_regions: @@ -150,15 +267,56 @@ static int snd_acp3x_probe(struct pci_dev *pci, return ret; } +static int snd_acp3x_suspend(struct device *dev) +{ + int ret; + struct acp3x_dev_data *adata; + + adata = dev_get_drvdata(dev); + ret = acp3x_deinit(adata->acp3x_base); + if (ret) + dev_err(dev, "ACP de-init failed\n"); + else + dev_dbg(dev, "ACP de-initialized\n"); + + return 0; +} + +static int snd_acp3x_resume(struct device *dev) +{ + int ret; + struct acp3x_dev_data *adata; + + adata = dev_get_drvdata(dev); + ret = acp3x_init(adata->acp3x_base); + if (ret) { + dev_err(dev, "ACP init failed\n"); + return ret; + } + return 0; +} + +static const struct dev_pm_ops acp3x_pm = { + .runtime_suspend = snd_acp3x_suspend, + .runtime_resume = snd_acp3x_resume, + .resume = snd_acp3x_resume, +}; + static void snd_acp3x_remove(struct pci_dev *pci) { - struct acp3x_dev_data *adata = pci_get_drvdata(pci); - int i; + struct acp3x_dev_data *adata; + int i, ret; + adata = pci_get_drvdata(pci); if (adata->acp3x_audio_mode == ACP3x_I2S_MODE) { - for (i = 0 ; i < ACP3x_DEVS ; i++) + for (i = 0; i < ACP3x_DEVS; i++) platform_device_unregister(adata->pdev[i]); } + ret = acp3x_deinit(adata->acp3x_base); + if (ret) + dev_err(&pci->dev, "ACP de-init failed\n"); + pm_runtime_disable(&pci->dev); + pm_runtime_get_noresume(&pci->dev); pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); @@ -177,6 +335,9 @@ static struct pci_driver acp3x_driver = { .id_table = snd_acp3x_ids, .probe = snd_acp3x_probe, .remove = snd_acp3x_remove, + .driver = { + .pm = &acp3x_pm, + } }; module_pci_driver(acp3x_driver); -- 2.7.4

1 0

[alsa-devel] [PATCH 4/6] ASoC: amd: Handle ACP3x I2S-SP Interrupts.
by Ravulapati Vishnu vardhan rao 27 Dec '19

27 Dec '19

Enabled support for I2S-SP interrupt handling. Previous to this implementation, driver supports only interrupts on BT instance. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index b67aea3..1bd4f76 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -177,6 +177,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) snd_pcm_period_elapsed(rv_i2s_data->play_stream); play_flag = 1; } + if ((val & BIT(I2S_TX_THRESHOLD)) && + rv_i2s_data->i2ssp_play_stream) { + rv_writel(BIT(I2S_TX_THRESHOLD), + rv_i2s_data->acp3x_base + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->i2ssp_play_stream); + play_flag = 1; + } if ((val & BIT(BT_RX_THRESHOLD)) && rv_i2s_data->capture_stream) { rv_writel(BIT(BT_RX_THRESHOLD), rv_i2s_data->acp3x_base + @@ -184,6 +191,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) snd_pcm_period_elapsed(rv_i2s_data->capture_stream); cap_flag = 1; } + if ((val & BIT(I2S_RX_THRESHOLD)) && + rv_i2s_data->i2ssp_capture_stream) { + rv_writel(BIT(I2S_RX_THRESHOLD), + rv_i2s_data->acp3x_base + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->i2ssp_capture_stream); + cap_flag = 1; + } if (play_flag | cap_flag) return IRQ_HANDLED; -- 2.7.4

1 0

[alsa-devel] [PATCH 3/6] ASoC: amd: add ACP3x TDM mode support
by Ravulapati Vishnu vardhan rao 27 Dec '19

27 Dec '19

ACP3x I2S (CPU DAI) can act in normal I2S and TDM modes. Added support for TDM mode. Desired mode can be selected from ASoC machine driver. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 28 ++++++++++++++++++++++------ sound/soc/amd/raven/acp3x.h | 1 + 2 files changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 7f05782..0856e0c 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -43,7 +43,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, u32 rx_mask, int slots, int slot_width) { struct i2s_dev_data *adata; - u32 val, reg_val, frmt_val, frm_len; + u32 val, reg_val, frmt_reg, frm_len; u16 slot_len; adata = snd_soc_dai_get_drvdata(cpu_dai); @@ -70,15 +70,31 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, frm_len = FRM_LEN | (slots << 15) | (slot_len << 18); if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) { - reg_val = mmACP_BTTDM_ITER; - frmt_val = mmACP_BTTDM_TXFRMT; + switch (adata->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_ITER; + frmt_reg = mmACP_BTTDM_TXFRMT; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_ITER; + frmt_reg = mmACP_I2STDM_TXFRMT; + } } else { - reg_val = mmACP_BTTDM_IRER; - frmt_val = mmACP_BTTDM_RXFRMT; + switch (adata->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_IRER; + frmt_reg = mmACP_BTTDM_RXFRMT; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_IRER; + frmt_reg = mmACP_I2STDM_RXFRMT; + } } val = rv_readl(adata->acp3x_base + reg_val); rv_writel(val | 0x2, adata->acp3x_base + reg_val); - rv_writel(frm_len, adata->acp3x_base + frmt_val); + rv_writel(frm_len, adata->acp3x_base + frmt_reg); adata->tdm_fmt = frm_len; return 0; } diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index a1cdc4e..43213ae 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -78,6 +78,7 @@ struct acp3x_platform_info { struct i2s_dev_data { bool tdm_mode; unsigned int i2s_irq; + u16 i2s_instance; u32 tdm_fmt; u32 substream_type; void __iomem *acp3x_base; -- 2.7.4

1 0

[alsa-devel] [PATCH 2/6] ASoC: amd: Enabling I2S instance in DMA and DAI
by Ravulapati Vishnu vardhan rao 27 Dec '19

27 Dec '19

This patch adds I2S SP support in ACP PCM DMA and DAI. Added I2S support in DMA and DAI probe,its hw_params handling its open and close functionalities. This enables to open and close on the SP instance for playback and capture. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 123 ++++++++++++++++---- sound/soc/amd/raven/acp3x-pcm-dma.c | 217 +++++++++++++++++++++++++----------- sound/soc/amd/raven/acp3x.h | 77 ++++++++++--- 3 files changed, 313 insertions(+), 104 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index cdc1c61..7f05782 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -28,10 +28,10 @@ static int acp3x_i2s_set_fmt(struct snd_soc_dai *cpu_dai, mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; switch (mode) { case SND_SOC_DAIFMT_I2S: - adata->tdm_mode = false; + adata->tdm_mode = TDM_DISABLE; break; case SND_SOC_DAIFMT_DSP_A: - adata->tdm_mode = true; + adata->tdm_mode = TDM_ENABLE; break; default: return -EINVAL; @@ -87,10 +87,22 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct i2s_stream_instance *rtd; + struct snd_soc_pcm_runtime *prtd; + struct snd_soc_card *card; + struct acp3x_platform_info *pinfo; u32 val; u32 reg_val; + prtd = substream->private_data; rtd = substream->runtime->private_data; + card = prtd->card; + pinfo = snd_soc_card_get_drvdata(card); + if (pinfo) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rtd->i2s_instance = pinfo->play_i2s_instance; + else + rtd->i2s_instance = pinfo->cap_i2s_instance; + } /* These values are as per Hardware Spec */ switch (params_format(params)) { @@ -110,11 +122,25 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - reg_val = mmACP_BTTDM_ITER; - else - reg_val = mmACP_BTTDM_IRER; - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_ITER; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_ITER; + } + } else { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_IRER; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_IRER; + } + } val = rv_readl(rtd->acp3x_base + reg_val); val = val | (rtd->xfer_resolution << 3); rv_writel(val, rtd->acp3x_base + reg_val); @@ -125,10 +151,21 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct i2s_stream_instance *rtd; - u32 val, period_bytes; - int ret, reg_val; + struct snd_soc_pcm_runtime *prtd; + struct snd_soc_card *card; + struct acp3x_platform_info *pinfo; + u32 ret, val, period_bytes, reg_val, ier_val, water_val; + prtd = substream->private_data; rtd = substream->runtime->private_data; + card = prtd->card; + pinfo = snd_soc_card_get_drvdata(card); + if (pinfo) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rtd->i2s_instance = pinfo->play_i2s_instance; + else + rtd->i2s_instance = pinfo->cap_i2s_instance; + } period_bytes = frames_to_bytes(substream->runtime, substream->runtime->period_size); switch (cmd) { @@ -138,31 +175,75 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, rtd->bytescount = acp_get_byte_count(rtd, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - reg_val = mmACP_BTTDM_ITER; - rv_writel(period_bytes, rtd->acp3x_base + - mmACP_BT_TX_INTR_WATERMARK_SIZE); + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + water_val = + mmACP_BT_TX_INTR_WATERMARK_SIZE; + reg_val = mmACP_BTTDM_ITER; + ier_val = mmACP_BTTDM_IER; + break; + case I2S_SP_INSTANCE: + default: + water_val = + mmACP_I2S_TX_INTR_WATERMARK_SIZE; + reg_val = mmACP_I2STDM_ITER; + ier_val = mmACP_I2STDM_IER; + } } else { - reg_val = mmACP_BTTDM_IRER; - rv_writel(period_bytes, rtd->acp3x_base + - mmACP_BT_RX_INTR_WATERMARK_SIZE); + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + water_val = + mmACP_BT_RX_INTR_WATERMARK_SIZE; + reg_val = mmACP_BTTDM_IRER; + ier_val = mmACP_BTTDM_IER; + break; + case I2S_SP_INSTANCE: + default: + water_val = + mmACP_I2S_RX_INTR_WATERMARK_SIZE; + reg_val = mmACP_I2STDM_IRER; + ier_val = mmACP_I2STDM_IER; + } } + rv_writel(period_bytes, rtd->acp3x_base + water_val); val = rv_readl(rtd->acp3x_base + reg_val); val = val | BIT(0); rv_writel(val, rtd->acp3x_base + reg_val); - rv_writel(1, rtd->acp3x_base + mmACP_BTTDM_IER); + rv_writel(1, rtd->acp3x_base + ier_val); + ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - reg_val = mmACP_BTTDM_ITER; - else - reg_val = mmACP_BTTDM_IRER; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_ITER; + ier_val = mmACP_BTTDM_IER; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_ITER; + ier_val = mmACP_I2STDM_IER; + } + } else { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_IRER; + ier_val = mmACP_BTTDM_IER; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_IRER; + ier_val = mmACP_I2STDM_IER; + } + } val = rv_readl(rtd->acp3x_base + reg_val); val = val & ~BIT(0); rv_writel(val, rtd->acp3x_base + reg_val); - rv_writel(0, rtd->acp3x_base + mmACP_BTTDM_IER); + rv_writel(0, rtd->acp3x_base + ier_val); + ret = 0; break; default: ret = -EINVAL; diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 351143b..b67aea3 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -194,15 +194,31 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) { u16 page_idx; - u32 low, high, val, acp_fifo_addr; - dma_addr_t addr = rtd->dma_addr; + u32 low, high, val, acp_fifo_addr, reg_fifo_addr; + u32 reg_ringbuf_size, reg_dma_size, reg_fifo_size; + dma_addr_t addr; - /* 8 scratch registers used to map one 64 bit address */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - val = 0; - else - val = rtd->num_pages * 8; + addr = rtd->dma_addr; + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + val = ACP_SRAM_BT_PB_PTE_OFFSET; + break; + case I2S_SP_INSTANCE: + default: + val = ACP_SRAM_SP_PB_PTE_OFFSET; + } + } else { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + val = ACP_SRAM_BT_CP_PTE_OFFSET; + break; + case I2S_SP_INSTANCE: + default: + val = ACP_SRAM_SP_CP_PTE_OFFSET; + } + } /* Group Enable */ rv_writel(ACP_SRAM_PTE_OFFSET | BIT(31), rtd->acp3x_base + mmACPAXI2AXI_ATU_BASE_ADDR_GRP_1); @@ -224,38 +240,61 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) } if (direction == SNDRV_PCM_STREAM_PLAYBACK) { - /* Config ringbuffer */ - rv_writel(MEM_WINDOW_START, rtd->acp3x_base + - mmACP_BT_TX_RINGBUFADDR); - rv_writel(MAX_BUFFER, rtd->acp3x_base + - mmACP_BT_TX_RINGBUFSIZE); - rv_writel(DMA_SIZE, rtd->acp3x_base + mmACP_BT_TX_DMA_SIZE); - - /* Config audio fifo */ - acp_fifo_addr = ACP_SRAM_PTE_OFFSET + (rtd->num_pages * 8) - + PLAYBACK_FIFO_ADDR_OFFSET; - rv_writel(acp_fifo_addr, rtd->acp3x_base + - mmACP_BT_TX_FIFOADDR); - rv_writel(FIFO_SIZE, rtd->acp3x_base + mmACP_BT_TX_FIFOSIZE); + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_ringbuf_size = mmACP_BT_TX_RINGBUFSIZE; + reg_dma_size = mmACP_BT_TX_DMA_SIZE; + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + BT_PB_FIFO_ADDR_OFFSET; + reg_fifo_addr = mmACP_BT_TX_FIFOADDR; + reg_fifo_size = mmACP_BT_TX_FIFOSIZE; + rv_writel(I2S_BT_TX_MEM_WINDOW_START, + rtd->acp3x_base + mmACP_BT_TX_RINGBUFADDR); + break; + + case I2S_SP_INSTANCE: + default: + reg_ringbuf_size = mmACP_I2S_TX_RINGBUFSIZE; + reg_dma_size = mmACP_I2S_TX_DMA_SIZE; + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + SP_PB_FIFO_ADDR_OFFSET; + reg_fifo_addr = mmACP_I2S_TX_FIFOADDR; + reg_fifo_size = mmACP_I2S_TX_FIFOSIZE; + rv_writel(I2S_SP_TX_MEM_WINDOW_START, + rtd->acp3x_base + mmACP_I2S_TX_RINGBUFADDR); + } } else { - /* Config ringbuffer */ - rv_writel(MEM_WINDOW_START + MAX_BUFFER, rtd->acp3x_base + - mmACP_BT_RX_RINGBUFADDR); - rv_writel(MAX_BUFFER, rtd->acp3x_base + - mmACP_BT_RX_RINGBUFSIZE); - rv_writel(DMA_SIZE, rtd->acp3x_base + mmACP_BT_RX_DMA_SIZE); - - /* Config audio fifo */ - acp_fifo_addr = ACP_SRAM_PTE_OFFSET + - (rtd->num_pages * 8) + CAPTURE_FIFO_ADDR_OFFSET; - rv_writel(acp_fifo_addr, rtd->acp3x_base + - mmACP_BT_RX_FIFOADDR); - rv_writel(FIFO_SIZE, rtd->acp3x_base + mmACP_BT_RX_FIFOSIZE); - } + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_ringbuf_size = mmACP_BT_RX_RINGBUFSIZE; + reg_dma_size = mmACP_BT_RX_DMA_SIZE; + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + BT_CAPT_FIFO_ADDR_OFFSET; + reg_fifo_addr = mmACP_BT_RX_FIFOADDR; + reg_fifo_size = mmACP_BT_RX_FIFOSIZE; + rv_writel(I2S_BT_RX_MEM_WINDOW_START, + rtd->acp3x_base + mmACP_BT_RX_RINGBUFADDR); + break; - /* Enable watermark/period interrupt to host */ - rv_writel(BIT(BT_TX_THRESHOLD) | BIT(BT_RX_THRESHOLD), - rtd->acp3x_base + mmACP_EXTERNAL_INTR_CNTL); + case I2S_SP_INSTANCE: + default: + reg_ringbuf_size = mmACP_I2S_RX_RINGBUFSIZE; + reg_dma_size = mmACP_I2S_RX_DMA_SIZE; + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + SP_CAPT_FIFO_ADDR_OFFSET; + reg_fifo_addr = mmACP_I2S_RX_FIFOADDR; + reg_fifo_size = mmACP_I2S_RX_FIFOSIZE; + rv_writel(I2S_SP_RX_MEM_WINDOW_START, + rtd->acp3x_base + mmACP_I2S_RX_RINGBUFADDR); + } + } + rv_writel(MAX_BUFFER, rtd->acp3x_base + reg_ringbuf_size); + rv_writel(DMA_SIZE, rtd->acp3x_base + reg_dma_size); + rv_writel(acp_fifo_addr, rtd->acp3x_base + reg_fifo_addr); + rv_writel(FIFO_SIZE, rtd->acp3x_base + reg_fifo_size); + rv_writel(BIT(I2S_RX_THRESHOLD) | BIT(BT_RX_THRESHOLD) + | BIT(I2S_TX_THRESHOLD) | BIT(BT_TX_THRESHOLD), + rtd->acp3x_base + mmACP_EXTERNAL_INTR_CNTL); } static int acp3x_dma_open(struct snd_soc_component *component, @@ -288,17 +327,21 @@ static int acp3x_dma_open(struct snd_soc_component *component, return ret; } - if (!adata->play_stream && !adata->capture_stream) + if (!adata->play_stream && !adata->capture_stream && + adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { adata->play_stream = substream; - else + adata->i2ssp_play_stream = substream; + } else { adata->capture_stream = substream; + adata->i2ssp_capture_stream = substream; + } i2s_data->acp3x_base = adata->acp3x_base; runtime->private_data = i2s_data; - return 0; + return ret; } @@ -306,13 +349,27 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { + struct i2s_stream_instance *rtd; + struct snd_soc_pcm_runtime *prtd; + struct snd_soc_card *card; + struct acp3x_platform_info *pinfo; u64 size; - struct snd_pcm_runtime *runtime = substream->runtime; - struct i2s_stream_instance *rtd = runtime->private_data; + prtd = substream->private_data; + card = prtd->card; + pinfo = snd_soc_card_get_drvdata(card); + rtd = substream->runtime->private_data; if (!rtd) return -EINVAL; + if (pinfo) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rtd->i2s_instance = pinfo->play_i2s_instance; + else + rtd->i2s_instance = pinfo->cap_i2s_instance; + else + pr_err("pinfo failed\n"); + size = params_buffer_bytes(params); rtd->dma_addr = substream->dma_buffer.addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); @@ -323,12 +380,25 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *prtd; + struct snd_soc_card *card; + struct acp3x_platform_info *pinfo; struct i2s_stream_instance *rtd; u32 pos; u32 buffersize; u64 bytescount; + prtd = substream->private_data; + card = prtd->card; rtd = substream->runtime->private_data; + pinfo = snd_soc_card_get_drvdata(card); + if (pinfo) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rtd->i2s_instance = pinfo->play_i2s_instance; + else + rtd->i2s_instance = pinfo->cap_i2s_instance; + } + buffersize = frames_to_bytes(substream->runtime, substream->runtime->buffer_size); bytescount = acp_get_byte_count(rtd, substream->stream); @@ -364,15 +434,19 @@ static int acp3x_dma_close(struct snd_soc_component *component, component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); adata = dev_get_drvdata(component->dev); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { adata->play_stream = NULL; - else + adata->i2ssp_play_stream = NULL; + } else { adata->capture_stream = NULL; + adata->i2ssp_capture_stream = NULL; + } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. */ - if (!adata->play_stream && !adata->capture_stream) + if (!adata->play_stream && !adata->capture_stream && + !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); return 0; } @@ -479,8 +553,10 @@ static int acp3x_resume(struct device *dev) { struct i2s_dev_data *adata; int status; - u32 val; + u32 val, reg_val, frmt_val; + reg_val = 0; + frmt_val = 0; adata = dev_get_drvdata(dev); status = acp3x_init(adata->acp3x_base); if (status) @@ -490,32 +566,39 @@ static int acp3x_resume(struct device *dev) struct i2s_stream_instance *rtd = adata->play_stream->runtime->private_data; config_acp3x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); - rv_writel((rtd->xfer_resolution << 3), - rtd->acp3x_base + mmACP_BTTDM_ITER); - if (adata->tdm_mode == true) { - rv_writel(adata->tdm_fmt, adata->acp3x_base + - mmACP_BTTDM_TXFRMT); - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); - rv_writel((val | 0x2), adata->acp3x_base + - mmACP_BTTDM_ITER); + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_ITER; + frmt_val = mmACP_BTTDM_TXFRMT; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_ITER; + frmt_val = mmACP_I2STDM_TXFRMT; } + rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); } - if (adata->capture_stream && adata->capture_stream->runtime) { struct i2s_stream_instance *rtd = adata->capture_stream->runtime->private_data; config_acp3x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); - rv_writel((rtd->xfer_resolution << 3), - rtd->acp3x_base + mmACP_BTTDM_IRER); - if (adata->tdm_mode == true) { - rv_writel(adata->tdm_fmt, adata->acp3x_base + - mmACP_BTTDM_RXFRMT); - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); - rv_writel((val | 0x2), adata->acp3x_base + - mmACP_BTTDM_IRER); + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + reg_val = mmACP_BTTDM_IRER; + frmt_val = mmACP_BTTDM_RXFRMT; + break; + case I2S_SP_INSTANCE: + default: + reg_val = mmACP_I2STDM_IRER; + frmt_val = mmACP_I2STDM_RXFRMT; } + rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + } + if (adata->tdm_mode == TDM_ENABLE) { + rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val); + val = rv_readl(adata->acp3x_base + reg_val); + rv_writel(val | 0x2, adata->acp3x_base + reg_val); } - rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); return 0; } @@ -525,8 +608,8 @@ static int acp3x_pcm_runtime_suspend(struct device *dev) { struct i2s_dev_data *adata; int status; - adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_deinit(adata->acp3x_base); if (status) dev_err(dev, "ACP de-init failed\n"); @@ -542,8 +625,8 @@ static int acp3x_pcm_runtime_resume(struct device *dev) { struct i2s_dev_data *adata; int status; - adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_init(adata->acp3x_base); if (status) return -ENODEV; diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index a6c6d63..a1cdc4e 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -6,6 +6,12 @@ */ #include "chip_offset_byte.h" +#include <sound/pcm.h> +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 + +#define TDM_ENABLE 1 +#define TDM_DISABLE 0 #define ACP3x_DEVS 3 #define ACP3x_PHY_BASE_ADDRESS 0x1240000 @@ -17,8 +23,11 @@ #define ACP3x_BT_TDM_REG_START 0x1242800 #define ACP3x_BT_TDM_REG_END 0x1242810 #define I2S_MODE 0x04 +#define I2S_RX_THRESHOLD 27 +#define I2S_TX_THRESHOLD 28 #define BT_TX_THRESHOLD 26 #define BT_RX_THRESHOLD 25 +#define ACP_ERR_INTR_MASK 29 #define ACP3x_POWER_ON 0x00 #define ACP3x_POWER_ON_IN_PROGRESS 0x01 #define ACP3x_POWER_OFF 0x02 @@ -26,19 +35,28 @@ #define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 #define ACP_SRAM_PTE_OFFSET 0x02050000 +#define ACP_SRAM_SP_PB_PTE_OFFSET 0x0 +#define ACP_SRAM_SP_CP_PTE_OFFSET 0x100 +#define ACP_SRAM_BT_PB_PTE_OFFSET 0x200 +#define ACP_SRAM_BT_CP_PTE_OFFSET 0x300 #define PAGE_SIZE_4K_ENABLE 0x2 -#define MEM_WINDOW_START 0x4000000 -#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 -#define CAPTURE_FIFO_ADDR_OFFSET 0x500 +#define I2S_SP_TX_MEM_WINDOW_START 0x4000000 +#define I2S_SP_RX_MEM_WINDOW_START 0x4020000 +#define I2S_BT_TX_MEM_WINDOW_START 0x4040000 +#define I2S_BT_RX_MEM_WINDOW_START 0x4060000 +#define SP_PB_FIFO_ADDR_OFFSET 0x500 +#define SP_CAPT_FIFO_ADDR_OFFSET 0x700 +#define BT_PB_FIFO_ADDR_OFFSET 0x900 +#define BT_CAPT_FIFO_ADDR_OFFSET 0xB00 #define PLAYBACK_MIN_NUM_PERIODS 2 #define PLAYBACK_MAX_NUM_PERIODS 8 -#define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 4096 +#define PLAYBACK_MAX_PERIOD_SIZE 8192 +#define PLAYBACK_MIN_PERIOD_SIZE 1024 #define CAPTURE_MIN_NUM_PERIODS 2 #define CAPTURE_MAX_NUM_PERIODS 8 -#define CAPTURE_MAX_PERIOD_SIZE 16384 -#define CAPTURE_MIN_PERIOD_SIZE 4096 +#define CAPTURE_MAX_PERIOD_SIZE 8192 +#define CAPTURE_MIN_PERIOD_SIZE 1024 #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER @@ -65,14 +83,20 @@ struct i2s_dev_data { void __iomem *acp3x_base; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; + struct snd_pcm_substream *i2ssp_play_stream; + struct snd_pcm_substream *i2ssp_capture_stream; }; struct i2s_stream_instance { u16 num_pages; + u16 i2s_instance; + u16 capture_channel; + u16 direction; u16 channels; u32 xfer_resolution; - u64 bytescount; + u32 val; dma_addr_t dma_addr; + u64 bytescount; void __iomem *acp3x_base; }; @@ -92,15 +116,36 @@ static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, u64 byte_count; if (direction == SNDRV_PCM_STREAM_PLAYBACK) { - byte_count = rv_readl(rtd->acp3x_base + - mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH); - byte_count |= rv_readl(rtd->acp3x_base + - mmACP_BT_TX_LINEARPOSITIONCNTR_LOW); + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_LOW); + break; + case I2S_SP_INSTANCE: + default: + byte_count = rv_readl(rtd->acp3x_base + + mmACP_I2S_TX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_I2S_TX_LINEARPOSITIONCNTR_LOW); + } + } else { - byte_count = rv_readl(rtd->acp3x_base + - mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH); - byte_count |= rv_readl(rtd->acp3x_base + - mmACP_BT_RX_LINEARPOSITIONCNTR_LOW); + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_LOW); + break; + case I2S_SP_INSTANCE: + default: + byte_count = rv_readl(rtd->acp3x_base + + mmACP_I2S_RX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_I2S_RX_LINEARPOSITIONCNTR_LOW); + } } return byte_count; } -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Enable the subwoofer of ASUS UX431FLC
by Chris Chiu 27 Dec '19

27 Dec '19

From: Chris Chiu <chiu(a)endlessm.com> ASUS reported that there's an additional speaker which serves as the subwoofer and uses DAC 0x02. It does not work with the commit 73a723348a43 ("ALSA: hda/realtek - Enable internal speaker of ASUS UX431FLC") which enables the amplifier and front speakers. This commit enables the subwoofer to improve the acoustic experience. Signed-off-by: Chris Chiu <chiu(a)endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++------ 1 file changed, 22 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fada1ff61353..0193a8722be2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5576,6 +5576,16 @@ static void alc295_fixup_disable_dac3(struct hda_codec *codec, } } +/* Fixed DAC (0x02) on NID 0x17 to enable the mono speaker */ +static void alc294_fixup_fixed_dac_subwoofer(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn[1] = { 0x02 }; + snd_hda_override_conn_list(codec, 0x17, 1, conn); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -5950,7 +5960,8 @@ enum { ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, - ALC294_FIXUP_ASUS_INTSPK_GPIO, + ALC294_FIXUP_ASUS_DUAL_SPEAKERS, + ALC294_FIXUP_FIXED_DAC_SUBWOOFER, }; static const struct hda_fixup alc269_fixups[] = { @@ -7097,10 +7108,9 @@ static const struct hda_fixup alc269_fixups[] = { { } } }, - [ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC] = { + [ALC294_FIXUP_ASUS_HEADSET_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x411111f0 }, /* disable confusing internal speaker */ { 0x19, 0x04a11150 }, /* use as headset mic, without its own jack detect */ { } }, @@ -7117,12 +7127,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE }, - [ALC294_FIXUP_ASUS_INTSPK_GPIO] = { + [ALC294_FIXUP_FIXED_DAC_SUBWOOFER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_fixed_dac_subwoofer, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, + [ALC294_FIXUP_ASUS_DUAL_SPEAKERS] = { .type = HDA_FIXUP_FUNC, /* The GPIO must be pulled to initialize the AMP */ .v.func = alc_fixup_gpio4, .chained = true, - .chain_id = ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC + .chain_id = ALC294_FIXUP_FIXED_DAC_SUBWOOFER }, }; @@ -7291,7 +7307,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_GPIO), + SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), -- 2.20.1

3 3

[alsa-devel] [DELL] Add Bass Speaker and fixed dac for bass Speaker
by Kailang 27 Dec '19

27 Dec '19

Hi Takashi, Dell has new platform which has dual speaker connecting. They want dual speaker which use same dac for output. Many Thanks. BR, Kailang

2 1

[alsa-devel] Applied "ASoC: amd: Create multiple I2S platform device endpoint" to the asoc tree
by Mark Brown 27 Dec '19

27 Dec '19

The patch ASoC: amd: Create multiple I2S platform device endpoint has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5def0136c609062091b03a91ac1d987f4afc5363 Mon Sep 17 00:00:00 2001 From: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Date: Thu, 5 Dec 2019 19:07:26 +0530 Subject: [PATCH] ASoC: amd: Create multiple I2S platform device endpoint Creates Platform Device endpoints for multiple I2S instances: SP and BT endpoints device. Pass PCI resources like MMIO, irq to the platform devices. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Reviewed-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/1575553053-18344-2-git-send-email-Vishnuvardhanra… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/raven/acp3x.h | 5 ++ sound/soc/amd/raven/pci-acp3x.c | 95 ++++++++++++++++++++++----------- 2 files changed, 68 insertions(+), 32 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 4f2cadd90a87..2f15fe10ead5 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -7,10 +7,15 @@ #include "chip_offset_byte.h" +#define ACP3x_DEVS 3 #define ACP3x_PHY_BASE_ADDRESS 0x1240000 #define ACP3x_I2S_MODE 0 #define ACP3x_REG_START 0x1240000 #define ACP3x_REG_END 0x1250200 +#define ACP3x_I2STDM_REG_START 0x1242400 +#define ACP3x_I2STDM_REG_END 0x1242410 +#define ACP3x_BT_TDM_REG_START 0x1242800 +#define ACP3x_BT_TDM_REG_END 0x1242810 #define I2S_MODE 0x04 #define BT_TX_THRESHOLD 26 #define BT_RX_THRESHOLD 25 diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec2472b34..94f5f21d9a53 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -16,17 +16,17 @@ struct acp3x_dev_data { void __iomem *acp3x_base; bool acp3x_audio_mode; struct resource *res; - struct platform_device *pdev; + struct platform_device *pdev[ACP3x_DEVS]; }; static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { - int ret; - u32 addr, val; struct acp3x_dev_data *adata; - struct platform_device_info pdevinfo; + struct platform_device_info pdevinfo[ACP3x_DEVS]; unsigned int irqflags; + int ret, i; + u32 addr, val; if (pci_enable_device(pci)) { dev_err(&pci->dev, "pci_enable_device failed\n"); @@ -56,10 +56,11 @@ static int snd_acp3x_probe(struct pci_dev *pci, irqflags = 0; addr = pci_resource_start(pci, 0); - adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); + adata->acp3x_base = devm_ioremap(&pci->dev, addr, + pci_resource_len(pci, 0)); if (!adata->acp3x_base) { ret = -ENOMEM; - goto release_regions; + goto disable_msi; } pci_set_master(pci); pci_set_drvdata(pci, adata); @@ -68,11 +69,11 @@ static int snd_acp3x_probe(struct pci_dev *pci, switch (val) { case I2S_MODE: adata->res = devm_kzalloc(&pci->dev, - sizeof(struct resource) * 2, + sizeof(struct resource) * 4, GFP_KERNEL); if (!adata->res) { ret = -ENOMEM; - goto unmap_mmio; + goto disable_msi; } adata->res[0].name = "acp3x_i2s_iomem"; @@ -80,40 +81,67 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->res[0].start = addr; adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); - adata->res[1].name = "acp3x_i2s_irq"; - adata->res[1].flags = IORESOURCE_IRQ; - adata->res[1].start = pci->irq; - adata->res[1].end = pci->irq; + adata->res[1].name = "acp3x_i2s_sp"; + adata->res[1].flags = IORESOURCE_MEM; + adata->res[1].start = addr + ACP3x_I2STDM_REG_START; + adata->res[1].end = addr + ACP3x_I2STDM_REG_END; + + adata->res[2].name = "acp3x_i2s_bt"; + adata->res[2].flags = IORESOURCE_MEM; + adata->res[2].start = addr + ACP3x_BT_TDM_REG_START; + adata->res[2].end = addr + ACP3x_BT_TDM_REG_END; + + adata->res[3].name = "acp3x_i2s_irq"; + adata->res[3].flags = IORESOURCE_IRQ; + adata->res[3].start = pci->irq; + adata->res[3].end = adata->res[3].start; adata->acp3x_audio_mode = ACP3x_I2S_MODE; memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo.name = "acp3x_rv_i2s"; - pdevinfo.id = 0; - pdevinfo.parent = &pci->dev; - pdevinfo.num_res = 2; - pdevinfo.res = adata->res; - pdevinfo.data = &irqflags; - pdevinfo.size_data = sizeof(irqflags); - - adata->pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(adata->pdev)) { - dev_err(&pci->dev, "cannot register %s device\n", - pdevinfo.name); - ret = PTR_ERR(adata->pdev); - goto unmap_mmio; + pdevinfo[0].name = "acp3x_rv_i2s_dma"; + pdevinfo[0].id = 0; + pdevinfo[0].parent = &pci->dev; + pdevinfo[0].num_res = 4; + pdevinfo[0].res = &adata->res[0]; + pdevinfo[0].data = &irqflags; + pdevinfo[0].size_data = sizeof(irqflags); + + pdevinfo[1].name = "acp3x_i2s_playcap"; + pdevinfo[1].id = 0; + pdevinfo[1].parent = &pci->dev; + pdevinfo[1].num_res = 1; + pdevinfo[1].res = &adata->res[1]; + + pdevinfo[2].name = "acp3x_i2s_playcap"; + pdevinfo[2].id = 1; + pdevinfo[2].parent = &pci->dev; + pdevinfo[2].num_res = 1; + pdevinfo[2].res = &adata->res[2]; + for (i = 0; i < ACP3x_DEVS ; i++) { + adata->pdev[i] = + platform_device_register_full(&pdevinfo[i]); + if (IS_ERR(adata->pdev[i])) { + dev_err(&pci->dev, "cannot register %s device\n", + pdevinfo[i].name); + ret = PTR_ERR(adata->pdev[i]); + goto unregister_devs; + } } break; default: dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); ret = -ENODEV; - goto unmap_mmio; + goto disable_msi; } return 0; -unmap_mmio: +unregister_devs: + if (val == I2S_MODE) + for (i = 0 ; i < ACP3x_DEVS ; i++) + platform_device_unregister(adata->pdev[i]); +disable_msi: pci_disable_msi(pci); - iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); disable_pci: @@ -125,10 +153,12 @@ static int snd_acp3x_probe(struct pci_dev *pci, static void snd_acp3x_remove(struct pci_dev *pci) { struct acp3x_dev_data *adata = pci_get_drvdata(pci); + int i; - platform_device_unregister(adata->pdev); - iounmap(adata->acp3x_base); - + if (adata->acp3x_audio_mode == ACP3x_I2S_MODE) { + for (i = 0 ; i < ACP3x_DEVS ; i++) + platform_device_unregister(adata->pdev[i]); + } pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); @@ -151,6 +181,7 @@ static struct pci_driver acp3x_driver = { module_pci_driver(acp3x_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_DESCRIPTION("AMD ACP3x PCI driver"); MODULE_LICENSE("GPL v2"); -- 2.20.1

1 0

[alsa-devel] [PATCH v14 2/7] ASoC: amd: Refactoring of DAI from DMA driver
by Ravulapati Vishnu vardhan rao 27 Dec '19

27 Dec '19

ASoC: PCM DMA driver should only have dma ops. So Removed all DAI related functionality.Refactoring the PCM DMA diver code.Added new file containing only DAI ops Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/Makefile | 2 + sound/soc/amd/raven/acp3x-i2s.c | 262 +++++++++++++++++++++++++++++++ sound/soc/amd/raven/acp3x-pcm-dma.c | 302 +++++++----------------------------- sound/soc/amd/raven/acp3x.h | 43 +++++ 4 files changed, 364 insertions(+), 245 deletions(-) create mode 100644 sound/soc/amd/raven/acp3x-i2s.c diff --git a/sound/soc/amd/raven/Makefile b/sound/soc/amd/raven/Makefile index 108d1ac..62c22b6 100644 --- a/sound/soc/amd/raven/Makefile +++ b/sound/soc/amd/raven/Makefile @@ -2,5 +2,7 @@ # Raven Ridge platform Support snd-pci-acp3x-objs := pci-acp3x.o snd-acp3x-pcm-dma-objs := acp3x-pcm-dma.o +snd-acp3x-i2s-objs := acp3x-i2s.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-pci-acp3x.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-pcm-dma.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-i2s.o diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c new file mode 100644 index 0000000..cdc1c61 --- /dev/null +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -0,0 +1,262 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// AMD ALSA SoC PCM Driver +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <linux/dma-mapping.h> + +#include "acp3x.h" + +#define DRV_NAME "acp3x-i2s" + +static int acp3x_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct i2s_dev_data *adata; + int mode; + + adata = snd_soc_dai_get_drvdata(cpu_dai); + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (mode) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = false; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = true; + break; + default: + return -EINVAL; + } + return 0; +} + +static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, + u32 tx_mask, u32 rx_mask, int slots, int slot_width) +{ + struct i2s_dev_data *adata; + u32 val, reg_val, frmt_val, frm_len; + u16 slot_len; + + adata = snd_soc_dai_get_drvdata(cpu_dai); + + /* These values are as per Hardware Spec */ + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + return -EINVAL; + } + + /* Enable I2S/BT channels TDM, respective TX/RX frame lengths.*/ + + frm_len = FRM_LEN | (slots << 15) | (slot_len << 18); + if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) { + reg_val = mmACP_BTTDM_ITER; + frmt_val = mmACP_BTTDM_TXFRMT; + } else { + reg_val = mmACP_BTTDM_IRER; + frmt_val = mmACP_BTTDM_RXFRMT; + } + val = rv_readl(adata->acp3x_base + reg_val); + rv_writel(val | 0x2, adata->acp3x_base + reg_val); + rv_writel(frm_len, adata->acp3x_base + frmt_val); + adata->tdm_fmt = frm_len; + return 0; +} + +static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct i2s_stream_instance *rtd; + u32 val; + u32 reg_val; + + rtd = substream->runtime->private_data; + + /* These values are as per Hardware Spec */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + rtd->xfer_resolution = 0x0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + default: + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg_val = mmACP_BTTDM_ITER; + else + reg_val = mmACP_BTTDM_IRER; + + val = rv_readl(rtd->acp3x_base + reg_val); + val = val | (rtd->xfer_resolution << 3); + rv_writel(val, rtd->acp3x_base + reg_val); + return 0; +} + +static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct i2s_stream_instance *rtd; + u32 val, period_bytes; + int ret, reg_val; + + rtd = substream->runtime->private_data; + period_bytes = frames_to_bytes(substream->runtime, + substream->runtime->period_size); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rtd->bytescount = acp_get_byte_count(rtd, + substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_val = mmACP_BTTDM_ITER; + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_TX_INTR_WATERMARK_SIZE); + } else { + reg_val = mmACP_BTTDM_IRER; + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_RX_INTR_WATERMARK_SIZE); + } + val = rv_readl(rtd->acp3x_base + reg_val); + val = val | BIT(0); + rv_writel(val, rtd->acp3x_base + reg_val); + rv_writel(1, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg_val = mmACP_BTTDM_ITER; + else + reg_val = mmACP_BTTDM_IRER; + + val = rv_readl(rtd->acp3x_base + reg_val); + val = val & ~BIT(0); + rv_writel(val, rtd->acp3x_base + reg_val); + rv_writel(0, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static struct snd_soc_dai_ops acp3x_i2s_dai_ops = { + .hw_params = acp3x_i2s_hwparams, + .trigger = acp3x_i2s_trigger, + .set_fmt = acp3x_i2s_set_fmt, + .set_tdm_slot = acp3x_i2s_set_tdm_slot, +}; + +static const struct snd_soc_component_driver acp3x_dai_component = { + .name = "acp3x-i2s", +}; + +static struct snd_soc_dai_driver acp3x_i2s_dai = { + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp3x_i2s_dai_ops, +}; + +static int acp3x_dai_probe(struct platform_device *pdev) +{ + struct resource *res; + struct i2s_dev_data *adata; + int ret; + + adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), + GFP_KERNEL); + if (!adata) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); + return -ENOMEM; + } + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (IS_ERR(adata->acp3x_base)) + return PTR_ERR(adata->acp3x_base); + + adata->i2s_irq = res->start; + dev_set_drvdata(&pdev->dev, adata); + ret = devm_snd_soc_register_component(&pdev->dev, + &acp3x_dai_component, &acp3x_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); + return -ENODEV; + } + return 0; +} + +static int acp3x_dai_remove(struct platform_device *pdev) +{ + /* As we use devm_ memory alloc there is nothing TBD here */ + + return 0; +} + +static struct platform_driver acp3x_dai_driver = { + .probe = acp3x_dai_probe, + .remove = acp3x_dai_remove, + .driver = { + .name = "acp3x_i2s_playcap", + }, +}; + +module_platform_driver(acp3x_dai_driver); + +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP 3.x PCM Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 60709e3..3150516 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -18,24 +18,6 @@ #define DRV_NAME "acp3x-i2s-audio" -struct i2s_dev_data { - bool tdm_mode; - unsigned int i2s_irq; - u32 tdm_fmt; - void __iomem *acp3x_base; - struct snd_pcm_substream *play_stream; - struct snd_pcm_substream *capture_stream; -}; - -struct i2s_stream_instance { - u16 num_pages; - u16 channels; - u32 xfer_resolution; - u64 bytescount; - dma_addr_t dma_addr; - void __iomem *acp3x_base; -}; - static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -178,10 +160,11 @@ static int acp3x_deinit(void __iomem *acp3x_base) static irqreturn_t i2s_irq_handler(int irq, void *dev_id) { + struct i2s_dev_data *rv_i2s_data; u16 play_flag, cap_flag; u32 val; - struct i2s_dev_data *rv_i2s_data = dev_id; + rv_i2s_data = dev_id; if (!rv_i2s_data) return IRQ_NONE; @@ -278,11 +261,17 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) static int acp3x_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - struct snd_pcm_runtime *runtime = substream->runtime; - struct i2s_dev_data *adata = dev_get_drvdata(component->dev); - struct i2s_stream_instance *i2s_data = kzalloc(sizeof(struct i2s_stream_instance), - GFP_KERNEL); + struct snd_pcm_runtime *runtime; + struct snd_soc_pcm_runtime *prtd; + struct i2s_dev_data *adata; + struct i2s_stream_instance *i2s_data; + int ret; + + runtime = substream->runtime; + prtd = substream->private_data; + component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + adata = dev_get_drvdata(component->dev); + i2s_data = kzalloc(sizeof(*i2s_data), GFP_KERNEL); if (!i2s_data) return -EINVAL; @@ -312,33 +301,18 @@ static int acp3x_dma_open(struct snd_soc_component *component, return 0; } -static u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) -{ - u64 byte_count; - - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { - byte_count = rv_readl(rtd->acp3x_base + - mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH); - byte_count |= rv_readl(rtd->acp3x_base + - mmACP_BT_TX_LINEARPOSITIONCNTR_LOW); - } else { - byte_count = rv_readl(rtd->acp3x_base + - mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH); - byte_count |= rv_readl(rtd->acp3x_base + - mmACP_BT_RX_LINEARPOSITIONCNTR_LOW); - } - return byte_count; -} static int acp3x_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { + struct snd_pcm_runtime *runtime; + struct i2s_stream_instance *rtd; int status; u64 size; - struct snd_pcm_runtime *runtime = substream->runtime; - struct i2s_stream_instance *rtd = runtime->private_data; + runtime = substream->runtime; + rtd = substream->private_data; if (!rtd) return -EINVAL; @@ -362,12 +336,12 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - u32 pos = 0; - u32 buffersize = 0; - u64 bytescount = 0; - struct i2s_stream_instance *rtd = - substream->runtime->private_data; + struct i2s_stream_instance *rtd; + u32 pos; + u32 buffersize; + u64 bytescount; + rtd = substream->runtime->private_data; buffersize = frames_to_bytes(substream->runtime, substream->runtime->buffer_size); bytescount = acp_get_byte_count(rtd, substream->stream); @@ -380,7 +354,10 @@ static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_soc_component *component, static int acp3x_dma_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - struct device *parent = component->dev->parent; + struct device *parent; + + component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + parent = component->dev->parent; snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, parent, MIN_BUFFER, MAX_BUFFER); return 0; @@ -402,8 +379,12 @@ static int acp3x_dma_mmap(struct snd_soc_component *component, static int acp3x_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct i2s_stream_instance *rtd = substream->runtime->private_data; - struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + struct snd_soc_pcm_runtime *prtd; + struct i2s_dev_data *adata; + + prtd = substream->private_data; + component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); + adata = dev_get_drvdata(component->dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) adata->play_stream = NULL; @@ -415,186 +396,9 @@ static int acp3x_dma_close(struct snd_soc_component *component, */ if (!adata->play_stream && !adata->capture_stream) rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - kfree(rtd); return 0; } -static int acp3x_dai_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) -{ - - struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - adata->tdm_mode = false; - break; - case SND_SOC_DAIFMT_DSP_A: - adata->tdm_mode = true; - break; - default: - return -EINVAL; - } - - return 0; -} - -static int acp3x_dai_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, - u32 rx_mask, int slots, int slot_width) -{ - u32 val = 0; - u16 slot_len; - - struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); - - switch (slot_width) { - case SLOT_WIDTH_8: - slot_len = 8; - break; - case SLOT_WIDTH_16: - slot_len = 16; - break; - case SLOT_WIDTH_24: - slot_len = 24; - break; - case SLOT_WIDTH_32: - slot_len = 0; - break; - default: - return -EINVAL; - } - - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); - rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_ITER); - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); - rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_IRER); - - val = (FRM_LEN | (slots << 15) | (slot_len << 18)); - rv_writel(val, adata->acp3x_base + mmACP_BTTDM_TXFRMT); - rv_writel(val, adata->acp3x_base + mmACP_BTTDM_RXFRMT); - - adata->tdm_fmt = val; - return 0; -} - -static int acp3x_dai_i2s_hwparams(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - u32 val = 0; - struct i2s_stream_instance *rtd = substream->runtime->private_data; - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: - rtd->xfer_resolution = 0x0; - break; - case SNDRV_PCM_FORMAT_S16_LE: - rtd->xfer_resolution = 0x02; - break; - case SNDRV_PCM_FORMAT_S24_LE: - rtd->xfer_resolution = 0x04; - break; - case SNDRV_PCM_FORMAT_S32_LE: - rtd->xfer_resolution = 0x05; - break; - default: - return -EINVAL; - } - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); - val = val | (rtd->xfer_resolution << 3); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); - else - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); - - return 0; -} - -static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - int ret = 0; - struct i2s_stream_instance *rtd = substream->runtime->private_data; - u32 val, period_bytes; - - period_bytes = frames_to_bytes(substream->runtime, - substream->runtime->period_size); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - rtd->bytescount = acp_get_byte_count(rtd, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - rv_writel(period_bytes, rtd->acp3x_base + - mmACP_BT_TX_INTR_WATERMARK_SIZE); - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); - val = val | BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); - } else { - rv_writel(period_bytes, rtd->acp3x_base + - mmACP_BT_RX_INTR_WATERMARK_SIZE); - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); - val = val | BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); - } - rv_writel(1, rtd->acp3x_base + mmACP_BTTDM_IER); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); - val = val & ~BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); - } else { - val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); - val = val & ~BIT(0); - rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); - } - rv_writel(0, rtd->acp3x_base + mmACP_BTTDM_IER); - break; - default: - ret = -EINVAL; - break; - } - - return ret; -} - -static struct snd_soc_dai_ops acp3x_dai_i2s_ops = { - .hw_params = acp3x_dai_i2s_hwparams, - .trigger = acp3x_dai_i2s_trigger, - .set_fmt = acp3x_dai_i2s_set_fmt, - .set_tdm_slot = acp3x_dai_set_tdm_slot, -}; - -static struct snd_soc_dai_driver acp3x_i2s_dai_driver = { - .playback = { - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .channels_min = 2, - .channels_max = 8, - - .rate_min = 8000, - .rate_max = 96000, - }, - .capture = { - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .channels_min = 2, - .channels_max = 2, - .rate_min = 8000, - .rate_max = 48000, - }, - .ops = &acp3x_dai_i2s_ops, -}; - static const struct snd_soc_component_driver acp3x_i2s_component = { .name = DRV_NAME, .open = acp3x_dma_open, @@ -609,10 +413,10 @@ static const struct snd_soc_component_driver acp3x_i2s_component = { static int acp3x_audio_probe(struct platform_device *pdev) { - int status; struct resource *res; struct i2s_dev_data *adata; unsigned int irqflags; + int status, ret; if (!pdev->dev.platform_data) { dev_err(&pdev->dev, "platform_data not retrieved\n"); @@ -622,7 +426,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); return -ENODEV; } @@ -632,6 +436,8 @@ static int acp3x_audio_probe(struct platform_device *pdev) adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!adata->acp3x_base) + return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (!res) { @@ -640,52 +446,54 @@ static int acp3x_audio_probe(struct platform_device *pdev) } adata->i2s_irq = res->start; - adata->play_stream = NULL; - adata->capture_stream = NULL; dev_set_drvdata(&pdev->dev, adata); /* Initialize ACP */ status = acp3x_init(adata->acp3x_base); if (status) return -ENODEV; + status = devm_snd_soc_register_component(&pdev->dev, &acp3x_i2s_component, - &acp3x_i2s_dai_driver, 1); + NULL, 0); if (status) { - dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); + dev_err(&pdev->dev, "Fail to register acp i2s component\n"); + ret = -ENODEV; goto dev_err; } status = devm_request_irq(&pdev->dev, adata->i2s_irq, i2s_irq_handler, irqflags, "ACP3x_I2S_IRQ", adata); if (status) { dev_err(&pdev->dev, "ACP3x I2S IRQ request failed\n"); + ret = -ENODEV; goto dev_err; } - pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_set_autosuspend_delay(&pdev->dev, 5000); pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_enable(&pdev->dev); return 0; + dev_err: status = acp3x_deinit(adata->acp3x_base); if (status) dev_err(&pdev->dev, "ACP de-init failed\n"); else - dev_info(&pdev->dev, "ACP de-initialized\n"); - /*ignore device status and return driver probe error*/ - return -ENODEV; + dev_dbg(&pdev->dev, "ACP de-initialized\n"); + return ret; } static int acp3x_audio_remove(struct platform_device *pdev) { + struct i2s_dev_data *adata; int ret; - struct i2s_dev_data *adata = dev_get_drvdata(&pdev->dev); + adata = dev_get_drvdata(&pdev->dev); ret = acp3x_deinit(adata->acp3x_base); if (ret) dev_err(&pdev->dev, "ACP de-init failed\n"); else - dev_info(&pdev->dev, "ACP de-initialized\n"); + dev_dbg(&pdev->dev, "ACP de-initialized\n"); pm_runtime_disable(&pdev->dev); return 0; @@ -693,10 +501,11 @@ static int acp3x_audio_remove(struct platform_device *pdev) static int acp3x_resume(struct device *dev) { + struct i2s_dev_data *adata; int status; u32 val; - struct i2s_dev_data *adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_init(adata->acp3x_base); if (status) return -ENODEV; @@ -738,14 +547,15 @@ static int acp3x_resume(struct device *dev) static int acp3x_pcm_runtime_suspend(struct device *dev) { + struct i2s_dev_data *adata; int status; - struct i2s_dev_data *adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_deinit(adata->acp3x_base); if (status) dev_err(dev, "ACP de-init failed\n"); else - dev_info(dev, "ACP de-initialized\n"); + dev_dbg(dev, "ACP de-initialized\n"); rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); @@ -754,8 +564,9 @@ static int acp3x_pcm_runtime_suspend(struct device *dev) static int acp3x_pcm_runtime_resume(struct device *dev) { + struct i2s_dev_data *adata; int status; - struct i2s_dev_data *adata = dev_get_drvdata(dev); + adata = dev_get_drvdata(dev); status = acp3x_init(adata->acp3x_base); if (status) @@ -774,13 +585,14 @@ static struct platform_driver acp3x_dma_driver = { .probe = acp3x_audio_probe, .remove = acp3x_audio_remove, .driver = { - .name = "acp3x_rv_i2s", + .name = "acp3x_rv_i2s_dma", .pm = &acp3x_pm_ops, }, }; module_platform_driver(acp3x_dma_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); MODULE_DESCRIPTION("AMD ACP 3.x PCM Driver"); diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 2f15fe1..a6c6d63 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -51,6 +51,30 @@ #define SLOT_WIDTH_24 0x18 #define SLOT_WIDTH_32 0x20 +struct acp3x_platform_info { + u16 play_i2s_instance; + u16 cap_i2s_instance; + u16 capture_channel; +}; + +struct i2s_dev_data { + bool tdm_mode; + unsigned int i2s_irq; + u32 tdm_fmt; + u32 substream_type; + void __iomem *acp3x_base; + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; +}; + +struct i2s_stream_instance { + u16 num_pages; + u16 channels; + u32 xfer_resolution; + u64 bytescount; + dma_addr_t dma_addr; + void __iomem *acp3x_base; +}; static inline u32 rv_readl(void __iomem *base_addr) { @@ -61,3 +85,22 @@ static inline void rv_writel(u32 val, void __iomem *base_addr) { writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); } + +static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, + int direction) +{ + u64 byte_count; + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_LOW); + } else { + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_LOW); + } + return byte_count; +} -- 2.7.4

2 1

[alsa-devel] [Regression drm-tip] Internal audio device missing
by Paul Menzel 26 Dec '19

26 Dec '19

Dear Linux folks, With $ git describe --tags drm-tip/drm-tip v5.5-rc3-1481-ga20d8cd6901a the internal audio device is not available, and just a dummy device. Running `alsa-info.sh` [1], the messages below are shown with the problematic Linux kernel. alsactl: get_controls:567: snd_ctl_open error: Sound protocol is not compatible cat: /tmp/alsa-info.ateDlDjrZX/alsactl.tmp: No such file or directory Please find the output of `alsa-info.sh` with Linux 5.5-rc3 and drm-tip attached. Kind regards, Paul [1]: https://www.alsa-project.org/alsa-info.sh

2 3

[alsa-devel] broadwell-rt286: add support for hardware volume, conformance fixes
by GitHub pull_request - edited 26 Dec '19

26 Dec '19

alsa-project/alsa-ucm-conf pull request #9 was edited from dpward: This commit by @perexg fixes a regression on broadwell-rt286 which causes PulseAudio to abort on startup with: `E: [pulseaudio] channelmap.c: Assertion 'pa_channels_valid(channels)' failed at pulse/channelmap.c:401, function pa_channel_map_init_extend(). Aborting.` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/9 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/9.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] broadwell-rt286: add support for hardware volume, conformance fixes
by GitHub pull_request - opened 26 Dec '19

26 Dec '19

alsa-project/alsa-ucm-conf pull request #9 was opened from dpward: This fixes a regression on broadwell-rt286. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/9 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/9.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [PATCH] ASoC: SOF: Intel: Fix build error
by YueHaibing 26 Dec '19

26 Dec '19

If SND_INTEL_DSP_CONFIG is m and SND_SOC_SOF_PCI is y, building fails: sound/soc/sof/sof-pci-dev.o: In function `sof_pci_probe': sof-pci-dev.c:(.text+0xb4): undefined reference to `snd_intel_dsp_driver_probe' Select SND_INTEL_DSP_CONFIG to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 71a0fc0..e0b04b5 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -14,6 +14,7 @@ config SND_SOC_SOF_PCI depends on PCI select SND_SOC_SOF select SND_SOC_ACPI if ACPI + select SND_INTEL_DSP_CONFIG help This adds support for PCI enumeration. This option is required to enable Intel Skylake+ devices -- 2.7.4

2 2

[alsa-devel] [PATCH v4] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 26 Dec '19

26 Dec '19

From: Jeff Chang <jeff_chang(a)richtek.com> The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper Signed-off-by: Jeff Chang <jeff_chang(a)richtek.com> --- Documentation/devicetree/bindings/sound/mt6660.txt | 53 ++ sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 628 +++++++++++++++++++++ sound/soc/codecs/mt6660.h | 75 +++ 5 files changed, 769 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6660.txt create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h changelogs between v4 & v3 - remove unnecessary kcontrols. - modify copy right header. - use dev_dbg instead of dev_info. - add necessary debug message. - add DT binding documentation. - add space before } at every table. changelogs between v3 & v2 - modify MT6660 Kconfig, remove unnecessary selection. - remove my own debug io interface. use standard regmap for debugging. - remove regmap volatile ops, we do not use cache. - remove component io read/write function, use snd_soc_component_init_regmap. - remove init setting write code. Using parsing dts to set them. - remove unnecessary pr_info log message. - remove mt6660_component_put_volsw. Using snd_soc_get_volsw. diff --git a/Documentation/devicetree/bindings/sound/mt6660.txt b/Documentation/devicetree/bindings/sound/mt6660.txt new file mode 100644 index 0000000..2a1736b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt6660.txt @@ -0,0 +1,53 @@ +MT6660 MediaTek Speaker Amplifier + +This device supports I2C mode only. + +Required properties: + + - compatible : "mediatek,mt6660" + + - reg : The I2C slave address + +Optional properties: + + - rt,init_setting_num : The initial register setting element number. + + - rt,init_setting_addr : the addreses array for INIT Setting table. + + - rt,init_setting_mask : the mask array for INIT Setting table. + + - rt,init_setting_val : the value array for INIT Setting table. + +Example: + + mt6660@34 { + status = "ok"; + compatible = "mediatek,mt6660"; + reg = <0x34>; + rt,init_setting_num = <26>; + rt,init_setting_addr = + <0x20 0x30 0x50 0xB1 + 0xD3 0xE0 0x98 0xB9 + 0xB7 0xB6 0x6B 0x07 + 0xBB 0x69 0xBD 0x70 + 0x7C 0x46 0x1A 0x1B + 0x51 0xA2 0x33 0x4C + 0x15 0x68>; + rt,init_setting_mask = + <0x80 0x01 0x1c 0x0c + 0x03 0x01 0x44 0xff + 0x7777 0x07 0xe0 0xff + 0xff 0xff 0xffff 0xff + 0xff 0xff 0xffffffff 0xffffffff + 0xff 0xff 0xffff 0xffff + 0x1800 0x1f>; + rt,init_setting_val = + <0x00 0x00 0x04 0x00 + 0x03 0x00 0x04 0x82 + 0x7273 0x03 0x20 0x70 + 0x20 0x40 0x17f8 0x15 + 0x00 0x1d 0x7fdb7ffe 0x7fdb7ffe + 0x58 0xce 0x7fff 0x0116 + 0x0800 0x07>; + }; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..f135fbb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,16 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..b8fc53b --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,628 @@ +// SPDX-License-Identifier: GPL-2.0 // + +// Copyright (c) 2019 MediaTek Inc. + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2 }, + { MT6660_REG_HCLIP_CTRL, 2 }, + { MT6660_REG_DA_GAIN, 2 }, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0, ret = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + ret = i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); + if (ret < 0) + return ret; + return 0; +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + return 0; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, +}; + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + dev_dbg(component->dev, "Amp on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_dbg(component->dev, "Amp off\n"); + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback" }, + { "PGA", NULL, "DAC" }, + { "ClassD", NULL, "PGA" }, + { "OUTP", NULL, "ClassD" }, + { "OUTN", NULL, "ClassD" }, + { "VI ADC", NULL, "ClassD" }, + { "aif_capture", NULL, "VI ADC" }, +}; + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, vol_ctl_tlv), + SOC_SINGLE("Hard Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0), + SOC_SINGLE("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0), + SOC_SINGLE("Boost Mode", MT6660_REG_BST_CTRL, 0, 3, 0), + SOC_SINGLE("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0), + SOC_SINGLE("DC Protect Switch", MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0), + SOC_SINGLE("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0), + SOC_SINGLE("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0), + SOC_SINGLE_EXT("T0 SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip Rev", MT6660_REG_DEVID, 8, 15, 0, + mt6660_component_get_volsw, NULL), +}; + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static int mt6660_apply_plat_data(struct mt6660_chip *chip, + struct snd_soc_component *component) +{ + size_t i = 0; + int num = chip->plat_data.init_setting_num; + int ret = 0; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + + for (i = 0; i < num; i++) { + ret = snd_soc_component_update_bits(component, + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + if (ret < 0) + return ret; + } + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + return 0; +} + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_init_regmap(component, chip->regmap); + + ret = mt6660_apply_plat_data(chip, component); + if (ret < 0) + dev_err(chip->dev, "mt6660 apply plat data failed\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + dev_dbg(component->dev, "%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) { + dev_err(chip->dev, "%s id(%x) not match\n", __func__, ret); + return -ENODEV; + } + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_parse_dt(struct mt6660_chip *chip, struct device *dev) +{ + struct device_node *np = dev->of_node; + u32 val; + size_t i = 0; + + if (!np) { + dev_err(dev, "no device node\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "rt,init_setting_num", &val)) { + dev_err(dev, "no init setting\n"); + chip->plat_data.init_setting_num = 0; + } else { + chip->plat_data.init_setting_num = val; + } + + chip->plat_data.init_setting_addr = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + chip->plat_data.init_setting_mask = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + chip->plat_data.init_setting_val = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + + if (of_property_read_u32_array(np, "rt,init_setting_addr", + chip->plat_data.init_setting_addr, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting addr\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_mask", + chip->plat_data.init_setting_mask, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting addr\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_val", + chip->plat_data.init_setting_val, + chip->plat_data.init_setting_num)) { + dev_err(dev, "no init setting addr\n"); + } + + dev_dbg(dev, "%s, init stting table, num = %d\n", __func__, + chip->plat_data.init_setting_num); + for (i = 0; i < chip->plat_data.init_setting_num; i++) { + dev_dbg(dev, "0x%02x, 0x%08x, 0x%08x\n", + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + } + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + dev_dbg(&client->dev, "%s\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + ret = mt6660_parse_dt(chip, &client->dev); + if (ret < 0) { + dev_err(&client->dev, "parsing dts failed\n"); + return ret; + } + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..6c40b40 --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,75 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +struct mt6660_platform_data { + u8 init_setting_num; + u32 *init_setting_addr; + u32 *init_setting_mask; + u32 *init_setting_val; +}; + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mt6660_platform_data plat_data; + struct mutex io_lock; + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

1 0

[alsa-devel] Applied "ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a6b748713205242d1483a78662bf147b1aa7d90b Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 19 Dec 2019 10:31:43 +0000 Subject: [PATCH] ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec This patch adds bindings for wcd9340/wcd9341 audio codec which can support both SLIMbus and I2S/I2C interface. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20191219103153.14875-2-srinivas.kandagatla@linaro… Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/qcom,wcd934x.yaml | 175 ++++++++++++++++++ 1 file changed, 175 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml new file mode 100644 index 000000000000..38eaf0c028f9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -0,0 +1,175 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wcd934x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bindings for Qualcomm WCD9340/WCD9341 Audio Codec + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +description: | + Qualcomm WCD9340/WCD9341 Codec is a standalone Hi-Fi audio codec IC. + It has in-built Soundwire controller, pin controller, interrupt mux and + supports both I2S/I2C and SLIMbus audio interfaces. + +properties: + compatible: + const: slim217,250 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + reset-gpios: + description: GPIO spec for reset line to use + maxItems: 1 + + slim-ifc-dev: true + + clocks: + maxItems: 1 + + clock-names: + const: extclk + + vdd-buck-supply: + description: A reference to the 1.8V buck supply + + vdd-buck-sido-supply: + description: A reference to the 1.8V SIDO buck supply + + vdd-rx-supply: + description: A reference to the 1.8V rx supply + + vdd-tx-supply: + description: A reference to the 1.8V tx supply + + vdd-vbat-supply: + description: A reference to the vbat supply + + vdd-io-supply: + description: A reference to the 1.8V I/O supply + + vdd-micbias-supply: + description: A reference to the micbias supply + + qcom,micbias1-microvolt: + description: micbias1 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias2-microvolt: + description: micbias2 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias3-microvolt: + description: micbias3 voltage + minimum: 1800000 + maximum: 2850000 + + qcom,micbias4-microvolt: + description: micbias4 voltage + minimum: 1800000 + maximum: 2850000 + + clock-output-names: + const: mclk + + clock-frequency: + description: Clock frequency of output clk in Hz + + interrupt-controller: true + + '#interrupt-cells': + const: 1 + + '#clock-cells': + const: 0 + + '#sound-dai-cells': + const: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + gpio@42: + type: object + allOf: + - $ref: ../gpio/qcom,wcd934x-gpio.yaml# + +patternProperties: + "^.*@[0-9a-f]+$": + type: object + description: | + WCD934x subnode for each slave devices. Bindings of each subnodes + depends on the specific driver providing the functionality and + documented in their respective bindings. + + properties: + reg: + maxItems: 1 + + required: + - reg + +required: + - compatible + - reg + - reset-gpios + - slim-ifc-dev + - interrupts + - interrupt-controller + - clock-frequency + - clock-output-names + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#interrupt-cells" + - "#clock-cells" + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + +examples: + - | + codec@1,0{ + compatible = "slim217,250"; + reg = <1 0>; + reset-gpios = <&tlmm 64 0>; + slim-ifc-dev = <&wcd9340_ifd>; + #sound-dai-cells = <1>; + interrupt-parent = <&tlmm>; + interrupts = <54 4>; + interrupt-controller; + #interrupt-cells = <1>; + #clock-cells = <0>; + clock-frequency = <9600000>; + clock-output-names = "mclk"; + qcom,micbias1-microvolt = <1800000>; + qcom,micbias2-microvolt = <1800000>; + qcom,micbias3-microvolt = <1800000>; + qcom,micbias4-microvolt = <1800000>; + clock-names = "extclk"; + clocks = <&rpmhcc 2>; + + #address-cells = <1>; + #size-cells = <1>; + + gpio@42 { + compatible = "qcom,wcd9340-gpio"; + reg = <0x42 0x2>; + gpio-controller; + #gpio-cells = <2>; + }; + }; + +... -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: wcd934x: add basic controls" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: wcd934x: add basic controls has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1cde8b82233275a72db34a8c1efa4cf6cc8c894f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 19 Dec 2019 10:31:46 +0000 Subject: [PATCH] ASoC: wcd934x: add basic controls This patch adds basic controls found in wcd934x codec. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20191219103153.14875-5-srinivas.kandagatla@linaro… Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 506 +++++++++++++++++++++++++++++++++++++ 1 file changed, 506 insertions(+) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 825cf057e8b7..27a864f2ff77 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -120,6 +120,21 @@ #define WCD934X_DEF_MICBIAS_MV 1800 #define WCD934X_MAX_MICBIAS_MV 2850 +#define WCD_IIR_FILTER_SIZE (sizeof(u32) * BAND_MAX) + +#define WCD_IIR_FILTER_CTL(xname, iidx, bidx) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = wcd934x_iir_filter_info, \ + .get = wcd934x_get_iir_band_audio_mixer, \ + .put = wcd934x_put_iir_band_audio_mixer, \ + .private_value = (unsigned long)&(struct wcd_iir_filter_ctl) { \ + .iir_idx = iidx, \ + .band_idx = bidx, \ + .bytes_ext = {.max = WCD_IIR_FILTER_SIZE, }, \ + } \ +} + enum { SIDO_SOURCE_INTERNAL, SIDO_SOURCE_RCO_BG, @@ -217,6 +232,35 @@ static const struct wcd934x_slim_ch wcd934x_rx_chs[WCD934X_RX_MAX] = { WCD934X_SLIM_RX_CH(12), }; +/* Codec supports 2 IIR filters */ +enum { + IIR0 = 0, + IIR1, + IIR_MAX, +}; + +/* Each IIR has 5 Filter Stages */ +enum { + BAND1 = 0, + BAND2, + BAND3, + BAND4, + BAND5, + BAND_MAX, +}; + +enum { + COMPANDER_1, /* HPH_L */ + COMPANDER_2, /* HPH_R */ + COMPANDER_3, /* LO1_DIFF */ + COMPANDER_4, /* LO2_DIFF */ + COMPANDER_5, /* LO3_SE - not used in Tavil */ + COMPANDER_6, /* LO4_SE - not used in Tavil */ + COMPANDER_7, /* SWR SPK CH1 */ + COMPANDER_8, /* SWR SPK CH2 */ + COMPANDER_MAX, +}; + enum { AIF1_PB = 0, AIF1_CAP, @@ -335,12 +379,112 @@ struct wcd934x_codec { int dmic_2_3_clk_cnt; int dmic_4_5_clk_cnt; int dmic_sample_rate; + int comp_enabled[COMPANDER_MAX]; int sysclk_users; struct mutex sysclk_mutex; }; #define to_wcd934x_codec(_hw) container_of(_hw, struct wcd934x_codec, hw) +struct wcd_iir_filter_ctl { + unsigned int iir_idx; + unsigned int band_idx; + struct soc_bytes_ext bytes_ext; +}; + +static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); +static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); +static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); + +/* Cutoff frequency for high pass filter */ +static const char * const cf_text[] = { + "CF_NEG_3DB_4HZ", "CF_NEG_3DB_75HZ", "CF_NEG_3DB_150HZ" +}; + +static const char * const rx_cf_text[] = { + "CF_NEG_3DB_4HZ", "CF_NEG_3DB_75HZ", "CF_NEG_3DB_150HZ", + "CF_NEG_3DB_0P48HZ" +}; + +static const char * const rx_hph_mode_mux_text[] = { + "Class H Invalid", "Class-H Hi-Fi", "Class-H Low Power", "Class-AB", + "Class-H Hi-Fi Low Power" +}; + +static const struct soc_enum cf_dec0_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX0_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX1_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec2_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX2_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec3_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX3_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec4_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX4_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec5_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX5_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec6_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX6_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec7_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX7_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_dec8_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX8_TX_PATH_CFG0, 5, 3, cf_text); + +static const struct soc_enum cf_int0_1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX0_RX_PATH_CFG2, 0, 4, rx_cf_text); + +static SOC_ENUM_SINGLE_DECL(cf_int0_2_enum, WCD934X_CDC_RX0_RX_PATH_MIX_CFG, 2, + rx_cf_text); + +static const struct soc_enum cf_int1_1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX1_RX_PATH_CFG2, 0, 4, rx_cf_text); + +static SOC_ENUM_SINGLE_DECL(cf_int1_2_enum, WCD934X_CDC_RX1_RX_PATH_MIX_CFG, 2, + rx_cf_text); + +static const struct soc_enum cf_int2_1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX2_RX_PATH_CFG2, 0, 4, rx_cf_text); + +static SOC_ENUM_SINGLE_DECL(cf_int2_2_enum, WCD934X_CDC_RX2_RX_PATH_MIX_CFG, 2, + rx_cf_text); + +static const struct soc_enum cf_int3_1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX3_RX_PATH_CFG2, 0, 4, rx_cf_text); + +static SOC_ENUM_SINGLE_DECL(cf_int3_2_enum, WCD934X_CDC_RX3_RX_PATH_MIX_CFG, 2, + rx_cf_text); + +static const struct soc_enum cf_int4_1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX4_RX_PATH_CFG2, 0, 4, rx_cf_text); + +static SOC_ENUM_SINGLE_DECL(cf_int4_2_enum, WCD934X_CDC_RX4_RX_PATH_MIX_CFG, 2, + rx_cf_text); + +static const struct soc_enum cf_int7_1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX7_RX_PATH_CFG2, 0, 4, rx_cf_text); + +static SOC_ENUM_SINGLE_DECL(cf_int7_2_enum, WCD934X_CDC_RX7_RX_PATH_MIX_CFG, 2, + rx_cf_text); + +static const struct soc_enum cf_int8_1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX8_RX_PATH_CFG2, 0, 4, rx_cf_text); + +static SOC_ENUM_SINGLE_DECL(cf_int8_2_enum, WCD934X_CDC_RX8_RX_PATH_MIX_CFG, 2, + rx_cf_text); + +static const struct soc_enum rx_hph_mode_mux_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rx_hph_mode_mux_text), + rx_hph_mode_mux_text); + static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) { @@ -1479,10 +1623,372 @@ static int wcd934x_comp_set_sysclk(struct snd_soc_component *comp, return clk_set_rate(wcd->extclk, freq); } +static uint32_t get_iir_band_coeff(struct snd_soc_component *component, + int iir_idx, int band_idx, int coeff_idx) +{ + u32 value = 0; + int reg, b2_reg; + + /* Address does not automatically update if reading */ + reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx; + b2_reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx; + + snd_soc_component_write(component, reg, + ((band_idx * BAND_MAX + coeff_idx) * + sizeof(uint32_t)) & 0x7F); + + value |= snd_soc_component_read32(component, b2_reg); + snd_soc_component_write(component, reg, + ((band_idx * BAND_MAX + coeff_idx) + * sizeof(uint32_t) + 1) & 0x7F); + + value |= (snd_soc_component_read32(component, b2_reg) << 8); + snd_soc_component_write(component, reg, + ((band_idx * BAND_MAX + coeff_idx) + * sizeof(uint32_t) + 2) & 0x7F); + + value |= (snd_soc_component_read32(component, b2_reg) << 16); + snd_soc_component_write(component, reg, + ((band_idx * BAND_MAX + coeff_idx) + * sizeof(uint32_t) + 3) & 0x7F); + + /* Mask bits top 2 bits since they are reserved */ + value |= (snd_soc_component_read32(component, b2_reg) << 24); + return value; +} + +static void set_iir_band_coeff(struct snd_soc_component *component, + int iir_idx, int band_idx, uint32_t value) +{ + int reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx; + + snd_soc_component_write(component, reg, (value & 0xFF)); + snd_soc_component_write(component, reg, (value >> 8) & 0xFF); + snd_soc_component_write(component, reg, (value >> 16) & 0xFF); + /* Mask top 2 bits, 7-8 are reserved */ + snd_soc_component_write(component, reg, (value >> 24) & 0x3F); +} + +static int wcd934x_put_iir_band_audio_mixer( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct wcd_iir_filter_ctl *ctl = + (struct wcd_iir_filter_ctl *)kcontrol->private_value; + struct soc_bytes_ext *params = &ctl->bytes_ext; + int iir_idx = ctl->iir_idx; + int band_idx = ctl->band_idx; + u32 coeff[BAND_MAX]; + int reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx; + + memcpy(&coeff[0], ucontrol->value.bytes.data, params->max); + + /* Mask top bit it is reserved */ + /* Updates addr automatically for each B2 write */ + snd_soc_component_write(component, reg, (band_idx * BAND_MAX * + sizeof(uint32_t)) & 0x7F); + + set_iir_band_coeff(component, iir_idx, band_idx, coeff[0]); + set_iir_band_coeff(component, iir_idx, band_idx, coeff[1]); + set_iir_band_coeff(component, iir_idx, band_idx, coeff[2]); + set_iir_band_coeff(component, iir_idx, band_idx, coeff[3]); + set_iir_band_coeff(component, iir_idx, band_idx, coeff[4]); + + return 0; +} + +static int wcd934x_get_iir_band_audio_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct wcd_iir_filter_ctl *ctl = + (struct wcd_iir_filter_ctl *)kcontrol->private_value; + struct soc_bytes_ext *params = &ctl->bytes_ext; + int iir_idx = ctl->iir_idx; + int band_idx = ctl->band_idx; + u32 coeff[BAND_MAX]; + + coeff[0] = get_iir_band_coeff(component, iir_idx, band_idx, 0); + coeff[1] = get_iir_band_coeff(component, iir_idx, band_idx, 1); + coeff[2] = get_iir_band_coeff(component, iir_idx, band_idx, 2); + coeff[3] = get_iir_band_coeff(component, iir_idx, band_idx, 3); + coeff[4] = get_iir_band_coeff(component, iir_idx, band_idx, 4); + + memcpy(ucontrol->value.bytes.data, &coeff[0], params->max); + + return 0; +} + +static int wcd934x_iir_filter_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct wcd_iir_filter_ctl *ctl = + (struct wcd_iir_filter_ctl *)kcontrol->private_value; + struct soc_bytes_ext *params = &ctl->bytes_ext; + + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; + ucontrol->count = params->max; + + return 0; +} + +static int wcd934x_compander_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kc); + int comp = ((struct soc_mixer_control *)kc->private_value)->shift; + struct wcd934x_codec *wcd = dev_get_drvdata(component->dev); + + ucontrol->value.integer.value[0] = wcd->comp_enabled[comp]; + + return 0; +} + +static int wcd934x_compander_set(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kc); + struct wcd934x_codec *wcd = dev_get_drvdata(component->dev); + int comp = ((struct soc_mixer_control *)kc->private_value)->shift; + int value = ucontrol->value.integer.value[0]; + int sel; + + wcd->comp_enabled[comp] = value; + sel = value ? WCD934X_HPH_GAIN_SRC_SEL_COMPANDER : + WCD934X_HPH_GAIN_SRC_SEL_REGISTER; + + /* Any specific register configuration for compander */ + switch (comp) { + case COMPANDER_1: + /* Set Gain Source Select based on compander enable/disable */ + snd_soc_component_update_bits(component, WCD934X_HPH_L_EN, + WCD934X_HPH_GAIN_SRC_SEL_MASK, + sel); + break; + case COMPANDER_2: + snd_soc_component_update_bits(component, WCD934X_HPH_R_EN, + WCD934X_HPH_GAIN_SRC_SEL_MASK, + sel); + break; + case COMPANDER_3: + case COMPANDER_4: + case COMPANDER_7: + case COMPANDER_8: + break; + default: + break; + }; + + return 0; +} + +static int wcd934x_rx_hph_mode_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kc); + struct wcd934x_codec *wcd = dev_get_drvdata(component->dev); + + ucontrol->value.enumerated.item[0] = wcd->hph_mode; + + return 0; +} + +static int wcd934x_rx_hph_mode_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kc); + struct wcd934x_codec *wcd = dev_get_drvdata(component->dev); + u32 mode_val; + + mode_val = ucontrol->value.enumerated.item[0]; + + if (mode_val == 0) { + dev_err(wcd->dev, "Invalid HPH Mode, default to ClSH HiFi\n"); + mode_val = CLS_H_LOHIFI; + } + wcd->hph_mode = mode_val; + + return 0; +} + +static const struct snd_kcontrol_new wcd934x_snd_controls[] = { + /* Gain Controls */ + SOC_SINGLE_TLV("EAR PA Volume", WCD934X_ANA_EAR, 4, 4, 1, ear_pa_gain), + SOC_SINGLE_TLV("HPHL Volume", WCD934X_HPH_L_EN, 0, 24, 1, line_gain), + SOC_SINGLE_TLV("HPHR Volume", WCD934X_HPH_R_EN, 0, 24, 1, line_gain), + SOC_SINGLE_TLV("LINEOUT1 Volume", WCD934X_DIFF_LO_LO1_COMPANDER, + 3, 16, 1, line_gain), + SOC_SINGLE_TLV("LINEOUT2 Volume", WCD934X_DIFF_LO_LO2_COMPANDER, + 3, 16, 1, line_gain), + + SOC_SINGLE_TLV("ADC1 Volume", WCD934X_ANA_AMIC1, 0, 20, 0, analog_gain), + SOC_SINGLE_TLV("ADC2 Volume", WCD934X_ANA_AMIC2, 0, 20, 0, analog_gain), + SOC_SINGLE_TLV("ADC3 Volume", WCD934X_ANA_AMIC3, 0, 20, 0, analog_gain), + SOC_SINGLE_TLV("ADC4 Volume", WCD934X_ANA_AMIC4, 0, 20, 0, analog_gain), + + SOC_SINGLE_S8_TLV("RX0 Digital Volume", WCD934X_CDC_RX0_RX_VOL_CTL, + -84, 40, digital_gain), /* -84dB min - 40dB max */ + SOC_SINGLE_S8_TLV("RX1 Digital Volume", WCD934X_CDC_RX1_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX2 Digital Volume", WCD934X_CDC_RX2_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX3 Digital Volume", WCD934X_CDC_RX3_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX4 Digital Volume", WCD934X_CDC_RX4_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX7 Digital Volume", WCD934X_CDC_RX7_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX8 Digital Volume", WCD934X_CDC_RX8_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX0 Mix Digital Volume", + WCD934X_CDC_RX0_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX1 Mix Digital Volume", + WCD934X_CDC_RX1_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX2 Mix Digital Volume", + WCD934X_CDC_RX2_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX3 Mix Digital Volume", + WCD934X_CDC_RX3_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX4 Mix Digital Volume", + WCD934X_CDC_RX4_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX7 Mix Digital Volume", + WCD934X_CDC_RX7_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX8 Mix Digital Volume", + WCD934X_CDC_RX8_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + + SOC_SINGLE_S8_TLV("DEC0 Volume", WCD934X_CDC_TX0_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC1 Volume", WCD934X_CDC_TX1_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC2 Volume", WCD934X_CDC_TX2_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC3 Volume", WCD934X_CDC_TX3_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC4 Volume", WCD934X_CDC_TX4_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC5 Volume", WCD934X_CDC_TX5_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC6 Volume", WCD934X_CDC_TX6_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC7 Volume", WCD934X_CDC_TX7_TX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("DEC8 Volume", WCD934X_CDC_TX8_TX_VOL_CTL, + -84, 40, digital_gain), + + SOC_SINGLE_S8_TLV("IIR0 INP0 Volume", + WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B1_CTL, -84, 40, + digital_gain), + SOC_SINGLE_S8_TLV("IIR0 INP1 Volume", + WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B2_CTL, -84, 40, + digital_gain), + SOC_SINGLE_S8_TLV("IIR0 INP2 Volume", + WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B3_CTL, -84, 40, + digital_gain), + SOC_SINGLE_S8_TLV("IIR0 INP3 Volume", + WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B4_CTL, -84, 40, + digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP0 Volume", + WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B1_CTL, -84, 40, + digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP1 Volume", + WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B2_CTL, -84, 40, + digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP2 Volume", + WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B3_CTL, -84, 40, + digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP3 Volume", + WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B4_CTL, -84, 40, + digital_gain), + + SOC_ENUM("TX0 HPF cut off", cf_dec0_enum), + SOC_ENUM("TX1 HPF cut off", cf_dec1_enum), + SOC_ENUM("TX2 HPF cut off", cf_dec2_enum), + SOC_ENUM("TX3 HPF cut off", cf_dec3_enum), + SOC_ENUM("TX4 HPF cut off", cf_dec4_enum), + SOC_ENUM("TX5 HPF cut off", cf_dec5_enum), + SOC_ENUM("TX6 HPF cut off", cf_dec6_enum), + SOC_ENUM("TX7 HPF cut off", cf_dec7_enum), + SOC_ENUM("TX8 HPF cut off", cf_dec8_enum), + + SOC_ENUM("RX INT0_1 HPF cut off", cf_int0_1_enum), + SOC_ENUM("RX INT0_2 HPF cut off", cf_int0_2_enum), + SOC_ENUM("RX INT1_1 HPF cut off", cf_int1_1_enum), + SOC_ENUM("RX INT1_2 HPF cut off", cf_int1_2_enum), + SOC_ENUM("RX INT2_1 HPF cut off", cf_int2_1_enum), + SOC_ENUM("RX INT2_2 HPF cut off", cf_int2_2_enum), + SOC_ENUM("RX INT3_1 HPF cut off", cf_int3_1_enum), + SOC_ENUM("RX INT3_2 HPF cut off", cf_int3_2_enum), + SOC_ENUM("RX INT4_1 HPF cut off", cf_int4_1_enum), + SOC_ENUM("RX INT4_2 HPF cut off", cf_int4_2_enum), + SOC_ENUM("RX INT7_1 HPF cut off", cf_int7_1_enum), + SOC_ENUM("RX INT7_2 HPF cut off", cf_int7_2_enum), + SOC_ENUM("RX INT8_1 HPF cut off", cf_int8_1_enum), + SOC_ENUM("RX INT8_2 HPF cut off", cf_int8_2_enum), + + SOC_ENUM_EXT("RX HPH Mode", rx_hph_mode_mux_enum, + wcd934x_rx_hph_mode_get, wcd934x_rx_hph_mode_put), + + SOC_SINGLE("IIR1 Band1 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL, + 0, 1, 0), + SOC_SINGLE("IIR1 Band2 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL, + 1, 1, 0), + SOC_SINGLE("IIR1 Band3 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL, + 2, 1, 0), + SOC_SINGLE("IIR1 Band4 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL, + 3, 1, 0), + SOC_SINGLE("IIR1 Band5 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL, + 4, 1, 0), + SOC_SINGLE("IIR2 Band1 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL, + 0, 1, 0), + SOC_SINGLE("IIR2 Band2 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL, + 1, 1, 0), + SOC_SINGLE("IIR2 Band3 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL, + 2, 1, 0), + SOC_SINGLE("IIR2 Band4 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL, + 3, 1, 0), + SOC_SINGLE("IIR2 Band5 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL, + 4, 1, 0), + WCD_IIR_FILTER_CTL("IIR0 Band1", IIR0, BAND1), + WCD_IIR_FILTER_CTL("IIR0 Band2", IIR0, BAND2), + WCD_IIR_FILTER_CTL("IIR0 Band3", IIR0, BAND3), + WCD_IIR_FILTER_CTL("IIR0 Band4", IIR0, BAND4), + WCD_IIR_FILTER_CTL("IIR0 Band5", IIR0, BAND5), + + WCD_IIR_FILTER_CTL("IIR1 Band1", IIR1, BAND1), + WCD_IIR_FILTER_CTL("IIR1 Band2", IIR1, BAND2), + WCD_IIR_FILTER_CTL("IIR1 Band3", IIR1, BAND3), + WCD_IIR_FILTER_CTL("IIR1 Band4", IIR1, BAND4), + WCD_IIR_FILTER_CTL("IIR1 Band5", IIR1, BAND5), + + SOC_SINGLE_EXT("COMP1 Switch", SND_SOC_NOPM, COMPANDER_1, 1, 0, + wcd934x_compander_get, wcd934x_compander_set), + SOC_SINGLE_EXT("COMP2 Switch", SND_SOC_NOPM, COMPANDER_2, 1, 0, + wcd934x_compander_get, wcd934x_compander_set), + SOC_SINGLE_EXT("COMP3 Switch", SND_SOC_NOPM, COMPANDER_3, 1, 0, + wcd934x_compander_get, wcd934x_compander_set), + SOC_SINGLE_EXT("COMP4 Switch", SND_SOC_NOPM, COMPANDER_4, 1, 0, + wcd934x_compander_get, wcd934x_compander_set), + SOC_SINGLE_EXT("COMP7 Switch", SND_SOC_NOPM, COMPANDER_7, 1, 0, + wcd934x_compander_get, wcd934x_compander_set), + SOC_SINGLE_EXT("COMP8 Switch", SND_SOC_NOPM, COMPANDER_8, 1, 0, + wcd934x_compander_get, wcd934x_compander_set), +}; + static const struct snd_soc_component_driver wcd934x_component_drv = { .probe = wcd934x_comp_probe, .remove = wcd934x_comp_remove, .set_sysclk = wcd934x_comp_set_sysclk, + .controls = wcd934x_snd_controls, + .num_controls = ARRAY_SIZE(wcd934x_snd_controls), }; static int wcd934x_codec_parse_data(struct wcd934x_codec *wcd) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: wcd934x: add playback dapm widgets" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: wcd934x: add playback dapm widgets has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dd9eb19b567303e4b92747dcfb5deedb182af111 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 19 Dec 2019 10:31:47 +0000 Subject: [PATCH] ASoC: wcd934x: add playback dapm widgets This patch adds required dapm widgets for playback. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20191219103153.14875-6-srinivas.kandagatla@linaro… Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1550 ++++++++++++++++++++++++++++++++++++ 1 file changed, 1550 insertions(+) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 27a864f2ff77..556c051f7f99 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -412,6 +412,94 @@ static const char * const rx_hph_mode_mux_text[] = { "Class-H Hi-Fi Low Power" }; +static const char *const slim_rx_mux_text[] = { + "ZERO", "AIF1_PB", "AIF2_PB", "AIF3_PB", "AIF4_PB", +}; + +static const char * const rx_int0_7_mix_mux_text[] = { + "ZERO", "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", + "RX6", "RX7", "PROXIMITY" +}; + +static const char * const rx_int_mix_mux_text[] = { + "ZERO", "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", + "RX6", "RX7" +}; + +static const char * const rx_prim_mix_text[] = { + "ZERO", "DEC0", "DEC1", "IIR0", "IIR1", "RX0", "RX1", "RX2", + "RX3", "RX4", "RX5", "RX6", "RX7" +}; + +static const char * const rx_sidetone_mix_text[] = { + "ZERO", "SRC0", "SRC1", "SRC_SUM" +}; + +static const char * const iir_inp_mux_text[] = { + "ZERO", "DEC0", "DEC1", "DEC2", "DEC3", "DEC4", "DEC5", "DEC6", + "DEC7", "DEC8", "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7" +}; + +static const char * const rx_int_dem_inp_mux_text[] = { + "NORMAL_DSM_OUT", "CLSH_DSM_OUT", +}; + +static const char * const rx_int0_1_interp_mux_text[] = { + "ZERO", "RX INT0_1 MIX1", +}; + +static const char * const rx_int1_1_interp_mux_text[] = { + "ZERO", "RX INT1_1 MIX1", +}; + +static const char * const rx_int2_1_interp_mux_text[] = { + "ZERO", "RX INT2_1 MIX1", +}; + +static const char * const rx_int3_1_interp_mux_text[] = { + "ZERO", "RX INT3_1 MIX1", +}; + +static const char * const rx_int4_1_interp_mux_text[] = { + "ZERO", "RX INT4_1 MIX1", +}; + +static const char * const rx_int7_1_interp_mux_text[] = { + "ZERO", "RX INT7_1 MIX1", +}; + +static const char * const rx_int8_1_interp_mux_text[] = { + "ZERO", "RX INT8_1 MIX1", +}; + +static const char * const rx_int0_2_interp_mux_text[] = { + "ZERO", "RX INT0_2 MUX", +}; + +static const char * const rx_int1_2_interp_mux_text[] = { + "ZERO", "RX INT1_2 MUX", +}; + +static const char * const rx_int2_2_interp_mux_text[] = { + "ZERO", "RX INT2_2 MUX", +}; + +static const char * const rx_int3_2_interp_mux_text[] = { + "ZERO", "RX INT3_2 MUX", +}; + +static const char * const rx_int4_2_interp_mux_text[] = { + "ZERO", "RX INT4_2 MUX", +}; + +static const char * const rx_int7_2_interp_mux_text[] = { + "ZERO", "RX INT7_2 MUX", +}; + +static const char * const rx_int8_2_interp_mux_text[] = { + "ZERO", "RX INT8_2 MUX", +}; + static const struct soc_enum cf_dec0_enum = SOC_ENUM_SINGLE(WCD934X_CDC_TX0_TX_PATH_CFG0, 5, 3, cf_text); @@ -485,6 +573,236 @@ static const struct soc_enum rx_hph_mode_mux_enum = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rx_hph_mode_mux_text), rx_hph_mode_mux_text); +static const struct soc_enum slim_rx_mux_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(slim_rx_mux_text), slim_rx_mux_text); + +static const struct soc_enum rx_int0_2_mux_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG1, 0, 10, + rx_int0_7_mix_mux_text); + +static const struct soc_enum rx_int1_2_mux_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG1, 0, 9, + rx_int_mix_mux_text); + +static const struct soc_enum rx_int2_2_mux_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG1, 0, 9, + rx_int_mix_mux_text); + +static const struct soc_enum rx_int3_2_mux_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG1, 0, 9, + rx_int_mix_mux_text); + +static const struct soc_enum rx_int4_2_mux_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG1, 0, 9, + rx_int_mix_mux_text); + +static const struct soc_enum rx_int7_2_mux_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG1, 0, 10, + rx_int0_7_mix_mux_text); + +static const struct soc_enum rx_int8_2_mux_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG1, 0, 9, + rx_int_mix_mux_text); + +static const struct soc_enum rx_int0_1_mix_inp0_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG0, 0, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int0_1_mix_inp1_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG0, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int0_1_mix_inp2_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG1, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int1_1_mix_inp0_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG0, 0, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int1_1_mix_inp1_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG0, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int1_1_mix_inp2_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG1, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int2_1_mix_inp0_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG0, 0, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int2_1_mix_inp1_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG0, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int2_1_mix_inp2_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG1, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int3_1_mix_inp0_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG0, 0, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int3_1_mix_inp1_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG0, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int3_1_mix_inp2_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG1, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int4_1_mix_inp0_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG0, 0, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int4_1_mix_inp1_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG0, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int4_1_mix_inp2_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG1, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int7_1_mix_inp0_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG0, 0, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int7_1_mix_inp1_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG0, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int7_1_mix_inp2_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG1, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int8_1_mix_inp0_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG0, 0, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int8_1_mix_inp1_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG0, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int8_1_mix_inp2_chain_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG1, 4, 13, + rx_prim_mix_text); + +static const struct soc_enum rx_int0_mix2_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 0, 4, + rx_sidetone_mix_text); + +static const struct soc_enum rx_int1_mix2_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 2, 4, + rx_sidetone_mix_text); + +static const struct soc_enum rx_int2_mix2_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 4, 4, + rx_sidetone_mix_text); + +static const struct soc_enum rx_int3_mix2_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 6, 4, + rx_sidetone_mix_text); + +static const struct soc_enum rx_int4_mix2_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG1, 0, 4, + rx_sidetone_mix_text); + +static const struct soc_enum rx_int7_mix2_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG1, 2, 4, + rx_sidetone_mix_text); + +static const struct soc_enum iir0_inp0_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG0, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum iir0_inp1_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG1, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum iir0_inp2_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG2, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum iir0_inp3_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG3, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum iir1_inp0_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG0, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum iir1_inp1_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG1, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum iir1_inp2_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG2, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum iir1_inp3_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG3, + 0, 18, iir_inp_mux_text); + +static const struct soc_enum rx_int0_dem_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX0_RX_PATH_SEC0, 0, + ARRAY_SIZE(rx_int_dem_inp_mux_text), + rx_int_dem_inp_mux_text); + +static const struct soc_enum rx_int1_dem_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX1_RX_PATH_SEC0, 0, + ARRAY_SIZE(rx_int_dem_inp_mux_text), + rx_int_dem_inp_mux_text); + +static const struct soc_enum rx_int2_dem_inp_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_RX2_RX_PATH_SEC0, 0, + ARRAY_SIZE(rx_int_dem_inp_mux_text), + rx_int_dem_inp_mux_text); +static const struct soc_enum rx_int0_1_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, + rx_int0_1_interp_mux_text); + +static const struct soc_enum rx_int1_1_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, + rx_int1_1_interp_mux_text); + +static const struct soc_enum rx_int2_1_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, + rx_int2_1_interp_mux_text); + +static const struct soc_enum rx_int3_1_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int3_1_interp_mux_text); + +static const struct soc_enum rx_int4_1_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int4_1_interp_mux_text); + +static const struct soc_enum rx_int7_1_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int7_1_interp_mux_text); + +static const struct soc_enum rx_int8_1_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int8_1_interp_mux_text); + +static const struct soc_enum rx_int0_2_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int0_2_interp_mux_text); + +static const struct soc_enum rx_int1_2_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int1_2_interp_mux_text); + +static const struct soc_enum rx_int2_2_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int2_2_interp_mux_text); + +static const struct soc_enum rx_int3_2_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int3_2_interp_mux_text); + +static const struct soc_enum rx_int4_2_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int4_2_interp_mux_text); + +static const struct soc_enum rx_int7_2_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int7_2_interp_mux_text); + +static const struct soc_enum rx_int8_2_interp_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int8_2_interp_mux_text); + static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) { @@ -639,6 +957,22 @@ static int __wcd934x_cdc_mclk_enable(struct wcd934x_codec *wcd, bool enable) return ret; } +static int wcd934x_codec_enable_mclk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kc, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return __wcd934x_cdc_mclk_enable(wcd, true); + case SND_SOC_DAPM_POST_PMD: + return __wcd934x_cdc_mclk_enable(wcd, false); + } + + return 0; +} + static int wcd934x_get_version(struct wcd934x_codec *wcd) { int val1, val2, ver, ret; @@ -1814,6 +2148,311 @@ static int wcd934x_rx_hph_mode_put(struct snd_kcontrol *kc, return 0; } +static int slim_rx_mux_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); + struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kc); + struct wcd934x_codec *wcd = dev_get_drvdata(dapm->dev); + + ucontrol->value.enumerated.item[0] = wcd->rx_port_value[w->shift]; + + return 0; +} + +static int slim_rx_mux_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kc); + struct wcd934x_codec *wcd = dev_get_drvdata(w->dapm->dev); + struct soc_enum *e = (struct soc_enum *)kc->private_value; + struct snd_soc_dapm_update *update = NULL; + u32 port_id = w->shift; + + if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0]) + return 0; + + wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; + + switch (wcd->rx_port_value[port_id]) { + case 0: + list_del_init(&wcd->rx_chs[port_id].list); + break; + case 1: + list_add_tail(&wcd->rx_chs[port_id].list, + &wcd->dai[AIF1_PB].slim_ch_list); + break; + case 2: + list_add_tail(&wcd->rx_chs[port_id].list, + &wcd->dai[AIF2_PB].slim_ch_list); + break; + case 3: + list_add_tail(&wcd->rx_chs[port_id].list, + &wcd->dai[AIF3_PB].slim_ch_list); + break; + case 4: + list_add_tail(&wcd->rx_chs[port_id].list, + &wcd->dai[AIF4_PB].slim_ch_list); + break; + default: + dev_err(wcd->dev, "Unknown AIF %d\n", + wcd->rx_port_value[port_id]); + goto err; + } + + snd_soc_dapm_mux_update_power(w->dapm, kc, wcd->rx_port_value[port_id], + e, update); + + return 0; +err: + return -EINVAL; +} + +static int wcd934x_int_dem_inp_mux_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *)kc->private_value; + struct snd_soc_component *component; + int reg, val, ret; + + component = snd_soc_dapm_kcontrol_component(kc); + val = ucontrol->value.enumerated.item[0]; + if (e->reg == WCD934X_CDC_RX0_RX_PATH_SEC0) + reg = WCD934X_CDC_RX0_RX_PATH_CFG0; + else if (e->reg == WCD934X_CDC_RX1_RX_PATH_SEC0) + reg = WCD934X_CDC_RX1_RX_PATH_CFG0; + else if (e->reg == WCD934X_CDC_RX2_RX_PATH_SEC0) + reg = WCD934X_CDC_RX2_RX_PATH_CFG0; + else + return -EINVAL; + + /* Set Look Ahead Delay */ + if (val) + snd_soc_component_update_bits(component, reg, + WCD934X_RX_DLY_ZN_EN_MASK, + WCD934X_RX_DLY_ZN_ENABLE); + else + snd_soc_component_update_bits(component, reg, + WCD934X_RX_DLY_ZN_EN_MASK, + WCD934X_RX_DLY_ZN_DISABLE); + + ret = snd_soc_dapm_put_enum_double(kc, ucontrol); + + return ret; +} + +static const struct snd_kcontrol_new rx_int0_2_mux = + SOC_DAPM_ENUM("RX INT0_2 MUX Mux", rx_int0_2_mux_chain_enum); + +static const struct snd_kcontrol_new rx_int1_2_mux = + SOC_DAPM_ENUM("RX INT1_2 MUX Mux", rx_int1_2_mux_chain_enum); + +static const struct snd_kcontrol_new rx_int2_2_mux = + SOC_DAPM_ENUM("RX INT2_2 MUX Mux", rx_int2_2_mux_chain_enum); + +static const struct snd_kcontrol_new rx_int3_2_mux = + SOC_DAPM_ENUM("RX INT3_2 MUX Mux", rx_int3_2_mux_chain_enum); + +static const struct snd_kcontrol_new rx_int4_2_mux = + SOC_DAPM_ENUM("RX INT4_2 MUX Mux", rx_int4_2_mux_chain_enum); + +static const struct snd_kcontrol_new rx_int7_2_mux = + SOC_DAPM_ENUM("RX INT7_2 MUX Mux", rx_int7_2_mux_chain_enum); + +static const struct snd_kcontrol_new rx_int8_2_mux = + SOC_DAPM_ENUM("RX INT8_2 MUX Mux", rx_int8_2_mux_chain_enum); + +static const struct snd_kcontrol_new rx_int0_1_mix_inp0_mux = + SOC_DAPM_ENUM("RX INT0_1 MIX1 INP0 Mux", rx_int0_1_mix_inp0_chain_enum); + +static const struct snd_kcontrol_new rx_int0_1_mix_inp1_mux = + SOC_DAPM_ENUM("RX INT0_1 MIX1 INP1 Mux", rx_int0_1_mix_inp1_chain_enum); + +static const struct snd_kcontrol_new rx_int0_1_mix_inp2_mux = + SOC_DAPM_ENUM("RX INT0_1 MIX1 INP2 Mux", rx_int0_1_mix_inp2_chain_enum); + +static const struct snd_kcontrol_new rx_int1_1_mix_inp0_mux = + SOC_DAPM_ENUM("RX INT1_1 MIX1 INP0 Mux", rx_int1_1_mix_inp0_chain_enum); + +static const struct snd_kcontrol_new rx_int1_1_mix_inp1_mux = + SOC_DAPM_ENUM("RX INT1_1 MIX1 INP1 Mux", rx_int1_1_mix_inp1_chain_enum); + +static const struct snd_kcontrol_new rx_int1_1_mix_inp2_mux = + SOC_DAPM_ENUM("RX INT1_1 MIX1 INP2 Mux", rx_int1_1_mix_inp2_chain_enum); + +static const struct snd_kcontrol_new rx_int2_1_mix_inp0_mux = + SOC_DAPM_ENUM("RX INT2_1 MIX1 INP0 Mux", rx_int2_1_mix_inp0_chain_enum); + +static const struct snd_kcontrol_new rx_int2_1_mix_inp1_mux = + SOC_DAPM_ENUM("RX INT2_1 MIX1 INP1 Mux", rx_int2_1_mix_inp1_chain_enum); + +static const struct snd_kcontrol_new rx_int2_1_mix_inp2_mux = + SOC_DAPM_ENUM("RX INT2_1 MIX1 INP2 Mux", rx_int2_1_mix_inp2_chain_enum); + +static const struct snd_kcontrol_new rx_int3_1_mix_inp0_mux = + SOC_DAPM_ENUM("RX INT3_1 MIX1 INP0 Mux", rx_int3_1_mix_inp0_chain_enum); + +static const struct snd_kcontrol_new rx_int3_1_mix_inp1_mux = + SOC_DAPM_ENUM("RX INT3_1 MIX1 INP1 Mux", rx_int3_1_mix_inp1_chain_enum); + +static const struct snd_kcontrol_new rx_int3_1_mix_inp2_mux = + SOC_DAPM_ENUM("RX INT3_1 MIX1 INP2 Mux", rx_int3_1_mix_inp2_chain_enum); + +static const struct snd_kcontrol_new rx_int4_1_mix_inp0_mux = + SOC_DAPM_ENUM("RX INT4_1 MIX1 INP0 Mux", rx_int4_1_mix_inp0_chain_enum); + +static const struct snd_kcontrol_new rx_int4_1_mix_inp1_mux = + SOC_DAPM_ENUM("RX INT4_1 MIX1 INP1 Mux", rx_int4_1_mix_inp1_chain_enum); + +static const struct snd_kcontrol_new rx_int4_1_mix_inp2_mux = + SOC_DAPM_ENUM("RX INT4_1 MIX1 INP2 Mux", rx_int4_1_mix_inp2_chain_enum); + +static const struct snd_kcontrol_new rx_int7_1_mix_inp0_mux = + SOC_DAPM_ENUM("RX INT7_1 MIX1 INP0 Mux", rx_int7_1_mix_inp0_chain_enum); + +static const struct snd_kcontrol_new rx_int7_1_mix_inp1_mux = + SOC_DAPM_ENUM("RX INT7_1 MIX1 INP1 Mux", rx_int7_1_mix_inp1_chain_enum); + +static const struct snd_kcontrol_new rx_int7_1_mix_inp2_mux = + SOC_DAPM_ENUM("RX INT7_1 MIX1 INP2 Mux", rx_int7_1_mix_inp2_chain_enum); + +static const struct snd_kcontrol_new rx_int8_1_mix_inp0_mux = + SOC_DAPM_ENUM("RX INT8_1 MIX1 INP0 Mux", rx_int8_1_mix_inp0_chain_enum); + +static const struct snd_kcontrol_new rx_int8_1_mix_inp1_mux = + SOC_DAPM_ENUM("RX INT8_1 MIX1 INP1 Mux", rx_int8_1_mix_inp1_chain_enum); + +static const struct snd_kcontrol_new rx_int8_1_mix_inp2_mux = + SOC_DAPM_ENUM("RX INT8_1 MIX1 INP2 Mux", rx_int8_1_mix_inp2_chain_enum); + +static const struct snd_kcontrol_new rx_int0_mix2_inp_mux = + SOC_DAPM_ENUM("RX INT0 MIX2 INP Mux", rx_int0_mix2_inp_mux_enum); + +static const struct snd_kcontrol_new rx_int1_mix2_inp_mux = + SOC_DAPM_ENUM("RX INT1 MIX2 INP Mux", rx_int1_mix2_inp_mux_enum); + +static const struct snd_kcontrol_new rx_int2_mix2_inp_mux = + SOC_DAPM_ENUM("RX INT2 MIX2 INP Mux", rx_int2_mix2_inp_mux_enum); + +static const struct snd_kcontrol_new rx_int3_mix2_inp_mux = + SOC_DAPM_ENUM("RX INT3 MIX2 INP Mux", rx_int3_mix2_inp_mux_enum); + +static const struct snd_kcontrol_new rx_int4_mix2_inp_mux = + SOC_DAPM_ENUM("RX INT4 MIX2 INP Mux", rx_int4_mix2_inp_mux_enum); + +static const struct snd_kcontrol_new rx_int7_mix2_inp_mux = + SOC_DAPM_ENUM("RX INT7 MIX2 INP Mux", rx_int7_mix2_inp_mux_enum); + +static const struct snd_kcontrol_new iir0_inp0_mux = + SOC_DAPM_ENUM("IIR0 INP0 Mux", iir0_inp0_mux_enum); +static const struct snd_kcontrol_new iir0_inp1_mux = + SOC_DAPM_ENUM("IIR0 INP1 Mux", iir0_inp1_mux_enum); +static const struct snd_kcontrol_new iir0_inp2_mux = + SOC_DAPM_ENUM("IIR0 INP2 Mux", iir0_inp2_mux_enum); +static const struct snd_kcontrol_new iir0_inp3_mux = + SOC_DAPM_ENUM("IIR0 INP3 Mux", iir0_inp3_mux_enum); + +static const struct snd_kcontrol_new iir1_inp0_mux = + SOC_DAPM_ENUM("IIR1 INP0 Mux", iir1_inp0_mux_enum); +static const struct snd_kcontrol_new iir1_inp1_mux = + SOC_DAPM_ENUM("IIR1 INP1 Mux", iir1_inp1_mux_enum); +static const struct snd_kcontrol_new iir1_inp2_mux = + SOC_DAPM_ENUM("IIR1 INP2 Mux", iir1_inp2_mux_enum); +static const struct snd_kcontrol_new iir1_inp3_mux = + SOC_DAPM_ENUM("IIR1 INP3 Mux", iir1_inp3_mux_enum); + +static const struct snd_kcontrol_new slim_rx_mux[WCD934X_RX_MAX] = { + SOC_DAPM_ENUM_EXT("SLIM RX0 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), + SOC_DAPM_ENUM_EXT("SLIM RX1 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), + SOC_DAPM_ENUM_EXT("SLIM RX2 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), + SOC_DAPM_ENUM_EXT("SLIM RX3 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), + SOC_DAPM_ENUM_EXT("SLIM RX4 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), + SOC_DAPM_ENUM_EXT("SLIM RX5 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), + SOC_DAPM_ENUM_EXT("SLIM RX6 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), + SOC_DAPM_ENUM_EXT("SLIM RX7 Mux", slim_rx_mux_enum, + slim_rx_mux_get, slim_rx_mux_put), +}; + +static const struct snd_kcontrol_new rx_int1_asrc_switch[] = { + SOC_DAPM_SINGLE("HPHL Switch", SND_SOC_NOPM, 0, 1, 0), +}; + +static const struct snd_kcontrol_new rx_int2_asrc_switch[] = { + SOC_DAPM_SINGLE("HPHR Switch", SND_SOC_NOPM, 0, 1, 0), +}; + +static const struct snd_kcontrol_new rx_int3_asrc_switch[] = { + SOC_DAPM_SINGLE("LO1 Switch", SND_SOC_NOPM, 0, 1, 0), +}; + +static const struct snd_kcontrol_new rx_int4_asrc_switch[] = { + SOC_DAPM_SINGLE("LO2 Switch", SND_SOC_NOPM, 0, 1, 0), +}; + +static const struct snd_kcontrol_new rx_int0_dem_inp_mux = + SOC_DAPM_ENUM_EXT("RX INT0 DEM MUX Mux", rx_int0_dem_inp_mux_enum, + snd_soc_dapm_get_enum_double, + wcd934x_int_dem_inp_mux_put); + +static const struct snd_kcontrol_new rx_int1_dem_inp_mux = + SOC_DAPM_ENUM_EXT("RX INT1 DEM MUX Mux", rx_int1_dem_inp_mux_enum, + snd_soc_dapm_get_enum_double, + wcd934x_int_dem_inp_mux_put); + +static const struct snd_kcontrol_new rx_int2_dem_inp_mux = + SOC_DAPM_ENUM_EXT("RX INT2 DEM MUX Mux", rx_int2_dem_inp_mux_enum, + snd_soc_dapm_get_enum_double, + wcd934x_int_dem_inp_mux_put); + +static const struct snd_kcontrol_new rx_int0_1_interp_mux = + SOC_DAPM_ENUM("RX INT0_1 INTERP Mux", rx_int0_1_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int1_1_interp_mux = + SOC_DAPM_ENUM("RX INT1_1 INTERP Mux", rx_int1_1_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int2_1_interp_mux = + SOC_DAPM_ENUM("RX INT2_1 INTERP Mux", rx_int2_1_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int3_1_interp_mux = + SOC_DAPM_ENUM("RX INT3_1 INTERP Mux", rx_int3_1_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int4_1_interp_mux = + SOC_DAPM_ENUM("RX INT4_1 INTERP Mux", rx_int4_1_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int7_1_interp_mux = + SOC_DAPM_ENUM("RX INT7_1 INTERP Mux", rx_int7_1_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int8_1_interp_mux = + SOC_DAPM_ENUM("RX INT8_1 INTERP Mux", rx_int8_1_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int0_2_interp_mux = + SOC_DAPM_ENUM("RX INT0_2 INTERP Mux", rx_int0_2_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int1_2_interp_mux = + SOC_DAPM_ENUM("RX INT1_2 INTERP Mux", rx_int1_2_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int2_2_interp_mux = + SOC_DAPM_ENUM("RX INT2_2 INTERP Mux", rx_int2_2_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int3_2_interp_mux = + SOC_DAPM_ENUM("RX INT3_2 INTERP Mux", rx_int3_2_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int4_2_interp_mux = + SOC_DAPM_ENUM("RX INT4_2 INTERP Mux", rx_int4_2_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int7_2_interp_mux = + SOC_DAPM_ENUM("RX INT7_2 INTERP Mux", rx_int7_2_interp_mux_enum); + +static const struct snd_kcontrol_new rx_int8_2_interp_mux = + SOC_DAPM_ENUM("RX INT8_2 INTERP Mux", rx_int8_2_interp_mux_enum); + static const struct snd_kcontrol_new wcd934x_snd_controls[] = { /* Gain Controls */ SOC_SINGLE_TLV("EAR PA Volume", WCD934X_ANA_EAR, 4, 4, 1, ear_pa_gain), @@ -1983,12 +2622,923 @@ static const struct snd_kcontrol_new wcd934x_snd_controls[] = { wcd934x_compander_get, wcd934x_compander_set), }; +static void wcd934x_codec_enable_int_port(struct wcd_slim_codec_dai_data *dai, + struct snd_soc_component *component) +{ + int port_num = 0; + unsigned short reg = 0; + unsigned int val = 0; + struct wcd934x_codec *wcd = dev_get_drvdata(component->dev); + struct wcd934x_slim_ch *ch; + + list_for_each_entry(ch, &dai->slim_ch_list, list) { + if (ch->port >= WCD934X_RX_START) { + port_num = ch->port - WCD934X_RX_START; + reg = WCD934X_SLIM_PGD_PORT_INT_EN0 + (port_num / 8); + } else { + port_num = ch->port; + reg = WCD934X_SLIM_PGD_PORT_INT_TX_EN0 + (port_num / 8); + } + + regmap_read(wcd->if_regmap, reg, &val); + if (!(val & BIT(port_num % 8))) + regmap_write(wcd->if_regmap, reg, + val | BIT(port_num % 8)); + } +} + +static int wcd934x_codec_enable_slim(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kc, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct wcd934x_codec *wcd = snd_soc_component_get_drvdata(comp); + struct wcd_slim_codec_dai_data *dai = &wcd->dai[w->shift]; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + wcd934x_codec_enable_int_port(dai, comp); + break; + } + + return 0; +} + +static void wcd934x_codec_hd2_control(struct snd_soc_component *component, + u16 interp_idx, int event) +{ + u16 hd2_scale_reg; + u16 hd2_enable_reg = 0; + + switch (interp_idx) { + case INTERP_HPHL: + hd2_scale_reg = WCD934X_CDC_RX1_RX_PATH_SEC3; + hd2_enable_reg = WCD934X_CDC_RX1_RX_PATH_CFG0; + break; + case INTERP_HPHR: + hd2_scale_reg = WCD934X_CDC_RX2_RX_PATH_SEC3; + hd2_enable_reg = WCD934X_CDC_RX2_RX_PATH_CFG0; + break; + default: + return; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + snd_soc_component_update_bits(component, hd2_scale_reg, + WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_MASK, + WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_0P3125); + snd_soc_component_update_bits(component, hd2_enable_reg, + WCD934X_CDC_RX_PATH_CFG_HD2_EN_MASK, + WCD934X_CDC_RX_PATH_CFG_HD2_ENABLE); + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + snd_soc_component_update_bits(component, hd2_enable_reg, + WCD934X_CDC_RX_PATH_CFG_HD2_EN_MASK, + WCD934X_CDC_RX_PATH_CFG_HD2_DISABLE); + snd_soc_component_update_bits(component, hd2_scale_reg, + WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_MASK, + WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_0P0000); + } +} + +static void wcd934x_codec_hphdelay_lutbypass(struct snd_soc_component *comp, + u16 interp_idx, int event) +{ + u8 hph_dly_mask; + u16 hph_lut_bypass_reg = 0; + u16 hph_comp_ctrl7 = 0; + + switch (interp_idx) { + case INTERP_HPHL: + hph_dly_mask = 1; + hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHL_COMP_LUT; + hph_comp_ctrl7 = WCD934X_CDC_COMPANDER1_CTL7; + break; + case INTERP_HPHR: + hph_dly_mask = 2; + hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHR_COMP_LUT; + hph_comp_ctrl7 = WCD934X_CDC_COMPANDER2_CTL7; + break; + default: + return; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + snd_soc_component_update_bits(comp, WCD934X_CDC_CLSH_TEST0, + hph_dly_mask, 0x0); + snd_soc_component_update_bits(comp, hph_lut_bypass_reg, + WCD934X_HPH_LUT_BYPASS_MASK, + WCD934X_HPH_LUT_BYPASS_ENABLE); + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + snd_soc_component_update_bits(comp, WCD934X_CDC_CLSH_TEST0, + hph_dly_mask, hph_dly_mask); + snd_soc_component_update_bits(comp, hph_lut_bypass_reg, + WCD934X_HPH_LUT_BYPASS_MASK, + WCD934X_HPH_LUT_BYPASS_DISABLE); + } +} + +static int wcd934x_config_compander(struct snd_soc_component *comp, + int interp_n, int event) +{ + struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev); + int compander; + u16 comp_ctl0_reg, rx_path_cfg0_reg; + + /* EAR does not have compander */ + if (!interp_n) + return 0; + + compander = interp_n - 1; + if (!wcd->comp_enabled[compander]) + return 0; + + comp_ctl0_reg = WCD934X_CDC_COMPANDER1_CTL0 + (compander * 8); + rx_path_cfg0_reg = WCD934X_CDC_RX1_RX_PATH_CFG0 + (compander * 20); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable Compander Clock */ + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_CLK_EN_MASK, + WCD934X_COMP_CLK_ENABLE); + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_SOFT_RST_MASK, + WCD934X_COMP_SOFT_RST_ENABLE); + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_SOFT_RST_MASK, + WCD934X_COMP_SOFT_RST_DISABLE); + snd_soc_component_update_bits(comp, rx_path_cfg0_reg, + WCD934X_HPH_CMP_EN_MASK, + WCD934X_HPH_CMP_ENABLE); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(comp, rx_path_cfg0_reg, + WCD934X_HPH_CMP_EN_MASK, + WCD934X_HPH_CMP_DISABLE); + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_HALT_MASK, + WCD934X_COMP_HALT); + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_SOFT_RST_MASK, + WCD934X_COMP_SOFT_RST_ENABLE); + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_SOFT_RST_MASK, + WCD934X_COMP_SOFT_RST_DISABLE); + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_CLK_EN_MASK, 0x0); + snd_soc_component_update_bits(comp, comp_ctl0_reg, + WCD934X_COMP_SOFT_RST_MASK, 0x0); + break; + } + + return 0; +} + +static int wcd934x_codec_enable_interp_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kc, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + int interp_idx = w->shift; + u16 main_reg = WCD934X_CDC_RX0_RX_PATH_CTL + (interp_idx * 20); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Clk enable */ + snd_soc_component_update_bits(comp, main_reg, + WCD934X_RX_CLK_EN_MASK, + WCD934X_RX_CLK_ENABLE); + wcd934x_codec_hd2_control(comp, interp_idx, event); + wcd934x_codec_hphdelay_lutbypass(comp, interp_idx, event); + wcd934x_config_compander(comp, interp_idx, event); + break; + case SND_SOC_DAPM_POST_PMD: + wcd934x_config_compander(comp, interp_idx, event); + wcd934x_codec_hphdelay_lutbypass(comp, interp_idx, event); + wcd934x_codec_hd2_control(comp, interp_idx, event); + /* Clk Disable */ + snd_soc_component_update_bits(comp, main_reg, + WCD934X_RX_CLK_EN_MASK, 0); + /* Reset enable and disable */ + snd_soc_component_update_bits(comp, main_reg, + WCD934X_RX_RESET_MASK, + WCD934X_RX_RESET_ENABLE); + snd_soc_component_update_bits(comp, main_reg, + WCD934X_RX_RESET_MASK, + WCD934X_RX_RESET_DISABLE); + /* Reset rate to 48K*/ + snd_soc_component_update_bits(comp, main_reg, + WCD934X_RX_PCM_RATE_MASK, + WCD934X_RX_PCM_RATE_F_48K); + break; + } + + return 0; +} + +static int wcd934x_codec_enable_mix_path(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kc, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + int offset_val = 0; + u16 gain_reg, mix_reg; + int val = 0; + + gain_reg = WCD934X_CDC_RX0_RX_VOL_MIX_CTL + + (w->shift * WCD934X_RX_PATH_CTL_OFFSET); + mix_reg = WCD934X_CDC_RX0_RX_PATH_MIX_CTL + + (w->shift * WCD934X_RX_PATH_CTL_OFFSET); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Clk enable */ + snd_soc_component_update_bits(comp, mix_reg, + WCD934X_CDC_RX_MIX_CLK_EN_MASK, + WCD934X_CDC_RX_MIX_CLK_ENABLE); + break; + + case SND_SOC_DAPM_POST_PMU: + val = snd_soc_component_read32(comp, gain_reg); + val += offset_val; + snd_soc_component_write(comp, gain_reg, val); + break; + }; + + return 0; +} + +static int wcd934x_codec_set_iir_gain(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + int reg = w->reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* B1 GAIN */ + snd_soc_component_write(comp, reg, + snd_soc_component_read32(comp, reg)); + /* B2 GAIN */ + reg++; + snd_soc_component_write(comp, reg, + snd_soc_component_read32(comp, reg)); + /* B3 GAIN */ + reg++; + snd_soc_component_write(comp, reg, + snd_soc_component_read32(comp, reg)); + /* B4 GAIN */ + reg++; + snd_soc_component_write(comp, reg, + snd_soc_component_read32(comp, reg)); + /* B5 GAIN */ + reg++; + snd_soc_component_write(comp, reg, + snd_soc_component_read32(comp, reg)); + break; + default: + break; + } + return 0; +} + +static int wcd934x_codec_enable_main_path(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + u16 gain_reg; + + gain_reg = WCD934X_CDC_RX0_RX_VOL_CTL + (w->shift * + WCD934X_RX_PATH_CTL_OFFSET); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_component_write(comp, gain_reg, + snd_soc_component_read32(comp, gain_reg)); + break; + }; + + return 0; +} + +static int wcd934x_codec_ear_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kc, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disable AutoChop timer during power up */ + snd_soc_component_update_bits(comp, + WCD934X_HPH_NEW_INT_HPH_TIMER1, + WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, 0x0); + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC, + WCD_CLSH_STATE_EAR, CLS_H_NORMAL); + + break; + case SND_SOC_DAPM_POST_PMD: + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, + WCD_CLSH_STATE_EAR, CLS_H_NORMAL); + break; + }; + + return 0; +} + +static int wcd934x_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev); + int hph_mode = wcd->hph_mode; + u8 dem_inp; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Read DEM INP Select */ + dem_inp = snd_soc_component_read32(comp, + WCD934X_CDC_RX1_RX_PATH_SEC0) & 0x03; + + if (((hph_mode == CLS_H_HIFI) || (hph_mode == CLS_H_LOHIFI) || + (hph_mode == CLS_H_LP)) && (dem_inp != 0x01)) { + return -EINVAL; + } + if (hph_mode != CLS_H_LP) + /* Ripple freq control enable */ + snd_soc_component_update_bits(comp, + WCD934X_SIDO_NEW_VOUT_D_FREQ2, + WCD934X_SIDO_RIPPLE_FREQ_EN_MASK, + WCD934X_SIDO_RIPPLE_FREQ_ENABLE); + /* Disable AutoChop timer during power up */ + snd_soc_component_update_bits(comp, + WCD934X_HPH_NEW_INT_HPH_TIMER1, + WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, 0x0); + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC, + WCD_CLSH_STATE_HPHL, hph_mode); + + break; + case SND_SOC_DAPM_POST_PMD: + /* 1000us required as per HW requirement */ + usleep_range(1000, 1100); + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, + WCD_CLSH_STATE_HPHL, hph_mode); + if (hph_mode != CLS_H_LP) + /* Ripple freq control disable */ + snd_soc_component_update_bits(comp, + WCD934X_SIDO_NEW_VOUT_D_FREQ2, + WCD934X_SIDO_RIPPLE_FREQ_EN_MASK, 0x0); + + break; + default: + break; + }; + + return 0; +} + +static int wcd934x_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev); + int hph_mode = wcd->hph_mode; + u8 dem_inp; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dem_inp = snd_soc_component_read32(comp, + WCD934X_CDC_RX2_RX_PATH_SEC0) & 0x03; + if (((hph_mode == CLS_H_HIFI) || (hph_mode == CLS_H_LOHIFI) || + (hph_mode == CLS_H_LP)) && (dem_inp != 0x01)) { + return -EINVAL; + } + if (hph_mode != CLS_H_LP) + /* Ripple freq control enable */ + snd_soc_component_update_bits(comp, + WCD934X_SIDO_NEW_VOUT_D_FREQ2, + WCD934X_SIDO_RIPPLE_FREQ_EN_MASK, + WCD934X_SIDO_RIPPLE_FREQ_ENABLE); + /* Disable AutoChop timer during power up */ + snd_soc_component_update_bits(comp, + WCD934X_HPH_NEW_INT_HPH_TIMER1, + WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, 0x0); + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC, + WCD_CLSH_STATE_HPHR, + hph_mode); + break; + case SND_SOC_DAPM_POST_PMD: + /* 1000us required as per HW requirement */ + usleep_range(1000, 1100); + + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, + WCD_CLSH_STATE_HPHR, hph_mode); + if (hph_mode != CLS_H_LP) + /* Ripple freq control disable */ + snd_soc_component_update_bits(comp, + WCD934X_SIDO_NEW_VOUT_D_FREQ2, + WCD934X_SIDO_RIPPLE_FREQ_EN_MASK, 0x0); + break; + default: + break; + }; + + return 0; +} + +static int wcd934x_codec_lineout_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kc, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC, + WCD_CLSH_STATE_LO, CLS_AB); + break; + case SND_SOC_DAPM_POST_PMD: + wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, + WCD_CLSH_STATE_LO, CLS_AB); + break; + } + + return 0; +} + +static int wcd934x_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* + * 7ms sleep is required after PA is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is needed. + */ + usleep_range(20000, 20100); + + snd_soc_component_update_bits(comp, WCD934X_HPH_L_TEST, + WCD934X_HPH_OCP_DET_MASK, + WCD934X_HPH_OCP_DET_ENABLE); + /* Remove Mute on primary path */ + snd_soc_component_update_bits(comp, WCD934X_CDC_RX1_RX_PATH_CTL, + WCD934X_RX_PATH_PGA_MUTE_EN_MASK, + 0); + /* Enable GM3 boost */ + snd_soc_component_update_bits(comp, WCD934X_HPH_CNP_WG_CTL, + WCD934X_HPH_GM3_BOOST_EN_MASK, + WCD934X_HPH_GM3_BOOST_ENABLE); + /* Enable AutoChop timer at the end of power up */ + snd_soc_component_update_bits(comp, + WCD934X_HPH_NEW_INT_HPH_TIMER1, + WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, + WCD934X_HPH_AUTOCHOP_TIMER_ENABLE); + /* Remove mix path mute */ + snd_soc_component_update_bits(comp, + WCD934X_CDC_RX1_RX_PATH_MIX_CTL, + WCD934X_CDC_RX_PGA_MUTE_EN_MASK, 0x00); + break; + case SND_SOC_DAPM_PRE_PMD: + /* Enable DSD Mute before PA disable */ + snd_soc_component_update_bits(comp, WCD934X_HPH_L_TEST, + WCD934X_HPH_OCP_DET_MASK, + WCD934X_HPH_OCP_DET_DISABLE); + snd_soc_component_update_bits(comp, WCD934X_CDC_RX1_RX_PATH_CTL, + WCD934X_RX_PATH_PGA_MUTE_EN_MASK, + WCD934X_RX_PATH_PGA_MUTE_ENABLE); + snd_soc_component_update_bits(comp, + WCD934X_CDC_RX1_RX_PATH_MIX_CTL, + WCD934X_RX_PATH_PGA_MUTE_EN_MASK, + WCD934X_RX_PATH_PGA_MUTE_ENABLE); + break; + case SND_SOC_DAPM_POST_PMD: + /* + * 5ms sleep is required after PA disable. If compander is + * disabled, then 20ms delay is needed after PA disable. + */ + usleep_range(20000, 20100); + break; + }; + + return 0; +} + +static int wcd934x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* + * 7ms sleep is required after PA is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is needed. + */ + usleep_range(20000, 20100); + snd_soc_component_update_bits(comp, WCD934X_HPH_R_TEST, + WCD934X_HPH_OCP_DET_MASK, + WCD934X_HPH_OCP_DET_ENABLE); + /* Remove mute */ + snd_soc_component_update_bits(comp, WCD934X_CDC_RX2_RX_PATH_CTL, + WCD934X_RX_PATH_PGA_MUTE_EN_MASK, + 0); + /* Enable GM3 boost */ + snd_soc_component_update_bits(comp, WCD934X_HPH_CNP_WG_CTL, + WCD934X_HPH_GM3_BOOST_EN_MASK, + WCD934X_HPH_GM3_BOOST_ENABLE); + /* Enable AutoChop timer at the end of power up */ + snd_soc_component_update_bits(comp, + WCD934X_HPH_NEW_INT_HPH_TIMER1, + WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, + WCD934X_HPH_AUTOCHOP_TIMER_ENABLE); + /* Remove mix path mute if it is enabled */ + if ((snd_soc_component_read32(comp, + WCD934X_CDC_RX2_RX_PATH_MIX_CTL)) & 0x10) + snd_soc_component_update_bits(comp, + WCD934X_CDC_RX2_RX_PATH_MIX_CTL, + WCD934X_CDC_RX_PGA_MUTE_EN_MASK, + WCD934X_CDC_RX_PGA_MUTE_DISABLE); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(comp, WCD934X_HPH_R_TEST, + WCD934X_HPH_OCP_DET_MASK, + WCD934X_HPH_OCP_DET_DISABLE); + snd_soc_component_update_bits(comp, WCD934X_CDC_RX2_RX_PATH_CTL, + WCD934X_RX_PATH_PGA_MUTE_EN_MASK, + WCD934X_RX_PATH_PGA_MUTE_ENABLE); + snd_soc_component_update_bits(comp, + WCD934X_CDC_RX2_RX_PATH_MIX_CTL, + WCD934X_CDC_RX_PGA_MUTE_EN_MASK, + WCD934X_CDC_RX_PGA_MUTE_ENABLE); + break; + case SND_SOC_DAPM_POST_PMD: + /* + * 5ms sleep is required after PA disable. If compander is + * disabled, then 20ms delay is needed after PA disable. + */ + usleep_range(20000, 20100); + break; + }; + + return 0; +} + +static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = { + /* Analog Outputs */ + SND_SOC_DAPM_OUTPUT("EAR"), + SND_SOC_DAPM_OUTPUT("HPHL"), + SND_SOC_DAPM_OUTPUT("HPHR"), + SND_SOC_DAPM_OUTPUT("LINEOUT1"), + SND_SOC_DAPM_OUTPUT("LINEOUT2"), + SND_SOC_DAPM_OUTPUT("SPK1 OUT"), + SND_SOC_DAPM_OUTPUT("SPK2 OUT"), + SND_SOC_DAPM_OUTPUT("ANC EAR"), + SND_SOC_DAPM_OUTPUT("ANC HPHL"), + SND_SOC_DAPM_OUTPUT("ANC HPHR"), + SND_SOC_DAPM_OUTPUT("WDMA3_OUT"), + SND_SOC_DAPM_OUTPUT("MAD_CPE_OUT1"), + SND_SOC_DAPM_OUTPUT("MAD_CPE_OUT2"), + SND_SOC_DAPM_AIF_IN_E("AIF1 PB", "AIF1 Playback", 0, SND_SOC_NOPM, + AIF1_PB, 0, wcd934x_codec_enable_slim, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("AIF2 PB", "AIF2 Playback", 0, SND_SOC_NOPM, + AIF2_PB, 0, wcd934x_codec_enable_slim, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("AIF3 PB", "AIF3 Playback", 0, SND_SOC_NOPM, + AIF3_PB, 0, wcd934x_codec_enable_slim, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("AIF4 PB", "AIF4 Playback", 0, SND_SOC_NOPM, + AIF4_PB, 0, wcd934x_codec_enable_slim, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX("SLIM RX0 MUX", SND_SOC_NOPM, WCD934X_RX0, 0, + &slim_rx_mux[WCD934X_RX0]), + SND_SOC_DAPM_MUX("SLIM RX1 MUX", SND_SOC_NOPM, WCD934X_RX1, 0, + &slim_rx_mux[WCD934X_RX1]), + SND_SOC_DAPM_MUX("SLIM RX2 MUX", SND_SOC_NOPM, WCD934X_RX2, 0, + &slim_rx_mux[WCD934X_RX2]), + SND_SOC_DAPM_MUX("SLIM RX3 MUX", SND_SOC_NOPM, WCD934X_RX3, 0, + &slim_rx_mux[WCD934X_RX3]), + SND_SOC_DAPM_MUX("SLIM RX4 MUX", SND_SOC_NOPM, WCD934X_RX4, 0, + &slim_rx_mux[WCD934X_RX4]), + SND_SOC_DAPM_MUX("SLIM RX5 MUX", SND_SOC_NOPM, WCD934X_RX5, 0, + &slim_rx_mux[WCD934X_RX5]), + SND_SOC_DAPM_MUX("SLIM RX6 MUX", SND_SOC_NOPM, WCD934X_RX6, 0, + &slim_rx_mux[WCD934X_RX6]), + SND_SOC_DAPM_MUX("SLIM RX7 MUX", SND_SOC_NOPM, WCD934X_RX7, 0, + &slim_rx_mux[WCD934X_RX7]), + + SND_SOC_DAPM_MIXER("SLIM RX0", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM RX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM RX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM RX3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM RX4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM RX5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM RX6", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM RX7", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX_E("RX INT0_2 MUX", SND_SOC_NOPM, INTERP_EAR, 0, + &rx_int0_2_mux, wcd934x_codec_enable_mix_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT1_2 MUX", SND_SOC_NOPM, INTERP_HPHL, 0, + &rx_int1_2_mux, wcd934x_codec_enable_mix_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT2_2 MUX", SND_SOC_NOPM, INTERP_HPHR, 0, + &rx_int2_2_mux, wcd934x_codec_enable_mix_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT3_2 MUX", SND_SOC_NOPM, INTERP_LO1, 0, + &rx_int3_2_mux, wcd934x_codec_enable_mix_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT4_2 MUX", SND_SOC_NOPM, INTERP_LO2, 0, + &rx_int4_2_mux, wcd934x_codec_enable_mix_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT7_2 MUX", SND_SOC_NOPM, INTERP_SPKR1, 0, + &rx_int7_2_mux, wcd934x_codec_enable_mix_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT8_2 MUX", SND_SOC_NOPM, INTERP_SPKR2, 0, + &rx_int8_2_mux, wcd934x_codec_enable_mix_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX("RX INT0_1 MIX1 INP0", SND_SOC_NOPM, 0, 0, + &rx_int0_1_mix_inp0_mux), + SND_SOC_DAPM_MUX("RX INT0_1 MIX1 INP1", SND_SOC_NOPM, 0, 0, + &rx_int0_1_mix_inp1_mux), + SND_SOC_DAPM_MUX("RX INT0_1 MIX1 INP2", SND_SOC_NOPM, 0, 0, + &rx_int0_1_mix_inp2_mux), + SND_SOC_DAPM_MUX("RX INT1_1 MIX1 INP0", SND_SOC_NOPM, 0, 0, + &rx_int1_1_mix_inp0_mux), + SND_SOC_DAPM_MUX("RX INT1_1 MIX1 INP1", SND_SOC_NOPM, 0, 0, + &rx_int1_1_mix_inp1_mux), + SND_SOC_DAPM_MUX("RX INT1_1 MIX1 INP2", SND_SOC_NOPM, 0, 0, + &rx_int1_1_mix_inp2_mux), + SND_SOC_DAPM_MUX("RX INT2_1 MIX1 INP0", SND_SOC_NOPM, 0, 0, + &rx_int2_1_mix_inp0_mux), + SND_SOC_DAPM_MUX("RX INT2_1 MIX1 INP1", SND_SOC_NOPM, 0, 0, + &rx_int2_1_mix_inp1_mux), + SND_SOC_DAPM_MUX("RX INT2_1 MIX1 INP2", SND_SOC_NOPM, 0, 0, + &rx_int2_1_mix_inp2_mux), + SND_SOC_DAPM_MUX("RX INT3_1 MIX1 INP0", SND_SOC_NOPM, 0, 0, + &rx_int3_1_mix_inp0_mux), + SND_SOC_DAPM_MUX("RX INT3_1 MIX1 INP1", SND_SOC_NOPM, 0, 0, + &rx_int3_1_mix_inp1_mux), + SND_SOC_DAPM_MUX("RX INT3_1 MIX1 INP2", SND_SOC_NOPM, 0, 0, + &rx_int3_1_mix_inp2_mux), + SND_SOC_DAPM_MUX("RX INT4_1 MIX1 INP0", SND_SOC_NOPM, 0, 0, + &rx_int4_1_mix_inp0_mux), + SND_SOC_DAPM_MUX("RX INT4_1 MIX1 INP1", SND_SOC_NOPM, 0, 0, + &rx_int4_1_mix_inp1_mux), + SND_SOC_DAPM_MUX("RX INT4_1 MIX1 INP2", SND_SOC_NOPM, 0, 0, + &rx_int4_1_mix_inp2_mux), + SND_SOC_DAPM_MUX("RX INT7_1 MIX1 INP0", SND_SOC_NOPM, 0, 0, + &rx_int7_1_mix_inp0_mux), + SND_SOC_DAPM_MUX("RX INT7_1 MIX1 INP1", SND_SOC_NOPM, 0, 0, + &rx_int7_1_mix_inp1_mux), + SND_SOC_DAPM_MUX("RX INT7_1 MIX1 INP2", SND_SOC_NOPM, 0, 0, + &rx_int7_1_mix_inp2_mux), + SND_SOC_DAPM_MUX("RX INT8_1 MIX1 INP0", SND_SOC_NOPM, 0, 0, + &rx_int8_1_mix_inp0_mux), + SND_SOC_DAPM_MUX("RX INT8_1 MIX1 INP1", SND_SOC_NOPM, 0, 0, + &rx_int8_1_mix_inp1_mux), + SND_SOC_DAPM_MUX("RX INT8_1 MIX1 INP2", SND_SOC_NOPM, 0, 0, + &rx_int8_1_mix_inp2_mux), + SND_SOC_DAPM_MIXER("RX INT0_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT0 SEC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT1_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT1 SEC MIX", SND_SOC_NOPM, 0, 0, + rx_int1_asrc_switch, + ARRAY_SIZE(rx_int1_asrc_switch)), + SND_SOC_DAPM_MIXER("RX INT2_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT2 SEC MIX", SND_SOC_NOPM, 0, 0, + rx_int2_asrc_switch, + ARRAY_SIZE(rx_int2_asrc_switch)), + SND_SOC_DAPM_MIXER("RX INT3_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT3 SEC MIX", SND_SOC_NOPM, 0, 0, + rx_int3_asrc_switch, + ARRAY_SIZE(rx_int3_asrc_switch)), + SND_SOC_DAPM_MIXER("RX INT4_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT4 SEC MIX", SND_SOC_NOPM, 0, 0, + rx_int4_asrc_switch, + ARRAY_SIZE(rx_int4_asrc_switch)), + SND_SOC_DAPM_MIXER("RX INT7_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT7 SEC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT8_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT8 SEC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT0 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT1 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT1 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT2 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT2 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT3 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT3 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT4 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("RX INT4 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("RX INT7 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("RX INT7 CHAIN", SND_SOC_NOPM, 0, 0, + NULL, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("RX INT8 CHAIN", SND_SOC_NOPM, 0, 0, + NULL, 0, NULL, 0), + SND_SOC_DAPM_MUX_E("RX INT0 MIX2 INP", WCD934X_CDC_RX0_RX_PATH_CFG0, 4, + 0, &rx_int0_mix2_inp_mux, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT1 MIX2 INP", WCD934X_CDC_RX1_RX_PATH_CFG0, 4, + 0, &rx_int1_mix2_inp_mux, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT2 MIX2 INP", WCD934X_CDC_RX2_RX_PATH_CFG0, 4, + 0, &rx_int2_mix2_inp_mux, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT3 MIX2 INP", WCD934X_CDC_RX3_RX_PATH_CFG0, 4, + 0, &rx_int3_mix2_inp_mux, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT4 MIX2 INP", WCD934X_CDC_RX4_RX_PATH_CFG0, 4, + 0, &rx_int4_mix2_inp_mux, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT7 MIX2 INP", WCD934X_CDC_RX7_RX_PATH_CFG0, 4, + 0, &rx_int7_mix2_inp_mux, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX("IIR0 INP0 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp0_mux), + SND_SOC_DAPM_MUX("IIR0 INP1 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp1_mux), + SND_SOC_DAPM_MUX("IIR0 INP2 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp2_mux), + SND_SOC_DAPM_MUX("IIR0 INP3 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp3_mux), + SND_SOC_DAPM_MUX("IIR1 INP0 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp0_mux), + SND_SOC_DAPM_MUX("IIR1 INP1 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp1_mux), + SND_SOC_DAPM_MUX("IIR1 INP2 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp2_mux), + SND_SOC_DAPM_MUX("IIR1 INP3 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp3_mux), + + SND_SOC_DAPM_PGA_E("IIR0", WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B1_CTL, + 0, 0, NULL, 0, wcd934x_codec_set_iir_gain, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("IIR1", WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B1_CTL, + 1, 0, NULL, 0, wcd934x_codec_set_iir_gain, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER("SRC0", WCD934X_CDC_SIDETONE_SRC0_ST_SRC_PATH_CTL, + 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SRC1", WCD934X_CDC_SIDETONE_SRC1_ST_SRC_PATH_CTL, + 4, 0, NULL, 0), + SND_SOC_DAPM_MUX("RX INT0 DEM MUX", SND_SOC_NOPM, 0, 0, + &rx_int0_dem_inp_mux), + SND_SOC_DAPM_MUX("RX INT1 DEM MUX", SND_SOC_NOPM, 0, 0, + &rx_int1_dem_inp_mux), + SND_SOC_DAPM_MUX("RX INT2 DEM MUX", SND_SOC_NOPM, 0, 0, + &rx_int2_dem_inp_mux), + + SND_SOC_DAPM_MUX_E("RX INT0_1 INTERP", SND_SOC_NOPM, INTERP_EAR, 0, + &rx_int0_1_interp_mux, + wcd934x_codec_enable_main_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT1_1 INTERP", SND_SOC_NOPM, INTERP_HPHL, 0, + &rx_int1_1_interp_mux, + wcd934x_codec_enable_main_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT2_1 INTERP", SND_SOC_NOPM, INTERP_HPHR, 0, + &rx_int2_1_interp_mux, + wcd934x_codec_enable_main_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT3_1 INTERP", SND_SOC_NOPM, INTERP_LO1, 0, + &rx_int3_1_interp_mux, + wcd934x_codec_enable_main_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT4_1 INTERP", SND_SOC_NOPM, INTERP_LO2, 0, + &rx_int4_1_interp_mux, + wcd934x_codec_enable_main_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT7_1 INTERP", SND_SOC_NOPM, INTERP_SPKR1, 0, + &rx_int7_1_interp_mux, + wcd934x_codec_enable_main_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("RX INT8_1 INTERP", SND_SOC_NOPM, INTERP_SPKR2, 0, + &rx_int8_1_interp_mux, + wcd934x_codec_enable_main_path, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX("RX INT0_2 INTERP", SND_SOC_NOPM, 0, 0, + &rx_int0_2_interp_mux), + SND_SOC_DAPM_MUX("RX INT1_2 INTERP", SND_SOC_NOPM, 0, 0, + &rx_int1_2_interp_mux), + SND_SOC_DAPM_MUX("RX INT2_2 INTERP", SND_SOC_NOPM, 0, 0, + &rx_int2_2_interp_mux), + SND_SOC_DAPM_MUX("RX INT3_2 INTERP", SND_SOC_NOPM, 0, 0, + &rx_int3_2_interp_mux), + SND_SOC_DAPM_MUX("RX INT4_2 INTERP", SND_SOC_NOPM, 0, 0, + &rx_int4_2_interp_mux), + SND_SOC_DAPM_MUX("RX INT7_2 INTERP", SND_SOC_NOPM, 0, 0, + &rx_int7_2_interp_mux), + SND_SOC_DAPM_MUX("RX INT8_2 INTERP", SND_SOC_NOPM, 0, 0, + &rx_int8_2_interp_mux), + SND_SOC_DAPM_DAC_E("RX INT0 DAC", NULL, SND_SOC_NOPM, + 0, 0, wcd934x_codec_ear_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("RX INT1 DAC", NULL, WCD934X_ANA_HPH, + 5, 0, wcd934x_codec_hphl_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("RX INT2 DAC", NULL, WCD934X_ANA_HPH, + 4, 0, wcd934x_codec_hphr_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("RX INT3 DAC", NULL, SND_SOC_NOPM, + 0, 0, wcd934x_codec_lineout_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("RX INT4 DAC", NULL, SND_SOC_NOPM, + 0, 0, wcd934x_codec_lineout_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("EAR PA", WCD934X_ANA_EAR, 7, 0, NULL, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("HPHL PA", WCD934X_ANA_HPH, 7, 0, NULL, 0, + wcd934x_codec_enable_hphl_pa, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("HPHR PA", WCD934X_ANA_HPH, 6, 0, NULL, 0, + wcd934x_codec_enable_hphr_pa, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LINEOUT1 PA", WCD934X_ANA_LO_1_2, 7, 0, NULL, 0, + NULL, 0), + SND_SOC_DAPM_PGA_E("LINEOUT2 PA", WCD934X_ANA_LO_1_2, 6, 0, NULL, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("RX_BIAS", WCD934X_ANA_RX_SUPPLIES, 0, 0, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("SBOOST0", WCD934X_CDC_RX7_RX_PATH_CFG1, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SBOOST0_CLK", WCD934X_CDC_BOOST0_BOOST_PATH_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SBOOST1", WCD934X_CDC_RX8_RX_PATH_CFG1, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SBOOST1_CLK", WCD934X_CDC_BOOST1_BOOST_PATH_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("INT0_CLK", SND_SOC_NOPM, INTERP_EAR, 0, + wcd934x_codec_enable_interp_clk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("INT1_CLK", SND_SOC_NOPM, INTERP_HPHL, 0, + wcd934x_codec_enable_interp_clk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("INT2_CLK", SND_SOC_NOPM, INTERP_HPHR, 0, + wcd934x_codec_enable_interp_clk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("INT3_CLK", SND_SOC_NOPM, INTERP_LO1, 0, + wcd934x_codec_enable_interp_clk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("INT4_CLK", SND_SOC_NOPM, INTERP_LO2, 0, + wcd934x_codec_enable_interp_clk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("INT7_CLK", SND_SOC_NOPM, INTERP_SPKR1, 0, + wcd934x_codec_enable_interp_clk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("INT8_CLK", SND_SOC_NOPM, INTERP_SPKR2, 0, + wcd934x_codec_enable_interp_clk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DSMDEM0_CLK", WCD934X_CDC_RX0_RX_PATH_DSMDEM_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DSMDEM1_CLK", WCD934X_CDC_RX1_RX_PATH_DSMDEM_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DSMDEM2_CLK", WCD934X_CDC_RX2_RX_PATH_DSMDEM_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DSMDEM3_CLK", WCD934X_CDC_RX3_RX_PATH_DSMDEM_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DSMDEM4_CLK", WCD934X_CDC_RX4_RX_PATH_DSMDEM_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DSMDEM7_CLK", WCD934X_CDC_RX7_RX_PATH_DSMDEM_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DSMDEM8_CLK", WCD934X_CDC_RX8_RX_PATH_DSMDEM_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, + wcd934x_codec_enable_mclk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + static const struct snd_soc_component_driver wcd934x_component_drv = { .probe = wcd934x_comp_probe, .remove = wcd934x_comp_remove, .set_sysclk = wcd934x_comp_set_sysclk, .controls = wcd934x_snd_controls, .num_controls = ARRAY_SIZE(wcd934x_snd_controls), + .dapm_widgets = wcd934x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wcd934x_dapm_widgets), }; static int wcd934x_codec_parse_data(struct wcd934x_codec *wcd) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: wcd934x: add capture dapm widgets" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: wcd934x: add capture dapm widgets has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a70d9245759a48e57bb1dc9f63213dcf3017db32 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 19 Dec 2019 10:31:48 +0000 Subject: [PATCH] ASoC: wcd934x: add capture dapm widgets This patch adds required dapm widgets for capture path. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20191219103153.14875-7-srinivas.kandagatla@linaro… Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1151 +++++++++++++++++++++++++++++++++++- 1 file changed, 1149 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 556c051f7f99..24f78a0b7f5a 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -135,6 +135,13 @@ } \ } +enum { + MIC_BIAS_1 = 1, + MIC_BIAS_2, + MIC_BIAS_3, + MIC_BIAS_4 +}; + enum { SIDO_SOURCE_INTERNAL, SIDO_SOURCE_RCO_BG, @@ -500,6 +507,83 @@ static const char * const rx_int8_2_interp_mux_text[] = { "ZERO", "RX INT8_2 MUX", }; +static const char * const dmic_mux_text[] = { + "ZERO", "DMIC0", "DMIC1", "DMIC2", "DMIC3", "DMIC4", "DMIC5" +}; + +static const char * const amic_mux_text[] = { + "ZERO", "ADC1", "ADC2", "ADC3", "ADC4" +}; + +static const char * const amic4_5_sel_text[] = { + "AMIC4", "AMIC5" +}; + +static const char * const adc_mux_text[] = { + "DMIC", "AMIC", "ANC_FB_TUNE1", "ANC_FB_TUNE2" +}; + +static const char * const cdc_if_tx0_mux_text[] = { + "ZERO", "RX_MIX_TX0", "DEC0", "DEC0_192" +}; + +static const char * const cdc_if_tx1_mux_text[] = { + "ZERO", "RX_MIX_TX1", "DEC1", "DEC1_192" +}; + +static const char * const cdc_if_tx2_mux_text[] = { + "ZERO", "RX_MIX_TX2", "DEC2", "DEC2_192" +}; + +static const char * const cdc_if_tx3_mux_text[] = { + "ZERO", "RX_MIX_TX3", "DEC3", "DEC3_192" +}; + +static const char * const cdc_if_tx4_mux_text[] = { + "ZERO", "RX_MIX_TX4", "DEC4", "DEC4_192" +}; + +static const char * const cdc_if_tx5_mux_text[] = { + "ZERO", "RX_MIX_TX5", "DEC5", "DEC5_192" +}; + +static const char * const cdc_if_tx6_mux_text[] = { + "ZERO", "RX_MIX_TX6", "DEC6", "DEC6_192" +}; + +static const char * const cdc_if_tx7_mux_text[] = { + "ZERO", "RX_MIX_TX7", "DEC7", "DEC7_192" +}; + +static const char * const cdc_if_tx8_mux_text[] = { + "ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192" +}; + +static const char * const cdc_if_tx9_mux_text[] = { + "ZERO", "DEC7", "DEC7_192" +}; + +static const char * const cdc_if_tx10_mux_text[] = { + "ZERO", "DEC6", "DEC6_192" +}; + +static const char * const cdc_if_tx11_mux_text[] = { + "DEC_0_5", "DEC_9_12", "MAD_AUDIO", "MAD_BRDCST" +}; + +static const char * const cdc_if_tx11_inp1_mux_text[] = { + "ZERO", "DEC0", "DEC1", "DEC2", "DEC3", "DEC4", + "DEC5", "RX_MIX_TX5", "DEC9_10", "DEC11_12" +}; + +static const char * const cdc_if_tx13_mux_text[] = { + "CDC_DEC_5", "MAD_BRDCST" +}; + +static const char * const cdc_if_tx13_inp1_mux_text[] = { + "ZERO", "DEC5", "DEC5_192" +}; + static const struct soc_enum cf_dec0_enum = SOC_ENUM_SINGLE(WCD934X_CDC_TX0_TX_PATH_CFG0, 5, 3, cf_text); @@ -758,6 +842,35 @@ static const struct soc_enum rx_int2_dem_inp_mux_enum = SOC_ENUM_SINGLE(WCD934X_CDC_RX2_RX_PATH_SEC0, 0, ARRAY_SIZE(rx_int_dem_inp_mux_text), rx_int_dem_inp_mux_text); + +static const struct soc_enum tx_adc_mux0_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1, 0, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1, 0, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux2_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG1, 0, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux3_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1, 0, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux4_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1, 2, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux5_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1, 2, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux6_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG1, 2, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux7_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1, 2, + ARRAY_SIZE(adc_mux_text), adc_mux_text); +static const struct soc_enum tx_adc_mux8_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1, 4, + ARRAY_SIZE(adc_mux_text), adc_mux_text); + static const struct soc_enum rx_int0_1_interp_mux_enum = SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int0_1_interp_mux_text); @@ -803,8 +916,122 @@ static const struct soc_enum rx_int7_2_interp_mux_enum = static const struct soc_enum rx_int8_2_interp_mux_enum = SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int8_2_interp_mux_text); -static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, - int sido_src) +static const struct soc_enum tx_dmic_mux0_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux2_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux3_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux4_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux5_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX5_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux6_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX6_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux7_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX7_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_dmic_mux8_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX8_CFG0, 3, 7, + dmic_mux_text); + +static const struct soc_enum tx_amic_mux0_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux1_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux2_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux3_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux4_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux5_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX5_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux6_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX6_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux7_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX7_CFG0, 0, 5, + amic_mux_text); +static const struct soc_enum tx_amic_mux8_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX8_CFG0, 0, 5, + amic_mux_text); + +static const struct soc_enum tx_amic4_5_enum = + SOC_ENUM_SINGLE(WCD934X_TX_NEW_AMIC_4_5_SEL, 7, 2, amic4_5_sel_text); + +static const struct soc_enum cdc_if_tx0_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 0, + ARRAY_SIZE(cdc_if_tx0_mux_text), cdc_if_tx0_mux_text); +static const struct soc_enum cdc_if_tx1_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 2, + ARRAY_SIZE(cdc_if_tx1_mux_text), cdc_if_tx1_mux_text); +static const struct soc_enum cdc_if_tx2_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 4, + ARRAY_SIZE(cdc_if_tx2_mux_text), cdc_if_tx2_mux_text); +static const struct soc_enum cdc_if_tx3_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 6, + ARRAY_SIZE(cdc_if_tx3_mux_text), cdc_if_tx3_mux_text); +static const struct soc_enum cdc_if_tx4_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 0, + ARRAY_SIZE(cdc_if_tx4_mux_text), cdc_if_tx4_mux_text); +static const struct soc_enum cdc_if_tx5_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 2, + ARRAY_SIZE(cdc_if_tx5_mux_text), cdc_if_tx5_mux_text); +static const struct soc_enum cdc_if_tx6_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 4, + ARRAY_SIZE(cdc_if_tx6_mux_text), cdc_if_tx6_mux_text); +static const struct soc_enum cdc_if_tx7_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 6, + ARRAY_SIZE(cdc_if_tx7_mux_text), cdc_if_tx7_mux_text); +static const struct soc_enum cdc_if_tx8_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG2, 0, + ARRAY_SIZE(cdc_if_tx8_mux_text), cdc_if_tx8_mux_text); +static const struct soc_enum cdc_if_tx9_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG2, 2, + ARRAY_SIZE(cdc_if_tx9_mux_text), cdc_if_tx9_mux_text); +static const struct soc_enum cdc_if_tx10_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG2, 4, + ARRAY_SIZE(cdc_if_tx10_mux_text), cdc_if_tx10_mux_text); +static const struct soc_enum cdc_if_tx11_inp1_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG3, 0, + ARRAY_SIZE(cdc_if_tx11_inp1_mux_text), + cdc_if_tx11_inp1_mux_text); +static const struct soc_enum cdc_if_tx11_mux_enum = + SOC_ENUM_SINGLE(WCD934X_DATA_HUB_SB_TX11_INP_CFG, 0, + ARRAY_SIZE(cdc_if_tx11_mux_text), cdc_if_tx11_mux_text); +static const struct soc_enum cdc_if_tx13_inp1_mux_enum = + SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG3, 4, + ARRAY_SIZE(cdc_if_tx13_inp1_mux_text), + cdc_if_tx13_inp1_mux_text); +static const struct soc_enum cdc_if_tx13_mux_enum = + SOC_ENUM_SINGLE(WCD934X_DATA_HUB_SB_TX13_INP_CFG, 0, + ARRAY_SIZE(cdc_if_tx13_mux_text), cdc_if_tx13_mux_text); + +static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) { if (sido_src == wcd->sido_input_src) return 0; @@ -2241,6 +2468,63 @@ static int wcd934x_int_dem_inp_mux_put(struct snd_kcontrol *kc, return ret; } +static int wcd934x_dec_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val; + u16 mic_sel_reg = 0; + u8 mic_sel; + + comp = snd_soc_dapm_kcontrol_component(kcontrol); + + val = ucontrol->value.enumerated.item[0]; + if (val > e->items - 1) + return -EINVAL; + + switch (e->reg) { + case WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1: + if (e->shift_l == 0) + mic_sel_reg = WCD934X_CDC_TX0_TX_PATH_CFG0; + else if (e->shift_l == 2) + mic_sel_reg = WCD934X_CDC_TX4_TX_PATH_CFG0; + else if (e->shift_l == 4) + mic_sel_reg = WCD934X_CDC_TX8_TX_PATH_CFG0; + break; + case WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1: + if (e->shift_l == 0) + mic_sel_reg = WCD934X_CDC_TX1_TX_PATH_CFG0; + else if (e->shift_l == 2) + mic_sel_reg = WCD934X_CDC_TX5_TX_PATH_CFG0; + break; + case WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG1: + if (e->shift_l == 0) + mic_sel_reg = WCD934X_CDC_TX2_TX_PATH_CFG0; + else if (e->shift_l == 2) + mic_sel_reg = WCD934X_CDC_TX6_TX_PATH_CFG0; + break; + case WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1: + if (e->shift_l == 0) + mic_sel_reg = WCD934X_CDC_TX3_TX_PATH_CFG0; + else if (e->shift_l == 2) + mic_sel_reg = WCD934X_CDC_TX7_TX_PATH_CFG0; + break; + default: + dev_err(comp->dev, "%s: e->reg: 0x%x not expected\n", + __func__, e->reg); + return -EINVAL; + } + + /* ADC: 0, DMIC: 1 */ + mic_sel = val ? 0x0 : 0x1; + if (mic_sel_reg) + snd_soc_component_update_bits(comp, mic_sel_reg, BIT(7), + mic_sel << 7); + + return snd_soc_dapm_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rx_int0_2_mux = SOC_DAPM_ENUM("RX INT0_2 MUX Mux", rx_int0_2_mux_chain_enum); @@ -2453,6 +2737,252 @@ static const struct snd_kcontrol_new rx_int7_2_interp_mux = static const struct snd_kcontrol_new rx_int8_2_interp_mux = SOC_DAPM_ENUM("RX INT8_2 INTERP Mux", rx_int8_2_interp_mux_enum); +static const struct snd_kcontrol_new tx_dmic_mux0 = + SOC_DAPM_ENUM("DMIC MUX0 Mux", tx_dmic_mux0_enum); + +static const struct snd_kcontrol_new tx_dmic_mux1 = + SOC_DAPM_ENUM("DMIC MUX1 Mux", tx_dmic_mux1_enum); + +static const struct snd_kcontrol_new tx_dmic_mux2 = + SOC_DAPM_ENUM("DMIC MUX2 Mux", tx_dmic_mux2_enum); + +static const struct snd_kcontrol_new tx_dmic_mux3 = + SOC_DAPM_ENUM("DMIC MUX3 Mux", tx_dmic_mux3_enum); + +static const struct snd_kcontrol_new tx_dmic_mux4 = + SOC_DAPM_ENUM("DMIC MUX4 Mux", tx_dmic_mux4_enum); + +static const struct snd_kcontrol_new tx_dmic_mux5 = + SOC_DAPM_ENUM("DMIC MUX5 Mux", tx_dmic_mux5_enum); + +static const struct snd_kcontrol_new tx_dmic_mux6 = + SOC_DAPM_ENUM("DMIC MUX6 Mux", tx_dmic_mux6_enum); + +static const struct snd_kcontrol_new tx_dmic_mux7 = + SOC_DAPM_ENUM("DMIC MUX7 Mux", tx_dmic_mux7_enum); + +static const struct snd_kcontrol_new tx_dmic_mux8 = + SOC_DAPM_ENUM("DMIC MUX8 Mux", tx_dmic_mux8_enum); + +static const struct snd_kcontrol_new tx_amic_mux0 = + SOC_DAPM_ENUM("AMIC MUX0 Mux", tx_amic_mux0_enum); + +static const struct snd_kcontrol_new tx_amic_mux1 = + SOC_DAPM_ENUM("AMIC MUX1 Mux", tx_amic_mux1_enum); + +static const struct snd_kcontrol_new tx_amic_mux2 = + SOC_DAPM_ENUM("AMIC MUX2 Mux", tx_amic_mux2_enum); + +static const struct snd_kcontrol_new tx_amic_mux3 = + SOC_DAPM_ENUM("AMIC MUX3 Mux", tx_amic_mux3_enum); + +static const struct snd_kcontrol_new tx_amic_mux4 = + SOC_DAPM_ENUM("AMIC MUX4 Mux", tx_amic_mux4_enum); + +static const struct snd_kcontrol_new tx_amic_mux5 = + SOC_DAPM_ENUM("AMIC MUX5 Mux", tx_amic_mux5_enum); + +static const struct snd_kcontrol_new tx_amic_mux6 = + SOC_DAPM_ENUM("AMIC MUX6 Mux", tx_amic_mux6_enum); + +static const struct snd_kcontrol_new tx_amic_mux7 = + SOC_DAPM_ENUM("AMIC MUX7 Mux", tx_amic_mux7_enum); + +static const struct snd_kcontrol_new tx_amic_mux8 = + SOC_DAPM_ENUM("AMIC MUX8 Mux", tx_amic_mux8_enum); + +static const struct snd_kcontrol_new tx_amic4_5 = + SOC_DAPM_ENUM("AMIC4_5 SEL Mux", tx_amic4_5_enum); + +static const struct snd_kcontrol_new tx_adc_mux0_mux = + SOC_DAPM_ENUM_EXT("ADC MUX0 Mux", tx_adc_mux0_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux1_mux = + SOC_DAPM_ENUM_EXT("ADC MUX1 Mux", tx_adc_mux1_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux2_mux = + SOC_DAPM_ENUM_EXT("ADC MUX2 Mux", tx_adc_mux2_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux3_mux = + SOC_DAPM_ENUM_EXT("ADC MUX3 Mux", tx_adc_mux3_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux4_mux = + SOC_DAPM_ENUM_EXT("ADC MUX4 Mux", tx_adc_mux4_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux5_mux = + SOC_DAPM_ENUM_EXT("ADC MUX5 Mux", tx_adc_mux5_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux6_mux = + SOC_DAPM_ENUM_EXT("ADC MUX6 Mux", tx_adc_mux6_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux7_mux = + SOC_DAPM_ENUM_EXT("ADC MUX7 Mux", tx_adc_mux7_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); +static const struct snd_kcontrol_new tx_adc_mux8_mux = + SOC_DAPM_ENUM_EXT("ADC MUX8 Mux", tx_adc_mux8_enum, + snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put); + +static const struct snd_kcontrol_new cdc_if_tx0_mux = + SOC_DAPM_ENUM("CDC_IF TX0 MUX Mux", cdc_if_tx0_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx1_mux = + SOC_DAPM_ENUM("CDC_IF TX1 MUX Mux", cdc_if_tx1_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx2_mux = + SOC_DAPM_ENUM("CDC_IF TX2 MUX Mux", cdc_if_tx2_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx3_mux = + SOC_DAPM_ENUM("CDC_IF TX3 MUX Mux", cdc_if_tx3_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx4_mux = + SOC_DAPM_ENUM("CDC_IF TX4 MUX Mux", cdc_if_tx4_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx5_mux = + SOC_DAPM_ENUM("CDC_IF TX5 MUX Mux", cdc_if_tx5_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx6_mux = + SOC_DAPM_ENUM("CDC_IF TX6 MUX Mux", cdc_if_tx6_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx7_mux = + SOC_DAPM_ENUM("CDC_IF TX7 MUX Mux", cdc_if_tx7_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx8_mux = + SOC_DAPM_ENUM("CDC_IF TX8 MUX Mux", cdc_if_tx8_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx9_mux = + SOC_DAPM_ENUM("CDC_IF TX9 MUX Mux", cdc_if_tx9_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx10_mux = + SOC_DAPM_ENUM("CDC_IF TX10 MUX Mux", cdc_if_tx10_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx11_mux = + SOC_DAPM_ENUM("CDC_IF TX11 MUX Mux", cdc_if_tx11_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx11_inp1_mux = + SOC_DAPM_ENUM("CDC_IF TX11 INP1 MUX Mux", cdc_if_tx11_inp1_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx13_mux = + SOC_DAPM_ENUM("CDC_IF TX13 MUX Mux", cdc_if_tx13_mux_enum); +static const struct snd_kcontrol_new cdc_if_tx13_inp1_mux = + SOC_DAPM_ENUM("CDC_IF TX13 INP1 MUX Mux", cdc_if_tx13_inp1_mux_enum); + +static int slim_tx_mixer_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); + struct wcd934x_codec *wcd = dev_get_drvdata(dapm->dev); + struct soc_mixer_control *mixer = + (struct soc_mixer_control *)kc->private_value; + int port_id = mixer->shift; + + ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id]; + + return 0; +} + +static int slim_tx_mixer_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc); + struct wcd934x_codec *wcd = dev_get_drvdata(widget->dapm->dev); + struct snd_soc_dapm_update *update = NULL; + struct soc_mixer_control *mixer = + (struct soc_mixer_control *)kc->private_value; + int enable = ucontrol->value.integer.value[0]; + int dai_id = widget->shift; + int port_id = mixer->shift; + + /* only add to the list if value not set */ + if (enable == wcd->tx_port_value[port_id]) + return 0; + + wcd->tx_port_value[port_id] = enable; + + if (enable) + list_add_tail(&wcd->tx_chs[port_id].list, + &wcd->dai[dai_id].slim_ch_list); + else + list_del_init(&wcd->tx_chs[port_id].list); + + snd_soc_dapm_mixer_update_power(widget->dapm, kc, enable, update); + + return 0; +} + +static const struct snd_kcontrol_new aif1_slim_cap_mixer[] = { + SOC_SINGLE_EXT("SLIM TX0", SND_SOC_NOPM, WCD934X_TX0, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX1", SND_SOC_NOPM, WCD934X_TX1, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX2", SND_SOC_NOPM, WCD934X_TX2, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX3", SND_SOC_NOPM, WCD934X_TX3, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX4", SND_SOC_NOPM, WCD934X_TX4, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX5", SND_SOC_NOPM, WCD934X_TX5, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX6", SND_SOC_NOPM, WCD934X_TX6, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX7", SND_SOC_NOPM, WCD934X_TX7, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX8", SND_SOC_NOPM, WCD934X_TX8, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX9", SND_SOC_NOPM, WCD934X_TX9, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX10", SND_SOC_NOPM, WCD934X_TX10, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX11", SND_SOC_NOPM, WCD934X_TX11, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX13", SND_SOC_NOPM, WCD934X_TX13, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), +}; + +static const struct snd_kcontrol_new aif2_slim_cap_mixer[] = { + SOC_SINGLE_EXT("SLIM TX0", SND_SOC_NOPM, WCD934X_TX0, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX1", SND_SOC_NOPM, WCD934X_TX1, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX2", SND_SOC_NOPM, WCD934X_TX2, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX3", SND_SOC_NOPM, WCD934X_TX3, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX4", SND_SOC_NOPM, WCD934X_TX4, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX5", SND_SOC_NOPM, WCD934X_TX5, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX6", SND_SOC_NOPM, WCD934X_TX6, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX7", SND_SOC_NOPM, WCD934X_TX7, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX8", SND_SOC_NOPM, WCD934X_TX8, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX9", SND_SOC_NOPM, WCD934X_TX9, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX10", SND_SOC_NOPM, WCD934X_TX10, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX11", SND_SOC_NOPM, WCD934X_TX11, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX13", SND_SOC_NOPM, WCD934X_TX13, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), +}; + +static const struct snd_kcontrol_new aif3_slim_cap_mixer[] = { + SOC_SINGLE_EXT("SLIM TX0", SND_SOC_NOPM, WCD934X_TX0, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX1", SND_SOC_NOPM, WCD934X_TX1, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX2", SND_SOC_NOPM, WCD934X_TX2, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX3", SND_SOC_NOPM, WCD934X_TX3, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX4", SND_SOC_NOPM, WCD934X_TX4, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX5", SND_SOC_NOPM, WCD934X_TX5, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX6", SND_SOC_NOPM, WCD934X_TX6, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX7", SND_SOC_NOPM, WCD934X_TX7, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX8", SND_SOC_NOPM, WCD934X_TX8, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX9", SND_SOC_NOPM, WCD934X_TX9, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX10", SND_SOC_NOPM, WCD934X_TX10, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX11", SND_SOC_NOPM, WCD934X_TX11, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), + SOC_SINGLE_EXT("SLIM TX13", SND_SOC_NOPM, WCD934X_TX13, 1, 0, + slim_tx_mixer_get, slim_tx_mixer_put), +}; + static const struct snd_kcontrol_new wcd934x_snd_controls[] = { /* Gain Controls */ SOC_SINGLE_TLV("EAR PA Volume", WCD934X_ANA_EAR, 4, 4, 1, ear_pa_gain), @@ -3193,6 +3723,455 @@ static int wcd934x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, return 0; } +static u32 wcd934x_get_dmic_sample_rate(struct snd_soc_component *comp, + unsigned int dmic, + struct wcd934x_codec *wcd) +{ + u8 tx_stream_fs; + u8 adc_mux_index = 0, adc_mux_sel = 0; + bool dec_found = false; + u16 adc_mux_ctl_reg, tx_fs_reg; + u32 dmic_fs; + + while (dec_found == 0 && adc_mux_index < WCD934X_MAX_VALID_ADC_MUX) { + if (adc_mux_index < 4) { + adc_mux_ctl_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 + + (adc_mux_index * 2); + } else if (adc_mux_index < WCD934X_INVALID_ADC_MUX) { + adc_mux_ctl_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 + + adc_mux_index - 4; + } else if (adc_mux_index == WCD934X_INVALID_ADC_MUX) { + ++adc_mux_index; + continue; + } + adc_mux_sel = ((snd_soc_component_read32(comp, adc_mux_ctl_reg) + & 0xF8) >> 3) - 1; + + if (adc_mux_sel == dmic) { + dec_found = true; + break; + } + + ++adc_mux_index; + } + + if (dec_found && adc_mux_index <= 8) { + tx_fs_reg = WCD934X_CDC_TX0_TX_PATH_CTL + (16 * adc_mux_index); + tx_stream_fs = snd_soc_component_read32(comp, tx_fs_reg) & 0x0F; + if (tx_stream_fs <= 4) { + if (wcd->dmic_sample_rate <= + WCD9XXX_DMIC_SAMPLE_RATE_2P4MHZ) + dmic_fs = wcd->dmic_sample_rate; + else + dmic_fs = WCD9XXX_DMIC_SAMPLE_RATE_2P4MHZ; + } else + dmic_fs = WCD9XXX_DMIC_SAMPLE_RATE_4P8MHZ; + } else { + dmic_fs = wcd->dmic_sample_rate; + } + + return dmic_fs; +} + +static u8 wcd934x_get_dmic_clk_val(struct snd_soc_component *comp, + u32 mclk_rate, u32 dmic_clk_rate) +{ + u32 div_factor; + u8 dmic_ctl_val; + + /* Default value to return in case of error */ + if (mclk_rate == WCD934X_MCLK_CLK_9P6MHZ) + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_2; + else + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_3; + + if (dmic_clk_rate == 0) { + dev_err(comp->dev, + "%s: dmic_sample_rate cannot be 0\n", + __func__); + goto done; + } + + div_factor = mclk_rate / dmic_clk_rate; + switch (div_factor) { + case 2: + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_2; + break; + case 3: + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_3; + break; + case 4: + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_4; + break; + case 6: + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_6; + break; + case 8: + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_8; + break; + case 16: + dmic_ctl_val = WCD934X_DMIC_CLK_DIV_16; + break; + default: + dev_err(comp->dev, + "%s: Invalid div_factor %u, clk_rate(%u), dmic_rate(%u)\n", + __func__, div_factor, mclk_rate, dmic_clk_rate); + break; + } + +done: + return dmic_ctl_val; +} + +static int wcd934x_codec_enable_dmic(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev); + u8 dmic_clk_en = 0x01; + u16 dmic_clk_reg; + s32 *dmic_clk_cnt; + u8 dmic_rate_val, dmic_rate_shift = 1; + unsigned int dmic; + u32 dmic_sample_rate; + int ret; + char *wname; + + wname = strpbrk(w->name, "012345"); + if (!wname) { + dev_err(comp->dev, "%s: widget not found\n", __func__); + return -EINVAL; + } + + ret = kstrtouint(wname, 10, &dmic); + if (ret < 0) { + dev_err(comp->dev, "%s: Invalid DMIC line on the codec\n", + __func__); + return -EINVAL; + } + + switch (dmic) { + case 0: + case 1: + dmic_clk_cnt = &wcd->dmic_0_1_clk_cnt; + dmic_clk_reg = WCD934X_CPE_SS_DMIC0_CTL; + break; + case 2: + case 3: + dmic_clk_cnt = &wcd->dmic_2_3_clk_cnt; + dmic_clk_reg = WCD934X_CPE_SS_DMIC1_CTL; + break; + case 4: + case 5: + dmic_clk_cnt = &wcd->dmic_4_5_clk_cnt; + dmic_clk_reg = WCD934X_CPE_SS_DMIC2_CTL; + break; + default: + dev_err(comp->dev, "%s: Invalid DMIC Selection\n", + __func__); + return -EINVAL; + }; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dmic_sample_rate = wcd934x_get_dmic_sample_rate(comp, dmic, + wcd); + dmic_rate_val = wcd934x_get_dmic_clk_val(comp, wcd->rate, + dmic_sample_rate); + (*dmic_clk_cnt)++; + if (*dmic_clk_cnt == 1) { + dmic_rate_val = dmic_rate_val << dmic_rate_shift; + snd_soc_component_update_bits(comp, dmic_clk_reg, + WCD934X_DMIC_RATE_MASK, + dmic_rate_val); + snd_soc_component_update_bits(comp, dmic_clk_reg, + dmic_clk_en, dmic_clk_en); + } + + break; + case SND_SOC_DAPM_POST_PMD: + (*dmic_clk_cnt)--; + if (*dmic_clk_cnt == 0) + snd_soc_component_update_bits(comp, dmic_clk_reg, + dmic_clk_en, 0); + break; + }; + + return 0; +} + +static int wcd934x_codec_find_amic_input(struct snd_soc_component *comp, + int adc_mux_n) +{ + u16 mask, shift, adc_mux_in_reg; + u16 amic_mux_sel_reg; + bool is_amic; + + if (adc_mux_n < 0 || adc_mux_n > WCD934X_MAX_VALID_ADC_MUX || + adc_mux_n == WCD934X_INVALID_ADC_MUX) + return 0; + + if (adc_mux_n < 3) { + adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1 + + adc_mux_n; + mask = 0x03; + shift = 0; + amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 + + 2 * adc_mux_n; + } else if (adc_mux_n < 4) { + adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1; + mask = 0x03; + shift = 0; + amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 + + 2 * adc_mux_n; + } else if (adc_mux_n < 7) { + adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1 + + (adc_mux_n - 4); + mask = 0x0C; + shift = 2; + amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 + + adc_mux_n - 4; + } else if (adc_mux_n < 8) { + adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1; + mask = 0x0C; + shift = 2; + amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 + + adc_mux_n - 4; + } else if (adc_mux_n < 12) { + adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1 + + ((adc_mux_n == 8) ? (adc_mux_n - 8) : + (adc_mux_n - 9)); + mask = 0x30; + shift = 4; + amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 + + adc_mux_n - 4; + } else if (adc_mux_n < 13) { + adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1; + mask = 0x30; + shift = 4; + amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 + + adc_mux_n - 4; + } else { + adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1; + mask = 0xC0; + shift = 6; + amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 + + adc_mux_n - 4; + } + + is_amic = (((snd_soc_component_read32(comp, adc_mux_in_reg) + & mask) >> shift) == 1); + if (!is_amic) + return 0; + + return snd_soc_component_read32(comp, amic_mux_sel_reg) & 0x07; +} + +static u16 wcd934x_codec_get_amic_pwlvl_reg(struct snd_soc_component *comp, + int amic) +{ + u16 pwr_level_reg = 0; + + switch (amic) { + case 1: + case 2: + pwr_level_reg = WCD934X_ANA_AMIC1; + break; + + case 3: + case 4: + pwr_level_reg = WCD934X_ANA_AMIC3; + break; + default: + break; + } + + return pwr_level_reg; +} + +static int wcd934x_codec_enable_dec(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + unsigned int decimator; + char *dec_adc_mux_name = NULL; + char *widget_name = NULL; + char *wname; + int ret = 0, amic_n; + u16 tx_vol_ctl_reg, pwr_level_reg = 0, dec_cfg_reg, hpf_gate_reg; + u16 tx_gain_ctl_reg; + char *dec; + u8 hpf_coff_freq; + + widget_name = kstrndup(w->name, 15, GFP_KERNEL); + if (!widget_name) + return -ENOMEM; + + wname = widget_name; + dec_adc_mux_name = strsep(&widget_name, " "); + if (!dec_adc_mux_name) { + dev_err(comp->dev, "%s: Invalid decimator = %s\n", + __func__, w->name); + ret = -EINVAL; + goto out; + } + dec_adc_mux_name = widget_name; + + dec = strpbrk(dec_adc_mux_name, "012345678"); + if (!dec) { + dev_err(comp->dev, "%s: decimator index not found\n", + __func__); + ret = -EINVAL; + goto out; + } + + ret = kstrtouint(dec, 10, &decimator); + if (ret < 0) { + dev_err(comp->dev, "%s: Invalid decimator = %s\n", + __func__, wname); + ret = -EINVAL; + goto out; + } + + tx_vol_ctl_reg = WCD934X_CDC_TX0_TX_PATH_CTL + 16 * decimator; + hpf_gate_reg = WCD934X_CDC_TX0_TX_PATH_SEC2 + 16 * decimator; + dec_cfg_reg = WCD934X_CDC_TX0_TX_PATH_CFG0 + 16 * decimator; + tx_gain_ctl_reg = WCD934X_CDC_TX0_TX_VOL_CTL + 16 * decimator; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + amic_n = wcd934x_codec_find_amic_input(comp, decimator); + if (amic_n) + pwr_level_reg = wcd934x_codec_get_amic_pwlvl_reg(comp, + amic_n); + + if (!pwr_level_reg) + break; + + switch ((snd_soc_component_read32(comp, pwr_level_reg) & + WCD934X_AMIC_PWR_LVL_MASK) >> + WCD934X_AMIC_PWR_LVL_SHIFT) { + case WCD934X_AMIC_PWR_LEVEL_LP: + snd_soc_component_update_bits(comp, dec_cfg_reg, + WCD934X_DEC_PWR_LVL_MASK, + WCD934X_DEC_PWR_LVL_LP); + break; + case WCD934X_AMIC_PWR_LEVEL_HP: + snd_soc_component_update_bits(comp, dec_cfg_reg, + WCD934X_DEC_PWR_LVL_MASK, + WCD934X_DEC_PWR_LVL_HP); + break; + case WCD934X_AMIC_PWR_LEVEL_DEFAULT: + case WCD934X_AMIC_PWR_LEVEL_HYBRID: + default: + snd_soc_component_update_bits(comp, dec_cfg_reg, + WCD934X_DEC_PWR_LVL_MASK, + WCD934X_DEC_PWR_LVL_DF); + break; + } + break; + case SND_SOC_DAPM_POST_PMU: + hpf_coff_freq = (snd_soc_component_read32(comp, dec_cfg_reg) & + TX_HPF_CUT_OFF_FREQ_MASK) >> 5; + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + CF_MIN_3DB_150HZ << 5); + snd_soc_component_update_bits(comp, hpf_gate_reg, + WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK, + WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ); + /* + * Minimum 1 clk cycle delay is required as per + * HW spec. + */ + usleep_range(1000, 1010); + snd_soc_component_update_bits(comp, hpf_gate_reg, + WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK, + 0); + } + /* apply gain after decimator is enabled */ + snd_soc_component_write(comp, tx_gain_ctl_reg, + snd_soc_component_read32(comp, + tx_gain_ctl_reg)); + break; + case SND_SOC_DAPM_PRE_PMD: + hpf_coff_freq = (snd_soc_component_read32(comp, dec_cfg_reg) & + TX_HPF_CUT_OFF_FREQ_MASK) >> 5; + + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + hpf_coff_freq << 5); + snd_soc_component_update_bits(comp, hpf_gate_reg, + WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK, + WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ); + /* + * Minimum 1 clk cycle delay is required as per + * HW spec. + */ + usleep_range(1000, 1010); + snd_soc_component_update_bits(comp, hpf_gate_reg, + WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK, + 0); + } + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(comp, tx_vol_ctl_reg, + 0x10, 0x00); + snd_soc_component_update_bits(comp, dec_cfg_reg, + WCD934X_DEC_PWR_LVL_MASK, + WCD934X_DEC_PWR_LVL_DF); + break; + }; +out: + kfree(wname); + return ret; +} + +static void wcd934x_codec_set_tx_hold(struct snd_soc_component *comp, + u16 amic_reg, bool set) +{ + u8 mask = 0x20; + u8 val; + + if (amic_reg == WCD934X_ANA_AMIC1 || + amic_reg == WCD934X_ANA_AMIC3) + mask = 0x40; + + val = set ? mask : 0x00; + + switch (amic_reg) { + case WCD934X_ANA_AMIC1: + case WCD934X_ANA_AMIC2: + snd_soc_component_update_bits(comp, WCD934X_ANA_AMIC2, + mask, val); + break; + case WCD934X_ANA_AMIC3: + case WCD934X_ANA_AMIC4: + snd_soc_component_update_bits(comp, WCD934X_ANA_AMIC4, + mask, val); + break; + default: + break; + } +} + +static int wcd934x_codec_enable_adc(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wcd934x_codec_set_tx_hold(comp, w->reg, true); + break; + default: + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = { /* Analog Outputs */ SND_SOC_DAPM_OUTPUT("EAR"), @@ -3529,6 +4508,174 @@ static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, wcd934x_codec_enable_mclk, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* TX */ + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("AMIC3"), + SND_SOC_DAPM_INPUT("AMIC4"), + SND_SOC_DAPM_INPUT("AMIC5"), + SND_SOC_DAPM_INPUT("DMIC0 Pin"), + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("DMIC3 Pin"), + SND_SOC_DAPM_INPUT("DMIC4 Pin"), + SND_SOC_DAPM_INPUT("DMIC5 Pin"), + + SND_SOC_DAPM_AIF_OUT_E("AIF1 CAP", "AIF1 Capture", 0, SND_SOC_NOPM, + AIF1_CAP, 0, wcd934x_codec_enable_slim, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT_E("AIF2 CAP", "AIF2 Capture", 0, SND_SOC_NOPM, + AIF2_CAP, 0, wcd934x_codec_enable_slim, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT_E("AIF3 CAP", "AIF3 Capture", 0, SND_SOC_NOPM, + AIF3_CAP, 0, wcd934x_codec_enable_slim, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("SLIM TX0", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX6", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX7", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX8", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX9", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX10", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX11", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SLIM TX13", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Mic Inputs */ + SND_SOC_DAPM_ADC_E("DMIC0", NULL, SND_SOC_NOPM, 0, 0, + wcd934x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC1", NULL, SND_SOC_NOPM, 0, 0, + wcd934x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC2", NULL, SND_SOC_NOPM, 0, 0, + wcd934x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC3", NULL, SND_SOC_NOPM, 0, 0, + wcd934x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC4", NULL, SND_SOC_NOPM, 0, 0, + wcd934x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC5", NULL, SND_SOC_NOPM, 0, 0, + wcd934x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_dmic_mux0), + SND_SOC_DAPM_MUX("DMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_dmic_mux1), + SND_SOC_DAPM_MUX("DMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_dmic_mux2), + SND_SOC_DAPM_MUX("DMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_dmic_mux3), + SND_SOC_DAPM_MUX("DMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_dmic_mux4), + SND_SOC_DAPM_MUX("DMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_dmic_mux5), + SND_SOC_DAPM_MUX("DMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_dmic_mux6), + SND_SOC_DAPM_MUX("DMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_dmic_mux7), + SND_SOC_DAPM_MUX("DMIC MUX8", SND_SOC_NOPM, 0, 0, &tx_dmic_mux8), + SND_SOC_DAPM_MUX("AMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_amic_mux0), + SND_SOC_DAPM_MUX("AMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_amic_mux1), + SND_SOC_DAPM_MUX("AMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_amic_mux2), + SND_SOC_DAPM_MUX("AMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_amic_mux3), + SND_SOC_DAPM_MUX("AMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_amic_mux4), + SND_SOC_DAPM_MUX("AMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_amic_mux5), + SND_SOC_DAPM_MUX("AMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_amic_mux6), + SND_SOC_DAPM_MUX("AMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_amic_mux7), + SND_SOC_DAPM_MUX("AMIC MUX8", SND_SOC_NOPM, 0, 0, &tx_amic_mux8), + SND_SOC_DAPM_MUX_E("ADC MUX0", WCD934X_CDC_TX0_TX_PATH_CTL, 5, 0, + &tx_adc_mux0_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX1", WCD934X_CDC_TX1_TX_PATH_CTL, 5, 0, + &tx_adc_mux1_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX2", WCD934X_CDC_TX2_TX_PATH_CTL, 5, 0, + &tx_adc_mux2_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX3", WCD934X_CDC_TX3_TX_PATH_CTL, 5, 0, + &tx_adc_mux3_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX4", WCD934X_CDC_TX4_TX_PATH_CTL, 5, 0, + &tx_adc_mux4_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX5", WCD934X_CDC_TX5_TX_PATH_CTL, 5, 0, + &tx_adc_mux5_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX6", WCD934X_CDC_TX6_TX_PATH_CTL, 5, 0, + &tx_adc_mux6_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX7", WCD934X_CDC_TX7_TX_PATH_CTL, 5, 0, + &tx_adc_mux7_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("ADC MUX8", WCD934X_CDC_TX8_TX_PATH_CTL, 5, 0, + &tx_adc_mux8_mux, wcd934x_codec_enable_dec, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADC1", NULL, WCD934X_ANA_AMIC1, 7, 0, + wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_ADC_E("ADC2", NULL, WCD934X_ANA_AMIC2, 7, 0, + wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_ADC_E("ADC3", NULL, WCD934X_ANA_AMIC3, 7, 0, + wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_ADC_E("ADC4", NULL, WCD934X_ANA_AMIC4, 7, 0, + wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("MIC BIAS1", WCD934X_ANA_MICB1, 6, 0, NULL, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("MIC BIAS2", WCD934X_ANA_MICB2, 6, 0, NULL, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("MIC BIAS3", WCD934X_ANA_MICB3, 6, 0, NULL, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("MIC BIAS4", WCD934X_ANA_MICB4, 6, 0, NULL, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX("AMIC4_5 SEL", SND_SOC_NOPM, 0, 0, &tx_amic4_5), + SND_SOC_DAPM_MUX("CDC_IF TX0 MUX", SND_SOC_NOPM, WCD934X_TX0, 0, + &cdc_if_tx0_mux), + SND_SOC_DAPM_MUX("CDC_IF TX1 MUX", SND_SOC_NOPM, WCD934X_TX1, 0, + &cdc_if_tx1_mux), + SND_SOC_DAPM_MUX("CDC_IF TX2 MUX", SND_SOC_NOPM, WCD934X_TX2, 0, + &cdc_if_tx2_mux), + SND_SOC_DAPM_MUX("CDC_IF TX3 MUX", SND_SOC_NOPM, WCD934X_TX3, 0, + &cdc_if_tx3_mux), + SND_SOC_DAPM_MUX("CDC_IF TX4 MUX", SND_SOC_NOPM, WCD934X_TX4, 0, + &cdc_if_tx4_mux), + SND_SOC_DAPM_MUX("CDC_IF TX5 MUX", SND_SOC_NOPM, WCD934X_TX5, 0, + &cdc_if_tx5_mux), + SND_SOC_DAPM_MUX("CDC_IF TX6 MUX", SND_SOC_NOPM, WCD934X_TX6, 0, + &cdc_if_tx6_mux), + SND_SOC_DAPM_MUX("CDC_IF TX7 MUX", SND_SOC_NOPM, WCD934X_TX7, 0, + &cdc_if_tx7_mux), + SND_SOC_DAPM_MUX("CDC_IF TX8 MUX", SND_SOC_NOPM, WCD934X_TX8, 0, + &cdc_if_tx8_mux), + SND_SOC_DAPM_MUX("CDC_IF TX9 MUX", SND_SOC_NOPM, WCD934X_TX9, 0, + &cdc_if_tx9_mux), + SND_SOC_DAPM_MUX("CDC_IF TX10 MUX", SND_SOC_NOPM, WCD934X_TX10, 0, + &cdc_if_tx10_mux), + SND_SOC_DAPM_MUX("CDC_IF TX11 MUX", SND_SOC_NOPM, WCD934X_TX11, 0, + &cdc_if_tx11_mux), + SND_SOC_DAPM_MUX("CDC_IF TX11 INP1 MUX", SND_SOC_NOPM, WCD934X_TX11, 0, + &cdc_if_tx11_inp1_mux), + SND_SOC_DAPM_MUX("CDC_IF TX13 MUX", SND_SOC_NOPM, WCD934X_TX13, 0, + &cdc_if_tx13_mux), + SND_SOC_DAPM_MUX("CDC_IF TX13 INP1 MUX", SND_SOC_NOPM, WCD934X_TX13, 0, + &cdc_if_tx13_inp1_mux), + SND_SOC_DAPM_MIXER("AIF1_CAP Mixer", SND_SOC_NOPM, AIF1_CAP, 0, + aif1_slim_cap_mixer, + ARRAY_SIZE(aif1_slim_cap_mixer)), + SND_SOC_DAPM_MIXER("AIF2_CAP Mixer", SND_SOC_NOPM, AIF2_CAP, 0, + aif2_slim_cap_mixer, + ARRAY_SIZE(aif2_slim_cap_mixer)), + SND_SOC_DAPM_MIXER("AIF3_CAP Mixer", SND_SOC_NOPM, AIF3_CAP, 0, + aif3_slim_cap_mixer, + ARRAY_SIZE(aif3_slim_cap_mixer)), }; static const struct snd_soc_component_driver wcd934x_component_drv = { -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: wcd934x: add audio routings" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: wcd934x: add audio routings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From da3e83f8bb866a91945ef5c47bdb25de189a381e Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 19 Dec 2019 10:31:49 +0000 Subject: [PATCH] ASoC: wcd934x: add audio routings This patch adds audio routing for both playback and capture. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20191219103153.14875-8-srinivas.kandagatla@linaro… Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 290 +++++++++++++++++++++++++++++++++++++ 1 file changed, 290 insertions(+) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 24f78a0b7f5a..158e878abd6c 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -135,6 +135,162 @@ } \ } +#define WCD934X_INTERPOLATOR_PATH(id) \ + {"RX INT" #id "_1 MIX1 INP0", "RX0", "SLIM RX0"}, \ + {"RX INT" #id "_1 MIX1 INP0", "RX1", "SLIM RX1"}, \ + {"RX INT" #id "_1 MIX1 INP0", "RX2", "SLIM RX2"}, \ + {"RX INT" #id "_1 MIX1 INP0", "RX3", "SLIM RX3"}, \ + {"RX INT" #id "_1 MIX1 INP0", "RX4", "SLIM RX4"}, \ + {"RX INT" #id "_1 MIX1 INP0", "RX5", "SLIM RX5"}, \ + {"RX INT" #id "_1 MIX1 INP0", "RX6", "SLIM RX6"}, \ + {"RX INT" #id "_1 MIX1 INP0", "RX7", "SLIM RX7"}, \ + {"RX INT" #id "_1 MIX1 INP0", "IIR0", "IIR0"}, \ + {"RX INT" #id "_1 MIX1 INP0", "IIR1", "IIR1"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX0", "SLIM RX0"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX1", "SLIM RX1"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX2", "SLIM RX2"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX3", "SLIM RX3"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX4", "SLIM RX4"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX5", "SLIM RX5"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX6", "SLIM RX6"}, \ + {"RX INT" #id "_1 MIX1 INP1", "RX7", "SLIM RX7"}, \ + {"RX INT" #id "_1 MIX1 INP1", "IIR0", "IIR0"}, \ + {"RX INT" #id "_1 MIX1 INP1", "IIR1", "IIR1"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX0", "SLIM RX0"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX1", "SLIM RX1"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX2", "SLIM RX2"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX3", "SLIM RX3"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX4", "SLIM RX4"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX5", "SLIM RX5"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX6", "SLIM RX6"}, \ + {"RX INT" #id "_1 MIX1 INP2", "RX7", "SLIM RX7"}, \ + {"RX INT" #id "_1 MIX1 INP2", "IIR0", "IIR0"}, \ + {"RX INT" #id "_1 MIX1 INP2", "IIR1", "IIR1"}, \ + {"RX INT" #id "_1 MIX1", NULL, "RX INT" #id "_1 MIX1 INP0"}, \ + {"RX INT" #id "_1 MIX1", NULL, "RX INT" #id "_1 MIX1 INP1"}, \ + {"RX INT" #id "_1 MIX1", NULL, "RX INT" #id "_1 MIX1 INP2"}, \ + {"RX INT" #id "_2 MUX", "RX0", "SLIM RX0"}, \ + {"RX INT" #id "_2 MUX", "RX1", "SLIM RX1"}, \ + {"RX INT" #id "_2 MUX", "RX2", "SLIM RX2"}, \ + {"RX INT" #id "_2 MUX", "RX3", "SLIM RX3"}, \ + {"RX INT" #id "_2 MUX", "RX4", "SLIM RX4"}, \ + {"RX INT" #id "_2 MUX", "RX5", "SLIM RX5"}, \ + {"RX INT" #id "_2 MUX", "RX6", "SLIM RX6"}, \ + {"RX INT" #id "_2 MUX", "RX7", "SLIM RX7"}, \ + {"RX INT" #id "_2 MUX", NULL, "INT" #id "_CLK"}, \ + {"RX INT" #id "_2 MUX", NULL, "DSMDEM" #id "_CLK"}, \ + {"RX INT" #id "_2 INTERP", NULL, "RX INT" #id "_2 MUX"}, \ + {"RX INT" #id " SEC MIX", NULL, "RX INT" #id "_2 INTERP"}, \ + {"RX INT" #id "_1 INTERP", NULL, "RX INT" #id "_1 MIX1"}, \ + {"RX INT" #id "_1 INTERP", NULL, "INT" #id "_CLK"}, \ + {"RX INT" #id "_1 INTERP", NULL, "DSMDEM" #id "_CLK"}, \ + {"RX INT" #id " SEC MIX", NULL, "RX INT" #id "_1 INTERP"} + +#define WCD934X_INTERPOLATOR_MIX2(id) \ + {"RX INT" #id " MIX2", NULL, "RX INT" #id " SEC MIX"}, \ + {"RX INT" #id " MIX2", NULL, "RX INT" #id " MIX2 INP"} + +#define WCD934X_SLIM_RX_AIF_PATH(id) \ + {"SLIM RX"#id" MUX", "AIF1_PB", "AIF1 PB"}, \ + {"SLIM RX"#id" MUX", "AIF2_PB", "AIF2 PB"}, \ + {"SLIM RX"#id" MUX", "AIF3_PB", "AIF3 PB"}, \ + {"SLIM RX"#id" MUX", "AIF4_PB", "AIF4 PB"}, \ + {"SLIM RX"#id, NULL, "SLIM RX"#id" MUX"} + +#define WCD934X_ADC_MUX(id) \ + {"ADC MUX" #id, "DMIC", "DMIC MUX" #id }, \ + {"ADC MUX" #id, "AMIC", "AMIC MUX" #id }, \ + {"DMIC MUX" #id, "DMIC0", "DMIC0"}, \ + {"DMIC MUX" #id, "DMIC1", "DMIC1"}, \ + {"DMIC MUX" #id, "DMIC2", "DMIC2"}, \ + {"DMIC MUX" #id, "DMIC3", "DMIC3"}, \ + {"DMIC MUX" #id, "DMIC4", "DMIC4"}, \ + {"DMIC MUX" #id, "DMIC5", "DMIC5"}, \ + {"AMIC MUX" #id, "ADC1", "ADC1"}, \ + {"AMIC MUX" #id, "ADC2", "ADC2"}, \ + {"AMIC MUX" #id, "ADC3", "ADC3"}, \ + {"AMIC MUX" #id, "ADC4", "ADC4"} + +#define WCD934X_IIR_INP_MUX(id) \ + {"IIR" #id, NULL, "IIR" #id " INP0 MUX"}, \ + {"IIR" #id " INP0 MUX", "DEC0", "ADC MUX0"}, \ + {"IIR" #id " INP0 MUX", "DEC1", "ADC MUX1"}, \ + {"IIR" #id " INP0 MUX", "DEC2", "ADC MUX2"}, \ + {"IIR" #id " INP0 MUX", "DEC3", "ADC MUX3"}, \ + {"IIR" #id " INP0 MUX", "DEC4", "ADC MUX4"}, \ + {"IIR" #id " INP0 MUX", "DEC5", "ADC MUX5"}, \ + {"IIR" #id " INP0 MUX", "DEC6", "ADC MUX6"}, \ + {"IIR" #id " INP0 MUX", "DEC7", "ADC MUX7"}, \ + {"IIR" #id " INP0 MUX", "DEC8", "ADC MUX8"}, \ + {"IIR" #id " INP0 MUX", "RX0", "SLIM RX0"}, \ + {"IIR" #id " INP0 MUX", "RX1", "SLIM RX1"}, \ + {"IIR" #id " INP0 MUX", "RX2", "SLIM RX2"}, \ + {"IIR" #id " INP0 MUX", "RX3", "SLIM RX3"}, \ + {"IIR" #id " INP0 MUX", "RX4", "SLIM RX4"}, \ + {"IIR" #id " INP0 MUX", "RX5", "SLIM RX5"}, \ + {"IIR" #id " INP0 MUX", "RX6", "SLIM RX6"}, \ + {"IIR" #id " INP0 MUX", "RX7", "SLIM RX7"}, \ + {"IIR" #id, NULL, "IIR" #id " INP1 MUX"}, \ + {"IIR" #id " INP1 MUX", "DEC0", "ADC MUX0"}, \ + {"IIR" #id " INP1 MUX", "DEC1", "ADC MUX1"}, \ + {"IIR" #id " INP1 MUX", "DEC2", "ADC MUX2"}, \ + {"IIR" #id " INP1 MUX", "DEC3", "ADC MUX3"}, \ + {"IIR" #id " INP1 MUX", "DEC4", "ADC MUX4"}, \ + {"IIR" #id " INP1 MUX", "DEC5", "ADC MUX5"}, \ + {"IIR" #id " INP1 MUX", "DEC6", "ADC MUX6"}, \ + {"IIR" #id " INP1 MUX", "DEC7", "ADC MUX7"}, \ + {"IIR" #id " INP1 MUX", "DEC8", "ADC MUX8"}, \ + {"IIR" #id " INP1 MUX", "RX0", "SLIM RX0"}, \ + {"IIR" #id " INP1 MUX", "RX1", "SLIM RX1"}, \ + {"IIR" #id " INP1 MUX", "RX2", "SLIM RX2"}, \ + {"IIR" #id " INP1 MUX", "RX3", "SLIM RX3"}, \ + {"IIR" #id " INP1 MUX", "RX4", "SLIM RX4"}, \ + {"IIR" #id " INP1 MUX", "RX5", "SLIM RX5"}, \ + {"IIR" #id " INP1 MUX", "RX6", "SLIM RX6"}, \ + {"IIR" #id " INP1 MUX", "RX7", "SLIM RX7"}, \ + {"IIR" #id, NULL, "IIR" #id " INP2 MUX"}, \ + {"IIR" #id " INP2 MUX", "DEC0", "ADC MUX0"}, \ + {"IIR" #id " INP2 MUX", "DEC1", "ADC MUX1"}, \ + {"IIR" #id " INP2 MUX", "DEC2", "ADC MUX2"}, \ + {"IIR" #id " INP2 MUX", "DEC3", "ADC MUX3"}, \ + {"IIR" #id " INP2 MUX", "DEC4", "ADC MUX4"}, \ + {"IIR" #id " INP2 MUX", "DEC5", "ADC MUX5"}, \ + {"IIR" #id " INP2 MUX", "DEC6", "ADC MUX6"}, \ + {"IIR" #id " INP2 MUX", "DEC7", "ADC MUX7"}, \ + {"IIR" #id " INP2 MUX", "DEC8", "ADC MUX8"}, \ + {"IIR" #id " INP2 MUX", "RX0", "SLIM RX0"}, \ + {"IIR" #id " INP2 MUX", "RX1", "SLIM RX1"}, \ + {"IIR" #id " INP2 MUX", "RX2", "SLIM RX2"}, \ + {"IIR" #id " INP2 MUX", "RX3", "SLIM RX3"}, \ + {"IIR" #id " INP2 MUX", "RX4", "SLIM RX4"}, \ + {"IIR" #id " INP2 MUX", "RX5", "SLIM RX5"}, \ + {"IIR" #id " INP2 MUX", "RX6", "SLIM RX6"}, \ + {"IIR" #id " INP2 MUX", "RX7", "SLIM RX7"}, \ + {"IIR" #id, NULL, "IIR" #id " INP3 MUX"}, \ + {"IIR" #id " INP3 MUX", "DEC0", "ADC MUX0"}, \ + {"IIR" #id " INP3 MUX", "DEC1", "ADC MUX1"}, \ + {"IIR" #id " INP3 MUX", "DEC2", "ADC MUX2"}, \ + {"IIR" #id " INP3 MUX", "DEC3", "ADC MUX3"}, \ + {"IIR" #id " INP3 MUX", "DEC4", "ADC MUX4"}, \ + {"IIR" #id " INP3 MUX", "DEC5", "ADC MUX5"}, \ + {"IIR" #id " INP3 MUX", "DEC6", "ADC MUX6"}, \ + {"IIR" #id " INP3 MUX", "DEC7", "ADC MUX7"}, \ + {"IIR" #id " INP3 MUX", "DEC8", "ADC MUX8"}, \ + {"IIR" #id " INP3 MUX", "RX0", "SLIM RX0"}, \ + {"IIR" #id " INP3 MUX", "RX1", "SLIM RX1"}, \ + {"IIR" #id " INP3 MUX", "RX2", "SLIM RX2"}, \ + {"IIR" #id " INP3 MUX", "RX3", "SLIM RX3"}, \ + {"IIR" #id " INP3 MUX", "RX4", "SLIM RX4"}, \ + {"IIR" #id " INP3 MUX", "RX5", "SLIM RX5"}, \ + {"IIR" #id " INP3 MUX", "RX6", "SLIM RX6"}, \ + {"IIR" #id " INP3 MUX", "RX7", "SLIM RX7"} + +#define WCD934X_SLIM_TX_AIF_PATH(id) \ + {"AIF1_CAP Mixer", "SLIM TX" #id, "SLIM TX" #id }, \ + {"AIF2_CAP Mixer", "SLIM TX" #id, "SLIM TX" #id }, \ + {"AIF3_CAP Mixer", "SLIM TX" #id, "SLIM TX" #id }, \ + {"SLIM TX" #id, NULL, "CDC_IF TX" #id " MUX"} + enum { MIC_BIAS_1 = 1, MIC_BIAS_2, @@ -4678,6 +4834,138 @@ static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = { ARRAY_SIZE(aif3_slim_cap_mixer)), }; +static const struct snd_soc_dapm_route wcd934x_audio_map[] = { + /* RX0-RX7 */ + WCD934X_SLIM_RX_AIF_PATH(0), + WCD934X_SLIM_RX_AIF_PATH(1), + WCD934X_SLIM_RX_AIF_PATH(2), + WCD934X_SLIM_RX_AIF_PATH(3), + WCD934X_SLIM_RX_AIF_PATH(4), + WCD934X_SLIM_RX_AIF_PATH(5), + WCD934X_SLIM_RX_AIF_PATH(6), + WCD934X_SLIM_RX_AIF_PATH(7), + + /* RX0 Ear out */ + WCD934X_INTERPOLATOR_PATH(0), + WCD934X_INTERPOLATOR_MIX2(0), + {"RX INT0 DEM MUX", "CLSH_DSM_OUT", "RX INT0 MIX2"}, + {"RX INT0 DAC", NULL, "RX INT0 DEM MUX"}, + {"RX INT0 DAC", NULL, "RX_BIAS"}, + {"EAR PA", NULL, "RX INT0 DAC"}, + {"EAR", NULL, "EAR PA"}, + + /* RX1 Headphone left */ + WCD934X_INTERPOLATOR_PATH(1), + WCD934X_INTERPOLATOR_MIX2(1), + {"RX INT1 MIX3", NULL, "RX INT1 MIX2"}, + {"RX INT1 DEM MUX", "CLSH_DSM_OUT", "RX INT1 MIX3"}, + {"RX INT1 DAC", NULL, "RX INT1 DEM MUX"}, + {"RX INT1 DAC", NULL, "RX_BIAS"}, + {"HPHL PA", NULL, "RX INT1 DAC"}, + {"HPHL", NULL, "HPHL PA"}, + + /* RX2 Headphone right */ + WCD934X_INTERPOLATOR_PATH(2), + WCD934X_INTERPOLATOR_MIX2(2), + {"RX INT2 MIX3", NULL, "RX INT2 MIX2"}, + {"RX INT2 DEM MUX", "CLSH_DSM_OUT", "RX INT2 MIX3"}, + {"RX INT2 DAC", NULL, "RX INT2 DEM MUX"}, + {"RX INT2 DAC", NULL, "RX_BIAS"}, + {"HPHR PA", NULL, "RX INT2 DAC"}, + {"HPHR", NULL, "HPHR PA"}, + + /* RX3 HIFi LineOut1 */ + WCD934X_INTERPOLATOR_PATH(3), + WCD934X_INTERPOLATOR_MIX2(3), + {"RX INT3 MIX3", NULL, "RX INT3 MIX2"}, + {"RX INT3 DAC", NULL, "RX INT3 MIX3"}, + {"RX INT3 DAC", NULL, "RX_BIAS"}, + {"LINEOUT1 PA", NULL, "RX INT3 DAC"}, + {"LINEOUT1", NULL, "LINEOUT1 PA"}, + + /* RX4 HIFi LineOut2 */ + WCD934X_INTERPOLATOR_PATH(4), + WCD934X_INTERPOLATOR_MIX2(4), + {"RX INT4 MIX3", NULL, "RX INT4 MIX2"}, + {"RX INT4 DAC", NULL, "RX INT4 MIX3"}, + {"RX INT4 DAC", NULL, "RX_BIAS"}, + {"LINEOUT2 PA", NULL, "RX INT4 DAC"}, + {"LINEOUT2", NULL, "LINEOUT2 PA"}, + + /* RX7 Speaker Left Out PA */ + WCD934X_INTERPOLATOR_PATH(7), + WCD934X_INTERPOLATOR_MIX2(7), + {"RX INT7 CHAIN", NULL, "RX INT7 MIX2"}, + {"RX INT7 CHAIN", NULL, "RX_BIAS"}, + {"RX INT7 CHAIN", NULL, "SBOOST0"}, + {"RX INT7 CHAIN", NULL, "SBOOST0_CLK"}, + {"SPK1 OUT", NULL, "RX INT7 CHAIN"}, + + /* RX8 Speaker Right Out PA */ + WCD934X_INTERPOLATOR_PATH(8), + {"RX INT8 CHAIN", NULL, "RX INT8 SEC MIX"}, + {"RX INT8 CHAIN", NULL, "RX_BIAS"}, + {"RX INT8 CHAIN", NULL, "SBOOST1"}, + {"RX INT8 CHAIN", NULL, "SBOOST1_CLK"}, + {"SPK2 OUT", NULL, "RX INT8 CHAIN"}, + + /* Tx */ + {"AIF1 CAP", NULL, "AIF1_CAP Mixer"}, + {"AIF2 CAP", NULL, "AIF2_CAP Mixer"}, + {"AIF3 CAP", NULL, "AIF3_CAP Mixer"}, + + WCD934X_SLIM_TX_AIF_PATH(0), + WCD934X_SLIM_TX_AIF_PATH(1), + WCD934X_SLIM_TX_AIF_PATH(2), + WCD934X_SLIM_TX_AIF_PATH(3), + WCD934X_SLIM_TX_AIF_PATH(4), + WCD934X_SLIM_TX_AIF_PATH(5), + WCD934X_SLIM_TX_AIF_PATH(6), + WCD934X_SLIM_TX_AIF_PATH(7), + WCD934X_SLIM_TX_AIF_PATH(8), + + WCD934X_ADC_MUX(0), + WCD934X_ADC_MUX(1), + WCD934X_ADC_MUX(2), + WCD934X_ADC_MUX(3), + WCD934X_ADC_MUX(4), + WCD934X_ADC_MUX(5), + WCD934X_ADC_MUX(6), + WCD934X_ADC_MUX(7), + WCD934X_ADC_MUX(8), + + {"CDC_IF TX0 MUX", "DEC0", "ADC MUX0"}, + {"CDC_IF TX1 MUX", "DEC1", "ADC MUX1"}, + {"CDC_IF TX2 MUX", "DEC2", "ADC MUX2"}, + {"CDC_IF TX3 MUX", "DEC3", "ADC MUX3"}, + {"CDC_IF TX4 MUX", "DEC4", "ADC MUX4"}, + {"CDC_IF TX5 MUX", "DEC5", "ADC MUX5"}, + {"CDC_IF TX6 MUX", "DEC6", "ADC MUX6"}, + {"CDC_IF TX7 MUX", "DEC7", "ADC MUX7"}, + {"CDC_IF TX8 MUX", "DEC8", "ADC MUX8"}, + + {"AMIC4_5 SEL", "AMIC4", "AMIC4"}, + {"AMIC4_5 SEL", "AMIC5", "AMIC5"}, + + { "DMIC0", NULL, "DMIC0 Pin" }, + { "DMIC1", NULL, "DMIC1 Pin" }, + { "DMIC2", NULL, "DMIC2 Pin" }, + { "DMIC3", NULL, "DMIC3 Pin" }, + { "DMIC4", NULL, "DMIC4 Pin" }, + { "DMIC5", NULL, "DMIC5 Pin" }, + + {"ADC1", NULL, "AMIC1"}, + {"ADC2", NULL, "AMIC2"}, + {"ADC3", NULL, "AMIC3"}, + {"ADC4", NULL, "AMIC4_5 SEL"}, + + WCD934X_IIR_INP_MUX(0), + WCD934X_IIR_INP_MUX(1), + + {"SRC0", NULL, "IIR0"}, + {"SRC1", NULL, "IIR1"}, +}; + static const struct snd_soc_component_driver wcd934x_component_drv = { .probe = wcd934x_comp_probe, .remove = wcd934x_comp_remove, @@ -4686,6 +4974,8 @@ static const struct snd_soc_component_driver wcd934x_component_drv = { .num_controls = ARRAY_SIZE(wcd934x_snd_controls), .dapm_widgets = wcd934x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wcd934x_dapm_widgets), + .dapm_routes = wcd934x_audio_map, + .num_dapm_routes = ARRAY_SIZE(wcd934x_audio_map), }; static int wcd934x_codec_parse_data(struct wcd934x_codec *wcd) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 834d899794f4ec3bbf9836c85e8f76b7de6e8a59 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 19 Dec 2019 10:31:52 +0000 Subject: [PATCH] ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga This patch adds compatible strings for DB845c and Lenovo Yoga C630 soundcard. Based on this compatible strings common machine driver will be in better position to setup board specific configuration. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Acked-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20191219103153.14875-11-srinivas.kandagatla@linar… Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/qcom,sdm845.txt | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt index 408c4837e6d5..ca8c89e88bfa 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt +++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt @@ -5,7 +5,10 @@ This binding describes the SDM845 sound card, which uses qdsp for audio. - compatible: Usage: required Value type: <stringlist> - Definition: must be "qcom,sdm845-sndcard" + Definition: must be one of this + "qcom,sdm845-sndcard" + "qcom,db845c-sndcard" + "lenovo,yoga-c630-sndcard" - audio-routing: Usage: Optional -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5caf64c633a3564f70e734868254281b25932fc0 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 19 Dec 2019 10:31:53 +0000 Subject: [PATCH] ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga This patch adds support to Lenovo Yoga c630 compatible strings and related setup to the sound machine driver. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20191219103153.14875-12-srinivas.kandagatla@linar… Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 86 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 85 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 28f3cef696e6..3b5547a27aad 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -24,6 +24,9 @@ #define RIGHT_SPK_TDM_TX_MASK 0xC0 #define SPK_TDM_RX_MASK 0x03 #define NUM_TDM_SLOTS 8 +#define SLIM_MAX_TX_PORTS 16 +#define SLIM_MAX_RX_PORTS 16 +#define WCD934X_DEFAULT_MCLK_RATE 9600000 struct sdm845_snd_data { struct snd_soc_jack jack; @@ -36,6 +39,39 @@ struct sdm845_snd_data { static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; +static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; + u32 rx_ch_cnt = 0, tx_ch_cnt = 0; + int ret = 0, i; + + for (i = 0 ; i < dai_link->num_codecs; i++) { + ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i], + &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); + + if (ret != 0 && ret != -ENOTSUPP) { + pr_err("failed to get codec chan map, err:%d\n", ret); + return ret; + } else if (ret == -ENOTSUPP) { + /* Ignore unsupported */ + continue; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, + rx_ch_cnt, rx_ch); + else + ret = snd_soc_dai_set_channel_map(cpu_dai, tx_ch_cnt, + tx_ch, 0, NULL); + } + + return 0; +} + static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -151,6 +187,11 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, case QUATERNARY_TDM_TX_0: ret = sdm845_tdm_snd_hw_params(substream, params); break; + case SLIMBUS_0_RX...SLIMBUS_6_TX: + ret = sdm845_slim_snd_hw_params(substream, params); + break; + case QUATERNARY_MI2S_RX: + break; default: pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); break; @@ -173,7 +214,20 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); struct snd_jack *jack; - int rval; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + /* + * Codec SLIMBUS configuration + * RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13 + * TX1, TX2, TX3, TX4, TX5, TX6, TX7, TX8, TX9, TX10, TX11, TX12, TX13 + * TX14, TX15, TX16 + */ + unsigned int rx_ch[SLIM_MAX_RX_PORTS] = {144, 145, 146, 147, 148, 149, + 150, 151, 152, 153, 154, 155, 156}; + unsigned int tx_ch[SLIM_MAX_TX_PORTS] = {128, 129, 130, 131, 132, 133, + 134, 135, 136, 137, 138, 139, + 140, 141, 142, 143}; + int rval, i; + if (!pdata->jack_setup) { rval = snd_soc_card_jack_new(card, "Headset Jack", @@ -211,6 +265,21 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) return rval; } break; + case SLIMBUS_0_RX...SLIMBUS_6_TX: + for (i = 0 ; i < dai_link->num_codecs; i++) { + rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i], + ARRAY_SIZE(tx_ch), + tx_ch, + ARRAY_SIZE(rx_ch), + rx_ch); + if (rval != 0 && rval != -ENOTSUPP) + return rval; + + snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0, + WCD934X_DEFAULT_MCLK_RATE, + SNDRV_PCM_STREAM_PLAYBACK); + } + break; default: break; } @@ -256,6 +325,14 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) } snd_soc_dai_set_fmt(cpu_dai, fmt); snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); + break; + case QUATERNARY_MI2S_RX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + + break; case QUATERNARY_TDM_RX_0: @@ -294,6 +371,8 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) } } break; + case SLIMBUS_0_RX...SLIMBUS_6_TX: + break; default: pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); @@ -338,6 +417,9 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) 0, SNDRV_PCM_STREAM_PLAYBACK); } break; + case SLIMBUS_0_RX...SLIMBUS_6_TX: + case QUATERNARY_MI2S_RX: + break; default: pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); @@ -451,6 +533,8 @@ static int sdm845_snd_platform_remove(struct platform_device *pdev) static const struct of_device_id sdm845_snd_device_id[] = { { .compatible = "qcom,sdm845-sndcard" }, + { .compatible = "qcom,db845c-sndcard" }, + { .compatible = "lenovo,yoga-c630-sndcard" }, {}, }; MODULE_DEVICE_TABLE(of, sdm845_snd_device_id); -- 2.20.1

1 0

Re: [alsa-devel] [PATCH] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Mark Brown 25 Dec '19

25 Dec '19

On Mon, Dec 23, 2019 at 02:10:12AM +0000, jeff_chang(張世佳) wrote: > --> When I check other driver at sound/soc/codecs/ folder, I just follow what others do. > It seems in .h --> /* SPDX-Liencese-Identifier: GPL-2.0 */ > In .c --> // SPDK-Liencese-Identifier: GPL-2.0 > Is it correct? Yes, headers use C style comments and source files use C++ style.

2 2

[alsa-devel] [PATCH 1/1] [PATCH] ALSA: usb-audio: Add support for Presonus Studio 1810c
by mickflemm＠gmail.com 25 Dec '19

25 Dec '19

This patch adds support for Presonus Studio 1810c, a usb interface that's UAC2 compliant with a few quirks and a few extra hw-specific controls. I've tested all 3 altsettings and the added switch controls and they work as expected. More infos on the card: https://www.presonus.com/products/Studio-1810c Note that this work is based on packet inspection with usbmon. I just wanted to get this card to work for using it on our open-source radio station: https://github.com/UoC-Radio Signed-off-by: Nick Kossifidis <mickflemm(a)gmail.com> --- sound/usb/Makefile | 1 + sound/usb/format.c | 17 ++ sound/usb/mixer_quirks.c | 5 + sound/usb/mixer_s1810c.c | 565 +++++++++++++++++++++++++++++++++++++++ sound/usb/mixer_s1810c.h | 6 + sound/usb/quirks.c | 34 +++ 6 files changed, 628 insertions(+) create mode 100644 sound/usb/mixer_s1810c.c create mode 100644 sound/usb/mixer_s1810c.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 78edd7d2f..56031026b 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_scarlett_gen2.o \ mixer_us16x08.o \ + mixer_s1810c.o \ pcm.o \ power.o \ proc.o \ diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db7130..edf3f2a55 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -262,6 +262,23 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, } for (rate = min; rate <= max; rate += res) { + + /* + * Presonus Studio 1810c anounces invalid + * sampling rates for its streams. + */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + ((rate > 48000 && fp->altsetting == 1) || + ((rate < 88200 || rate > 96000) + && fp->altsetting == 2) || + ((rate < 176400 || rate > 192000) + && fp->altsetting == 3))) { + if (res) + continue; + else + break; + } + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 39e27ae6c..71e705fab 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -34,6 +34,7 @@ #include "mixer_scarlett.h" #include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" +#include "mixer_s1810c.h" #include "helper.h" struct std_mono_table { @@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fd4): /* RME */ err = snd_rme_controls_create(mixer); break; + + case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + err = snd_sc1810_init_mixer(mixer); + break; } return err; diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c new file mode 100644 index 000000000..ff86e4aab --- /dev/null +++ b/sound/usb/mixer_s1810c.c @@ -0,0 +1,565 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm(a)gmail.com> + * + * Based on reverse engineering of the communication protocol + * between the windows driver / Univeral Control (UC) program + * and the device, through usbmon. + * + * For now this bypasses the mixer, with all channels split, + * so that the software can mix with greater flexibility. + * It also adds controls for the 4 buttons on the front of + * the device. + */ +#include <linux/usb.h> +#include <linux/usb/audio-v2.h> +#include <sound/control.h> +#include <linux/slab.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "helper.h" + +#define SC1810C_CMD_REQ 160 +#define SC1810C_CMD_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT) +#define SC1810C_CMD_F1 0x50617269 +#define SC1810C_CMD_F2 0x14 + +/* + * DISCLAIMER: These are just guesses based on the + * dumps I got. + * + * It seems like a selects between + * device (0), mixer (0x64) and output (0x65) + * + * For mixer (0x64): + * * b selects an input channel (see below). + * * c selects an output channel pair (see below). + * * d selects left (0) or right (1) of that pair. + * * e 0-> disconnect, 0x01000000-> connect, + * 0x0109-> used for stereo-linking channels, + * e is also used for setting volume levels + * in which case b is also set so I guess + * this way it is possible to set the volume + * level from the specified input to the + * specified output. + * + * IN Channels: + * 0 - 7 Mic/Inst/Line (Analog inputs) + * 8 - 9 S/PDIF + * 10 - 17 ADAT + * 18 - 35 DAW (Inputs from the host) + * + * OUT Channels (pairs): + * 0 -> Main out + * 1 -> Line1/2 + * 2 -> Line3/4 + * 3 -> S/PDIF + * 4 -> ADAT? + * + * For device (0): + * * b and c are not used, at least not on the + * dumps I got. + * * d sets the control id to be modified + * (see below). + * * e sets the setting for that control. + * (so for the switches I was interested + * in it's 0/1) + * + * For output (0x65): + * * b is the output channel (see above). + * * c is zero. + * * e I guess the same as with mixer except 0x0109 + * which I didn't see in my dumps. + * + * The two fixed fields have the same values for + * mixer and output but a different set for device. + */ +struct s1810c_ctl_packet { + uint32_t a; + uint32_t b; + uint32_t fixed1; + uint32_t fixed2; + uint32_t c; + uint32_t d; + uint32_t e; +}; + +#define SC1810C_CTL_LINE_SW 0 +#define SC1810C_CTL_MUTE_SW 1 +#define SC1810C_CTL_AB_SW 3 +#define SC1810C_CTL_48V_SW 4 + +#define SC1810C_SET_STATE_REQ 161 +#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE +#define SC1810C_SET_STATE_F1 0x64656D73 +#define SC1810C_SET_STATE_F2 0xF4 + +#define SC1810C_GET_STATE_REQ 162 +#define SC1810C_GET_STATE_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN) +#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1 +#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2 + +#define SC1810C_STATE_F1_IDX 2 +#define SC1810C_STATE_F2_IDX 3 + +/* + * This packet includes mixer volumes and + * various other fields, it's an extended + * version of ctl_packet, with a and b + * being zero and different f1/f2. + */ +struct s1810c_state_packet { + uint32_t fields[63]; +}; + +#define SC1810C_STATE_48V_SW 58 +#define SC1810C_STATE_LINE_SW 59 +#define SC1810C_STATE_MUTE_SW 60 +#define SC1810C_STATE_AB_SW 62 + +struct s1810_mixer_state { + uint16_t seqnum; + struct mutex usb_mutex; +}; + +static int +snd_s1810c_send_ctl_packet(struct usb_device *dev, uint32_t a, + uint32_t b, uint32_t c, uint32_t d, uint32_t e) +{ + struct s1810c_ctl_packet pkt = { 0 }; + int ret = 0; + + pkt.fixed1 = SC1810C_CMD_F1; + pkt.fixed2 = SC1810C_CMD_F2; + + pkt.a = a; + pkt.b = b; + pkt.c = c; + pkt.d = d; + /* + * Value for settings 0/1 for this + * output channel is always 0 (probably because + * there is no ADAT output on 1810c) + */ + pkt.e = (c == 4) ? 0 : e; + + ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + SC1810C_CMD_REQ, + SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send ctl packet\n"); + return ret; + } + return 0; +} + +/* + * When opening Universal Control the program periodicaly + * sends and receives state packets for syncinc state between + * the device and the host. + * + * Note that if we send only the request to get data back we'll + * get an error, we need to first send an empty state packet and + * then ask to receive a filled. Their seqnumbers must also match. + */ +static int +snd_sc1810c_get_status_field(struct usb_device *dev, + uint32_t *field, int field_idx, uint16_t *seqnum) +{ + struct s1810c_state_packet pkt_out = { 0 }; + struct s1810c_state_packet pkt_in = { 0 }; + int ret = 0; + + pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; + pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2; + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_SET_STATE_REQ, + SC1810C_SET_STATE_REQTYPE, + (*seqnum), 0, &pkt_out, sizeof(pkt_out)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send state packet (%d)\n", ret); + return ret; + } + + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_GET_STATE_REQ, + SC1810C_GET_STATE_REQTYPE, + (*seqnum), 0, &pkt_in, sizeof(pkt_in)); + if (ret < 0) { + dev_warn(&dev->dev, "could not get state field %u (%d)\n", + field_idx, ret); + return ret; + } + + (*field) = pkt_in.fields[field_idx]; + (*seqnum)++; + return 0; +} + +/* + * This is what I got when bypassing the mixer with + * all channels split. I'm not 100% sure of what's going + * on, I could probably clean this up based on my observations + * but I prefer to keep the same behavior as the windows driver. + */ +static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) +{ + uint32_t a, b, c, e, n, off; + struct usb_device *dev = chip->dev; + + /* Set initial volume levels ? */ + a = 0x64; + e = 0xbc; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off, c = 0; b < 18 + off; b++) { + /* This channel to all outputs ? */ + for (c = 0; c <= 8; c++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + /* This channel to main output (again) */ + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + /* + * I noticed on UC that DAW channels have different + * initial volumes (they can go both above and below + * 0db), so this makes sense. + */ + e = 0xb53bf0; + } + + /* Connect analog outputs ? */ + a = 0x65; + e = 0x01000000; + for (b = 1; b < 3; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e); + + /* Set initial volume levels for S/PDIF mappings ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF output ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Connect all outputs (again) ? */ + a = 0x65; + e = 0x01000000; + for (b = 0; b < 4; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + + /* Basic routing to get sound out of the device */ + a = 0x64; + e = 0x01000000; + for (c = 0; c < 4; c++) { + for (b = 0; b < 36; b++) { + if ((c == 0 && b == 18) || /* DAW1/2 -> Main */ + (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */ + (c == 2 && b == 22) || /* DAW4/5 -> Line5/4 */ + (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */ + /* Left */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + b++; + /* Right */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } else { + /* Leave the rest disconnected */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + } + } + } + + /* Set initial volume levels for S/PDIF (again) ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF outputs (again) ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Again ? */ + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + return 0; +} + +/* + * Sync state with the device and retrieve the requested field, + * whose index is specified in (kctl->private_value & 0xFF), + * from the received fields array. + */ +static int +snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, uint32_t *state) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + uint32_t field = 0; + uint32_t ctl_idx = (uint32_t) (kctl->private_value & 0xFF); + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_sc1810c_get_status_field(chip->dev, &field, + ctl_idx, &private->seqnum); + if (ret < 0) + goto unlock; + + *state = field; + unlock: + mutex_unlock(&private->usb_mutex); + return ret ? ret : 0; +} + +/* + * Send a control packet to the device for the control id + * specified in (kctl->private_value >> 8) with value + * specified in (kctl->private_value >> 16). + */ +static int +snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + uint32_t pval = (uint32_t) kctl->private_value; + uint32_t ctl_id = (pval >> 8) & 0xFF; + uint32_t ctl_val = (pval >> 16) & 0x1; + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val); + mutex_unlock(&private->usb_mutex); + return ret; +} + +/* Generic get/set/init functions for switch controls */ + +static int +snd_s1810c_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + uint32_t state = 0; + int ret = 0; + + ret = snd_s1810c_get_switch_state(mixer, kctl, &state); + if (ret < 0) + return ret; + + ctl_elem->value.integer.value[0] = (long)state; + + return 0; +} + +static int +snd_s1810c_switch_set(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + uint32_t curval = 0; + uint32_t newval = (uint32_t) ctl_elem->value.integer.value[0]; + int ret = 0; + + ret = snd_s1810c_get_switch_state(mixer, kctl, &curval); + if (ret < 0) + return 0; + + if (curval == newval) + return 0; + + kctl->private_value &= ~(0x1 << 16); + kctl->private_value |= (unsigned int)(newval & 0x1) << 16; + ret = snd_s1810c_set_switch_state(mixer, kctl); + if (ret < 0) + return 0; + + return 1; +} + +static int +snd_s1810c_switch_init(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *new_kctl) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + + elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = 0; + elem->head.id = 0; + elem->channels = 1; + + kctl = snd_ctl_new1(new_kctl, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + return snd_usb_mixer_add_control(&elem->head, kctl); +} + +static int +snd_s1810c_line_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { "Preamp on (Mic/Inst)", + "Preamp off (Line in)" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static struct snd_kcontrol_new snd_s1810c_line_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line 1/2 source type", + .info = snd_s1810c_line_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8) +}; + +static struct snd_kcontrol_new snd_s1810c_mute_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mute Main Out", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8) +}; + +static struct snd_kcontrol_new snd_s1810c_48v_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "48V Phantom power on Mic inputs", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8) +}; + +static int +snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { "1/2", + "3/4" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static struct snd_kcontrol_new snd_s1810c_ab_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone 1 source selector", + .info = snd_s1810c_ab_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8) +}; + +static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = mixer->private_data; + vfree(private); + mixer->private_data = NULL; +} + +/* Entry point, called from mixer_quirks.c */ +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = NULL; + struct snd_usb_audio *chip = mixer->chip; + struct usb_device *dev = chip->dev; + int ret = 0; + + /* Run this only once */ + if (!list_empty(&chip->mixer_list)) + return 0; + + dev_info(&dev->dev, + "Presonus Studio 1810c, device_setup: %u\n", chip->setup); + if (chip->setup == 1) + dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + else if (chip->setup == 2) + dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + else + dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + + ret = snd_s1810c_init_mixer_maps(chip); + if (ret < 0) + return ret; + + private = vzalloc(sizeof(struct s1810_mixer_state)); + if (!private) { + dev_err(&dev->dev, "could not allocate mixer state\n"); + return -ENOMEM; + } + + mutex_init(&private->usb_mutex); + + mixer->private_data = private; + mixer->private_free = snd_sc1810_mixer_state_free; + + private->seqnum = 1; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + if (ret < 0) + return ret; + return ret; +} diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h new file mode 100644 index 000000000..2a9ae0922 --- /dev/null +++ b/sound/usb/mixer_s1810c.h @@ -0,0 +1,6 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm(a)gmail.com> + */ +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 349e1e529..9a0ceaaf9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1216,6 +1216,36 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* + * Playback: + * 1: 6 Analog + 2 S/PDIF + * 2: 6 Analog + 2 S/PDIF + * 3: 6 Analog + * + * Capture: + * 1: 8 Analog + 2 S/PDIF + 8 ADAT + * 2: 8 Analog + 2 S/PDIF + 4 ADAT + * 3: 8 Analog + */ + + /* + * I'll leave 2 as the default one and + * use device_setup to switch to the + * other two. + */ + if (chip->setup == 1 && altno != 1) + return 1; + else if (chip->setup == 2 && altno != 3) + return 1; + else if (altno != 2) + return 1; + + return 0; +} + int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1229,6 +1259,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, /* fasttrackpro usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); + /* presonus studio 1810c: skip altsets incompatible with device_setup */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + return s1810c_skip_setting_quirk(chip, iface, altno); + return 0; } -- 2.23.0

1 0

[alsa-devel] [PATCH] ASoC: SOF: imx8: fix memory allocation failure check on priv->pd_dev
by Colin King 25 Dec '19

25 Dec '19

From: Colin Ian King <colin.king(a)canonical.com> The memory allocation failure check for priv->pd_dev is incorrectly pointer checking priv instead of priv->pd_dev. Fix this. Addresses-Coverity: ("Logically dead code") Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/imx/imx8.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index cfefcfd92798..9d926b1df0d7 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -209,7 +209,7 @@ static int imx8_probe(struct snd_sof_dev *sdev) priv->pd_dev = devm_kmalloc_array(&pdev->dev, priv->num_domains, sizeof(*priv->pd_dev), GFP_KERNEL); - if (!priv) + if (!priv->pd_dev) return -ENOMEM; priv->link = devm_kmalloc_array(&pdev->dev, priv->num_domains, -- 2.24.0

4 3

[alsa-devel] [PATCH] ASoC: soc-core: Set dpcm_playback / dpcm_capture
by Daniel Baluta 25 Dec '19

25 Dec '19

When converting a normal link to a DPCM link we need to set dpcm_playback / dpcm_capture otherwise playback/capture streams will not be created resulting in errors like this: [ 36.039111] sai1-wm8960-hifi: ASoC: no backend playback stream Fixes: a655de808cbde ("ASoC: core: Allow topology to override machine driver FE DAI link config") Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 977a7bfad519..f89cf9d0860c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1872,6 +1872,8 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) /* convert non BE into BE */ dai_link->no_pcm = 1; + dai_link->dpcm_playback = 1; + dai_link->dpcm_capture = 1; /* override any BE fixups */ dai_link->be_hw_params_fixup = -- 2.17.1

5 4

[alsa-devel] [PATCH 0/3] ASoC: stm32: spdifrx: fix race conditions
by Olivier Moysan 25 Dec '19

25 Dec '19

This patchset fixes some race conditions on STM32 SPDIFRX driver. Olivier Moysan (3): ASoC: stm32: spdifrx: fix inconsistent lock state ASoC: stm32: spdifrx: fix race condition in irq handler ASoC: stm32: spdifrx: fix input pin state management sound/soc/stm/stm32_spdifrx.c | 40 +++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 14 deletions(-) -- 2.17.1

2 6

[alsa-devel] Applied "ASoC: fsl_audmix: add missed pm_runtime_disable" to the asoc tree
by Mark Brown 25 Dec '19

25 Dec '19

The patch ASoC: fsl_audmix: add missed pm_runtime_disable has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 77fffa742285f2b587648d6c72b5c705633f146f Mon Sep 17 00:00:00 2001 From: Chuhong Yuan <hslester96(a)gmail.com> Date: Tue, 3 Dec 2019 19:13:03 +0800 Subject: [PATCH] ASoC: fsl_audmix: add missed pm_runtime_disable The driver forgets to call pm_runtime_disable in probe failure and remove. Add the missed calls to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20191203111303.12933-1-hslester96@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_audmix.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index a1db1bce330f..5faecbeb5497 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -505,15 +505,20 @@ static int fsl_audmix_probe(struct platform_device *pdev) ARRAY_SIZE(fsl_audmix_dai)); if (ret) { dev_err(dev, "failed to register ASoC DAI\n"); - return ret; + goto err_disable_pm; } priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); if (IS_ERR(priv->pdev)) { ret = PTR_ERR(priv->pdev); dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); + goto err_disable_pm; } + return 0; + +err_disable_pm: + pm_runtime_disable(dev); return ret; } @@ -521,6 +526,8 @@ static int fsl_audmix_remove(struct platform_device *pdev) { struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev); + pm_runtime_disable(&pdev->dev); + if (priv->pdev) platform_device_unregister(priv->pdev); -- 2.20.1

1 0

[alsa-devel] [PATCH 0/3] ASoC: SOF: fixes for kernel oopses/use-after-free
by Pierre-Louis Bossart 25 Dec '19

25 Dec '19

More stability fixes for corner cases. It wasn't straightforward to add a Fixes tag for the two Intel patches, it's likely issues that can be applied to 5.3, possibly earlier. For Dragos' patch Ranjani mentioned this may be due to da704f26ba376 ('ASoC: soc-core: merge snd_soc_remove_dai_link() and soc_unbind_dai_link()'), but Morimoto-san may need to confirm. Dragos Tarcatu (1): ASoC: topology: Prevent use-after-free in snd_soc_get_pcm_runtime() Kai Vehmanen (1): ASoC: SOF: fix fault at driver unload after failed probe Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda: hda-dai: fix oops on hda_link .hw_free sound/soc/soc-topology.c | 6 +++--- sound/soc/sof/intel/hda-dai.c | 11 +++++++++-- sound/soc/sof/ipc.c | 3 +++ 3 files changed, 15 insertions(+), 5 deletions(-) -- 2.20.1

3 5

[alsa-devel] [PATCH] ASoC: SOF: imx8: Fix dsp_box offset
by Pierre-Louis Bossart 25 Dec '19

25 Dec '19

From: Daniel Baluta <daniel.baluta(a)nxp.com> dsp_box is used to keep DSP initiated messages. The value of dsp_offset is set by the DSP with the first message, so we need a way to bootstrap it in order to get the first message. We do this by setting the correct default dsp_box offset which on i.MX8 is not zero. Very interesting is why it has worked until now. On i.MX8, DSP communicates with ARM core using a shared SDRAM memory area. Actually, there are two shared areas: * SDRAM0 - starting at 0x92400000, size 0x800000 * SDRAM1 - starting at 0x92C00000, size 0x800000 SDRAM0 keeps the data sections, starting with .rodata. By chance fw_ready structure was placed at the beginning of .rodata. dsp_box_base is defined as SDRAM0 + dsp_box_offset and it is placed at the beginning of SDRAM1 (dsp_box_offset should be 0x800000). But because it is zero initialized by default it points to SDRAM0 where by chance the fw_ready was placed in the SOF firmware. Anyhow, SOF commit 7466bee378dd811b ("clk: make freq arrays constant") fw_ready is no longer at the beginning of SDRAM0 and everything shows how lucky we were until now. Fix this by properly setting the default dsp_box offset. Fixes: 202acc565a1f050 ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx8.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index cfefcfd92798..dca85cf07ffa 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -304,6 +304,9 @@ static int imx8_probe(struct snd_sof_dev *sdev) } sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM; + /* set default mailbox offset for FW ready message */ + sdev->dsp_box.offset = MBOX_OFFSET; + return 0; exit_pdev_unregister: -- 2.20.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: SOF: Intel: add support for TigerLake 4th iDISP link
by Pierre-Louis Bossart 25 Dec '19

25 Dec '19

TigerLake has 4 iDISP connections, not 3 as in previous generations. The patches add the relevant DAI definition. This was tested as having no impact on previous generations. Support for 4 links with the legacy HDaudio driver is already planned and will be release soon by Kai. Sathyanarayana Nujella (2): ASoC: SOF: Intel: hda: Add iDisp4 DAI ASoC: hdac_hda: Update hdac hda dai table to include intel-hdmi-hifi4 sound/soc/codecs/hdac_hda.c | 16 ++++++++++++++++ sound/soc/codecs/hdac_hda.h | 3 ++- sound/soc/sof/intel/hda-dai.c | 4 ++++ 3 files changed, 22 insertions(+), 1 deletion(-) -- 2.20.1

2 4

[alsa-devel] [PATCH v4 0/2] Add the broadmobi BM818
by Angus Ainslie (Purism) 25 Dec '19

25 Dec '19

The broadmobi uses slightly different parameters from the option modems so add the paramters and document them. Changes since v3: Dropped unrelated indentation fix. Fixed const warning. Changes since v2: Use of_device_get_match_data to get the dai data. Updated subject styles. Changes since v1: Cut back the CC list. Use data from compatible match instead of explicit compatible match. Angus Ainslie (Purism) (2): ASoC: gtm601: add Broadmobi bm818 sound profile dt-bindings: sound: gtm601: add the broadmobi interface .../devicetree/bindings/sound/gtm601.txt | 10 +++++-- sound/soc/codecs/gtm601.c | 30 +++++++++++++++++-- 2 files changed, 35 insertions(+), 5 deletions(-) -- 2.17.1

2 4

[alsa-devel] [PATCH -next] ASoC: Intel: kbl_da7219_max98357a: remove unused variable 'constraints_16000' and 'ch_mono'
by YueHaibing 25 Dec '19

25 Dec '19

sound/soc/intel/boards/kbl_da7219_max98357a.c:343:48: warning: constraints_16000 defined but not used [-Wunused-const-variable=] sound/soc/intel/boards/kbl_da7219_max98357a.c:348:27: warning: ch_mono defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a889..0d55319 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -336,19 +336,6 @@ static struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: ctl: allow TLV read operation for callback type of element in locked case
by Takashi Sakamoto 24 Dec '19

24 Dec '19

A design of ALSA control core allows applications to execute three operations for TLV feature; read, write and command. Furthermore, it allows driver developers to process the operations by two ways; allocated array or callback function. In the former, read operation is just allowed, thus developers uses the latter when device driver supports variety of models or the target model is expected to dynamically change information stored in TLV container. The core also allows applications to lock any element so that the other applications can't perform write operation to the element for element value and TLV information. When the element is locked, write and command operation for TLV information are prohibited as well as element value. Any read operation should be allowed in the case. At present, when an element has callback function for TLV information, TLV read operation returns EPERM if the element is locked. On the other hand, the read operation is success when an element has allocated array for TLV information. In both cases, read operation is success for element value expectedly. This commit fixes the bug. This change can be backported to v4.14 kernel or later. --- sound/core/control.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 3fa1171dc1c2..49f3428dfacb 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1398,8 +1398,9 @@ static int call_tlv_handler(struct snd_ctl_file *file, int op_flag, if (kctl->tlv.c == NULL) return -ENXIO; - /* When locked, this is unavailable. */ - if (vd->owner != NULL && vd->owner != file) + // Write and command operations are not allowed for locked element. + if (op_flag != SNDRV_CTL_TLV_OP_READ && + vd->owner != NULL && vd->owner != file) return -EPERM; return kctl->tlv.c(kctl, op_flag, size, buf); -- 2.20.1

3 5

[alsa-devel] [PATCH] xfer-opitons: Fix -Wformat=2 warnings
by Rosen Penev 24 Dec '19

24 Dec '19

Allows the compiler to check types. format string functions expect constant expressions, not constant variables. Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- axfer/xfer-options.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/axfer/xfer-options.c b/axfer/xfer-options.c index 3740b16..d4e5ff2 100644 --- a/axfer/xfer-options.c +++ b/axfer/xfer-options.c @@ -422,8 +422,6 @@ static int generate_path_with_suffix(struct xfer_context *xfer, const char *template, unsigned int index, const char *suffix) { - static const char *const single_format = "%s%s"; - static const char *const multiple_format = "%s-%i%s"; unsigned int len; len = strlen(template) + strlen(suffix) + 1; @@ -435,10 +433,10 @@ static int generate_path_with_suffix(struct xfer_context *xfer, return -ENOMEM; if (xfer->path_count == 1) { - snprintf(xfer->paths[index], len, single_format, template, + snprintf(xfer->paths[index], len, "%s%s", template, suffix); } else { - snprintf(xfer->paths[index], len, multiple_format, template, + snprintf(xfer->paths[index], len, "%s-%i%s", template, index, suffix); } @@ -449,8 +447,6 @@ static int generate_path_without_suffix(struct xfer_context *xfer, const char *template, unsigned int index, const char *suffix) { - static const char *const single_format = "%s"; - static const char *const multiple_format = "%s-%i"; unsigned int len; len = strlen(template) + 1; @@ -462,9 +458,9 @@ static int generate_path_without_suffix(struct xfer_context *xfer, return -ENOMEM; if (xfer->path_count == 1) { - snprintf(xfer->paths[index], len, single_format, template); + snprintf(xfer->paths[index], len, "%s", template); } else { - snprintf(xfer->paths[index], len, multiple_format, template, + snprintf(xfer->paths[index], len, "%s-%i", template, index); } -- 2.24.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Apply sync-write workaround to old Intel platforms, too
by Takashi Iwai 24 Dec '19

24 Dec '19

Klaus Ethgen reported occasional high CPU usages in his system that seem caused by HD-audio driver. The perf output revealed that it's in the unsolicited event handling in the workqueue, and the problem seems triggered by some communication stall between the controller and the codec at the runtime or system resume. Actually a similar phenomenon was seen in the past for other Intel platforms, and we already applied the workaround to enforce sync-write for CORB/RIRB verbs for Skylake and newer chipsets (commit 2756d9143aa5 "ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips"). Fortunately, the same workaround is applicable to the old chipset, and the experiment showed the positive effect. Based on the experiment result, this patch enables the sync-write workaround for all Intel chipsets. The only reason I hesitated to apply this workaround was about the possibly slightly higher CPU usage. But if the lack of sync causes a much severer problem even for quite old chip, we should think this would be necessary for all Intel chips. Reported-by: Klaus Ethgen <Klaus(a)ethgen.ch> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20191223171833.GA17053@chua Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f69c8de64bd6..5b92f290cbb0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -282,12 +282,13 @@ enum { /* quirks for old Intel chipsets */ #define AZX_DCAPS_INTEL_ICH \ - (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE) + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\ + AZX_DCAPS_SYNC_WRITE) /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_BASE \ (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ - AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) /* PCH up to IVB; no runtime PM; bind with i915 gfx */ #define AZX_DCAPS_INTEL_PCH_NOPM \ @@ -302,13 +303,13 @@ enum { #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) #define AZX_DCAPS_INTEL_BAYTRAIL \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT) -- 2.16.4

1 0

[alsa-devel] CPU hook snd_hda_intel
by Klaus Ethgen 24 Dec '19

24 Dec '19

Hello, Please keep me in Cc when you answer as I am not subscribed to the list. I use the most current kernel 5.4.1 (but I did have that problem since several 5.x versions) on gentoo on a X61s Lenovo laptop. What I see is that from time to time one CPU is constantly working and draining my battery with a process kworker/*-events. Today I was able to identify the source of the problem. The module snd-hda-intel seems to produce that load. When I unload the module, the CPU hook is gone. Even if I load it again, I have a good change to solve the problem. It also helps often to put the laptop to sleep and waking it up again. My relevant dmesg part is: [ 7.186766] snd_hda_intel 0000:00:1b.0: probe_mask set to 0x1 for device 17aa:20ac [ 7.563277] snd_hda_codec_generic hdaudioC0D0: autoconfig for Generic: line_outs=1 (0x12/0x0/0x0/0x0/0x0) type:speaker [ 7.563281] snd_hda_codec_generic hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 7.563284] snd_hda_codec_generic hdaudioC0D0: hp_outs=1 (0x11/0x0/0x0/0x0/0x0) [ 7.563286] snd_hda_codec_generic hdaudioC0D0: mono: mono_out=0x0 [ 7.563288] snd_hda_codec_generic hdaudioC0D0: inputs: [ 7.563292] snd_hda_codec_generic hdaudioC0D0: Internal Mic=0x15 [ 7.563294] snd_hda_codec_generic hdaudioC0D0: Mic=0x14 [ 7.563297] snd_hda_codec_generic hdaudioC0D0: Dock Mic=0x1c For reference I attach my kernel config to this mail. I am not sure if it is needed but better give as much as informations that I have. So please, if you have any idea how to narrow that problem or even solve it, help would be appreciated. Regards Klaus -- Klaus Ethgen http://www.ethgen.ch/ pub 4096R/4E20AF1C 2011-05-16 Klaus Ethgen <Klaus(a)Ethgen.ch> Fingerprint: 85D4 CA42 952C 949B 1753 62B3 79D0 B06F 4E20 AF1C

2 16

[alsa-devel] [PATCH alsa-lib, alsa-plugins] Update the attributes.m4 macro file from xine
by David Ward 23 Dec '19

23 Dec '19

This file was imported from the xine project. Update it to the current revision, which resolves the "no AC_LANG_SOURCE call detected in body" warnings with Autoconf 2.68 or later. Signed-off-by: David Ward <david.ward(a)ll.mit.edu> --- m4/attributes.m4 | 33 ++++++++++++++++++++------------- 1 file changed, 20 insertions(+), 13 deletions(-) diff --git a/m4/attributes.m4 b/m4/attributes.m4 index e86456a..3d9c256 100644 --- a/m4/attributes.m4 +++ b/m4/attributes.m4 @@ -1,6 +1,6 @@ dnl Macros to check the presence of generic (non-typed) symbols. -dnl Copyright (c) 2006-2007 Diego Pettenò <flameeyes(a)gmail.com> -dnl Copyright (c) 2006-2007 xine project +dnl Copyright (c) 2006-2008 Diego Pettenò <flameeyes(a)gmail.com> +dnl Copyright (c) 2006-2008 xine project dnl dnl This program is free software; you can redistribute it and/or modify dnl it under the terms of the GNU General Public License as published by @@ -25,7 +25,7 @@ dnl License when using or distributing such scripts, even though portions dnl of the text of the Macro appear in them. The GNU General Public dnl License (GPL) does govern all other use of the material that dnl constitutes the Autoconf Macro. -dnl +dnl dnl This special exception to the GPL applies to versions of the dnl Autoconf Macro released by this project. When you make and dnl distribute a modified version of the Autoconf Macro, you may extend @@ -39,7 +39,7 @@ AC_DEFUN([CC_CHECK_CFLAGS_SILENT], [ AC_CACHE_VAL(AS_TR_SH([cc_cv_cflags_$1]), [ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS $1" - AC_COMPILE_IFELSE([int a;], + AC_COMPILE_IFELSE([AC_LANG_SOURCE([int a;])], [eval "AS_TR_SH([cc_cv_cflags_$1])='yes'"], [eval "AS_TR_SH([cc_cv_cflags_$1])='no'"]) CFLAGS="$ac_save_CFLAGS" @@ -71,7 +71,7 @@ AC_DEFUN([CC_CHECK_CFLAG_APPEND], [ ) AS_IF([eval test x$]AS_TR_SH([cc_cv_cflags_$1])[ = xyes], - [CFLAGS="$CFLAGS $1"; $2], [$3]) + [CFLAGS="$CFLAGS $1"; DEBUG_CFLAGS="$DEBUG_CFLAGS $1"; $2], [$3]) ]) dnl CC_CHECK_CFLAGS_APPEND([FLAG1 FLAG2], [action-if-found], [action-if-not]) @@ -89,7 +89,7 @@ AC_DEFUN([CC_CHECK_LDFLAGS], [ AS_TR_SH([cc_cv_ldflags_$1]), [ac_save_LDFLAGS="$LDFLAGS" LDFLAGS="$LDFLAGS $1" - AC_LINK_IFELSE([int main() { return 1; }], + AC_LINK_IFELSE([AC_LANG_SOURCE([int main() { return 1; }])], [eval "AS_TR_SH([cc_cv_ldflags_$1])='yes'"], [eval "AS_TR_SH([cc_cv_ldflags_$1])="]) LDFLAGS="$ac_save_LDFLAGS" @@ -109,14 +109,21 @@ AC_DEFUN([CC_NOUNDEFINED], [ dnl FreeBSD (et al.) does not complete linking for shared objects when pthreads dnl are requested, as different implementations are present; to avoid problems dnl use -Wl,-z,defs only for those platform not behaving this way. - *-freebsd*) ;; + dnl + dnl MinGW platforms: for libraries required -no-undefined, + dnl use it only for libraries in mingw32-w64 + + *-freebsd* | *-openbsd*) ;; + *-mingw*) + LDFLAGS_NOUNDEFINED="-no-undefined" + ;; *) dnl First of all check for the --no-undefined variant of GNU ld. This allows dnl for a much more readable commandline, so that people can understand what dnl it does without going to look for what the heck -z defs does. - for possible_flags in "-Wl,--no-undefined" "-Wl,-z,defs"; do + for possible_flags in "-Wl,--no-undefined" "-Wl,-z,defs"; do CC_CHECK_LDFLAGS([$possible_flags], [LDFLAGS_NOUNDEFINED="$possible_flags"]) - break + if test "x$LDFLAGS_NOUNDEFINED" = "x"; then break; fi done ;; esac @@ -147,7 +154,7 @@ AC_DEFUN([CC_CHECK_ATTRIBUTE], [ AS_TR_SH([cc_cv_attribute_$1]), [ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS $cc_cv_werror" - AC_COMPILE_IFELSE([$3], + AC_COMPILE_IFELSE([AC_LANG_SOURCE([$3])], [eval "AS_TR_SH([cc_cv_attribute_$1])='yes'"], [eval "AS_TR_SH([cc_cv_attribute_$1])='no'"]) CFLAGS="$ac_save_CFLAGS" @@ -257,7 +264,7 @@ AC_DEFUN([CC_FLAG_VISIBILITY], [ cc_cv_flag_visibility='yes', cc_cv_flag_visibility='no') CFLAGS="$cc_flag_visibility_save_CFLAGS"]) - + AS_IF([test "x$cc_cv_flag_visibility" = "xyes"], [AC_DEFINE([SUPPORT_FLAG_VISIBILITY], 1, [Define this if the compiler supports the -fvisibility flag]) @@ -295,11 +302,11 @@ AC_DEFUN([CC_ATTRIBUTE_ALIGNED], [ [ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS $cc_cv_werror" for cc_attribute_align_try in 64 32 16 8 4 2; do - AC_COMPILE_IFELSE([ + AC_COMPILE_IFELSE([AC_LANG_SOURCE([ int main() { static char c __attribute__ ((aligned($cc_attribute_align_try))) = 0; return c; - }], [cc_cv_attribute_aligned=$cc_attribute_align_try; break]) + }])], [cc_cv_attribute_aligned=$cc_attribute_align_try; break]) done CFLAGS="$ac_save_CFLAGS" ]) -- 1.8.3.1

1 0

[alsa-devel] [PATCH for-5.6 0/4] ALSA: ctl: UAPI updates for obsoleted members
by Takashi Sakamoto 23 Dec '19

23 Dec '19

Hi, As a result of applying patchset for Y2038 issue[1], 'snd_ctl_elem_value' structure is changed to lost 'tstamp' member. This change should corresponds to removal of an access flag of 'snd_ctl_elem_info' structure. These changes brings slight lose of backward compatibility, thus it's better to bump protocol version. Additionally, for recent few year I work to obsolete 'dimen' member from 'snd_ctl_elem_info' structure. It's good timing to attempt toward it. This patchset includes these changes and bump the version up to v2.1.0. [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2019-December/160082.… Takashi Sakamoto (4): ALSA: ctl: remove unused macro for timestamping of elem_value ALSA: pci: echoaudio: remove usage of dimen menber of elem_value structure ALSA: ctl: remove dimen member from elem_info structure ALSA: ctl: bump protocol version up to v2.1.0 include/uapi/sound/asound.h | 10 +++------- sound/core/control.c | 32 -------------------------------- sound/pci/echoaudio/echoaudio.c | 11 ----------- 3 files changed, 3 insertions(+), 50 deletions(-) -- 2.20.1

2 5

[alsa-devel] [PATCH] ASoC: rt5682: Add option to select pulse IRQ in jack detect
by Akshu Agrawal 23 Dec '19

23 Dec '19

Some SoC need to set IRQ type as pulse along with other JD1 options. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 3 +++ sound/soc/codecs/rt5682.h | 2 ++ 3 files changed, 6 insertions(+) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index bc2c31734df1..64cfa77ec9ee 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -22,6 +22,7 @@ enum rt5682_dmic1_clk_pin { enum rt5682_jd_src { RT5682_JD_NULL, RT5682_JD1, + RT5682_JD2, }; struct rt5682_platform_data { diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index ae6f6121bc1b..5135d7757361 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1009,6 +1009,9 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, } switch (rt5682->pdata.jd_src) { + case RT5682_JD2: + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_PULSE_MASK, RT5682_JD1_PULSE_EN); case RT5682_JD1: snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 18faaa2a49a0..434b1c9778b2 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1091,6 +1091,8 @@ #define RT5682_JD1_POL_MASK (0x1 << 13) #define RT5682_JD1_POL_NOR (0x0 << 13) #define RT5682_JD1_POL_INV (0x1 << 13) +#define RT5682_JD1_PULSE_MASK (0x1 << 10) +#define RT5682_JD1_PULSE_EN (0x1 << 10) /* IRQ Control 3 (0x00b8) */ #define RT5682_IL_IRQ_MASK (0x1 << 7) -- 2.17.1

2 1

[alsa-devel] How to install on debian buster?
by GitHub issues - opened 23 Dec '19

23 Dec '19

alsa-project/alsa-ucm-conf issue #8 was opened from kopax: I use kernel 5.3 on debian buster and I am following thos offical Lenovo tutorial to install the microphone on my x1c7: https://forums.lenovo.com/lnv/attachments/lnv/lx02_en/3199/1/Enabling%20aud… It says: ![image](https://user-images.githubusercontent.com/1866564/71328607-1a5e8180-251a-11ea-8621-1655d7746c60.png) Now the [link](https://github.com/alsa-project/alsa-ucm-conf/tree/master/ucm/sof-skl… is 404, what do I need to get now ? Thanks in advance and merry xmas! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/8 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [PATCH v3 0/2] Add the broadmobi BM818
by Angus Ainslie (Purism) 22 Dec '19

22 Dec '19

The broadmobi uses slightly different parameters from the option modems so add the paramters and document them. Changes since v2: Use of_device_get_match_data to get the dai data. Updated subject styles. Changes since v1: Cut back the CC list. Use data from compatible match instead of explicit compatible match. Angus Ainslie (Purism) (2): ASoC: gtm601: add Broadmobi bm818 sound profile dt-bindings: sound: gtm601: add the broadmobi interface .../devicetree/bindings/sound/gtm601.txt | 10 ++++-- sound/soc/codecs/gtm601.c | 31 ++++++++++++++++--- 2 files changed, 35 insertions(+), 6 deletions(-) -- 2.17.1

3 5

[alsa-devel] [sound:for-next 163/166] ./usr/include/sound/hdspm.h:44:2: error: unknown type name '__u32'
by kbuild test robot 22 Dec '19

22 Dec '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: bfea224d9250e56ed5475bf72190783a7e1d8e43 commit: 4fa406caf950fd46ae06bccf9a4c72171401d203 [163/166] ALSA: hdspm: Drop linux/types.h inclusion in uapi header config: i386-allyesconfig (attached as .config) compiler: gcc-7 (Debian 7.5.0-3) 7.5.0 reproduce: git checkout 4fa406caf950fd46ae06bccf9a4c72171401d203 # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from <command-line>:32:0: >> ./usr/include/sound/hdspm.h:44:2: error: unknown type name '__u32' __u32 input_peaks[64]; ^~~~~ ./usr/include/sound/hdspm.h:45:2: error: unknown type name '__u32' __u32 playback_peaks[64]; ^~~~~ ./usr/include/sound/hdspm.h:46:2: error: unknown type name '__u32' __u32 output_peaks[64]; ^~~~~ >> ./usr/include/sound/hdspm.h:48:2: error: unknown type name '__u64' __u64 input_rms[64]; ^~~~~ ./usr/include/sound/hdspm.h:49:2: error: unknown type name '__u64' __u64 playback_rms[64]; ^~~~~ ./usr/include/sound/hdspm.h:50:2: error: unknown type name '__u64' __u64 output_rms[64]; ^~~~~ >> ./usr/include/sound/hdspm.h:52:2: error: unknown type name '__u8' __u8 speed; /* enum {ss, ds, qs} */ ^~~~ ./usr/include/sound/hdspm.h:153:2: error: unknown type name '__u8' __u8 card_type; /* enum hdspm_io_type */ ^~~~ ./usr/include/sound/hdspm.h:156:2: error: unknown type name '__u64' __u64 card_clock; ^~~~~ ./usr/include/sound/hdspm.h:157:2: error: unknown type name '__u32' __u32 master_period; ^~~~~ ./usr/include/sound/hdspm.h:161:4: error: unknown type name '__u8' __u8 sync_wc; /* enum hdspm_sync */ ^~~~ ./usr/include/sound/hdspm.h:162:4: error: unknown type name '__u8' __u8 sync_madi; /* enum hdspm_sync */ ^~~~ ./usr/include/sound/hdspm.h:163:4: error: unknown type name '__u8' __u8 sync_tco; /* enum hdspm_sync */ ^~~~ ./usr/include/sound/hdspm.h:164:4: error: unknown type name '__u8' __u8 sync_in; /* enum hdspm_sync */ ^~~~ ./usr/include/sound/hdspm.h:165:4: error: unknown type name '__u8' __u8 madi_input; /* enum hdspm_madi_input */ ^~~~ ./usr/include/sound/hdspm.h:166:4: error: unknown type name '__u8' __u8 channel_format; /* enum hdspm_madi_channel_format */ ^~~~ ./usr/include/sound/hdspm.h:167:4: error: unknown type name '__u8' __u8 frame_format; /* enum hdspm_madi_frame_format */ ^~~~ ./usr/include/sound/hdspm.h:182:2: error: unknown type name '__u8' __u8 card_type; /* enum hdspm_io_type */ ^~~~ --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org Intel Corporation

1 0

[alsa-devel] ctl: elem_id/elem_value: add APIs to check whether two instances have the same meaning
by GitHub pull_request - opened 22 Dec '19

22 Dec '19

alsa-project/alsa-gobject pull request #2 was opened from takaswie: For some cases, it's convenient for users to check whether two instances of elem_id/elem_value have the same meaning. In glib, APIs for comparison of content of objects are named as equal. This commit adds equal APIs for this purpose. Request URL : https://github.com/alsa-project/alsa-gobject/pull/2 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/2.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[alsa-devel] ctl: elem_info_int: fix wrong assignment to private value
by GitHub pull_request - opened 22 Dec '19

22 Dec '19

alsa-project/alsa-gobject pull request #1 was opened from takaswie: ALSACtl::ElemInfoInt has properties for 'min', 'max' and 'step' but their assignments are not proper. This commit fixes the bug. Fixes: aac870451a89 ("ctl: elem_info_int: add properties for type-dependent parameters and constructor") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/1 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[alsa-devel] [sound:for-next 162/166] ./usr/include/sound/hdsp.h:35:2: error: unknown type name '__u32'
by kbuild test robot 22 Dec '19

22 Dec '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: bfea224d9250e56ed5475bf72190783a7e1d8e43 commit: d63e63d4210713ddc7eb1b2d2afc4a8d42ccf91e [162/166] ALSA: hdsp: Make uapi/hdsp.h compilable again config: i386-allyesconfig (attached as .config) compiler: gcc-7 (Debian 7.5.0-3) 7.5.0 reproduce: git checkout d63e63d4210713ddc7eb1b2d2afc4a8d42ccf91e # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from <command-line>:32:0: >> ./usr/include/sound/hdsp.h:35:2: error: unknown type name '__u32' __u32 input_peaks[26]; ^~~~~ ./usr/include/sound/hdsp.h:36:2: error: unknown type name '__u32' __u32 playback_peaks[26]; ^~~~~ ./usr/include/sound/hdsp.h:37:2: error: unknown type name '__u32' __u32 output_peaks[28]; ^~~~~ >> ./usr/include/sound/hdsp.h:38:2: error: unknown type name '__u64' __u64 input_rms[26]; ^~~~~ ./usr/include/sound/hdsp.h:39:2: error: unknown type name '__u64' __u64 playback_rms[26]; ^~~~~ ./usr/include/sound/hdsp.h:41:2: error: unknown type name '__u64' __u64 output_rms[26]; ^~~~~ --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org Intel Corporation

1 0

[alsa-devel] [sound:for-next 161/166] usr/include/sound/emu10k1.h:322: found __[us]{8, 16, 32, 64} type without #include <linux/types.h>
by kbuild test robot 21 Dec '19

21 Dec '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: bfea224d9250e56ed5475bf72190783a7e1d8e43 commit: 2e4688676392767e16c1adeca4cc2c083e2db13f [161/166] ALSA: emu10k1: Make uapi/emu10k1.h compilable again config: i386-allyesconfig (attached as .config) compiler: gcc-7 (Debian 7.5.0-3) 7.5.0 reproduce: git checkout 2e4688676392767e16c1adeca4cc2c083e2db13f # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): >> usr/include/sound/emu10k1.h:322: found __[us]{8,16,32,64} type without #include <linux/types.h> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: hda/hdmi - fix atpx_present when CLASS is not VGA
by Alex Deucher 21 Dec '19

21 Dec '19

You can't use PCI_BASE_CLASS with pci_get_class(). This happens to work by luck on devices with PCI_CLASS_DISPLAY_VGA, but misses PCI_CLASS_DISPLAY_OTHER. Add a check for those as well. Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/pci/hda/hda_intel.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b856b89378ac..f69c8de64bd6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1410,7 +1410,17 @@ static bool atpx_present(void) acpi_handle dhandle, atpx_handle; acpi_status status; - while ((pdev = pci_get_class(PCI_BASE_CLASS_DISPLAY << 16, pdev)) != NULL) { + while ((pdev = pci_get_class(PCI_CLASS_DISPLAY_VGA << 8, pdev)) != NULL) { + dhandle = ACPI_HANDLE(&pdev->dev); + if (dhandle) { + status = acpi_get_handle(dhandle, "ATPX", &atpx_handle); + if (!ACPI_FAILURE(status)) { + pci_dev_put(pdev); + return true; + } + } + } + while ((pdev = pci_get_class(PCI_CLASS_DISPLAY_OTHER << 8, pdev)) != NULL) { dhandle = ACPI_HANDLE(&pdev->dev); if (dhandle) { status = acpi_get_handle(dhandle, "ATPX", &atpx_handle); -- 2.24.1

2 1

[alsa-devel] [PATCH v10 0/2] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 20 Dec '19

20 Dec '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v9: - Program pa_gain settings in put function, rather than delaying it. Srinivas Kandagatla (2): dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 68 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1182 +++++++++++++++++ 4 files changed, 1262 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

2 5

[alsa-devel] [PATCH alsa-lib] uapi: Move typedefs from uapi to sound/*
by Takashi Iwai 20 Dec '19

20 Dec '19

For keeping uapi/*.h cleaner. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/emu10k1.h | 9 +++++++++ include/sound/hdsp.h | 10 ++++++++++ include/sound/hdspm.h | 8 ++++++++ include/sound/sb16_csp.h | 4 ++++ include/sound/uapi/emu10k1.h | 10 ---------- include/sound/uapi/hdsp.h | 9 --------- include/sound/uapi/hdspm.h | 8 -------- 7 files changed, 31 insertions(+), 27 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index f06ecee51c2b..0832f95bd995 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1,2 +1,11 @@ #include <alsa/sound/type_compat.h> #include <alsa/sound/uapi/emu10k1.h> +#ifndef __emu10k1_type_defined +#define __emu10k1_type_defined +typedef struct snd_emu10k1_fx8010_info emu10k1_fx8010_info_t; +typedef struct snd_emu10k1_fx8010_control_gpr emu10k1_fx8010_control_gpr_t; +typedef struct snd_emu10k1_fx8010_code emu10k1_fx8010_code_t; +typedef struct snd_emu10k1_fx8010_tram emu10k1_fx8010_tram_t; +typedef struct snd_emu10k1_fx8010_pcm_rec emu10k1_fx8010_pcm_t; +typedef struct emu10k1_ctl_elem_id emu10k1_ctl_elem_id_t; +#endif diff --git a/include/sound/hdsp.h b/include/sound/hdsp.h index 78fb745b607a..e8f931569548 100644 --- a/include/sound/hdsp.h +++ b/include/sound/hdsp.h @@ -1,2 +1,12 @@ #include <alsa/sound/type_compat.h> #include <alsa/sound/uapi/hdsp.h> +#ifndef __hdsp_type_defined +#define __hdsp_type_defined +typedef enum HDSP_IO_Type HDSP_IO_Type; +typedef struct hdsp_peak_rms hdsp_peak_rms_t; +typedef struct hdsp_config_info hdsp_config_info_t; +typedef struct hdsp_firmware hdsp_firmware_t; +typedef struct hdsp_version hdsp_version_t; +typedef struct hdsp_mixer hdsp_mixer_t; +typedef struct hdsp_9632_aeb hdsp_9632_aeb_t; +#endif diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h index af6d19eda87f..d9095d9f10bb 100644 --- a/include/sound/hdspm.h +++ b/include/sound/hdspm.h @@ -1,2 +1,10 @@ #include <alsa/sound/type_compat.h> #include <alsa/sound/uapi/hdspm.h> +#ifndef __hdspm_type_defined +#define __hdspm_type_defined +typedef struct hdspm_peak_rms hdspm_peak_rms_t; +typedef struct hdspm_config_info hdspm_config_info_t; +typedef struct hdspm_version hdspm_version_t; +typedef struct hdspm_channelfader snd_hdspm_channelfader_t; +typedef struct hdspm_mixer hdspm_mixer_t; +#endif diff --git a/include/sound/sb16_csp.h b/include/sound/sb16_csp.h index 24121fcbd888..fd02bc565e21 100644 --- a/include/sound/sb16_csp.h +++ b/include/sound/sb16_csp.h @@ -1 +1,5 @@ #include <alsa/sound/uapi/sb16_csp.h> +#ifndef __sb16_csp_type_defined +#define __sb16_csp_type_defined +typedef struct snd_sb_csp_microcode snd_sb_csp_microcode_t; +#endif diff --git a/include/sound/uapi/emu10k1.h b/include/sound/uapi/emu10k1.h index c1150e4d0231..6bcd76f64c1c 100644 --- a/include/sound/uapi/emu10k1.h +++ b/include/sound/uapi/emu10k1.h @@ -382,14 +382,4 @@ struct snd_emu10k1_fx8010_pcm_rec { #define SNDRV_EMU10K1_IOCTL_SINGLE_STEP _IOW ('H', 0x83, int) #define SNDRV_EMU10K1_IOCTL_DBG_READ _IOR ('H', 0x84, int) -#ifndef __KERNEL__ -/* typedefs for compatibility to user-space */ -typedef struct snd_emu10k1_fx8010_info emu10k1_fx8010_info_t; -typedef struct snd_emu10k1_fx8010_control_gpr emu10k1_fx8010_control_gpr_t; -typedef struct snd_emu10k1_fx8010_code emu10k1_fx8010_code_t; -typedef struct snd_emu10k1_fx8010_tram emu10k1_fx8010_tram_t; -typedef struct snd_emu10k1_fx8010_pcm_rec emu10k1_fx8010_pcm_t; -typedef struct emu10k1_ctl_elem_id emu10k1_ctl_elem_id_t; -#endif - #endif /* _UAPI__SOUND_EMU10K1_H */ diff --git a/include/sound/uapi/hdsp.h b/include/sound/uapi/hdsp.h index 88c92a3fb477..7ac2d3f2a9b3 100644 --- a/include/sound/uapi/hdsp.h +++ b/include/sound/uapi/hdsp.h @@ -97,13 +97,4 @@ struct hdsp_9632_aeb { #define SNDRV_HDSP_IOCTL_GET_9632_AEB _IOR('H', 0x45, struct hdsp_9632_aeb) -/* typedefs for compatibility to user-space */ -typedef enum HDSP_IO_Type HDSP_IO_Type; -typedef struct hdsp_peak_rms hdsp_peak_rms_t; -typedef struct hdsp_config_info hdsp_config_info_t; -typedef struct hdsp_firmware hdsp_firmware_t; -typedef struct hdsp_version hdsp_version_t; -typedef struct hdsp_mixer hdsp_mixer_t; -typedef struct hdsp_9632_aeb hdsp_9632_aeb_t; - #endif /* __SOUND_HDSP_H */ diff --git a/include/sound/uapi/hdspm.h b/include/sound/uapi/hdspm.h index 2d91f90eb5e1..3fbfd9dc5f51 100644 --- a/include/sound/uapi/hdspm.h +++ b/include/sound/uapi/hdspm.h @@ -219,12 +219,4 @@ struct hdspm_mixer_ioctl { /* use indirect access due to the limit of ioctl bit size */ #define SNDRV_HDSPM_IOCTL_GET_MIXER _IOR('H', 0x44, struct hdspm_mixer_ioctl) -/* typedefs for compatibility to user-space */ -typedef struct hdspm_peak_rms hdspm_peak_rms_t; -typedef struct hdspm_config_info hdspm_config_info_t; -typedef struct hdspm_version hdspm_version_t; -typedef struct hdspm_channelfader snd_hdspm_channelfader_t; -typedef struct hdspm_mixer hdspm_mixer_t; - - #endif -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: uapi: Drop unneeded typedefs
by Takashi Iwai 20 Dec '19

20 Dec '19

We kept some typedefs in uapi/sound/*.h so that the programs in alsa-tools can be built. Now that alsa-lib takes these and applies the workarounds in its own, we don't need these typedefs any longer in the kernel uapi side. Let's drop them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/emu10k1.h | 10 ---------- include/uapi/sound/hdsp.h | 9 --------- include/uapi/sound/hdspm.h | 8 -------- 3 files changed, 27 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index c1150e4d0231..6bcd76f64c1c 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -382,14 +382,4 @@ struct snd_emu10k1_fx8010_pcm_rec { #define SNDRV_EMU10K1_IOCTL_SINGLE_STEP _IOW ('H', 0x83, int) #define SNDRV_EMU10K1_IOCTL_DBG_READ _IOR ('H', 0x84, int) -#ifndef __KERNEL__ -/* typedefs for compatibility to user-space */ -typedef struct snd_emu10k1_fx8010_info emu10k1_fx8010_info_t; -typedef struct snd_emu10k1_fx8010_control_gpr emu10k1_fx8010_control_gpr_t; -typedef struct snd_emu10k1_fx8010_code emu10k1_fx8010_code_t; -typedef struct snd_emu10k1_fx8010_tram emu10k1_fx8010_tram_t; -typedef struct snd_emu10k1_fx8010_pcm_rec emu10k1_fx8010_pcm_t; -typedef struct emu10k1_ctl_elem_id emu10k1_ctl_elem_id_t; -#endif - #endif /* _UAPI__SOUND_EMU10K1_H */ diff --git a/include/uapi/sound/hdsp.h b/include/uapi/sound/hdsp.h index 88c92a3fb477..7ac2d3f2a9b3 100644 --- a/include/uapi/sound/hdsp.h +++ b/include/uapi/sound/hdsp.h @@ -97,13 +97,4 @@ struct hdsp_9632_aeb { #define SNDRV_HDSP_IOCTL_GET_9632_AEB _IOR('H', 0x45, struct hdsp_9632_aeb) -/* typedefs for compatibility to user-space */ -typedef enum HDSP_IO_Type HDSP_IO_Type; -typedef struct hdsp_peak_rms hdsp_peak_rms_t; -typedef struct hdsp_config_info hdsp_config_info_t; -typedef struct hdsp_firmware hdsp_firmware_t; -typedef struct hdsp_version hdsp_version_t; -typedef struct hdsp_mixer hdsp_mixer_t; -typedef struct hdsp_9632_aeb hdsp_9632_aeb_t; - #endif /* __SOUND_HDSP_H */ diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 2d91f90eb5e1..3fbfd9dc5f51 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -219,12 +219,4 @@ struct hdspm_mixer_ioctl { /* use indirect access due to the limit of ioctl bit size */ #define SNDRV_HDSPM_IOCTL_GET_MIXER _IOR('H', 0x44, struct hdspm_mixer_ioctl) -/* typedefs for compatibility to user-space */ -typedef struct hdspm_peak_rms hdspm_peak_rms_t; -typedef struct hdspm_config_info hdspm_config_info_t; -typedef struct hdspm_version hdspm_version_t; -typedef struct hdspm_channelfader snd_hdspm_channelfader_t; -typedef struct hdspm_mixer hdspm_mixer_t; - - #endif -- 2.16.4

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix VDDA and VDDIO comparison
by Marek Vasut 20 Dec '19

20 Dec '19

Comparing the voltage of VDDA and VDDIO to determine whether or not to enable VDDC manual override is insufficient. This is a problem in case the VDDA is supplied from different regulator than VDDIO, while both report the same voltage to the regulator framework. In that case where VDDA and VDDIO is supplied by different regulators, the VDDC manual override must not be applied. Fixes: b6319b061ba2 ("ASoC: sgtl5000: Fix charge pump source assignment") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Igor Opaniuk <igor.opaniuk(a)toradex.com> Cc: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> --- sound/soc/codecs/sgtl5000.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 93da6b58c97df..b4568f417d78e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -28,6 +28,8 @@ #include "sgtl5000.h" +#include "../../../drivers/regulator/internal.h" + #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A @@ -1344,7 +1346,8 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component) * if vddio == vdda the source of charge pump should be * assigned manually to VDDIO */ - if (vddio == vdda) { + if (sgtl5000->supplies[VDDA].consumer->rdev == + sgtl5000->supplies[VDDIO].consumer->rdev) { lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << SGTL5000_VDDC_MAN_ASSN_SHIFT; -- 2.24.1

2 5

[alsa-devel] [PATCH] hdspmixer: Use __u32 and __u64 for RMS array types
by Takashi Iwai 20 Dec '19

20 Dec '19

Some variable types are referring to the field in ioctl struct, which are actually __u32 or __u64 instead of uint32_t or uint64_t. This inconsistency may result in the compile error. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- hdspmixer/src/HDSPMixerWindow.cxx | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/hdspmixer/src/HDSPMixerWindow.cxx b/hdspmixer/src/HDSPMixerWindow.cxx index 342efb2b857f..9efc25d0d5c0 100644 --- a/hdspmixer/src/HDSPMixerWindow.cxx +++ b/hdspmixer/src/HDSPMixerWindow.cxx @@ -35,8 +35,8 @@ static void readregisters_cb(void *arg) hdsp_peak_rms_t hdsp_peak_rms; struct hdspm_peak_rms hdspm_peak_rms; bool isMADI = false; - uint32_t *input_peaks, *playback_peaks, *output_peaks; - uint64_t *input_rms, *playback_rms, *output_rms; + __u32 *input_peaks, *playback_peaks, *output_peaks; + __u64 *input_rms, *playback_rms, *output_rms; HDSPMixerWindow *w = (HDSPMixerWindow *)arg; -- 2.16.4

1 0

[alsa-devel] [PATCH alsa-lib 2/2] type_compat: Add missing __s64 and __u64 definitions for non-Linux
by Takashi Iwai 20 Dec '19

20 Dec '19

Just for the case without Linux-compatible definitions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/type_compat.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/type_compat.h b/include/sound/type_compat.h index d4790c1f0e64..5b765f447e2e 100644 --- a/include/sound/type_compat.h +++ b/include/sound/type_compat.h @@ -9,9 +9,11 @@ typedef uint8_t __u8; typedef uint16_t __u16; typedef uint32_t __u32; +typedef uint64_t __u64; typedef int8_t __s8; typedef int16_t __s16; typedef int32_t __s32; +typedef int64_t __s64; #include <endian.h> #include <byteswap.h> -- 2.16.4

1 0

[alsa-devel] [PATCH] ASoC: max98090: Remove empty suspend function
by Mark Brown 20 Dec '19

20 Dec '19

The suspend function is empty so can be removed. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98090.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index da23810f958e..4c7b16d557e2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2838,17 +2838,12 @@ static int max98090_resume(struct device *dev) return 0; } - -static int max98090_suspend(struct device *dev) -{ - return 0; -} #endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, max98090_runtime_resume, NULL) - SET_SYSTEM_SLEEP_PM_OPS(max98090_suspend, max98090_resume) + SET_SYSTEM_SLEEP_PM_OPS(NULL, max98090_resume) }; static const struct i2c_device_id max98090_i2c_id[] = { -- 2.20.1

2 2

[alsa-devel] [PATCH] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 20 Dec '19

20 Dec '19

From: Jeff Chang <jeff_chang(a)richtek.com> The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper Signed-off-by: Jeff Chang <jeff_chang(a)richtek.com> --- sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 653 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 75 ++++++ 4 files changed, 741 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h changelogs between v3 & v2 - modify MT6660 Kconfig, remove unnecessary selection. - remove my own debug io interface. use standard regmap for debugging. - remove regmap volatile ops, we do not use cache. - remove component io read/write function, use snd_soc_component_init_regmap. - remove init setting write code. Using parsing dts to set them. - remove unnecessary pr_info log message. - remove mt6660_component_put_volsw. Using snd_soc_get_volsw. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..f135fbb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,16 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..5f3b9262 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,653 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0, ret = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + ret = i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); + if (ret < 0) + return ret; + return 0; +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + return 0; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, +}; + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + dev_dbg(component->dev, "Amp on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_dbg(component->dev, "Amp off\n"); + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "OUTP", NULL, "ClassD"}, + { "OUTN", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, snd_soc_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT Switch", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("Hard Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("DC_Protect Switch", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("audio input selection", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF AUD IN Switch", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("AUD LOOP BACK Switch", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("Mute Switch", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("Bypass CS Comp Switch", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, snd_soc_put_volsw), + SOC_SINGLE_EXT("T0 SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static int mt6660_apply_plat_data(struct mt6660_chip *chip, + struct snd_soc_component *component) +{ + size_t i = 0; + int num = chip->plat_data.init_setting_num; + int ret = 0; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + + for (i = 0; i < num; i++) { + ret = snd_soc_component_update_bits(component, + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + if (ret < 0) + return ret; + } + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(chip->dev, "%s power on failed\n", __func__); + return ret; + } + return 0; +} + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_info(component->dev, "%s\n", __func__); + snd_soc_component_init_regmap(component, chip->regmap); + + ret = mt6660_apply_plat_data(chip, component); + if (ret < 0) + dev_err(chip->dev, "mt6660 apply plat data failed\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + dev_info(component->dev, "%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_parse_dt(struct mt6660_chip *chip, struct device *dev) +{ + struct device_node *np = dev->of_node; + u32 val; + size_t i = 0; + + if (!np) { + dev_err(dev, "no device node\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "rt,init_setting_num", &val)) { + dev_info(dev, "no init setting\n"); + chip->plat_data.init_setting_num = 0; + } else { + chip->plat_data.init_setting_num = val; + } + + chip->plat_data.init_setting_addr = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + chip->plat_data.init_setting_mask = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + chip->plat_data.init_setting_val = + devm_kzalloc(dev, sizeof(u32) * + chip->plat_data.init_setting_num, GFP_KERNEL); + + if (of_property_read_u32_array(np, "rt,init_setting_addr", + chip->plat_data.init_setting_addr, + chip->plat_data.init_setting_num)) { + dev_info(dev, "no init setting addr\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_mask", + chip->plat_data.init_setting_mask, + chip->plat_data.init_setting_num)) { + dev_info(dev, "no init setting addr\n"); + } + if (of_property_read_u32_array(np, "rt,init_setting_val", + chip->plat_data.init_setting_val, + chip->plat_data.init_setting_num)) { + dev_info(dev, "no init setting addr\n"); + } + + dev_info(dev, "%s, init stting table, num = %d\n", __func__, + chip->plat_data.init_setting_num); + for (i = 0; i < chip->plat_data.init_setting_num; i++) { + dev_info(dev, "0x%02x, 0x%08x, 0x%08x\n", + chip->plat_data.init_setting_addr[i], + chip->plat_data.init_setting_mask[i], + chip->plat_data.init_setting_val[i]); + } + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + pr_info("%s start\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + + ret = mt6660_parse_dt(chip, &client->dev); + if (ret < 0) { + dev_err(&client->dev, "parsing dts failed\n"); + return ret; + } + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + pr_info("%s end, ret = %d\n", __func__, ret); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_info(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_info(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..6c40b40 --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,75 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +struct mt6660_platform_data { + u8 init_setting_num; + u32 *init_setting_addr; + u32 *init_setting_mask; + u32 *init_setting_val; +}; + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mt6660_platform_data plat_data; + struct mutex io_lock; + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix set_format altsetting sanity check
by Johan Hovold 20 Dec '19

20 Dec '19

Make sure to check the return value of usb_altnum_to_altsetting() to avoid dereferencing a NULL pointer when the requested alternate settings is missing. The format altsetting number may come from a quirk table and there does not seem to be any other validation of it (the corresponding index is checked however). Fixes: b099b9693d23 ("ALSA: usb-audio: Avoid superfluous usb_set_interface() calls") Cc: stable <stable(a)vger.kernel.org> # 4.18 Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/usb/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9c8930bb00c8..73dd9d21bb42 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -506,9 +506,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (WARN_ON(!iface)) return -EINVAL; alts = usb_altnum_to_altsetting(iface, fmt->altsetting); - altsd = get_iface_desc(alts); - if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) + if (WARN_ON(!alts)) return -EINVAL; + altsd = get_iface_desc(alts); if (fmt == subs->cur_audiofmt) return 0; -- 2.24.1

2 3

[alsa-devel] configure.ac: remove an unnecessary libtool fix
by GitHub pull_request - opened 20 Dec '19

20 Dec '19

alsa-project/alsa-lib pull request #19 was opened from tanuk: This code was added in commit 75d393a563efb578c79364a277087c6326267f52 without explaining why. I assume it was a mistake, since it looks like the libtool problem should have gone away a long time ago. The referenced wiki page https://wiki.debian.org/RpathIssue says: Since libtool 1.5.2 (released 2004-01-25), on Linux libtool no longer sets RPATH for any directories in the dynamic linker search path, so this should no longer be an issue unless upstream used a really old version of libtool when creating their distribution tarball. This code caused problems in OpenEmbedded, where the libtool script is named "x86_64-oe-linux-libtool" or similar rather than just "libtool", so the sed command failed with a file not found error. Rather than adapting the code to OpenEmbedded's peculiarities, it seems best to just remove the unnecessary code altogether. Signed-off-by: Tanu Kaskinen <tanuk(a)iki.fi> Request URL : https://github.com/alsa-project/alsa-lib/pull/19 Patch URL : https://github.com/alsa-project/alsa-lib/pull/19.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [alsa-lib][PATCH] conf: SOF-Intel: map pcm device for sof driver
by Hui Wang 20 Dec '19

20 Dec '19

On the machines with HDA codec, the sof driver will use the legacy hda codec driver, as a result, most mixer controls are same as before. There are only 2 difference: - HDMI pcm index is 3, 4 and 5 instead of 3, 7, 8... - dmic is on device 6, 7 instead of 0 After applying this patch, we will get (aplay -L): hdmi:CARD=sofsklhdacard,DEV=0 sof-skl_hda_card, HDMI Audio Output hdmi:CARD=sofsklhdacard,DEV=1 sof-skl_hda_card, HDMI Audio Output hdmi:CARD=sofsklhdacard,DEV=2 sof-skl_hda_card, HDMI Audio Output and (arecord -L): dmic:CARD=sofsklhdacard,DEV=0 sof-skl_hda_card, Digital Mic connected to the PCH directly dmic:CARD=sofsklhdacard,DEV=1 sof-skl_hda_card, Digital Mic connected to the PCH directly Then we could add minor changes in the pulseaudio configuration, after that, all output and input devices will work under pulseaudio. This is a supplement for ucm, the ucm for sof driver has higher priority than this conf, if there is no ucm for sof driver, this conf will take effect. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- src/conf/alsa.conf | 4 + src/conf/cards/Makefile.am | 3 +- src/conf/cards/SOF-Intel.conf | 345 ++++++++++++++++++++++++++++++++++ src/conf/cards/aliases.conf | 1 + src/conf/pcm/Makefile.am | 2 +- src/conf/pcm/dmic.conf | 54 ++++++ 6 files changed, 407 insertions(+), 2 deletions(-) create mode 100644 src/conf/cards/SOF-Intel.conf create mode 100644 src/conf/pcm/dmic.conf diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf index 09980586..28c11c81 100644 --- a/src/conf/alsa.conf +++ b/src/conf/alsa.conf @@ -101,6 +101,9 @@ defaults.pcm.iec958.card defaults.pcm.card defaults.pcm.iec958.device defaults.pcm.device defaults.pcm.modem.card defaults.pcm.card defaults.pcm.modem.device defaults.pcm.device +defaults.pcm.dmic.card defaults.pcm.card +defaults.pcm.dmic.device 6 + # truncate files via file or tee PCM defaults.pcm.file_format "raw" defaults.pcm.file_truncate true @@ -141,6 +144,7 @@ pcm.dmix cards.pcm.dmix pcm.dsnoop cards.pcm.dsnoop pcm.modem cards.pcm.modem pcm.phoneline cards.pcm.phoneline +pcm.dmic cards.pcm.dmic pcm.hw { @args [ CARD DEV SUBDEV ] diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index 00999f01..b22386a0 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -58,7 +58,8 @@ cfg_files = aliases.conf \ VIA8237.conf \ VX222.conf \ VXPocket.conf \ - VXPocket440.conf + VXPocket440.conf \ + SOF-Intel.conf if BUILD_ALISP cfg_files += aliases.alisp diff --git a/src/conf/cards/SOF-Intel.conf b/src/conf/cards/SOF-Intel.conf new file mode 100644 index 00000000..fe280d72 --- /dev/null +++ b/src/conf/cards/SOF-Intel.conf @@ -0,0 +1,345 @@ +# +# Configuration for the Intel HD audio (with sof driver) +# + +<confdir:pcm/front.conf> + +SOF-Intel.pcm.front.0 { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type softvol + slave.pcm { + type hw + card $CARD + subdevice 0 + } + control { + name "PCM Playback Volume" + card $CARD + } + } + capture.pcm { + type hw + card $CARD + } +} + +# default with dmix+softvol & dsnoop +SOF-Intel.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + type softvol + slave.pcm { + @func concat + strings [ "dmix:" $CARD ] + } + control { + name "PCM Playback Volume" + card $CARD + } + } + } + capture.pcm { + type plug + slave.pcm { + type softvol + slave.pcm { + @func concat + strings [ "dsnoop:" $CARD ] + } + control { + name "Digital Capture Volume" + card $CARD + } + min_dB -30.0 + max_dB 30.0 + resolution 121 + } + # to avoid possible phase inversions with digital mics + route_policy copy + } + hint.device 0 +} + +<confdir:pcm/surround21.conf> +<confdir:pcm/surround40.conf> +<confdir:pcm/surround41.conf> +<confdir:pcm/surround50.conf> +<confdir:pcm/surround51.conf> +<confdir:pcm/surround71.conf> + +SOF-Intel.pcm.surround40.0 cards.SOF-Intel.pcm.front.0 +SOF-Intel.pcm.surround51.0 cards.SOF-Intel.pcm.front.0 +SOF-Intel.pcm.surround71.0 cards.SOF-Intel.pcm.front.0 + +<confdir:pcm/iec958.conf> + +SOF-Intel.pcm.iec958.0 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type asym + playback.pcm { + type hooks + slave.pcm { + type hw + card $CARD + device 1 + } + hooks.0 { + type ctl_elems + hook_args [ + { + name "IEC958 Playback Default" + index 16 + optional true + lock true + preserve true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + { + name "IEC958 Playback Switch" + index 16 + optional true + value true + # if this element is present, skip the rest + skip_rest true + } + { + name "IEC958 Playback Default" + lock true + preserve true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + { + name "IEC958 Playback Switch" + value true + } + ] + } + } + capture.pcm { + type hooks + slave.pcm { + type hw + card $CARD + device 1 + } + hooks.0 { + type ctl_elems + hook_args [ + { + name "IEC958 Capture Switch" + lock true + preserve true + value true + } + ] + } + } + hint.device 1 +} + +<confdir:pcm/hdmi.conf> + +SOF-Intel.pcm.hdmi.common { + @args [ CARD DEVICE CTLINDEX AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.DEVICE { + type integer + } + @args.CTLINDEX { + type integer + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type hooks + slave.pcm { + type hw + card $CARD + device $DEVICE + } + hooks.0 { + type ctl_elems + hook_args [ + { + name "IEC958 Playback Default" + index $CTLINDEX + lock true + preserve true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + { + name "IEC958 Playback Switch" + index $CTLINDEX + value true + } + ] + } + hint.device $DEVICE +} + +SOF-Intel.pcm.hdmi.0 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.SOF-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=3," + "CTLINDEX=0," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +SOF-Intel.pcm.hdmi.1 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.SOF-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=4," + "CTLINDEX=1," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +SOF-Intel.pcm.hdmi.2 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.SOF-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=5," + "CTLINDEX=2," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +<confdir:pcm/dmic.conf> + +SOF-Intel.pcm.dmic.common { + @args [ CARD DEVICE ] + @args.CARD { + type string + } + @args.DEVICE { + type integer + } + type plug + slave.pcm { + type hw + card $CARD + device $DEVICE + } + + hint.device $DEVICE +} + +SOF-Intel.pcm.dmic.0 { + @args [ CARD ] + @args.CARD { type string } + @func refer + name { + @func concat + strings [ + "cards.SOF-Intel.pcm.dmic.common:" + "CARD=" $CARD "," + "DEVICE=6" + ] + } +} + +SOF-Intel.pcm.dmic.1 { + @args [ CARD ] + @args.CARD { type string } + @func refer + name { + @func concat + strings [ + "cards.SOF-Intel.pcm.dmic.common:" + "CARD=" $CARD "," + "DEVICE=7" + ] + } +} + +<confdir:pcm/modem.conf> + +SOF-Intel.pcm.modem.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 6 + hint.show off +} diff --git a/src/conf/cards/aliases.conf b/src/conf/cards/aliases.conf index 18a920f4..0c44c1aa 100644 --- a/src/conf/cards/aliases.conf +++ b/src/conf/cards/aliases.conf @@ -57,6 +57,7 @@ CMI8786 cards.CMI8788 CMI8787 cards.CMI8788 pistachio cards.pistachio-card VC4-HDMI cards.vc4-hdmi +sof-skl_hda_car cards.SOF-Intel <confdir:pcm/default.conf> <confdir:pcm/dmix.conf> diff --git a/src/conf/pcm/Makefile.am b/src/conf/pcm/Makefile.am index c548660d..20da0448 100644 --- a/src/conf/pcm/Makefile.am +++ b/src/conf/pcm/Makefile.am @@ -3,7 +3,7 @@ cfg_files = default.conf front.conf rear.conf center_lfe.conf side.conf\ surround50.conf surround51.conf \ surround71.conf iec958.conf hdmi.conf modem.conf \ dmix.conf dsnoop.conf \ - dpl.conf + dpl.conf dmic.conf EXTRA_DIST = $(cfg_files) diff --git a/src/conf/pcm/dmic.conf b/src/conf/pcm/dmic.conf new file mode 100644 index 00000000..1dce37c9 --- /dev/null +++ b/src/conf/pcm/dmic.conf @@ -0,0 +1,54 @@ +# +# input from digital mic +# + +pcm.!dmic { + @args [ CARD DEV ] + @args.CARD { + type string + default { + @func getenv + vars [ + ALSA_PCM_CARD + ALSA_CARD + ] + default { + @func refer + name defaults.pcm.dmic.card + } + } + } + @args.DEV { + type integer + default { + default { + @func refer + name defaults.pcm.dmic.device + } + } + } + type empty + slave.pcm { + @func refer + name { + @func concat + strings [ + "cards." + { + @func card_driver + card $CARD + } + ".pcm.dmic." $DEV ":" + "CARD=" $CARD + ] + } + } + hint { + show { + @func refer + name defaults.namehint.basic + } + description "Digital Mic connected to the PCH directly" + device $DEV + } +} -- 2.17.1

2 2

[alsa-devel] [PATCH AUTOSEL 5.4 197/350] ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency
by Sasha Levin 19 Dec '19

19 Dec '19

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f9ad75468453b019b92c5296e6a04bf7c37f49e4 ] updated solution to the problem reported with randconfig: CONFIG_SND_SOC_SOF_IMX depends on CONFIG_SND_SOC_SOF, but is in turn referenced by the sof-of-dev driver. This creates a reverse dependency that manifests in a link error when CONFIG_SND_SOC_SOF_OF is built-in but CONFIG_SND_SOC_SOF_IMX=m: sound/soc/sof/sof-of-dev.o:(.data+0x118): undefined reference to `sof_imx8_ops' use def_trisate to propagate the right settings without select. Fixes: f4df4e4042b0 ("ASoC: SOF: imx8: Fix COMPILE_TEST error") Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191101173045.27099-6-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/imx/Kconfig | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750d..71f318bc2c74f 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -11,8 +11,8 @@ config SND_SOC_SOF_IMX_TOPLEVEL if SND_SOC_SOF_IMX_TOPLEVEL -config SND_SOC_SOF_IMX8 - tristate "SOF support for i.MX8" +config SND_SOC_SOF_IMX8_SUPPORT + bool "SOF support for i.MX8" depends on IMX_SCU depends on IMX_DSP help @@ -20,4 +20,8 @@ config SND_SOC_SOF_IMX8 Say Y if you have such a device. If unsure select "N". +config SND_SOC_SOF_IMX8 + def_tristate SND_SOC_SOF_OF + depends on SND_SOC_SOF_IMX8_SUPPORT + endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL -- 2.20.1

2 2

[alsa-devel] [PATCH v9 0/2] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 19 Dec '19

19 Dec '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v8: - Moved soundwire stream handling into this codec as there is strong requirements on sequence of PA and soundwire port enables from hw side. Srinivas Kandagatla (2): dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 68 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1206 +++++++++++++++++ 4 files changed, 1286 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

2 4

[alsa-devel] [PATCH v14 7/7] ASoC: amd MMAP_INTERLEAVED Support
by Ravulapati Vishnu vardhan rao 19 Dec '19

19 Dec '19

ACP-I2S device support MMAP_INTERLEAVED. Added support for the same. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 916f0d8..28a3081 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -22,6 +22,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | @@ -42,7 +43,8 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: max98090: Remove empty suspend function
by Mark Brown 19 Dec '19

19 Dec '19

The suspend function is empty so can be removed. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98090.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index da23810f958e..68271042f586 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2838,15 +2838,10 @@ static int max98090_resume(struct device *dev) return 0; } - -static int max98090_suspend(struct device *dev) -{ - return 0; -} #endif static const struct dev_pm_ops max98090_pm = { - SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + SET_RUNTIME_PM_OPS(NULL, max98090_runtime_resume, NULL) SET_SYSTEM_SLEEP_PM_OPS(max98090_suspend, max98090_resume) }; -- 2.20.1

2 2

[alsa-devel] [PATCH] ALSA: hda: Use standard waitqueue for RIRB wakeup
by Takashi Iwai 19 Dec '19

19 Dec '19

The HD-audio CORB/RIRB communication was programmed in a way that was documented in the reference in decades ago, which is essentially a polling in the waiter side. It's working fine but costs CPU cycles on some platforms that support only slow communications. Also, for some platforms that had unreliable communications, we put longer wait time (2 ms), which accumulate quite long time if you execute many verbs in a shot (e.g. at the initialization or resume phase). This patch attempts to improve the situation by introducing the standard waitqueue in the RIRB waiter side instead of polling. The test results on my machine show significant improvements. The time spent for "cat /proc/asound/card*/codec#*" were changed like: * Intel SKL + Realtek codec before the patch: 0.00user 0.04system 0:00.10elapsed 40.0%CPU after the patch: 0.00user 0.01system 0:00.10elapsed 10.0%CPU * Nvidia GP107GL + Nvidia HDMI codec before the patch: 0.00user 0.00system 0:02.76elapsed 0.0%CPU after the patch: 0.00user 0.00system 0:00.01elapsed 17.0%CPU So, for Intel chips, the total time is same, while the total time is greatly reduced (from 2.76 to 0.01s) for Nvidia chips. The only negative data here is the increase of CPU time for Nvidia, but this is the unavoidable cost for faster wakeups, supposedly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_bus.c | 1 + sound/hda/hdac_controller.c | 3 +++ sound/pci/hda/hda_controller.c | 29 ++++++++++++++--------------- 4 files changed, 19 insertions(+), 15 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index e05b95e83d5a..81373a2efd96 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -317,6 +317,7 @@ struct hdac_bus { struct hdac_rb corb; struct hdac_rb rirb; unsigned int last_cmd[HDA_MAX_CODECS]; /* last sent command */ + wait_queue_head_t rirb_wq; /* CORB/RIRB and position buffers */ struct snd_dma_buffer rb; diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 8f19876244eb..48b227fff204 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -43,6 +43,7 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, mutex_init(&bus->cmd_mutex); mutex_init(&bus->lock); INIT_LIST_HEAD(&bus->hlink_list); + init_waitqueue_head(&bus->rirb_wq); bus->irq = -1; return 0; } diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 7e7be8e4dcf9..cd1c3b282657 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -216,6 +216,9 @@ void snd_hdac_bus_update_rirb(struct hdac_bus *bus) else if (bus->rirb.cmds[addr]) { bus->rirb.res[addr] = res; bus->rirb.cmds[addr]--; + if (!bus->rirb.cmds[addr] && + waitqueue_active(&bus->rirb_wq)) + wake_up(&bus->rirb_wq); } else { dev_err_ratelimited(bus->dev, "spurious response %#x:%#x, last cmd=%#08x\n", diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2f3b7a35f2d9..f30a053d981e 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -792,21 +792,25 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, struct hda_bus *hbus = &chip->bus; unsigned long timeout; unsigned long loopcounter; - int do_poll = 0; + wait_queue_entry_t wait; bool warned = false; + init_wait_entry(&wait, 0); again: timeout = jiffies + msecs_to_jiffies(1000); for (loopcounter = 0;; loopcounter++) { spin_lock_irq(&bus->reg_lock); - if (bus->polling_mode || do_poll) + if (!bus->polling_mode) + prepare_to_wait(&bus->rirb_wq, &wait, + TASK_UNINTERRUPTIBLE); + if (bus->polling_mode) snd_hdac_bus_update_rirb(bus); if (!bus->rirb.cmds[addr]) { - if (!do_poll) - bus->poll_count = 0; if (res) *res = bus->rirb.res[addr]; /* the last value */ + if (!bus->polling_mode) + finish_wait(&bus->rirb_wq, &wait); spin_unlock_irq(&bus->reg_lock); return 0; } @@ -814,7 +818,9 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, if (time_after(jiffies, timeout)) break; #define LOOP_COUNT_MAX 3000 - if (hbus->needs_damn_long_delay || + if (!bus->polling_mode) { + schedule_timeout(msecs_to_jiffies(2)); + } else if (hbus->needs_damn_long_delay || loopcounter > LOOP_COUNT_MAX) { if (loopcounter > LOOP_COUNT_MAX && !warned) { dev_dbg_ratelimited(chip->card->dev, @@ -829,19 +835,12 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, } } + if (!bus->polling_mode) + finish_wait(&bus->rirb_wq, &wait); + if (hbus->no_response_fallback) return -EIO; - if (!bus->polling_mode && bus->poll_count < 2) { - dev_dbg(chip->card->dev, - "azx_get_response timeout, polling the codec once: last cmd=0x%08x\n", - bus->last_cmd[addr]); - do_poll = 1; - bus->poll_count++; - goto again; - } - - if (!bus->polling_mode) { dev_warn(chip->card->dev, "azx_get_response timeout, switching to polling mode: last cmd=0x%08x\n", -- 2.16.4

3 9

[alsa-devel] [PATCH] slimbus: qcom-ngd-ctrl: Use dma_request_chan() instead dma_request_slave_channel()
by Peter Ujfalusi 19 Dec '19

19 Dec '19

dma_request_slave_channel() is a wrapper on top of dma_request_chan() eating up the error code. By using dma_request_chan() directly the driver can support deferred probing against DMA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 29fbab55c3b3..e3f5ebc0c05e 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -666,10 +666,12 @@ static int qcom_slim_ngd_init_rx_msgq(struct qcom_slim_ngd_ctrl *ctrl) struct device *dev = ctrl->dev; int ret, size; - ctrl->dma_rx_channel = dma_request_slave_channel(dev, "rx"); - if (!ctrl->dma_rx_channel) { - dev_err(dev, "Failed to request dma channels"); - return -EINVAL; + ctrl->dma_rx_channel = dma_request_chan(dev, "rx"); + if (IS_ERR(ctrl->dma_rx_channel)) { + dev_err(dev, "Failed to request RX dma channel"); + ret = PTR_ERR(ctrl->dma_rx_channel); + ctrl->dma_rx_channel = NULL; + return ret; } size = QCOM_SLIM_NGD_DESC_NUM * SLIM_MSGQ_BUF_LEN; @@ -703,10 +705,12 @@ static int qcom_slim_ngd_init_tx_msgq(struct qcom_slim_ngd_ctrl *ctrl) int ret = 0; int size; - ctrl->dma_tx_channel = dma_request_slave_channel(dev, "tx"); - if (!ctrl->dma_tx_channel) { - dev_err(dev, "Failed to request dma channels"); - return -EINVAL; + ctrl->dma_tx_channel = dma_request_chan(dev, "tx"); + if (IS_ERR(ctrl->dma_tx_channel)) { + dev_err(dev, "Failed to request TX dma channel"); + ret = PTR_ERR(ctrl->dma_tx_channel); + ctrl->dma_tx_channel = NULL; + return ret; } size = ((QCOM_SLIM_NGD_DESC_NUM + 1) * SLIM_MSGQ_BUF_LEN); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: set the interface format after resume on Dell WD19
by Hui Wang 19 Dec '19

19 Dec '19

Recently we found the headset-mic on the Dell Dock WD19 doesn't work anymore after s3 (s2i or deep), this problem could be workarounded by closeing (pcm_close) the app and then reopening (pcm_open) the app, so this bug is not easy to be detected by users. When problem happens, retire_capture_urb() could still be called periodically, but the size of captured data is always 0, it could be a firmware bug on the dock. Anyway I found after resuming, the snd_usb_pcm_prepare() will be called, and if we forcibly run set_format() to set the interface and its endpoint, the capture size will be normal again. This problem and workaound also apply to playback. To fix it in the kernel, add a quirk to let set_format() run forcibly once after resume. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/usb/card.h | 1 + sound/usb/pcm.c | 21 +++++++++++++++++++-- sound/usb/quirks-table.h | 3 ++- sound/usb/quirks.c | 11 +++++++++++ sound/usb/usbaudio.h | 3 ++- 5 files changed, 35 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 2991b9986f66..395403a2d33f 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -145,6 +145,7 @@ struct snd_usb_substream { struct snd_usb_endpoint *sync_endpoint; unsigned long flags; bool need_setup_ep; /* (re)configure EP at prepare? */ + bool need_setup_fmt; /* (re)configure fmt after resume? */ unsigned int speed; /* USB_SPEED_XXX */ u64 formats; /* format bitmasks (all or'ed) */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 8a52996041e9..4ac571b2e9a6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -523,11 +523,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) return -EINVAL; - if (fmt == subs->cur_audiofmt) + if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt) return 0; /* close the old interface */ - if (subs->interface >= 0 && subs->interface != fmt->iface) { + if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) { if (!subs->stream->chip->keep_iface) { err = usb_set_interface(subs->dev, subs->interface, 0); if (err < 0) { @@ -541,6 +541,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->altset_idx = 0; } + if (subs->need_setup_fmt) + subs->need_setup_fmt = false; + /* set interface */ if (iface->cur_altsetting != alts) { err = snd_usb_select_mode_quirk(subs, fmt); @@ -1734,6 +1737,13 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea subs->data_endpoint->retire_data_urb = retire_playback_urb; subs->running = 0; return 0; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (subs->stream->chip->setup_fmt_after_resume_quirk) { + stop_endpoints(subs, true); + subs->need_setup_fmt = true; + return 0; + } + break; } return -EINVAL; @@ -1766,6 +1776,13 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream subs->data_endpoint->retire_data_urb = retire_capture_urb; subs->running = 1; return 0; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (subs->stream->chip->setup_fmt_after_resume_quirk) { + stop_endpoints(subs, true); + subs->need_setup_fmt = true; + return 0; + } + break; } return -EINVAL; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 70c338f3ae24..d187aa6d50db 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3466,7 +3466,8 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .vendor_name = "Dell", .product_name = "WD19 Dock", .profile_name = "Dell-WD15-Dock", - .ifnum = QUIRK_NO_INTERFACE + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_FMT_AFTER_RESUME } }, /* MOTU Microbook II */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 349e1e52996d..a81c2066499f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -508,6 +508,16 @@ static int create_standard_mixer_quirk(struct snd_usb_audio *chip, return snd_usb_create_mixer(chip, quirk->ifnum, 0); } + +static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + chip->setup_fmt_after_resume_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + /* * audio-interface quirks * @@ -546,6 +556,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, + [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk, }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ff3cbf653de8..6fe3ab582ec6 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -33,7 +33,7 @@ struct snd_usb_audio { wait_queue_head_t shutdown_wait; unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ - + unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; @@ -98,6 +98,7 @@ enum quirk_type { QUIRK_AUDIO_EDIROL_UAXX, QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_AUDIO_STANDARD_MIXER, + QUIRK_SETUP_FMT_AFTER_RESUME, QUIRK_TYPE_COUNT }; -- 2.17.1

3 6

[alsa-devel] [PATCH v3 0/8] ASoC: soc-core cleanup step8
by Kuninori Morimoto 19 Dec '19

19 Dec '19

Hi Mark # did I lost email ?? These are v3 of soc-core cleanup step8. The diff from v2 are - add Reported-by, Reviewed-By for each patches - [2/8] is added - rebase to latest for-v5.6 branch Kuninori Morimoto (8): ASoC: soc-core: remove snd_soc_rtdcom_list ASoC: soc-core: rename snd_soc_rtdcom_add() to snd_soc_rtd_add_component() ASoC: soc-core: care .ignore_suspend for Component suspend ASoC: soc-core: remove duplicate pinctrl operation when suspend ASoC: soc-core: do pinctrl_pm_select_xxx() as component ASoC: soc-core: add snd_soc_close_delayed_work() ASoC: soc-dapm: add snd_soc_dapm_stream_stop() ASoC: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP include/sound/soc-dapm.h | 1 + include/sound/soc.h | 19 +++--- sound/soc/soc-component.c | 33 +++++---- sound/soc/soc-compress.c | 124 ++++++++++------------------------ sound/soc/soc-core.c | 167 +++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 23 +++++++ sound/soc/soc-pcm.c | 115 ++++++++----------------------- 7 files changed, 191 insertions(+), 291 deletions(-) -- 2.7.4

1 8

[alsa-devel] [PATCH v3 0/8] ASoC: soc-core cleanup step8
by Kuninori Morimoto 19 Dec '19

19 Dec '19

Hi Mark These are v3 of soc-core cleanup step8. The difference from v2 are... - add Reviewed-By, Reported-by for each patches - [2/8] is added - rebase to latest for-5.6 branch Kuninori Morimoto (8): 1) ASoC: soc-core: remove snd_soc_rtdcom_list 2) ASoC: soc-core: rename snd_soc_rtdcom_add() to snd_soc_rtd_add_component() 3) ASoC: soc-core: care .ignore_suspend for Component suspend 4) ASoC: soc-core: remove duplicate pinctrl operation when suspend 5) ASoC: soc-core: do pinctrl_pm_select_xxx() as component 6) ASoC: soc-core: add snd_soc_close_delayed_work() 7) ASoC: soc-dapm: add snd_soc_dapm_stream_stop() 8) ASoC: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP include/sound/soc-dapm.h | 1 + include/sound/soc.h | 19 +++--- sound/soc/soc-component.c | 33 +++++---- sound/soc/soc-compress.c | 124 ++++++++++------------------------ sound/soc/soc-core.c | 167 +++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 23 +++++++ sound/soc/soc-pcm.c | 115 ++++++++----------------------- 7 files changed, 191 insertions(+), 291 deletions(-) -- 2.7.4

1 0

[alsa-devel] [PATCH 0/7 v2] ASoC: soc-core cleanup step8
by Kuninori Morimoto 19 Dec '19

19 Dec '19

Hi Mark These are v2 of soc-core cleanup step8. There is no relationship between each patches. - [6/7] renamed function name - [7/7] new patch which cares SND_SOC_DAPM_STREAM_STOP Kuninori Morimoto (7): 1) ASoC: soc-core: remove snd_soc_rtdcom_list 2) ASoC: soc-core: care .ignore_suspend for Component suspend 3) ASoC: soc-core: remove duplicate pinctrl operation when suspend 4) ASoC: soc-core: do pinctrl_pm_select_xxx() as component 5) ASoC: soc-core: add snd_soc_close_delayed_work() 6) ASoC: soc-dapm: add snd_soc_dapm_stream_stop() 7) ASoC: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP include/sound/soc-dapm.h | 1 + include/sound/soc.h | 19 +++--- sound/soc/soc-component.c | 33 +++++----- sound/soc/soc-compress.c | 112 +++++++++------------------------ sound/soc/soc-core.c | 157 +++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 23 +++++++ sound/soc/soc-pcm.c | 115 ++++++++------------------------- 7 files changed, 181 insertions(+), 279 deletions(-) -- 2.7.4

5 17

[alsa-devel] [PATCH] ASoC: SOF: Intel: add PCI ID for CometLake-S
by Pierre-Louis Bossart 18 Dec '19

18 Dec '19

Mirror ID added for legacy HDaudio Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/Kconfig | 16 ++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 4 ++++ 2 files changed, 20 insertions(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index cc09bb606f7d..91c8bbcc015a 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -27,6 +27,7 @@ config SND_SOC_SOF_INTEL_PCI select SND_SOC_SOF_ICELAKE if SND_SOC_SOF_ICELAKE_SUPPORT select SND_SOC_SOF_COMETLAKE_LP if SND_SOC_SOF_COMETLAKE_LP_SUPPORT select SND_SOC_SOF_COMETLAKE_H if SND_SOC_SOF_COMETLAKE_H_SUPPORT + select SND_SOC_SOF_COMETLAKE_S if SND_SOC_SOF_COMETLAKE_S_SUPPORT select SND_SOC_SOF_TIGERLAKE if SND_SOC_SOF_TIGERLAKE_SUPPORT select SND_SOC_SOF_ELKHARTLAKE if SND_SOC_SOF_ELKHARTLAKE_SUPPORT select SND_SOC_SOF_JASPERLAKE if SND_SOC_SOF_JASPERLAKE_SUPPORT @@ -231,6 +232,21 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT Say Y if you have such a device. If unsure select "N". +config SND_SOC_SOF_COMETLAKE_S + tristate + select SND_SOC_SOF_HDA_COMMON + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_COMETLAKE_S_SUPPORT + bool "SOF support for CometLake-S" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Cometlake-H processors. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_SOF_TIGERLAKE_SUPPORT bool "SOF support for Tigerlake" help diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bbeffd932de7..0b39ea6117cf 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -417,6 +417,10 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0x06c8), .driver_data = (unsigned long)&cml_desc}, #endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_S) + { PCI_DEVICE(0x8086, 0xa3f0), + .driver_data = (unsigned long)&cml_desc}, +#endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) { PCI_DEVICE(0x8086, 0xa0c8), .driver_data = (unsigned long)&tgl_desc}, -- 2.20.1

3 10

[alsa-devel] [PATCH v2 0/3] More stability fixes for corner cases.
by Pierre-Louis Bossart 18 Dec '19

18 Dec '19

It wasn't straightforward to add a Fixes tag for the two Intel patches, it's likely issues that can be applied to 5.3, possibly earlier. For Dragos' patch Ranjani mentioned this may be due to da704f26ba376 ('ASoC: soc-core: merge snd_soc_remove_dai_link() and soc_unbind_dai_link()'), but Morimoto-san may need to confirm. Changes since v1: Fixed conflict in 3rd patch with Morimoto-san changes (rename of snd_soc_remove_dai_link) Dragos Tarcatu (1): ASoC: topology: Prevent use-after-free in snd_soc_get_pcm_runtime() Kai Vehmanen (1): ASoC: SOF: fix fault at driver unload after failed probe Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda: hda-dai: fix oops on hda_link .hw_free sound/soc/soc-topology.c | 8 ++++---- sound/soc/sof/intel/hda-dai.c | 11 +++++++++-- sound/soc/sof/ipc.c | 3 +++ 3 files changed, 16 insertions(+), 6 deletions(-) -- 2.20.1

3 7

[alsa-devel] [PATCH] ASoC: Intel: cml_rt1011_rt5682: fix codec_conf by removing legacy style
by Pierre-Louis Bossart 18 Dec '19

18 Dec '19

Now that the legacy style is removed, we have to use the new macros for the codec configuration. This change was missed in the initial series. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Fixes: ee8f537fd8b71c ("ASoC: soc-core: remove legacy style of codec_conf") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index a22f97234201..ab1196108d23 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -406,19 +406,19 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { static struct snd_soc_codec_conf rt1011_conf[] = { { - .dev_name = "i2c-10EC1011:00", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:00"), .name_prefix = "WL", }, { - .dev_name = "i2c-10EC1011:01", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, { - .dev_name = "i2c-10EC1011:02", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), .name_prefix = "TL", }, { - .dev_name = "i2c-10EC1011:03", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), .name_prefix = "TR", }, }; -- 2.20.1

4 4

[alsa-devel] [PATCH 0/8] ASoC: SOF: Intel: add module namespace support
by Pierre-Louis Bossart 18 Dec '19

18 Dec '19

A lot of the existing SOF/Intel drivers expose ops structures or helpers that should not be visible or loadable by other drivers. During the last round of SoundWire reviews, GregKH suggested the use of module namespaces [1] to avoid exposing symbols that are only meant to be used in a restricted context. This patchset suggests the use of namespaces for the different Intel parts, largely modeled after the existing hiearchical Kconfig options. While several SOF reviewers suggested on GitHub that this use of namespaces was too fine-grained, it can also be argued that this level of detail helps identify cases where the existing structures are not properly defined. We found such a case with the Xtensa architecture abstraction and another code partitioning issue for the SoundWire driver yesterday. Module namespaces are currently not used widely in kernel circles, this is a first attempt to test the water for audio drivers and gather comments from maintainers/reviewers. [1] https://www.kernel.org/doc/html/latest/core-api/symbol-namespaces.html Pierre-Louis Bossart (8): ASoC: SOF: Intel: add module namespace for legacy IPC ASoC: SOF: Intel: add namespaces for BAYTRAIL and MERRIFIELD ASoC: SOF: Intel: add namespace for BROADWELL ASoC: SOF: remove references to Haswell ASoC: SOF: Intel: add namespace for HDA_COMMON ASoC: SOF: Intel: hda: add namespace for hda-codec functionality ASoC: SOF: move arch_ops under ops ASoC: SOF: Intel: add namespace for XTENSA include/sound/sof.h | 1 - sound/soc/sof/Kconfig | 2 +- sound/soc/sof/intel/apl.c | 6 ++++-- sound/soc/sof/intel/bdw.c | 8 ++++++-- sound/soc/sof/intel/byt.c | 20 ++++++++++++++------ sound/soc/sof/intel/cnl.c | 14 ++++++++------ sound/soc/sof/intel/hda-codec.c | 14 +++++++------- sound/soc/sof/intel/hda.c | 3 +++ sound/soc/sof/intel/intel-ipc.c | 8 ++++---- sound/soc/sof/intel/shim.h | 6 ++---- sound/soc/sof/sof-acpi-dev.c | 26 ++------------------------ sound/soc/sof/sof-pci-dev.c | 12 ++---------- sound/soc/sof/sof-priv.h | 4 +++- sound/soc/sof/xtensa/core.c | 2 +- 14 files changed, 57 insertions(+), 69 deletions(-) -- 2.20.1

2 16

[alsa-devel] [PATCH 0/8] ASoC: SOF: last patches for 2019
by Pierre-Louis Bossart 18 Dec '19

18 Dec '19

This patchset adds a set of somewhat independent changes for i.MX, Intel, boards and core. Guido Roncarolo (2): ASoC: SOF: imx: Describe SAI parameters to be sent to DSP ASoC: SOF: imx: Read SAI parameters and send them to DSP Kai Vehmanen (2): ASoC: SOF: Intel: drop HDA codec upon probe failure ASoC: Intel: boards: fix incorrect HDMI Kconfig dependency Karol Trzcinski (2): ASoC: SOF: define struct with compiler name and version ASoC: SOF: log compiler name and version information Ranjani Sridharan (1): ASoC: SOF: Introduce state machine for FW boot Yong Zhi (1): ASoC: Intel: Add machine driver for da7219_max98373 include/sound/sof/dai-imx.h | 20 + include/sound/sof/dai.h | 1 + include/sound/sof/info.h | 15 + include/uapi/sound/sof/tokens.h | 3 +- sound/soc/intel/boards/Kconfig | 28 +- sound/soc/intel/boards/Makefile | 3 + sound/soc/intel/boards/sof_da7219_max98373.c | 371 ++++++++++++++++++ .../intel/common/soc-acpi-intel-jsl-match.c | 7 + sound/soc/sof/core.c | 50 ++- sound/soc/sof/intel/hda-codec.c | 12 +- sound/soc/sof/intel/hda-loader.c | 1 - sound/soc/sof/intel/hda.c | 4 +- sound/soc/sof/intel/hda.h | 4 +- sound/soc/sof/ipc.c | 17 +- sound/soc/sof/loader.c | 58 ++- sound/soc/sof/pcm.c | 8 + sound/soc/sof/pm.c | 21 +- sound/soc/sof/sof-priv.h | 12 +- sound/soc/sof/topology.c | 69 +++- 19 files changed, 662 insertions(+), 42 deletions(-) create mode 100644 sound/soc/intel/boards/sof_da7219_max98373.c -- 2.20.1

2 16

[alsa-devel] [PATCH 0/4] ASoC: meson: axg-fifo: fix depth handling
by Jerome Brunet 18 Dec '19

18 Dec '19

This patchset improves the handling the depth of the different fifos found in amlogic audio subsystems derived from the axg family. Initially, it was reported that, when the period is less than 8kB on sm1 and with TODDR A, several period are coming at once. This is because the field related to the fifo request threshold changed on the sm1 SoC and also because TODDR A instance is significantly larger compared to the other instances. With this initial issue fixed, I used the opportunity to improve the driver and properly deal with the different fifo depth found in these SoCs. Jerome Brunet (4): ASoC: meson: axg-fifo: fix fifo threshold setup ASoC: meson: axg-fifo: add fifo depth to the bindings documentation ASoC: meson: axg-fifo: improve depth handling ASoC: meson: axg-fifo: relax period size constraints .../bindings/sound/amlogic,axg-fifo.txt | 4 ++ sound/soc/meson/axg-fifo.c | 50 ++++++++++++++++--- sound/soc/meson/axg-fifo.h | 9 ++-- sound/soc/meson/axg-frddr.c | 33 ++++++------ sound/soc/meson/axg-toddr.c | 21 ++++---- 5 files changed, 76 insertions(+), 41 deletions(-) -- 2.23.0

2 10

[alsa-devel] [PATCH v4] ASoC: intel: Add Broadwell rt5650 machine driver
by Curtis Malainey 18 Dec '19

18 Dec '19

From: Ben Zhang <benzh(a)chromium.org> Add machine driver for Broadwell + rt5650. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Jon Flatley <jflat(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- v2: * Addressed Pierre's comments * Added SOF dependencies * Added platform override * Fixed Kconfig * Moved to devm register v3: * Removed THIS_MODULE * Added PM Ops v4: * Fixed sparse warnings sound/soc/intel/boards/Kconfig | 12 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 319 ++++++++++++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 7 + 4 files changed, 340 insertions(+) create mode 100644 sound/soc/intel/boards/bdw-rt5650.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6c9fd9ad566ed..de613ae5052bf 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -31,6 +31,18 @@ endif ## SND_SOC_INTEL_HASWELL if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL +config SND_SOC_INTEL_BDW_RT5650_MACH + tristate "Broadwell with RT5650 codec" + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT5645 + help + This adds the ASoC machine driver for Intel Broadwell platforms with + the RT5650 codec. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index ba1aa89db09d5..4efb3689503f8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o @@ -37,6 +38,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da721 obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c new file mode 100644 index 0000000000000..ba3fc1ef900ab --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,319 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * ASoC machine driver for Intel Broadwell platforms with RT5650 codec + * + * Copyright 2019, The Chromium OS Authors. All rights reserved. + */ + +#include <linux/delay.h> +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt5645.h" + +struct bdw_rt5650_priv { + struct gpio_desc *gpio_hp_en; + struct snd_soc_component *component; +}; + +static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("DMIC Pair1", NULL), + SND_SOC_DAPM_MIC("DMIC Pair2", NULL), +}; + +static const struct snd_soc_dapm_route bdw_rt5650_map[] = { + /* Speakers */ + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + + /* Headset jack connectors */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + + /* Digital MICs + * DMIC Pair1 are the two DMICs connected on the DMICN1 connector. + * DMIC Pair2 are the two DMICs connected on the DMICN2 connector. + * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2 + * are on the right side. + */ + {"DMIC L1", NULL, "DMIC Pair1"}, + {"DMIC R1", NULL, "DMIC Pair1"}, + {"DMIC L2", NULL, "DMIC Pair2"}, + {"DMIC R2", NULL, "DMIC Pair2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static const struct snd_kcontrol_new bdw_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("DMIC Pair1"), + SOC_DAPM_PIN_SWITCH("DMIC Pair2"), +}; + + +static struct snd_soc_jack headphone_jack; +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin headphone_jack_pin = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, max 4-channels */ + rate->min = rate->max = 48000; + channels->min = 2; + channels->max = 4; + + /* set SSP0 to 24 bit */ + snd_mask_set_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Workaround: set codec PLL to 19.2MHz that PLL source is + * from MCLK(24MHz) to conform 2.4MHz DMIC clock. + */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + 24000000, 19200000); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* The actual MCLK freq is 24MHz. The codec is told that MCLK is + * 24.576MHz to satisfy the requirement of rl6231_get_clk_info. + * ASRC is enabled on AD and DA filters to ensure good audio quality. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops bdw_rt5650_ops = { + .hw_params = bdw_rt5650_hw_params, +}; + +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct sst_pdata *pdata = dev_get_platdata(component->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings + * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz + */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + return 0; +} +#endif + +static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) +{ + struct bdw_rt5650_priv *bdw_rt5650 = + snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. + * The ASRC clock source is clk_i2s1_asrc. + */ + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER | + RT5645_AD_MONO_L_FILTER | + RT5645_AD_MONO_R_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* Create and initialize headphone jack */ + if (snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &headphone_jack, + &headphone_jack_pin, 1)) { + dev_err(component->dev, "Can't create headphone jack\n"); + } + + /* Create and initialize mic jack */ + if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &mic_jack, &mic_jack_pin, 1)) { + dev_err(component->dev, "Can't create mic jack\n"); + } + + rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL); + + bdw_rt5650->component = component; + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +SND_SOC_DAILINK_DEF(dummy, + DAILINK_COMP_ARRAY(COMP_DUMMY())); + +SND_SOC_DAILINK_DEF(fe, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); + +SND_SOC_DAILINK_DEF(be, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); + +static struct snd_soc_dai_link bdw_rt5650_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .dynamic = 1, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + .init = bdw_rt5650_rtd_init, +#endif + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(fe, dummy, platform), + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &bdw_rt5650_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = bdw_rt5650_init, + SND_SOC_DAILINK_REG(dummy, be, dummy), + }, +}; + +/* ASoC machine driver for Broadwell DSP + RT5650 */ +static struct snd_soc_card bdw_rt5650_card = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + .dai_link = bdw_rt5650_dais, + .num_links = ARRAY_SIZE(bdw_rt5650_dais), + .dapm_widgets = bdw_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets), + .dapm_routes = bdw_rt5650_map, + .num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map), + .controls = bdw_rt5650_controls, + .num_controls = ARRAY_SIZE(bdw_rt5650_controls), + .fully_routed = true, +}; + +static int bdw_rt5650_probe(struct platform_device *pdev) +{ + struct bdw_rt5650_priv *bdw_rt5650; + struct snd_soc_acpi_mach *mach; + int ret; + + bdw_rt5650_card.dev = &pdev->dev; + + /* Allocate driver private struct */ + bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv), + GFP_KERNEL); + if (!bdw_rt5650) + return -ENOMEM; + + /* override plaform name, if required */ + mach = (&pdev->dev)->platform_data; + ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card, + mach->mach_params.platform); + + if (ret) + return ret; + + snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650); + + return devm_snd_soc_register_card(&pdev->dev, &bdw_rt5650_card); +} + +static struct platform_driver bdw_rt5650_audio = { + .probe = bdw_rt5650_probe, + .driver = { + .name = "bdw-rt5650", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(bdw_rt5650_audio) + +/* Module information */ +MODULE_AUTHOR("Ben Zhang <benzh(a)chromium.org>"); +MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bdw-rt5650"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 34eb0baaa9517..35958553652ec 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -29,6 +29,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .sof_fw_filename = "sof-bdw.ri", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, + { + .id = "10EC5650", + .drv_name = "bdw-rt5650", + .fw_filename = "intel/IntcSST2.bin", + .sof_fw_filename = "sof-bdw.ri", + .sof_tplg_filename = "sof-bdw-rt5650.tplg", + }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", -- 2.24.1.735.g03f4e72817-goog

3 2

[alsa-devel] [PATCH] ALSA: ice1724: Fix sleep-in-atomic in Infrasonic Quartet support code
by Takashi Iwai 18 Dec '19

18 Dec '19

Jia-Ju Bai reported a possible sleep-in-atomic scenario in the ice1724 driver with Infrasonic Quartet support code: namely, ice->set_rate callback gets called inside ice->reg_lock spinlock, while the callback in quartet.c holds ice->gpio_mutex. This patch fixes the invalid call: it simply moves the calls of ice->set_rate and ice->set_mclk callbacks outside the spinlock. Reported-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/5d43135e-73b9-a46a-2155-9e91d0dcdf83@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ice1712/ice1724.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index c80a16ee6e76..242542e23d28 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -647,6 +647,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, unsigned long flags; unsigned char mclk_change; unsigned int i, old_rate; + bool call_set_rate = false; if (rate > ice->hw_rates->list[ice->hw_rates->count - 1]) return -EINVAL; @@ -670,7 +671,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, * setting clock rate for internal clock mode */ old_rate = ice->get_rate(ice); if (force || (old_rate != rate)) - ice->set_rate(ice, rate); + call_set_rate = true; else if (rate == ice->cur_rate) { spin_unlock_irqrestore(&ice->reg_lock, flags); return 0; @@ -678,12 +679,14 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, } ice->cur_rate = rate; + spin_unlock_irqrestore(&ice->reg_lock, flags); + + if (call_set_rate) + ice->set_rate(ice, rate); /* setting master clock */ mclk_change = ice->set_mclk(ice, rate); - spin_unlock_irqrestore(&ice->reg_lock, flags); - if (mclk_change && ice->gpio.i2s_mclk_changed) ice->gpio.i2s_mclk_changed(ice); if (ice->gpio.set_pro_rate) -- 2.16.4

1 0

[alsa-devel] [BUG] ALSA: ice1712: a possible sleep-in-atomic-context bug in snd_vt1724_set_pro_rate()
by Jia-Ju Bai 18 Dec '19

18 Dec '19

The driver may sleep while holding a read lock. The function call path (from bottom to top) in Linux 4.19 is: sound/pci/ice1712/quartet.c, 414: mutex_lock in reg_write sound/pci/ice1712/quartet.c, 485: reg_write in set_cpld sound/pci/ice1712/quartet.c, 876: set_cpld in qtet_set_rate sound/pci/ice1712/ice1724.c, 687: (FUNC_PTR) qtet_set_rate in snd_vt1724_set_pro_rate sound/pci/ice1712/ice1724.c, 668: _raw_spin_lock_irqsave in snd_vt1724_set_pro_rate (FUNC_PTR) means a function pointer is called. mutex_lock() can sleep at runtime. I am not sure how to properly fix this possible bug, so I only report it. This bug is found by a static analysis tool STCheck written by myself. Best wishes, Jia-Ju Bai

2 1

[alsa-devel] Playback issue while recording simultaneously
by Mohan Prasath Ramamoorthy 18 Dec '19

18 Dec '19

Dear All, We are working on external modules of microphone and speaker and we are using the kernel version 3.10.96 We are recording with commands like arecord and aplay for recording and playback respectively. While doing the below experiments we could able to observe an issue as below, 1. With only recording and not playback we haven't faced any issues. 2. With only playback and not recording we haven't faced any issues. 3. By starting an infinite recording in the background and by doing a playback of the audio file(30seconds, the sampling rate of 22050, the channel is stereo), after some iterations, we are facing a below error and we could not able to listen to the audio. Error: aplay: pcm_write:1940: write error: Input/output error Below are the respective commands for recording and playback, 1. arecord -D hw:0,0 -c 2 -f s16_le -r 8000 /home/ubuntu/mic_testing.wav & 2. aplay -D hw:0,3 audio_22k_c2.wav But when the issues come, we just stopped recording and restarted the recording in the background. After this, the playback is working fine. Also, during the issue came, we have checked for the below properties, 1. I2S has been enabled. Checked under the file: /sys/kernel/debug/clock/clock_tree 2. Driver callbacks have been called. (Checked with the kernel logs[dmesg]) We would like to know the reason behind this error.? -- With Best Regards, Mohan Prasath VVDN Technologies Pvt Ltd Cell: +91 9629768919 Web: www.vvdntech.com -- _Disclaimer: _© 2019 VVDN Technologies Pvt. Ltd. This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful._ _ _ _ __

2 4

[alsa-devel] [INTERNAL REVIEW] asoc: sti: fix possible sleep-in-atomic
by Arnaud Pouliquen 18 Dec '19

18 Dec '19

Change mutex and spinlock management to avoid sleep in atomic issue. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 48ea915b24ba..2ed92c990b97 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -226,7 +226,6 @@ static void uni_player_set_channel_status(struct uniperif *player, * sampling frequency. If no sample rate is already specified, then * set one. */ - mutex_lock(&player->ctrl_lock); if (runtime) { switch (runtime->rate) { case 22050: @@ -303,7 +302,6 @@ static void uni_player_set_channel_status(struct uniperif *player, player->stream_settings.iec958.status[3 + (n * 4)] << 24; SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status); } - mutex_unlock(&player->ctrl_lock); /* Update the channel status */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -365,8 +363,10 @@ static int uni_player_prepare_iec958(struct uniperif *player, SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player); + mutex_lock(&player->ctrl_lock); /* Update the channel status */ uni_player_set_channel_status(player, runtime); + mutex_unlock(&player->ctrl_lock); /* Clear the user validity user bits */ SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); @@ -598,7 +598,6 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[1] = ucontrol->value.iec958.status[1]; iec958->status[2] = ucontrol->value.iec958.status[2]; iec958->status[3] = ucontrol->value.iec958.status[3]; - mutex_unlock(&player->ctrl_lock); spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) @@ -608,6 +607,8 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, uni_player_set_channel_status(player, NULL); spin_unlock_irqrestore(&player->irq_lock, flags); + mutex_unlock(&player->ctrl_lock); + return 0; } -- 2.17.1

1 0

[alsa-devel] [BUG] ALSA: soc: sti: a possible sleep-in-atomic-context bug in uni_player_ctl_iec958_put()
by Jia-Ju Bai 18 Dec '19

18 Dec '19

The driver may sleep while holding a spinlock. The function call path (from bottom to top) in Linux 4.19 is: sound/soc/sti/uniperif_player.c, 229: mutex_lock in uni_player_set_channel_status sound/soc/sti/uniperif_player.c, 608: uni_player_set_channel_status in uni_player_ctl_iec958_put sound/soc/sti/uniperif_player.c, 603: _raw_spin_lock_irqsave in uni_player_ctl_iec958_put mutex_lock() can sleep at runtime. I am not sure how to properly fix this possible bug, so I only report it. This bug is found by a static analysis tool STCheck written by myself. Best wishes, Jia-Ju Bai

2 1

[alsa-devel] [PATCH 0/6] ASoC: soc-core cleanup step8
by Kuninori Morimoto 18 Dec '19

18 Dec '19

Hi Mark These are soc-core cleanup step8. There is no relationship between each patches. Kuninori Morimoto (6): ASoC: soc-core: remove snd_soc_rtdcom_list ASoC: soc-core: care .ignore_suspend for Component suspend ASoC: soc-core: remove duplicate pinctrl operation when suspend ASoC: soc-core: do pinctrl_pm_select_xxx() as component ASoC: soc-core: add snd_soc_close_delayed_work() ASoC: soc-core: add snd_soc_stream_stop() include/sound/soc.h | 20 +++--- sound/soc/soc-component.c | 33 +++++---- sound/soc/soc-compress.c | 110 ++++++++-------------------- sound/soc/soc-core.c | 180 +++++++++++++++++++++++++--------------------- sound/soc/soc-pcm.c | 113 +++++++---------------------- 5 files changed, 179 insertions(+), 277 deletions(-) -- 2.7.4

3 13

[alsa-devel] Undecalared Variable "p" when compile alsa-lib with TPLG_DEBUG defined
by GitHub issues - opened 18 Dec '19

18 Dec '19

alsa-project/alsa-lib issue #18 was opened from aiChaoSONG: Undecalared Variable "p" when compile alsa-lib with TPLG_DEBUG defined, the pointer 'p' in line 152 of src/topology/data.c seem not declared. see code below. ``` static void dump_priv_data(struct tplg_elem *elem) { struct snd_soc_tplg_private *priv = elem->data; unsigned int i, j = 0; tplg_dbg(" elem size = %d, priv data size = %d\n", elem->size, priv->size); for (i = 0; i < priv->size; i++) { if (j++ % 8 == 0) tplg_dbg("\n"); *******tplg_dbg(" 0x%x", *p++);******* } tplg_dbg("\n\n"); } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/18 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] TI McASP: Independent RX & TX
by Sven Schwermer 18 Dec '19

18 Dec '19

Hi, I’m looking at hooking up two codecs to a single AM3358’s McASP instance. One codec will be hooked up to the receive section while the other one will be hooked up to the transmit section. BCLK, MCLK and format will not necessarily be shared and may differ. As far as I understand, the McASP allows to operate receive and transmit pretty much independently. Looking at the ASoC driver (davinci-mcasp.c), however, it seems to me that it is not really possible to use receive and transmit independently. Is that correct? Is that particular use case not within the scope of the current driver or is my initial assumption wrong that transmit and receive are basically independent? Any pointers are appreciated. Best regards, Sven

1 0

[alsa-devel] [PATCH] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Nathan Chancellor 18 Dec '19

18 Dec '19

Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index d1caa8ed9e68..9985fc139487 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -119,7 +119,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.24.1

2 1

[alsa-devel] [PATCH] ASoC: soc-core: rename snd_soc_rtdcom_add() to snd_soc_rtd_add_component()
by Kuninori Morimoto 18 Dec '19

18 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We don't have snd_soc_rtdcom_list anymore. Let's rename snd_soc_rtdcom_add() to more understandable snd_soc_rtd_add_component() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This is based on my previous soc-core cleanup patch-set. sound/soc/soc-core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f0217b8..554409d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -257,8 +257,8 @@ static inline void snd_soc_debugfs_exit(void) #endif -static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, - struct snd_soc_component *component) +static int snd_soc_rtd_add_component(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_component *component) { struct snd_soc_component *comp; int i; @@ -1010,7 +1010,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, cpu->dai_name); goto _err_defer; } - snd_soc_rtdcom_add(rtd, asoc_cpu_dai(rtd, i)->component); + snd_soc_rtd_add_component(rtd, asoc_cpu_dai(rtd, i)->component); } /* Find CODEC from registered CODECs */ @@ -1022,7 +1022,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, goto _err_defer; } - snd_soc_rtdcom_add(rtd, asoc_codec_dai(rtd, i)->component); + snd_soc_rtd_add_component(rtd, asoc_codec_dai(rtd, i)->component); } /* Find PLATFORM from registered PLATFORMs */ @@ -1031,7 +1031,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, if (!snd_soc_is_matching_component(platform, component)) continue; - snd_soc_rtdcom_add(rtd, component); + snd_soc_rtd_add_component(rtd, component); } } -- 2.7.4

1 0

[alsa-devel] [PATCH v8 6/9] ALSA: Avoid using timespec for struct snd_timer_tread
by Arnd Bergmann 17 Dec '19

17 Dec '19

From: Baolin Wang <baolin.wang(a)linaro.org> The struct snd_timer_tread will use 'timespec' type variables to record timestamp, which is not year 2038 safe on 32bits system. Since the struct snd_timer_tread is passed through read() rather than ioctl(), and the read syscall has no command number that lets us pick between the 32-bit or 64-bit version of this structure. Thus we introduced one new command SNDRV_TIMER_IOCTL_TREAD64 and new struct snd_timer_tread64 replacing timespec with s64 type to handle 64bit time_t. That means we will set tu->tread = TREAD_FORMAT_64BIT when user space has a 64bit time_t, then we will copy to user with struct snd_timer_tread64. Otherwise we will use 32bit time_t variables when copying to user. Moreover this patch replaces timespec type with timespec64 type and related y2038 safe APIs. Signed-off-by: Baolin Wang <baolin.wang(a)linaro.org> Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- This v8 is what is in the pull request "[GIT PULL, v8] Fix year 2038 issue for sound subsystem", posted here for reference include/uapi/sound/asound.h | 15 +++- sound/core/timer.c | 150 +++++++++++++++++++++++++++--------- sound/core/timer_compat.c | 5 +- 3 files changed, 131 insertions(+), 39 deletions(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index e0ada33afa1e..ad86c5a7a1e2 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -783,7 +783,7 @@ struct snd_timer_status { #define SNDRV_TIMER_IOCTL_PVERSION _IOR('T', 0x00, int) #define SNDRV_TIMER_IOCTL_NEXT_DEVICE _IOWR('T', 0x01, struct snd_timer_id) -#define SNDRV_TIMER_IOCTL_TREAD _IOW('T', 0x02, int) +#define SNDRV_TIMER_IOCTL_TREAD_OLD _IOW('T', 0x02, int) #define SNDRV_TIMER_IOCTL_GINFO _IOWR('T', 0x03, struct snd_timer_ginfo) #define SNDRV_TIMER_IOCTL_GPARAMS _IOW('T', 0x04, struct snd_timer_gparams) #define SNDRV_TIMER_IOCTL_GSTATUS _IOWR('T', 0x05, struct snd_timer_gstatus) @@ -796,6 +796,15 @@ struct snd_timer_status { #define SNDRV_TIMER_IOCTL_STOP _IO('T', 0xa1) #define SNDRV_TIMER_IOCTL_CONTINUE _IO('T', 0xa2) #define SNDRV_TIMER_IOCTL_PAUSE _IO('T', 0xa3) +#define SNDRV_TIMER_IOCTL_TREAD64 _IOW('T', 0xa4, int) + +#if __BITS_PER_LONG == 64 +#define SNDRV_TIMER_IOCTL_TREAD SNDRV_TIMER_IOCTL_TREAD_OLD +#else +#define SNDRV_TIMER_IOCTL_TREAD ((sizeof(__kernel_long_t) >= sizeof(time_t)) ? \ + SNDRV_TIMER_IOCTL_TREAD_OLD : \ + SNDRV_TIMER_IOCTL_TREAD64) +#endif struct snd_timer_read { unsigned int resolution; @@ -821,11 +830,15 @@ enum { SNDRV_TIMER_EVENT_MRESUME = SNDRV_TIMER_EVENT_RESUME + 10, }; +#ifndef __KERNEL__ struct snd_timer_tread { int event; + __time_pad pad1; struct timespec tstamp; unsigned int val; + __time_pad pad2; }; +#endif /**************************************************************************** * * diff --git a/sound/core/timer.c b/sound/core/timer.c index d2f69039f941..4cfd8e691903 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -44,6 +44,28 @@ MODULE_PARM_DESC(timer_tstamp_monotonic, "Use posix monotonic clock source for t MODULE_ALIAS_CHARDEV(CONFIG_SND_MAJOR, SNDRV_MINOR_TIMER); MODULE_ALIAS("devname:snd/timer"); +enum timer_tread_format { + TREAD_FORMAT_NONE = 0, + TREAD_FORMAT_TIME64, + TREAD_FORMAT_TIME32, +}; + +struct snd_timer_tread32 { + int event; + s32 tstamp_sec; + s32 tstamp_nsec; + unsigned int val; +}; + +struct snd_timer_tread64 { + int event; + u8 pad1[4]; + s64 tstamp_sec; + s64 tstamp_nsec; + unsigned int val; + u8 pad2[4]; +}; + struct snd_timer_user { struct snd_timer_instance *timeri; int tread; /* enhanced read with timestamps and events */ @@ -55,7 +77,7 @@ struct snd_timer_user { int queue_size; bool disconnected; struct snd_timer_read *queue; - struct snd_timer_tread *tqueue; + struct snd_timer_tread64 *tqueue; spinlock_t qlock; unsigned long last_resolution; unsigned int filter; @@ -1329,7 +1351,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } static void snd_timer_user_append_to_tqueue(struct snd_timer_user *tu, - struct snd_timer_tread *tread) + struct snd_timer_tread64 *tread) { if (tu->qused >= tu->queue_size) { tu->overrun++; @@ -1346,7 +1368,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, unsigned long resolution) { struct snd_timer_user *tu = timeri->callback_data; - struct snd_timer_tread r1; + struct snd_timer_tread64 r1; unsigned long flags; if (event >= SNDRV_TIMER_EVENT_START && @@ -1356,7 +1378,8 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, return; memset(&r1, 0, sizeof(r1)); r1.event = event; - r1.tstamp = timespec64_to_timespec(*tstamp); + r1.tstamp_sec = tstamp->tv_sec; + r1.tstamp_nsec = tstamp->tv_nsec; r1.val = resolution; spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); @@ -1378,7 +1401,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, unsigned long ticks) { struct snd_timer_user *tu = timeri->callback_data; - struct snd_timer_tread *r, r1; + struct snd_timer_tread64 *r, r1; struct timespec64 tstamp; int prev, append = 0; @@ -1399,7 +1422,8 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && tu->last_resolution != resolution) { r1.event = SNDRV_TIMER_EVENT_RESOLUTION; - r1.tstamp = timespec64_to_timespec(tstamp); + r1.tstamp_sec = tstamp.tv_sec; + r1.tstamp_nsec = tstamp.tv_nsec; r1.val = resolution; snd_timer_user_append_to_tqueue(tu, &r1); tu->last_resolution = resolution; @@ -1413,14 +1437,16 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, prev = tu->qtail == 0 ? tu->queue_size - 1 : tu->qtail - 1; r = &tu->tqueue[prev]; if (r->event == SNDRV_TIMER_EVENT_TICK) { - r->tstamp = timespec64_to_timespec(tstamp); + r->tstamp_sec = tstamp.tv_sec; + r->tstamp_nsec = tstamp.tv_nsec; r->val += ticks; append++; goto __wake; } } r1.event = SNDRV_TIMER_EVENT_TICK; - r1.tstamp = timespec64_to_timespec(tstamp); + r1.tstamp_sec = tstamp.tv_sec; + r1.tstamp_nsec = tstamp.tv_nsec; r1.val = ticks; snd_timer_user_append_to_tqueue(tu, &r1); append++; @@ -1435,7 +1461,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, static int realloc_user_queue(struct snd_timer_user *tu, int size) { struct snd_timer_read *queue = NULL; - struct snd_timer_tread *tqueue = NULL; + struct snd_timer_tread64 *tqueue = NULL; if (tu->tread) { tqueue = kcalloc(size, sizeof(*tqueue), GFP_KERNEL); @@ -1874,11 +1900,11 @@ static int snd_timer_user_params(struct file *file, tu->qhead = tu->qtail = tu->qused = 0; if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) { if (tu->tread) { - struct snd_timer_tread tread; + struct snd_timer_tread64 tread; memset(&tread, 0, sizeof(tread)); tread.event = SNDRV_TIMER_EVENT_EARLY; - tread.tstamp.tv_sec = 0; - tread.tstamp.tv_nsec = 0; + tread.tstamp_sec = 0; + tread.tstamp_nsec = 0; tread.val = 0; snd_timer_user_append_to_tqueue(tu, &tread); } else { @@ -2008,6 +2034,36 @@ static int snd_timer_user_pause(struct file *file) return 0; } +static int snd_timer_user_tread(void __user *argp, struct snd_timer_user *tu, + unsigned int cmd, bool compat) +{ + int __user *p = argp; + int xarg, old_tread; + + if (tu->timeri) /* too late */ + return -EBUSY; + if (get_user(xarg, p)) + return -EFAULT; + + old_tread = tu->tread; + + if (!xarg) + tu->tread = TREAD_FORMAT_NONE; + else if (cmd == SNDRV_TIMER_IOCTL_TREAD64 || + (IS_ENABLED(CONFIG_64BIT) && !compat)) + tu->tread = TREAD_FORMAT_TIME64; + else + tu->tread = TREAD_FORMAT_TIME32; + + if (tu->tread != old_tread && + realloc_user_queue(tu, tu->queue_size) < 0) { + tu->tread = old_tread; + return -ENOMEM; + } + + return 0; +} + enum { SNDRV_TIMER_IOCTL_START_OLD = _IO('T', 0x20), SNDRV_TIMER_IOCTL_STOP_OLD = _IO('T', 0x21), @@ -2016,7 +2072,7 @@ enum { }; static long __snd_timer_user_ioctl(struct file *file, unsigned int cmd, - unsigned long arg) + unsigned long arg, bool compat) { struct snd_timer_user *tu; void __user *argp = (void __user *)arg; @@ -2028,23 +2084,9 @@ static long __snd_timer_user_ioctl(struct file *file, unsigned int cmd, return put_user(SNDRV_TIMER_VERSION, p) ? -EFAULT : 0; case SNDRV_TIMER_IOCTL_NEXT_DEVICE: return snd_timer_user_next_device(argp); - case SNDRV_TIMER_IOCTL_TREAD: - { - int xarg, old_tread; - - if (tu->timeri) /* too late */ - return -EBUSY; - if (get_user(xarg, p)) - return -EFAULT; - old_tread = tu->tread; - tu->tread = xarg ? 1 : 0; - if (tu->tread != old_tread && - realloc_user_queue(tu, tu->queue_size) < 0) { - tu->tread = old_tread; - return -ENOMEM; - } - return 0; - } + case SNDRV_TIMER_IOCTL_TREAD_OLD: + case SNDRV_TIMER_IOCTL_TREAD64: + return snd_timer_user_tread(argp, tu, cmd, compat); case SNDRV_TIMER_IOCTL_GINFO: return snd_timer_user_ginfo(file, argp); case SNDRV_TIMER_IOCTL_GPARAMS: @@ -2084,7 +2126,7 @@ static long snd_timer_user_ioctl(struct file *file, unsigned int cmd, long ret; mutex_lock(&tu->ioctl_lock); - ret = __snd_timer_user_ioctl(file, cmd, arg); + ret = __snd_timer_user_ioctl(file, cmd, arg, false); mutex_unlock(&tu->ioctl_lock); return ret; } @@ -2100,13 +2142,29 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, size_t count, loff_t *offset) { + struct snd_timer_tread64 *tread; + struct snd_timer_tread32 tread32; struct snd_timer_user *tu; long result = 0, unit; int qhead; int err = 0; tu = file->private_data; - unit = tu->tread ? sizeof(struct snd_timer_tread) : sizeof(struct snd_timer_read); + switch (tu->tread) { + case TREAD_FORMAT_TIME64: + unit = sizeof(struct snd_timer_tread64); + break; + case TREAD_FORMAT_TIME32: + unit = sizeof(struct snd_timer_tread32); + break; + case TREAD_FORMAT_NONE: + unit = sizeof(struct snd_timer_read); + break; + default: + WARN_ONCE(1, "Corrupt snd_timer_user\n"); + return -ENOTSUPP; + } + mutex_lock(&tu->ioctl_lock); spin_lock_irq(&tu->qlock); while ((long)count - result >= unit) { @@ -2145,14 +2203,34 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, tu->qused--; spin_unlock_irq(&tu->qlock); - if (tu->tread) { - if (copy_to_user(buffer, &tu->tqueue[qhead], - sizeof(struct snd_timer_tread))) + tread = &tu->tqueue[qhead]; + + switch (tu->tread) { + case TREAD_FORMAT_TIME64: + if (copy_to_user(buffer, tread, + sizeof(struct snd_timer_tread64))) err = -EFAULT; - } else { + break; + case TREAD_FORMAT_TIME32: + memset(&tread32, 0, sizeof(tread32)); + tread32 = (struct snd_timer_tread32) { + .event = tread->event, + .tstamp_sec = tread->tstamp_sec, + .tstamp_sec = tread->tstamp_nsec, + .val = tread->val, + }; + + if (copy_to_user(buffer, &tread32, sizeof(tread32))) + err = -EFAULT; + break; + case TREAD_FORMAT_NONE: if (copy_to_user(buffer, &tu->queue[qhead], sizeof(struct snd_timer_read))) err = -EFAULT; + break; + default: + err = -ENOTSUPP; + break; } spin_lock_irq(&tu->qlock); diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 20eef5bc304b..0103d16f6f9f 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -83,7 +83,8 @@ static long __snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, switch (cmd) { case SNDRV_TIMER_IOCTL_PVERSION: - case SNDRV_TIMER_IOCTL_TREAD: + case SNDRV_TIMER_IOCTL_TREAD_OLD: + case SNDRV_TIMER_IOCTL_TREAD64: case SNDRV_TIMER_IOCTL_GINFO: case SNDRV_TIMER_IOCTL_GSTATUS: case SNDRV_TIMER_IOCTL_SELECT: @@ -97,7 +98,7 @@ static long __snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, case SNDRV_TIMER_IOCTL_PAUSE: case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: - return __snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + return __snd_timer_user_ioctl(file, cmd, (unsigned long)argp, true); case SNDRV_TIMER_IOCTL_GPARAMS32: return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: -- 2.20.0

1 0

Re: [alsa-devel] [PATCH v3 08/15] ASoC: tegra: Add audio mclk control through clk_out_1 and extern1
by Dmitry Osipenko 17 Dec '19

17 Dec '19

17.12.2019 19:39, Sowjanya Komatineni пишет: > > On 12/17/19 8:16 AM, Dmitry Osipenko wrote: >> 17.12.2019 19:12, Sowjanya Komatineni пишет: >>> On 12/17/19 7:36 AM, Dmitry Osipenko wrote: >>>> 17.12.2019 04:29, Sowjanya Komatineni пишет: >>>>> On 12/7/19 11:20 AM, Sowjanya Komatineni wrote: >>>>>> On 12/7/19 6:58 AM, Dmitry Osipenko wrote: >>>>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>>>> Current ASoC driver uses extern1 as cdev1 clock from Tegra30 >>>>>>>> onwards >>>>>>>> through device tree. >>>>>>>> >>>>>>>> Actual audio mclk is clk_out_1 and to use PLLA for mclk rate >>>>>>>> control, >>>>>>>> need to clk_out_1_mux parent to extern1 and extern1 parent to >>>>>>>> PLLA_OUT0. >>>>>>>> >>>>>>>> Currently Tegra clock driver init sets the parents and enables both >>>>>>>> clk_out_1 and extern1 clocks. But these clocks parent and enables >>>>>>>> should >>>>>>>> be controlled by ASoC driver. >>>>>>>> >>>>>>>> Clock parents can be specified in device tree using assigned-clocks >>>>>>>> and assigned-clock-parents. >>>>>>>> >>>>>>>> To enable audio mclk, both clk_out_1 and extern1 clocks need to be >>>>>>>> enabled. >>>>>>>> >>>>>>>> This patch configures parents for clk_out_1 and extern1 clocks if >>>>>>>> device >>>>>>>> tree does not specify clock parents inorder to support old device >>>>>>>> tree >>>>>>>> and controls mclk using both clk_out_1 and extern1 clocks. >>>>>>>> >>>>>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>>>>> --- >>>>>>>> sound/soc/tegra/tegra_asoc_utils.c | 66 >>>>>>>> ++++++++++++++++++++++++++++++++++++++ >>>>>>>> sound/soc/tegra/tegra_asoc_utils.h | 1 + >>>>>>>> 2 files changed, 67 insertions(+) >>>>>>>> >>>>>>>> diff --git a/sound/soc/tegra/tegra_asoc_utils.c >>>>>>>> b/sound/soc/tegra/tegra_asoc_utils.c >>>>>>>> index 536a578e9512..8e3a3740df7c 100644 >>>>>>>> --- a/sound/soc/tegra/tegra_asoc_utils.c >>>>>>>> +++ b/sound/soc/tegra/tegra_asoc_utils.c >>>>>>>> @@ -60,6 +60,7 @@ int tegra_asoc_utils_set_rate(struct >>>>>>>> tegra_asoc_utils_data *data, int srate, >>>>>>>> data->set_mclk = 0; >>>>>>>> clk_disable_unprepare(data->clk_cdev1); >>>>>>>> + clk_disable_unprepare(data->clk_extern1); >>>>>>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>>>>>> clk_disable_unprepare(data->clk_pll_a); >>>>>>>> @@ -89,6 +90,14 @@ int tegra_asoc_utils_set_rate(struct >>>>>>>> tegra_asoc_utils_data *data, int srate, >>>>>>>> return err; >>>>>>>> } >>>>>>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>>>>>> + err = clk_prepare_enable(data->clk_extern1); >>>>>>>> + if (err) { >>>>>>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>>>>>> + return err; >>>>>>>> + } >>>>>>>> + } >>>>>>>> + >>>>>>>> err = clk_prepare_enable(data->clk_cdev1); >>>>>>>> if (err) { >>>>>>>> dev_err(data->dev, "Can't enable cdev1: %d\n", err); >>>>>>>> @@ -109,6 +118,7 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>>>>>> tegra_asoc_utils_data *data) >>>>>>>> int err; >>>>>>>> clk_disable_unprepare(data->clk_cdev1); >>>>>>>> + clk_disable_unprepare(data->clk_extern1); >>>>>>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>>>>>> clk_disable_unprepare(data->clk_pll_a); >>>>>>>> @@ -142,6 +152,14 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>>>>>> tegra_asoc_utils_data *data) >>>>>>>> return err; >>>>>>>> } >>>>>>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>>>>>> + err = clk_prepare_enable(data->clk_extern1); >>>>>>>> + if (err) { >>>>>>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>>>>>> + return err; >>>>>>>> + } >>>>>>>> + } >>>>>>> Why this is needed given that clk_extern1 is either a child of >>>>>>> MCLK or >>>>>>> MCLK itself (on T20)? The child clocks are enabled when the >>>>>>> parent is >>>>>>> enabled. >>>>>> For T30 and later, clk_extern1 is one of the source for >>>>>> clk_out_1_mux. >>>>>> clk_extern1 is in CAR and it has its own gate and mux. >>>>>> >>>>>> As audio mclk related clocks (clk_out_1, clk_out_1_mux, and extern1) >>>>>> are moved into ASoC driver from clock driver >>>>>> >>>>>> need to enable extern1 gate as well along with clk_out1 for T30 >>>>>> through T210. >>>>>> >>>>>> Just FYI, extern1 enable here happens only when data->clk_extern1 is >>>>>> available which is for T30 onwards. >>>>>> >>>>>>>> err = clk_prepare_enable(data->clk_cdev1); >>>>>>>> if (err) { >>>>>>>> dev_err(data->dev, "Can't enable cdev1: %d\n", err); >>>>>>>> @@ -158,6 +176,7 @@ >>>>>>>> EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_ac97_rate); >>>>>>>> int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, >>>>>>>> struct device *dev) >>>>>>>> { >>>>>>>> + struct clk *clk_out_1_mux; >>>>>>>> int ret; >>>>>>>> data->dev = dev; >>>>>>>> @@ -196,6 +215,51 @@ int tegra_asoc_utils_init(struct >>>>>>>> tegra_asoc_utils_data *data, >>>>>>>> goto err_put_pll_a_out0; >>>>>>>> } >>>>>>> In a previous patch you added fallback to EXTPERIPH when >>>>>>> clk_get(MCLK) >>>>>>> fails. This will work perfectly fine for the older kernels which >>>>>>> have >>>>>>> all clocks in the same single CaR driver, but this may not work that >>>>>>> great for the newer kernels because PMC driver isn't registered >>>>>>> early >>>>>>> during boot and thus it is possible to get a legit -EPROBE_DEFER >>>>>>> which >>>>>>> shouldn't be ignored. In other words, you need to add into this >>>>>>> patch a >>>>>>> check for the error code returned by clk_get(MCLK) and fallback >>>>>>> only for >>>>>>> -EINVAL. >>>>>> yeah right, will add check in next version. >>>>>>>> + /* >>>>>>>> + * If clock parents are not set in DT, configure here to use >>>>>>>> clk_out_1 >>>>>>>> + * as mclk and extern1 as parent for Tegra30 and higher. >>>>>>>> + */ >>>>>>>> + if (!of_find_property(dev->of_node, "assigned-clock-parents", >>>>>>>> NULL) && >>>>>>>> + data->soc > TEGRA_ASOC_UTILS_SOC_TEGRA20) { >>>>>>>> + data->clk_extern1 = clk_get_sys("clk_out_1", "extern1"); >>>>>>>> + if (IS_ERR(data->clk_extern1)) { >>>>>>>> + dev_err(data->dev, "Can't retrieve clk extern1\n"); >>>>>>>> + ret = PTR_ERR(data->clk_extern1); >>>>>>>> + goto err_put_cdev1; >>>>>>>> + } >>>>>>>> + >>>>>>>> + ret = clk_set_parent(data->clk_extern1, >>>>>>>> data->clk_pll_a_out0); >>>>>>>> + if (ret < 0) { >>>>>>>> + dev_err(data->dev, >>>>>>>> + "Set parent failed for clk extern1: %d\n", >>>>>>>> + ret); >>>>>>>> + goto err_put_cdev1; >>>>>>>> + } >>>>>>>> + >>>>>>>> + clk_out_1_mux = clk_get_sys(NULL, "clk_out_1_mux"); >>>>>>> Note1: clk_get(dev, "clk_out_1_mux") should work here by letting clk >>>>>>> core to fall back to the clk_get_sys() by itself. Either way should >>>>>>> be good. >>>>> clk_get uses device rather and dev_id will be name of this device and >>>>> when clk_get fall back to __clk_get_sys() it still will use dev id of >>>>> this device rather than actual dev_id that pmc clocks are added to the >>>>> lookup. So clk_get_sys() seems to be correct to use as we can specify >>>>> exact dev_id and con_id. >>>> It should be better to use something "resource managed", thus >>>> devm_clk_get() should be a better choice. >>>> >>>>> Also, clk_find retrieves clock from lookup only when it finds matching >>>>> clock with both dev_id and con_id as pmc clocks are registered with >>>>> both >>>>> dev_id and con_id. >>>>> >>>>> I see existing clock driver adds both extern and pmc clocks >>>>> (clk_out) to >>>>> lookup with same dev_id of clk_out_1/2/3 and con_id of extern1/2/3 and >>>>> with this always extern clock will be retrieved and this is probably >>>>> because old DT and audio driver always uses extern1 rather than actual >>>>> clk_out_1 >>>>> >>>>> But this need to be fixed now as we changed to use clk_out directly >>>>> rather than extern (even for other pmc clocks) to match actual hw >>>>> design. >>>>> >>>>> Will fix this as well to register pmc clocks using con_id as >>>>> clk_out_1/2/3 in pmc driver and extern clocks using con_id of >>>>> extern1/2/3 with dev_id being NULL so we can retrieve these clocks by >>>>> just using con_id only using clk_get_sys as we switched to use >>>>> clk_out_1 >>>>> directly as pmc clock rather than extern from DT and no longer need to >>>>> pair pmc clocks to extern clocks. >>>> I'm not sure it's worth the effort to care about con_ids if implicit >>>> fallback to clk_get_sys(NULL, "...") does the right thing for the audio >>>> driver. >>>> >>>> IIRC, CCF uses variant of matching clocks by names, although I'm not >>>> sure whether that applies to older stable kernels. >>>> >>>> [snip] >>> Current clock driver adds EXTERN clock to lookup with dev_id as >>> clk_out_1/2/3 and con_id as extern_1/2/3 >>> >>> With this we can retrieve clock from lookup only with clk_get_sys where >>> we can pass dev_id as clk_out_1/2/3 and con_id as extern_1/2/3. >>> >>> We cant use devm_clk_get() or clk_get() for retrieving clocks from >>> lookup by passing device object from sound driver as dev_id will be diff >>> and clk_find will return NULL. >> Have you actually tried to test that it fails? I think it's a false >> assumption. > > Yes I tried and looking at devm_clk_get it falls back to __clk_get_sys > which uses dev_id as dev_name of the device we pass from audio driver. > > looking into clk_find API implementation it doesn't find the clock from > lookup unless both dev_id and con_id matches if a clock is added to > lookup with both dev_id and clk_id > > >>> But with the fix of having dev_id as NULL and use only con_id to add to >>> lookup, we can use resource managed APIs devm_clk_get. >>> >>> So was saying will fix this in clock driver as part of removing >>> clk_out_1/2/3 ids and pmc init from clock driver so we can use >>> devm_clk_get API in audio driver. >>> Ok

1 0

Re: [alsa-devel] linux-next: Tree for Dec 17 (sound/soc/intel/boards/cml_rt1011_rt5682.o)
by Randy Dunlap 17 Dec '19

17 Dec '19

On 12/16/19 6:08 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191216: > on x86_64: CC sound/soc/intel/boards/cml_rt1011_rt5682.o ../sound/soc/intel/boards/cml_rt1011_rt5682.c:409:4: error: ‘struct snd_soc_codec_conf’ has no member named ‘dev_name’ .dev_name = "i2c-10EC1011:00", ^~~~~~~~ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:407:50: warning: missing braces around initializer [-Wmissing-braces] static struct snd_soc_codec_conf rt1011_conf[] = { ^ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:409:15: .dev_name = "i2c-10EC1011:00", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:413:4: error: ‘struct snd_soc_codec_conf’ has no member named ‘dev_name’ .dev_name = "i2c-10EC1011:01", ^~~~~~~~ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:407:50: warning: missing braces around initializer [-Wmissing-braces] static struct snd_soc_codec_conf rt1011_conf[] = { ^ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:409:15: .dev_name = "i2c-10EC1011:00", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:413:15: .dev_name = "i2c-10EC1011:01", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:417:4: error: ‘struct snd_soc_codec_conf’ has no member named ‘dev_name’ .dev_name = "i2c-10EC1011:02", ^~~~~~~~ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:407:50: warning: missing braces around initializer [-Wmissing-braces] static struct snd_soc_codec_conf rt1011_conf[] = { ^ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:409:15: .dev_name = "i2c-10EC1011:00", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:413:15: .dev_name = "i2c-10EC1011:01", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:417:15: .dev_name = "i2c-10EC1011:02", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:421:4: error: ‘struct snd_soc_codec_conf’ has no member named ‘dev_name’ .dev_name = "i2c-10EC1011:03", ^~~~~~~~ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:407:50: warning: missing braces around initializer [-Wmissing-braces] static struct snd_soc_codec_conf rt1011_conf[] = { ^ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:409:15: .dev_name = "i2c-10EC1011:00", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:413:15: .dev_name = "i2c-10EC1011:01", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:417:15: .dev_name = "i2c-10EC1011:02", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:421:15: .dev_name = "i2c-10EC1011:03", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:407:50: warning: missing braces around initializer [-Wmissing-braces] static struct snd_soc_codec_conf rt1011_conf[] = { ^ ../sound/soc/intel/boards/cml_rt1011_rt5682.c:409:15: .dev_name = "i2c-10EC1011:00", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:413:15: .dev_name = "i2c-10EC1011:01", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:417:15: .dev_name = "i2c-10EC1011:02", { } ../sound/soc/intel/boards/cml_rt1011_rt5682.c:421:15: .dev_name = "i2c-10EC1011:03", { } make[5]: *** [../scripts/Makefile.build:266: sound/soc/intel/boards/cml_rt1011_rt5682.o] Error 1 -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

2 1

[alsa-devel] [PATCH] ALSA: cmipci: Allow disabling MPU port via module option
by Takashi Iwai 17 Dec '19

17 Dec '19

Patrick May reported that his sound card with CMI8378 chip causes a crash / reboot when accessing the MIDI port that isn't actually present on the board. Moreover, despite of the documentation, passing mpu_port=0 doesn't disable the MIDI port on this board. It implies that the chip is a newer revision and the MPU401 port is integrated and mapped on the PCI register. For this chip model, the driver enables the MPU port unconditionally, so far. Although fixing the unexpected reboot would be the best solution, it's not so trivial to identify the cause. So, as a plan B, this patch extends the existing mpu_port option usage to allow disabling the port by specifying the value 0, just like we applied for fm_port option in commit 2f24d159d5ac ("[ALSA] cmipci - Allow to disable integrated FM port"). As default, the MPU port is still enabled, but user can pass mpu_port=0 to disable it. Reported-and-tested-by: Patrick May <dusthillresident(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa-configuration.rst | 3 ++- sound/pci/cmipci.c | 5 +++-- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 02aacd69ab96..392875a1b94e 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -495,7 +495,8 @@ Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. mpu_port port address of MIDI interface (8338 only): 0x300,0x310,0x320,0x330 = legacy port, - 0 = disable (default) + 1 = integrated PCI port (default on 8738), + 0 = disable fm_port port address of OPL-3 FM synthesizer (8x38 only): 0x388 = legacy port, diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 4bfab21c53f4..266c4cf28b78 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -42,7 +42,7 @@ MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8738}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ -static long mpu_port[SNDRV_CARDS]; +static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; static long fm_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; static bool soft_ac3[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; #ifdef SUPPORT_JOYSTICK @@ -3132,7 +3132,8 @@ static int snd_cmipci_create(struct snd_card *card, struct pci_dev *pci, if (cm->chip_version >= 39) { val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1); if (val != 0x00 && val != 0xff) { - iomidi = cm->iobase + CM_REG_MPU_PCI; + if (mpu_port[dev]) + iomidi = cm->iobase + CM_REG_MPU_PCI; integrated_midi = 1; } } -- 2.16.4

2 1

Re: [alsa-devel] [PATCH v3 08/15] ASoC: tegra: Add audio mclk control through clk_out_1 and extern1
by Dmitry Osipenko 17 Dec '19

17 Dec '19

17.12.2019 19:12, Sowjanya Komatineni пишет: > > On 12/17/19 7:36 AM, Dmitry Osipenko wrote: >> 17.12.2019 04:29, Sowjanya Komatineni пишет: >>> On 12/7/19 11:20 AM, Sowjanya Komatineni wrote: >>>> On 12/7/19 6:58 AM, Dmitry Osipenko wrote: >>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>> Current ASoC driver uses extern1 as cdev1 clock from Tegra30 onwards >>>>>> through device tree. >>>>>> >>>>>> Actual audio mclk is clk_out_1 and to use PLLA for mclk rate control, >>>>>> need to clk_out_1_mux parent to extern1 and extern1 parent to >>>>>> PLLA_OUT0. >>>>>> >>>>>> Currently Tegra clock driver init sets the parents and enables both >>>>>> clk_out_1 and extern1 clocks. But these clocks parent and enables >>>>>> should >>>>>> be controlled by ASoC driver. >>>>>> >>>>>> Clock parents can be specified in device tree using assigned-clocks >>>>>> and assigned-clock-parents. >>>>>> >>>>>> To enable audio mclk, both clk_out_1 and extern1 clocks need to be >>>>>> enabled. >>>>>> >>>>>> This patch configures parents for clk_out_1 and extern1 clocks if >>>>>> device >>>>>> tree does not specify clock parents inorder to support old device >>>>>> tree >>>>>> and controls mclk using both clk_out_1 and extern1 clocks. >>>>>> >>>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>>> --- >>>>>> sound/soc/tegra/tegra_asoc_utils.c | 66 >>>>>> ++++++++++++++++++++++++++++++++++++++ >>>>>> sound/soc/tegra/tegra_asoc_utils.h | 1 + >>>>>> 2 files changed, 67 insertions(+) >>>>>> >>>>>> diff --git a/sound/soc/tegra/tegra_asoc_utils.c >>>>>> b/sound/soc/tegra/tegra_asoc_utils.c >>>>>> index 536a578e9512..8e3a3740df7c 100644 >>>>>> --- a/sound/soc/tegra/tegra_asoc_utils.c >>>>>> +++ b/sound/soc/tegra/tegra_asoc_utils.c >>>>>> @@ -60,6 +60,7 @@ int tegra_asoc_utils_set_rate(struct >>>>>> tegra_asoc_utils_data *data, int srate, >>>>>> data->set_mclk = 0; >>>>>> clk_disable_unprepare(data->clk_cdev1); >>>>>> + clk_disable_unprepare(data->clk_extern1); >>>>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>>>> clk_disable_unprepare(data->clk_pll_a); >>>>>> @@ -89,6 +90,14 @@ int tegra_asoc_utils_set_rate(struct >>>>>> tegra_asoc_utils_data *data, int srate, >>>>>> return err; >>>>>> } >>>>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>>>> + err = clk_prepare_enable(data->clk_extern1); >>>>>> + if (err) { >>>>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>>>> + return err; >>>>>> + } >>>>>> + } >>>>>> + >>>>>> err = clk_prepare_enable(data->clk_cdev1); >>>>>> if (err) { >>>>>> dev_err(data->dev, "Can't enable cdev1: %d\n", err); >>>>>> @@ -109,6 +118,7 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>>>> tegra_asoc_utils_data *data) >>>>>> int err; >>>>>> clk_disable_unprepare(data->clk_cdev1); >>>>>> + clk_disable_unprepare(data->clk_extern1); >>>>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>>>> clk_disable_unprepare(data->clk_pll_a); >>>>>> @@ -142,6 +152,14 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>>>> tegra_asoc_utils_data *data) >>>>>> return err; >>>>>> } >>>>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>>>> + err = clk_prepare_enable(data->clk_extern1); >>>>>> + if (err) { >>>>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>>>> + return err; >>>>>> + } >>>>>> + } >>>>> Why this is needed given that clk_extern1 is either a child of MCLK or >>>>> MCLK itself (on T20)? The child clocks are enabled when the parent is >>>>> enabled. >>>> For T30 and later, clk_extern1 is one of the source for clk_out_1_mux. >>>> clk_extern1 is in CAR and it has its own gate and mux. >>>> >>>> As audio mclk related clocks (clk_out_1, clk_out_1_mux, and extern1) >>>> are moved into ASoC driver from clock driver >>>> >>>> need to enable extern1 gate as well along with clk_out1 for T30 >>>> through T210. >>>> >>>> Just FYI, extern1 enable here happens only when data->clk_extern1 is >>>> available which is for T30 onwards. >>>> >>>>>> err = clk_prepare_enable(data->clk_cdev1); >>>>>> if (err) { >>>>>> dev_err(data->dev, "Can't enable cdev1: %d\n", err); >>>>>> @@ -158,6 +176,7 @@ >>>>>> EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_ac97_rate); >>>>>> int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, >>>>>> struct device *dev) >>>>>> { >>>>>> + struct clk *clk_out_1_mux; >>>>>> int ret; >>>>>> data->dev = dev; >>>>>> @@ -196,6 +215,51 @@ int tegra_asoc_utils_init(struct >>>>>> tegra_asoc_utils_data *data, >>>>>> goto err_put_pll_a_out0; >>>>>> } >>>>> In a previous patch you added fallback to EXTPERIPH when clk_get(MCLK) >>>>> fails. This will work perfectly fine for the older kernels which have >>>>> all clocks in the same single CaR driver, but this may not work that >>>>> great for the newer kernels because PMC driver isn't registered early >>>>> during boot and thus it is possible to get a legit -EPROBE_DEFER which >>>>> shouldn't be ignored. In other words, you need to add into this >>>>> patch a >>>>> check for the error code returned by clk_get(MCLK) and fallback >>>>> only for >>>>> -EINVAL. >>>> yeah right, will add check in next version. >>>>>> + /* >>>>>> + * If clock parents are not set in DT, configure here to use >>>>>> clk_out_1 >>>>>> + * as mclk and extern1 as parent for Tegra30 and higher. >>>>>> + */ >>>>>> + if (!of_find_property(dev->of_node, "assigned-clock-parents", >>>>>> NULL) && >>>>>> + data->soc > TEGRA_ASOC_UTILS_SOC_TEGRA20) { >>>>>> + data->clk_extern1 = clk_get_sys("clk_out_1", "extern1"); >>>>>> + if (IS_ERR(data->clk_extern1)) { >>>>>> + dev_err(data->dev, "Can't retrieve clk extern1\n"); >>>>>> + ret = PTR_ERR(data->clk_extern1); >>>>>> + goto err_put_cdev1; >>>>>> + } >>>>>> + >>>>>> + ret = clk_set_parent(data->clk_extern1, >>>>>> data->clk_pll_a_out0); >>>>>> + if (ret < 0) { >>>>>> + dev_err(data->dev, >>>>>> + "Set parent failed for clk extern1: %d\n", >>>>>> + ret); >>>>>> + goto err_put_cdev1; >>>>>> + } >>>>>> + >>>>>> + clk_out_1_mux = clk_get_sys(NULL, "clk_out_1_mux"); >>>>> Note1: clk_get(dev, "clk_out_1_mux") should work here by letting clk >>>>> core to fall back to the clk_get_sys() by itself. Either way should >>>>> be good. >>> clk_get uses device rather and dev_id will be name of this device and >>> when clk_get fall back to __clk_get_sys() it still will use dev id of >>> this device rather than actual dev_id that pmc clocks are added to the >>> lookup. So clk_get_sys() seems to be correct to use as we can specify >>> exact dev_id and con_id. >> It should be better to use something "resource managed", thus >> devm_clk_get() should be a better choice. >> >>> Also, clk_find retrieves clock from lookup only when it finds matching >>> clock with both dev_id and con_id as pmc clocks are registered with both >>> dev_id and con_id. >>> >>> I see existing clock driver adds both extern and pmc clocks (clk_out) to >>> lookup with same dev_id of clk_out_1/2/3 and con_id of extern1/2/3 and >>> with this always extern clock will be retrieved and this is probably >>> because old DT and audio driver always uses extern1 rather than actual >>> clk_out_1 >>> >>> But this need to be fixed now as we changed to use clk_out directly >>> rather than extern (even for other pmc clocks) to match actual hw >>> design. >>> >>> Will fix this as well to register pmc clocks using con_id as >>> clk_out_1/2/3 in pmc driver and extern clocks using con_id of >>> extern1/2/3 with dev_id being NULL so we can retrieve these clocks by >>> just using con_id only using clk_get_sys as we switched to use clk_out_1 >>> directly as pmc clock rather than extern from DT and no longer need to >>> pair pmc clocks to extern clocks. >> I'm not sure it's worth the effort to care about con_ids if implicit >> fallback to clk_get_sys(NULL, "...") does the right thing for the audio >> driver. >> >> IIRC, CCF uses variant of matching clocks by names, although I'm not >> sure whether that applies to older stable kernels. >> >> [snip] > > Current clock driver adds EXTERN clock to lookup with dev_id as > clk_out_1/2/3 and con_id as extern_1/2/3 > > With this we can retrieve clock from lookup only with clk_get_sys where > we can pass dev_id as clk_out_1/2/3 and con_id as extern_1/2/3. > > We cant use devm_clk_get() or clk_get() for retrieving clocks from > lookup by passing device object from sound driver as dev_id will be diff > and clk_find will return NULL. Have you actually tried to test that it fails? I think it's a false assumption. > But with the fix of having dev_id as NULL and use only con_id to add to > lookup, we can use resource managed APIs devm_clk_get. > > So was saying will fix this in clock driver as part of removing > clk_out_1/2/3 ids and pmc init from clock driver so we can use > devm_clk_get API in audio driver. > >>

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Re: [alsa-devel] [PATCH v3 08/15] ASoC: tegra: Add audio mclk control through clk_out_1 and extern1
by Dmitry Osipenko 17 Dec '19

17 Dec '19

17.12.2019 04:29, Sowjanya Komatineni пишет: > > On 12/7/19 11:20 AM, Sowjanya Komatineni wrote: >> >> On 12/7/19 6:58 AM, Dmitry Osipenko wrote: >>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>> Current ASoC driver uses extern1 as cdev1 clock from Tegra30 onwards >>>> through device tree. >>>> >>>> Actual audio mclk is clk_out_1 and to use PLLA for mclk rate control, >>>> need to clk_out_1_mux parent to extern1 and extern1 parent to >>>> PLLA_OUT0. >>>> >>>> Currently Tegra clock driver init sets the parents and enables both >>>> clk_out_1 and extern1 clocks. But these clocks parent and enables >>>> should >>>> be controlled by ASoC driver. >>>> >>>> Clock parents can be specified in device tree using assigned-clocks >>>> and assigned-clock-parents. >>>> >>>> To enable audio mclk, both clk_out_1 and extern1 clocks need to be >>>> enabled. >>>> >>>> This patch configures parents for clk_out_1 and extern1 clocks if >>>> device >>>> tree does not specify clock parents inorder to support old device tree >>>> and controls mclk using both clk_out_1 and extern1 clocks. >>>> >>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>> --- >>>> sound/soc/tegra/tegra_asoc_utils.c | 66 >>>> ++++++++++++++++++++++++++++++++++++++ >>>> sound/soc/tegra/tegra_asoc_utils.h | 1 + >>>> 2 files changed, 67 insertions(+) >>>> >>>> diff --git a/sound/soc/tegra/tegra_asoc_utils.c >>>> b/sound/soc/tegra/tegra_asoc_utils.c >>>> index 536a578e9512..8e3a3740df7c 100644 >>>> --- a/sound/soc/tegra/tegra_asoc_utils.c >>>> +++ b/sound/soc/tegra/tegra_asoc_utils.c >>>> @@ -60,6 +60,7 @@ int tegra_asoc_utils_set_rate(struct >>>> tegra_asoc_utils_data *data, int srate, >>>> data->set_mclk = 0; >>>> clk_disable_unprepare(data->clk_cdev1); >>>> + clk_disable_unprepare(data->clk_extern1); >>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>> clk_disable_unprepare(data->clk_pll_a); >>>> @@ -89,6 +90,14 @@ int tegra_asoc_utils_set_rate(struct >>>> tegra_asoc_utils_data *data, int srate, >>>> return err; >>>> } >>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>> + err = clk_prepare_enable(data->clk_extern1); >>>> + if (err) { >>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>> + return err; >>>> + } >>>> + } >>>> + >>>> err = clk_prepare_enable(data->clk_cdev1); >>>> if (err) { >>>> dev_err(data->dev, "Can't enable cdev1: %d\n", err); >>>> @@ -109,6 +118,7 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>> tegra_asoc_utils_data *data) >>>> int err; >>>> clk_disable_unprepare(data->clk_cdev1); >>>> + clk_disable_unprepare(data->clk_extern1); >>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>> clk_disable_unprepare(data->clk_pll_a); >>>> @@ -142,6 +152,14 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>> tegra_asoc_utils_data *data) >>>> return err; >>>> } >>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>> + err = clk_prepare_enable(data->clk_extern1); >>>> + if (err) { >>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>> + return err; >>>> + } >>>> + } >>> Why this is needed given that clk_extern1 is either a child of MCLK or >>> MCLK itself (on T20)? The child clocks are enabled when the parent is >>> enabled. >> >> For T30 and later, clk_extern1 is one of the source for clk_out_1_mux. >> clk_extern1 is in CAR and it has its own gate and mux. >> >> As audio mclk related clocks (clk_out_1, clk_out_1_mux, and extern1) >> are moved into ASoC driver from clock driver >> >> need to enable extern1 gate as well along with clk_out1 for T30 >> through T210. >> >> Just FYI, extern1 enable here happens only when data->clk_extern1 is >> available which is for T30 onwards. >> >>>> err = clk_prepare_enable(data->clk_cdev1); >>>> if (err) { >>>> dev_err(data->dev, "Can't enable cdev1: %d\n", err); >>>> @@ -158,6 +176,7 @@ EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_ac97_rate); >>>> int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, >>>> struct device *dev) >>>> { >>>> + struct clk *clk_out_1_mux; >>>> int ret; >>>> data->dev = dev; >>>> @@ -196,6 +215,51 @@ int tegra_asoc_utils_init(struct >>>> tegra_asoc_utils_data *data, >>>> goto err_put_pll_a_out0; >>>> } >>> In a previous patch you added fallback to EXTPERIPH when clk_get(MCLK) >>> fails. This will work perfectly fine for the older kernels which have >>> all clocks in the same single CaR driver, but this may not work that >>> great for the newer kernels because PMC driver isn't registered early >>> during boot and thus it is possible to get a legit -EPROBE_DEFER which >>> shouldn't be ignored. In other words, you need to add into this patch a >>> check for the error code returned by clk_get(MCLK) and fallback only for >>> -EINVAL. >> yeah right, will add check in next version. >>>> + /* >>>> + * If clock parents are not set in DT, configure here to use >>>> clk_out_1 >>>> + * as mclk and extern1 as parent for Tegra30 and higher. >>>> + */ >>>> + if (!of_find_property(dev->of_node, "assigned-clock-parents", >>>> NULL) && >>>> + data->soc > TEGRA_ASOC_UTILS_SOC_TEGRA20) { >>>> + data->clk_extern1 = clk_get_sys("clk_out_1", "extern1"); >>>> + if (IS_ERR(data->clk_extern1)) { >>>> + dev_err(data->dev, "Can't retrieve clk extern1\n"); >>>> + ret = PTR_ERR(data->clk_extern1); >>>> + goto err_put_cdev1; >>>> + } >>>> + >>>> + ret = clk_set_parent(data->clk_extern1, data->clk_pll_a_out0); >>>> + if (ret < 0) { >>>> + dev_err(data->dev, >>>> + "Set parent failed for clk extern1: %d\n", >>>> + ret); >>>> + goto err_put_cdev1; >>>> + } >>>> + >>>> + clk_out_1_mux = clk_get_sys(NULL, "clk_out_1_mux"); >>> Note1: clk_get(dev, "clk_out_1_mux") should work here by letting clk >>> core to fall back to the clk_get_sys() by itself. Either way should >>> be good. > > clk_get uses device rather and dev_id will be name of this device and > when clk_get fall back to __clk_get_sys() it still will use dev id of > this device rather than actual dev_id that pmc clocks are added to the > lookup. So clk_get_sys() seems to be correct to use as we can specify > exact dev_id and con_id. It should be better to use something "resource managed", thus devm_clk_get() should be a better choice. > Also, clk_find retrieves clock from lookup only when it finds matching > clock with both dev_id and con_id as pmc clocks are registered with both > dev_id and con_id. > > I see existing clock driver adds both extern and pmc clocks (clk_out) to > lookup with same dev_id of clk_out_1/2/3 and con_id of extern1/2/3 and > with this always extern clock will be retrieved and this is probably > because old DT and audio driver always uses extern1 rather than actual > clk_out_1 > > But this need to be fixed now as we changed to use clk_out directly > rather than extern (even for other pmc clocks) to match actual hw design. > > Will fix this as well to register pmc clocks using con_id as > clk_out_1/2/3 in pmc driver and extern clocks using con_id of > extern1/2/3 with dev_id being NULL so we can retrieve these clocks by > just using con_id only using clk_get_sys as we switched to use clk_out_1 > directly as pmc clock rather than extern from DT and no longer need to > pair pmc clocks to extern clocks. I'm not sure it's worth the effort to care about con_ids if implicit fallback to clk_get_sys(NULL, "...") does the right thing for the audio driver. IIRC, CCF uses variant of matching clocks by names, although I'm not sure whether that applies to older stable kernels. [snip]

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[alsa-devel] [PATCH] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 17 Dec '19

17 Dec '19

The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 1063 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 95 ++++ 4 files changed, 1174 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..806cd03 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,19 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + select CRC32 + select CRYPTO_SHA256 + select CRYTO_RSA + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..5ba8ec5 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,1063 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +#if GENERIC_DEBUGFS +static int mt6660_dbg_io_read(void *drvdata, u16 reg, void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int ret = 0, i = 0; + unsigned int data; + u8 *_val = val; + + if (size != 1 && size != 2 && size != 4) { + dev_err(chip->dev, "%s size not match\n", __func__); + return -EINVAL; + } + ret = regmap_read(chip->regmap, reg, &data); + if (reg < 0) + return ret; + + for (i = 0; i < size; i++) + _val[size-i-1] = (data >> (8*i))&0xff; + + return 0; +} + +static int mt6660_dbg_io_write(void *drvdata, u16 reg, + const void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int reg_size = mt6660_get_reg_size(reg); + int i = 0; + unsigned int regval = 0; + u8 *_val = (u8 *)val; + + if (size != reg_size) { + dev_err(chip->dev, + "%s size not match reg_size(%d), size(%d)\n", + __func__, reg_size, size); + return -EINVAL; + } + + for (i = 0; i < size; i++) { + regval <<= 8; + regval |= _val[i]; + } + + return regmap_write(chip->regmap, reg, regval); +} +#endif /* GENERIC_DEBUGFS */ + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + + return ret; +} + +static bool mt6660_volatile_reg(struct device *dev, unsigned int reg) +{ + return true; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, + .volatile_reg = mt6660_volatile_reg, +}; + +static unsigned int mt6660_component_io_read( + struct snd_soc_component *component, unsigned int reg) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret; + + ret = regmap_read(chip->regmap, reg, &val); + if (ret < 0) /* ret success -> >= 0, fail -> < - */ + return ret; + pr_err("%s val = 0x%x\n", __func__, val); + return val; +} + +static int mt6660_component_io_write(struct snd_soc_component *component, + unsigned int reg, unsigned int data) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + pr_err("%s data = 0x%x\n", __func__, data); + return regmap_write(chip->regmap, reg, data); +} + +static const int mt6660_dump_table[] = { + MT6660_REG_DEVID, + MT6660_REG_SYSTEM_CTRL, + MT6660_REG_IRQ_STATUS1, + MT6660_REG_ADDA_CLOCK, + MT6660_REG_SERIAL_CFG1, + MT6660_REG_DATAO_SEL, + MT6660_REG_TDM_CFG3, + MT6660_REG_HPF_CTRL, + MT6660_REG_HPF1_COEF, + MT6660_REG_HPF2_COEF, + MT6660_REG_PATH_BYPASS, + MT6660_REG_WDT_CTRL, + MT6660_REG_HCLIP_CTRL, + MT6660_REG_VOL_CTRL, + MT6660_REG_SPS_CTRL, + MT6660_REG_SIGMAX, + MT6660_REG_CALI_T0, + MT6660_REG_BST_CTRL, + MT6660_REG_PROTECTION_CFG, + MT6660_REG_DA_GAIN, + MT6660_REG_AUDIO_IN2_SEL, + MT6660_REG_SIG_GAIN, + MT6660_REG_PLL_CFG1, + MT6660_REG_DRE_CTRL, + MT6660_REG_DRE_THDMODE, + MT6660_REG_DRE_CORASE, + MT6660_REG_PWM_CTRL, + MT6660_REG_DC_PROTECT_CTRL, + MT6660_REG_ADC_USB_MODE, + MT6660_REG_INTERNAL_CFG, + MT6660_REG_RESV0, + MT6660_REG_RESV1, + MT6660_REG_RESV2, + MT6660_REG_RESV3, + MT6660_REG_RESV7, + MT6660_REG_RESV10, + MT6660_REG_RESV11, + MT6660_REG_RESV16, + MT6660_REG_RESV17, + MT6660_REG_RESV19, + MT6660_REG_RESV21, + MT6660_REG_RESV23, + MT6660_REG_RESV31, + MT6660_REG_RESV40, +}; + +#if GENERIC_DEBUGFS +#ifdef CONFIG_DEBUG_FS +/* reg/size/data/bustype */ +#define PREALLOC_RBUFFER_SIZE (32) +#define PREALLOC_WBUFFER_SIZE (1000) + +static int data_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + int i, ret; + + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* read transfer */ + if (!di->io_read) + return -EPERM; + ret = di->io_read(di->io_drvdata, d->reg, d->data_buffer, d->size); + if (ret < 0) + return ret; + pdata = d->data_buffer; + seq_printf(s, "0x"); + for (i = 0; i < d->size; i++) + seq_printf(s, "%02x,", *(pdata + i)); + seq_printf(s, "\n"); + return 0; +} + +static int data_debug_open(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_open(file, data_debug_show, inode->i_private); + return simple_open(inode, file); +} + +static ssize_t data_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + char buf[PREALLOC_WBUFFER_SIZE + 1], *token, *cur; + int val_cnt = 0, ret; + + if (cnt > PREALLOC_WBUFFER_SIZE) + return -ENOMEM; + if (copy_from_user(buf, user_buf, cnt)) + return -EFAULT; + buf[cnt] = 0; + /* buffer size check */ + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* data parsing */ + cur = buf; + pdata = d->data_buffer; + while ((token = strsep(&cur, ",\n")) != NULL) { + if (!*token) + break; + if (val_cnt++ >= d->size) + break; + if (kstrtou8(token, 16, pdata++)) + return -EINVAL; + } + if (val_cnt != d->size) + return -EINVAL; + /* write transfer */ + if (!di->io_write) + return -EPERM; + ret = di->io_write(di->io_drvdata, d->reg, d->data_buffer, d->size); + return (ret < 0) ? ret : cnt; +} + +static int data_debug_release(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_release(inode, file); + return 0; +} + +static const struct file_operations data_debug_fops = { + .open = data_debug_open, + .read = seq_read, + .write = data_debug_write, + .llseek = seq_lseek, + .release = data_debug_release, +}; + +static int type_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + + seq_printf(s, "%s,%s\n", di->typestr, di->devname); + return 0; +} + +static int type_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, type_debug_show, inode->i_private); +} + +static const struct file_operations type_debug_fops = { + .open = type_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static int dump_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct mt6660_chip *chip = + container_of(di, struct mt6660_chip, dbg_info); + int i = 0, j = 0, ret = 0; + unsigned int val; + int size = 0; + + if (!chip) { + pr_err("%s chip is null\n", __func__); + return -ENODEV; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_dump_table); i++) { + ret = regmap_read(chip->regmap, mt6660_dump_table[i], &val); + size = mt6660_get_reg_size(mt6660_dump_table[i]); + seq_printf(s, "reg0x%02x:0x", mt6660_dump_table[i]); + for (j = size - 1; j >= 0; j--) + seq_printf(s, "%x,", (val >> 8*j)&0xff); + seq_printf(s, "\n"); + } + return 0; +} + +static int dump_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, dump_debug_show, inode->i_private); +} + +static const struct file_operations dump_debug_fops = { + .open = dump_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static ssize_t lock_debug_read(struct file *file, + char __user *user_buf, size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + char buf[10]; + + snprintf(buf, sizeof(buf), "%d\n", mutex_is_locked(&d->io_lock)); + return simple_read_from_buffer(user_buf, cnt, loff, buf, strlen(buf)); +} + +static ssize_t lock_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + u32 lock; + int ret; + + ret = kstrtou32_from_user(user_buf, cnt, 0, &lock); + if (ret < 0) + return ret; + lock ? mutex_lock(&d->io_lock) : mutex_unlock(&d->io_lock); + return cnt; +} + +static const struct file_operations lock_debug_fops = { + .open = simple_open, + .read = lock_debug_read, + .write = lock_debug_write, +}; + +static int generic_debugfs_init(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + /* valid check */ + if (!di->dirname || !di->devname || !di->typestr) + return -EINVAL; + d->data_buffer_size = PREALLOC_RBUFFER_SIZE; + d->data_buffer = kzalloc(PREALLOC_RBUFFER_SIZE, GFP_KERNEL); + if (!d->data_buffer) + return -ENOMEM; + /* create debugfs */ + d->rt_root = debugfs_lookup("ext_dev_io", NULL); + if (!d->rt_root) { + d->rt_root = debugfs_create_dir("ext_dev_io", NULL); + if (!d->rt_root) + return -ENODEV; + d->rt_dir_create = true; + } + d->ic_root = debugfs_create_dir(di->dirname, d->rt_root); + if (!d->ic_root) + goto err_cleanup_rt; + if (!debugfs_create_u16("reg", 0644, d->ic_root, &d->reg)) + goto err_cleanup_ic; + if (!debugfs_create_u16("size", 0644, d->ic_root, &d->size)) + goto err_cleanup_ic; + if (!debugfs_create_file("data", 0644, + d->ic_root, di, &data_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("type", 0444, + d->ic_root, di, &type_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("lock", 0644, + d->ic_root, di, &lock_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("dumps", 0444, + d->ic_root, di, &dump_debug_fops)) + goto err_cleanup_ic; + mutex_init(&d->io_lock); + return 0; +err_cleanup_ic: + debugfs_remove_recursive(d->ic_root); +err_cleanup_rt: + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); + return -ENODEV; +} + +static void generic_debugfs_exit(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + mutex_destroy(&d->io_lock); + debugfs_remove_recursive(d->ic_root); + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); +} +#else +static inline int generic_debugfs_init(struct dbg_info *di) +{ + return 0; +} + +static inline void generic_debugfs_exit(struct dbg_info *di) {} +#endif /* CONFIG_DEBUG_FS */ +#endif /* GENERIC_DEBUGFS */ + +/* + * MT6660 Generic Setting make this chip work normally. + * it is tuned by Richtek RDs. + */ +static const struct codec_reg_val generic_reg_inits[] = { + { MT6660_REG_WDT_CTRL, 0x80, 0x00 }, + { MT6660_REG_SPS_CTRL, 0x01, 0x00 }, + { MT6660_REG_AUDIO_IN2_SEL, 0x1c, 0x04 }, + { MT6660_REG_RESV11, 0x0c, 0x00 }, + { MT6660_REG_RESV31, 0x03, 0x03 }, + { MT6660_REG_RESV40, 0x01, 0x00 }, + { MT6660_REG_RESV0, 0x44, 0x04 }, + { MT6660_REG_RESV19, 0xff, 0x82 }, + { MT6660_REG_RESV17, 0x7777, 0x7273 }, + { MT6660_REG_RESV16, 0x07, 0x03 }, + { MT6660_REG_DRE_CORASE, 0xe0, 0x20 }, + { MT6660_REG_ADDA_CLOCK, 0xff, 0x70 }, + { MT6660_REG_RESV21, 0xff, 0x20 }, + { MT6660_REG_DRE_THDMODE, 0xff, 0x40 }, + { MT6660_REG_RESV23, 0xffff, 0x17f8 }, + { MT6660_REG_PWM_CTRL, 0xff, 0x15 }, + { MT6660_REG_ADC_USB_MODE, 0xff, 0x00 }, + { MT6660_REG_PROTECTION_CFG, 0xff, 0x1d }, + { MT6660_REG_HPF1_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_HPF2_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_SIG_GAIN, 0xff, 0x58 }, + { MT6660_REG_RESV6, 0xff, 0xce }, + { MT6660_REG_SIGMAX, 0xffff, 0x7fff }, + { MT6660_REG_DA_GAIN, 0xffff, 0x0116 }, + { MT6660_REG_TDM_CFG3, 0x1800, 0x0800 }, + { MT6660_REG_DRE_CTRL, 0x1f, 0x07 }, +}; + +static int mt6660_component_init_setting(struct snd_soc_component *component) +{ + int i, len, ret; + const struct codec_reg_val *init_table; + + pr_info("%s start\n", __func__); + init_table = generic_reg_inits; + len = ARRAY_SIZE(generic_reg_inits); + + for (i = 0; i < len; i++) { + ret = snd_soc_component_update_bits(component, + init_table[i].addr, + init_table[i].mask, init_table[i].data); + if (ret < 0) + return ret; + } + pr_info("%s end\n", __func__); + return 0; +} + +static int mt6660_component_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + ret = regmap_read(chip->regmap, MT6660_REG_IRQ_STATUS1, &val); + dev_info(component->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + break; + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + /* Fall Through */ + case SND_SOC_BIAS_STANDBY: + default: + break; + } + dapm->bias_level = level; + dev_dbg(component->dev, "c bias_level = %d\n", level); + return 0; +} + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_PRE_PMD: + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "OUTP", NULL, "ClassD"}, + { "OUTN", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + int put_ret = 0; + + pm_runtime_get_sync(component->dev); + put_ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (put_ret < 0) + return put_ret; + pm_runtime_put(component->dev); + return put_ret; +} + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, mt6660_component_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT Switch", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Hard Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DC_Protect Switch", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("audio input selection", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF AUD IN Switch", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("AUD LOOP BACK Switch", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Mute Switch", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Bypass CS Comp Switch", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("T0 SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + pr_info("%s\n", __func__); + //snd_soc_component_init_regmap(component, chip->regmap); + + /* init chip setting */ + ret = mt6660_component_init_setting(component); + if (ret < 0) + dev_err(chip->dev, "chip i2c init setting fail\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + pr_info("%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + .read = mt6660_component_io_read, + .write = mt6660_component_io_write, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .set_bias_level = mt6660_component_set_bias_level, + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + pr_info("%s start\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + +#if GENERIC_DEBUGFS + /* debugfs interface */ + chip->dbg_info.dirname = devm_kasprintf(&client->dev, + GFP_KERNEL, "MT6660.%s", + dev_name(&client->dev)); + chip->dbg_info.devname = dev_name(&client->dev); + chip->dbg_info.typestr = devm_kasprintf(&client->dev, + GFP_KERNEL, "I2C,MT6660"); + chip->dbg_info.io_drvdata = chip; + chip->dbg_info.io_read = mt6660_dbg_io_read; + chip->dbg_info.io_write = mt6660_dbg_io_write; + + ret = generic_debugfs_init(&chip->dbg_info); + if (ret < 0) { + dev_err(&client->dev, "generic dbg init fail\n"); + return -EINVAL; + } +#endif /* GENERIC_DEBUGFS */ + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + pr_info("%s end, ret = %d\n", __func__, ret); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); +#if GENERIC_DEBUGFS + generic_debugfs_exit(&chip->dbg_info); +#endif /* GENERIC_DEBUGFS */ + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..9e62dcc --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,95 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +#define GENERIC_DEBUGFS 0 + +#if GENERIC_DEBUGFS +struct dbg_internal { + struct dentry *rt_root; + struct dentry *ic_root; + bool rt_dir_create; + struct mutex io_lock; + u16 reg; + u16 size; + u16 data_buffer_size; + void *data_buffer; +}; + +struct dbg_info { + const char *dirname; + const char *devname; + const char *typestr; + void *io_drvdata; + int (*io_read)(void *drvdata, u16 reg, void *val, u16 size); + int (*io_write)(void *drvdata, u16 reg, const void *val, u16 size); + struct dbg_internal internal; +}; +#endif /* GENERIC_DEBUGFS */ + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mutex io_lock; +#if GENERIC_DEBUGFS + struct dbg_info dbg_info; +#endif /* GENERIC_DEBUGFS */ + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

4 3

[alsa-devel] [BUG] ALSA: seq: a possible sleep-in-atomic-context bug in snd_virmidi_dev_receive_event()
by Jia-Ju Bai 17 Dec '19

17 Dec '19

The driver may sleep while holding a read lock. The function call path (from bottom to top) in Linux 4.19 is: sound/core/seq/seq_memory.c, 96: copy_from_user in snd_seq_dump_var_event sound/core/seq/seq_virmidi.c, 97: snd_seq_dump_var_event in snd_virmidi_dev_receive_event sound/core/seq/seq_virmidi.c, 88: _raw_read_lock in snd_virmidi_dev_receive_event copy_from_user() can sleep at runtime. I am not sure how to properly fix this possible bug, so I only report it. This bug is found by a static analysis tool STCheck written by myself. Best wishes, Jia-Ju Bai

3 3

[alsa-devel] [GIT PULL] ASoC fixes for v5.5
by Mark Brown 17 Dec '19

17 Dec '19

The following changes since commit 39870b0dec68ed7dd814beb697e541670975c7d8: ASoC: DMI long name - avoid to add board name if matches with product name (2019-11-22 19:54:50 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.5-rc2 for you to fetch changes up to 556672d75ff486e0b6786056da624131679e0576: ASoC: wm8962: fix lambda value (2019-12-16 11:47:28 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.5 A collection of fixes since the merge window, mostly driver specific but there's a few in the core that clean up fallout from the refactorings done in the last cycle. ---------------------------------------------------------------- Amery Song (1): ASoC: Intel: common: work-around incorrect ACPI HID for CML boards Curtis Malainey (3): ASoC: core: Init pcm runtime work early to avoid warnings ASoC: core: only flush inited work during free ASoC: SOF: Intel: split cht and byt debug window sizes Daniel Baluta (2): ASoC: SOF: topology: Fix unload for SAI/ESAI ASoC: simple-card: Don't create separate link when platform is present Dragos Tarcatu (2): ASoC: topology: Check return value for snd_soc_add_dai_link() ASoC: topology: Check return value for soc_tplg_pcm_create() Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Update quirk for Teclast X89 Karol Trzcinski (2): ASoC: SOF: loader: snd_sof_fw_parse_ext_data log warning on unknown header ASoC: SOF: loader: fix snd_sof_fw_parse_ext_data Keyon Jie (1): ASoC: SOF: Intel: BYT: fix a copy/paste mistake in byt_dump() Michael Walle (1): ASoC: wm8904: fix automatic sysclk configuration Shengjiu Wang (1): ASoC: wm8962: fix lambda value Shuming Fan (1): ASoC: rt5682: fix i2c arbitration lost issue Tzung-Bi Shih (3): ASoC: max98090: remove msleep in PLL unlocked workaround ASoC: max98090: exit workaround earlier if PLL is locked ASoC: max98090: fix possible race conditions Yu-Hsuan Hsu (1): ASoC: AMD: Enable clk in startup intead of hw_params YueHaibing (2): ASoC: rt5677: Fix build error without CONFIG_SPI ASoC: Intel: sst: Add missing include <linux/io.h> include/sound/soc.h | 1 + sound/soc/amd/acp-da7219-max98357a.c | 46 +++++++---------------- sound/soc/codecs/max98090.c | 30 ++++++++++----- sound/soc/codecs/max98090.h | 1 - sound/soc/codecs/rt5677-spi.h | 16 ++++++++ sound/soc/codecs/rt5682.c | 2 + sound/soc/codecs/wm8904.c | 6 +++ sound/soc/codecs/wm8962.c | 4 +- sound/soc/generic/simple-card.c | 6 +++ sound/soc/intel/atom/sst/sst.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 8 +++- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 41 +++++++++++--------- sound/soc/soc-compress.c | 6 +-- sound/soc/soc-core.c | 13 ++++++- sound/soc/soc-pcm.c | 11 ++---- sound/soc/soc-topology.c | 27 +++++++++---- sound/soc/sof/intel/byt.c | 25 ++++++------ sound/soc/sof/loader.c | 7 +++- sound/soc/sof/topology.c | 4 +- 19 files changed, 158 insertions(+), 97 deletions(-)

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[alsa-devel] [PATCH] ASoC: intel: skylake: Remove superfluous bus ops
by Takashi Iwai 17 Dec '19

17 Dec '19

The bus_core_ops values set in skl driver are same as the default values, so we can drop it and pass NULL to snd_hdac_ext_bus_init() instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/skylake/skl.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 58ba3e9469ba..f755ca2484cf 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -773,11 +773,6 @@ static void skl_codec_create(struct hdac_bus *bus) } } -static const struct hdac_bus_ops bus_core_ops = { - .command = snd_hdac_bus_send_cmd, - .get_response = snd_hdac_bus_get_response, -}; - static int skl_i915_init(struct hdac_bus *bus) { int err; @@ -888,7 +883,7 @@ static int skl_create(struct pci_dev *pci, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif - snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, ext_ops); + snd_hdac_ext_bus_init(bus, &pci->dev, NULL, ext_ops); bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); -- 2.16.4

2 1

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 17 Dec '19

17 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. I recommend using 32bit signed integers for sample rate conversion. Currently, once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 17 Dec '19

17 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Maybe, double, unsigned int, or unsigned int64? Once sample rate conversion is done, using S32_LE is meaningless. Does dmix also use 16bits internally? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] [PATCH for-5.6 00/14] media: ALSA PCM API updates
by Takashi Iwai 17 Dec '19

17 Dec '19

Hi, this is a patch set to adapt the latest ALSA PCM API to media drivers. Basically these are merely cleanups, as shown in diffstat, and there should be no functional changes. As the corresponding ALSA PCM API change is found in 5.5-rc1, please apply these on 5.5-rc1 or later. Or if you prefer, I can merge them through sound tree, too. Let me know. thanks, Takashi === Takashi Iwai (14): media: solo6x10: Use managed buffer allocation media: tw686x: Use managed buffer allocation media: usbtv: Use managed buffer allocation media: cobalt: Clean up ALSA PCM API usages media: cx18: Clean up ALSA PCM API usages media: ivtv: Clean up ALSA PCM API usages media: cs231xx: Clean up ALSA PCM API usages media: em28xx: Clean up ALSA PCM API usages media: go7007: Clean up ALSA PCM API usages media: tm6000: Clean up ALSA PCM API usages media: cobalt: Drop superfluous ioctl PCM ops media: cx18: Drop superfluous ioctl PCM ops media: ivtv: Drop superfluous ioctl PCM ops media: Drop superfluous ioctl PCM ops drivers/media/pci/cobalt/cobalt-alsa-pcm.c | 69 ++---------------------- drivers/media/pci/cx18/cx18-alsa-pcm.c | 75 +------------------------- drivers/media/pci/cx23885/cx23885-alsa.c | 1 - drivers/media/pci/cx25821/cx25821-alsa.c | 1 - drivers/media/pci/cx88/cx88-alsa.c | 1 - drivers/media/pci/ivtv/ivtv-alsa-pcm.c | 76 +------------------------- drivers/media/pci/saa7134/saa7134-alsa.c | 1 - drivers/media/pci/solo6x10/solo6x10-g723.c | 24 ++------- drivers/media/pci/tw686x/tw686x-audio.c | 16 +----- drivers/media/usb/cx231xx/cx231xx-audio.c | 79 +-------------------------- drivers/media/usb/em28xx/em28xx-audio.c | 87 +----------------------------- drivers/media/usb/go7007/snd-go7007.c | 25 ++------- drivers/media/usb/tm6000/tm6000-alsa.c | 82 +--------------------------- drivers/media/usb/usbtv/usbtv-audio.c | 29 +--------- 14 files changed, 19 insertions(+), 547 deletions(-) -- 2.16.4

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[alsa-devel] [PATCH v4 00/15] soundwire: intel: implement new ASoC interfaces
by Pierre-Louis Bossart 17 Dec '19

17 Dec '19

This patchset applies on top of soundwire/next, now that the interface definitions are merged. The changes are essentially a removal of the platform devices and the implementation of the new interfaces required to scan the ACPI tables, probe the links and start them. The missing prepare, trigger and setup ASoC callbacks are also implemented. The hw_params and free callbacks use the new interfaces as well. While there are quite a few lines of code changed, this is mostly about interface changes. The next series will contain more functional changes and deal with race conditions on probe, enumeration and suspend/resume issues. Changes since v3: One line change to re-add EXPORT_SYMBOL Add missing driver_registration Changes since v2: moved uevent handling to slave_type (Vinod) Changes since v1: fix typo (Vinod) removed uevent open for Master (Vinod) clarified commit messages (Cezary) no functionality change Bard Liao (1): soundwire: register master device driver Pierre-Louis Bossart (11): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: slave: move uevent handling to slave soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and provide new interface soundwire: intel: free all resources on hw_free() soundwire: intel_init: add implementation of sdw_intel_enable_irq() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus.h | 2 + drivers/soundwire/bus_type.c | 63 +++--- drivers/soundwire/intel.c | 282 +++++++++++++++++++++----- drivers/soundwire/intel.h | 8 +- drivers/soundwire/intel_init.c | 309 ++++++++++++++++++++++------- drivers/soundwire/master.c | 63 ++++++ drivers/soundwire/slave.c | 10 +- include/linux/soundwire/sdw.h | 39 +++- include/linux/soundwire/sdw_type.h | 34 +++- 12 files changed, 657 insertions(+), 161 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

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[alsa-devel] alsa-info: lots of mostly syntactic changes
by GitHub pull_request - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-utils pull request #29 was edited from E5ten: - remove all bashisms and change shebang to #!/bin/sh - replace use of which with command -v for portability - remove awk dependency - general cosmetic consistency: - indentation - always put then or do on the same line as if, for, or while - quote much more, especially when the variable refers to a file - replace many uses of >> $FILE with command grouping and single redirects of the command groups - simplify text processing pipelines, mostly by turning them into a single sed command Request URL : https://github.com/alsa-project/alsa-utils/pull/29 Patch URL : https://github.com/alsa-project/alsa-utils/pull/29.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] Free-running MMAP Mode
by Fletcher Woodruff 16 Dec '19

16 Dec '19

Hi alsa-devel, I'm working on implementing memory-mapped audio for a project that uses the alsa user-space API. My understanding is that typically, mmap mode requires explicit calls to mmap_begin/mmap_commit before and after every read/write in order to notify the driver of newly read/written frames. Is there any way for a user to request a sort of "free-running" MMAP, where alsa just assumes that the data it needs has been written to the circular buffer and doesn't expect those begin/commit calls? Thanks, Fletcher

1 0

[alsa-devel] [PATCH v2 0/2] Add the broadmobi BM818
by Angus Ainslie (Purism) 16 Dec '19

16 Dec '19

he broadmobi uses slightly different parameters from the option modems so add the paramters and document them. Changes since v1: Cut back the CC list. Use data from compatible match instead of explicit compatible match. Angus Ainslie (Purism) (2): sound: codecs: gtm601: add Broadmobi bm818 sound profile dt-bindings: sound: gtm601: add the broadmobi interface .../devicetree/bindings/sound/gtm601.txt | 10 +++- sound/soc/codecs/gtm601.c | 46 +++++++++++++++---- 2 files changed, 46 insertions(+), 10 deletions(-) -- 2.17.1

3 4

Re: [alsa-devel] [PATCH v3 03/15] soc: tegra: Add Tegra PMC clock registrations into PMC driver
by Dmitry Osipenko 16 Dec '19

16 Dec '19

10.12.2019 19:53, Sowjanya Komatineni пишет: > > On 12/9/19 3:03 PM, Sowjanya Komatineni wrote: >> >> On 12/9/19 12:46 PM, Sowjanya Komatineni wrote: >>> >>> On 12/9/19 12:12 PM, Dmitry Osipenko wrote: >>>> 08.12.2019 00:36, Sowjanya Komatineni пишет: >>>>> On 12/7/19 11:59 AM, Sowjanya Komatineni wrote: >>>>>> On 12/7/19 8:00 AM, Dmitry Osipenko wrote: >>>>>>> 07.12.2019 18:53, Dmitry Osipenko пишет: >>>>>>>> 07.12.2019 18:47, Dmitry Osipenko пишет: >>>>>>>>> 07.12.2019 17:28, Dmitry Osipenko пишет: >>>>>>>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>>>>>>> Tegra210 and prior Tegra PMC has clk_out_1, clk_out_2, clk_out_3 >>>>>>>>>>> with >>>>>>>>>>> mux and gate for each of these clocks. >>>>>>>>>>> >>>>>>>>>>> Currently these PMC clocks are registered by Tegra clock driver >>>>>>>>>>> using >>>>>>>>>>> clk_register_mux and clk_register_gate by passing PMC base >>>>>>>>>>> address >>>>>>>>>>> and register offsets and PMC programming for these clocks >>>>>>>>>>> happens >>>>>>>>>>> through direct PMC access by the clock driver. >>>>>>>>>>> >>>>>>>>>>> With this, when PMC is in secure mode any direct PMC access >>>>>>>>>>> from the >>>>>>>>>>> non-secure world does not go through and these clocks will >>>>>>>>>>> not be >>>>>>>>>>> functional. >>>>>>>>>>> >>>>>>>>>>> This patch adds these clocks registration with PMC as a clock >>>>>>>>>>> provider >>>>>>>>>>> for these clocks. clk_ops callback implementations for these >>>>>>>>>>> clocks >>>>>>>>>>> uses tegra_pmc_readl and tegra_pmc_writel which supports PMC >>>>>>>>>>> programming >>>>>>>>>>> in secure mode and non-secure mode. >>>>>>>>>>> >>>>>>>>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>>>>>>>> --- >>>>>>>>> [snip] >>>>>>>>> >>>>>>>>>>> + >>>>>>>>>>> +static const struct clk_ops pmc_clk_gate_ops = { >>>>>>>>>>> + .is_enabled = pmc_clk_is_enabled, >>>>>>>>>>> + .enable = pmc_clk_enable, >>>>>>>>>>> + .disable = pmc_clk_disable, >>>>>>>>>>> +}; >>>>>>>>>> What's the benefit of separating GATE from the MUX? >>>>>>>>>> >>>>>>>>>> I think it could be a single clock. >>>>>>>>> According to TRM: >>>>>>>>> >>>>>>>>> 1. GATE and MUX are separate entities. >>>>>>>>> >>>>>>>>> 2. GATE is the parent of MUX (see PMC's CLK_OUT paths diagram >>>>>>>>> in TRM). >>>>>>>>> >>>>>>>>> 3. PMC doesn't gate EXTPERIPH clock but could "force-enable" it, >>>>>>>>> correct? >>>>> Was following existing clk-tegra-pmc as I am not sure of reason for >>>>> having these clocks registered as separate mux and gate clocks. >>>>> >>>>> Yes, PMC clocks can be registered as single clock and can use clk_ops >>>>> for set/get parent and enable/disable. >>>>> >>>>> enable/disable of PMC clocks is for force-enable to force the clock to >>>>> run regardless of ACCEPT_REQ or INVERT_REQ. >>>>> >>>>>>>> 4. clk_m_div2/4 are internal PMC OSC dividers and thus these clocks >>>>>>>> should belong to PMC. >>>>>>> Also, it should be "osc" and not "clk_m". >>>>>> I followed the same parents as it were in existing clk-tegra-pmc >>>>>> driver. >>>>>> >>>>>> Yeah they are wrong and they should be from osc and not clk_m. >>>>>> >>>>>> Will fix in next version. >>>>>> >> Reg clk_m_div2/3, they are dividers at OSC pad and not really internal >> to PMC block. >> >> current clock driver creates clk_m_div clocks which should actually be >> osc_div2/osc_div4 clocks with osc as parent. >> >> There are no clk_m_div2 and clk_m_div4 from clk_m >> >> Will fix this in next version. >> >>>> Could you please describe the full EXTPERIPH clock topology and how the >>>> pinmux configuration is related to it all? >>>> >>>> What is internal to the Tegra chip and what are the external outputs? >>>> >>>> Is it possible to bypass PMC on T30+ for the EXTPERIPH clocks? >>> >>> PMC CLK1/2/3 possible sources are OSC_DIV1, OSC_DIV2, OSC_DIV4, >>> EXTPERIPH from CAR. >>> >>> OSC_DIV1/2/4 are with internal dividers at the OSC Pads >>> >>> EXTPERIPH is from CAR and it has reset and enable controls along with >>> clock source selections to choose one of the PLLA_OUT0, CLK_S, >>> PLLP_OUT0, CLK_M, PLLE_OUT0 >>> >>> So, PMC CLK1/2/4 possible parents are OSC_DIV1, OSC_DIV2, OSC_DIV4, >>> EXTERN. >>> >>> >>> CLK1/2/3 also has Pinmux to route EXTPERIPH output on to these pins. >>> >>> >>> When EXTERN output clock is selected for these PMC clocks thru >>> CLKx_SRC_SEL, output clock is from driver by EXTPERIPH from CAR via >>> Pinmux logic or driven as per CLKx_SRC_SEL bypassing pinmux based on >>> CLKx_ACCEPT_REQ bit. >>> >>> >>> PMC Clock control register has bit CLKx_ACCEPT_REQ >>> When CLKx_ACCEPT_REQ = 0, output clock driver is from by EXTPERIPH >>> through the pinmux >>> When CLKx_ACCEPT_REQ = 1, output clock is based on CLKx_SRC_SEL bits >>> (OSC_DIV1/2/4 and EXTPERIPH clock bypassing the pinmux) >>> >>> FORCE_EN bit in PMC CLock control register forces the clock to run >>> regardless of this. > > PMC clock gate is based on the state of CLKx_ACCEPT_REQ and FORCE_EN > like explained above. > > CLKx_ACCEPT_REQ is 0 default and FORCE_EN acts as gate to enable/disable > EXTPERIPH clock output to PMC CLK_OUT_1/2/3. [and to enable OSC as well] > So I believe we need to register as MUX and Gate rather than as a single > clock. Please confirm. 1. The force-enabling is applied to both OSC and EXTERN sources of PMC_CLK_OUT_x by PMC at once. 2. Both of PMC's force-enabling and OSC/EXTERN selection is internal to PMC. Should be better to define it as a single "pmc_clk_out_x". I don't see any good reasons for differentiating PMC's Gate from the MUX, it's a single hardware unit from a point of view of the rest of the system. Peter, do you have any objections?

2 7

[alsa-devel] [PATCH] ALSA: hda - Downgrade error message for single-cmd fallback
by Takashi Iwai 16 Dec '19

16 Dec '19

We made the error message for the CORB/RIRB communication clearer by upgrading to dev_WARN() so that user can notice better. But this struck us like a boomerang: now it caught syzbot and reported back as a fatal issue although it's not really any too serious bug that worth for stopping the whole system. OK, OK, let's be softy, downgrade it to the standard dev_err() again. Fixes: dd65f7e19c69 ("ALSA: hda - Show the fatal CORB/RIRB error more clearly") Reported-by: syzbot+b3028ac3933f5c466389(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2f3b7a35f2d9..ba56b59b3e17 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -883,7 +883,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, return -EAGAIN; /* give a chance to retry */ } - dev_WARN(chip->card->dev, + dev_err(chip->card->dev, "azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n", bus->last_cmd[addr]); chip->single_cmd = 1; -- 2.16.4

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: mediatek: Use managed buffer allocation
by Takashi Iwai 16 Dec '19

16 Dec '19

Clean up the drivers with the new managed buffer allocation API. The superfluous snd_pcm_lib_malloc_pages() and snd_pcm_lib_free_pages() calls are dropped, as well as the superfluous snd_pcm_lib_preallocate_free_for_all() call. As of the result, hw_free and pcm_destruct ops became empty and got removed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 6 +----- sound/soc/mediatek/common/mtk-afe-platform-driver.c | 12 ++---------- sound/soc/mediatek/common/mtk-afe-platform-driver.h | 2 -- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 1 - sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 1 - 5 files changed, 3 insertions(+), 19 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index e761cb66be5d..8f314e15ce73 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -129,10 +129,6 @@ int mtk_afe_fe_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(params); snd_pcm_format_t format = params_format(params); - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; - if (afe->request_dram_resource) afe->request_dram_resource(afe->dev); @@ -193,7 +189,7 @@ int mtk_afe_fe_hw_free(struct snd_pcm_substream *substream, if (afe->release_dram_resource) afe->release_dram_resource(afe->dev); - return snd_pcm_lib_free_pages(substream); + return 0; } EXPORT_SYMBOL_GPL(mtk_afe_fe_hw_free); diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index b6624d8d084b..719d43bb0b8f 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -120,25 +120,17 @@ int mtk_afe_pcm_new(struct snd_soc_component *component, struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); size = afe->mtk_afe_hardware->buffer_bytes_max; - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - afe->dev, size, size); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + afe->dev, size, size); return 0; } EXPORT_SYMBOL_GPL(mtk_afe_pcm_new); -void mtk_afe_pcm_free(struct snd_soc_component *component, - struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} -EXPORT_SYMBOL_GPL(mtk_afe_pcm_free); - const struct snd_soc_component_driver mtk_afe_pcm_platform = { .name = AFE_PCM_NAME, .ioctl = snd_soc_pcm_lib_ioctl, .pointer = mtk_afe_pcm_pointer, .pcm_construct = mtk_afe_pcm_new, - .pcm_destruct = mtk_afe_pcm_free, }; EXPORT_SYMBOL_GPL(mtk_afe_pcm_platform); diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.h b/sound/soc/mediatek/common/mtk-afe-platform-driver.h index e550d11568c3..fcc923b88f12 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.h +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.h @@ -21,8 +21,6 @@ snd_pcm_uframes_t mtk_afe_pcm_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream); int mtk_afe_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd); -void mtk_afe_pcm_free(struct snd_soc_component *component, - struct snd_pcm *pcm); int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component); diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 033c07fb599c..0743c699d8c1 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -715,7 +715,6 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .ioctl = snd_soc_pcm_lib_ioctl, .pointer = mtk_afe_pcm_pointer, .pcm_construct = mtk_afe_pcm_new, - .pcm_destruct = mtk_afe_pcm_free, }; static int mt6797_dai_memif_register(struct mtk_base_afe *afe) diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 76af09d8f1af..6b5f407de3d0 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1053,7 +1053,6 @@ static const struct snd_soc_component_driver mt8183_afe_component = { .ioctl = snd_soc_pcm_lib_ioctl, .pointer = mtk_afe_pcm_pointer, .pcm_construct = mtk_afe_pcm_new, - .pcm_destruct = mtk_afe_pcm_free, }; static int mt8183_dai_memif_register(struct mtk_base_afe *afe) -- 2.16.4

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[alsa-devel] [PATCH] ASoC: wm8962: fix lambda value
by Shengjiu Wang 16 Dec '19

16 Dec '19

According to user manual, it is required that FLL_LAMBDA > 0 in all cases (Integer and Franctional modes). Fixes: 9a76f1ff6e29 ("ASoC: Add initial WM8962 CODEC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3e5c69fbc33a..d9d59f45833f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2788,7 +2788,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2858,7 +2858,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ -- 2.21.0

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[alsa-devel] [PATCH v2 00/15] ASoC: soc-core cleanup step7
by Kuninori Morimoto 16 Dec '19

16 Dec '19

Hi Mark These are v2 of codec_conf cleanup. ASoC has codec_conf, and we are using original dev_name, of_node for it to finding codec. But, we already have snd_soc_dai_link_component method. We can reuse it for codec_conf, too. This patches are for it. Big change from v1 is - COMP_CODEC_CNF(_name) + COMP_CODEC_CONF(_name) Kuninori Morimoto (15): ASoC: soc-core: support snd_soc_dai_link_component for codec_conf ASoC: fsl: imx-audmix: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_da7219_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_rt5663_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_rt5663_rt5514_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: skl_nau88l25_ssm4567: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8173-rt5650-rt5514: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8173-rt5650-rt5676: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8183-da7219-max98357: use snd_soc_dai_link_component for codec_conf ASoC: samsung: bells: use snd_soc_dai_link_component for codec_conf ASoC: samsung: lowland: use snd_soc_dai_link_component for codec_conf ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for codec_conf ASoC: samsung: speyside: use snd_soc_dai_link_component for codec_conf ASoC: ti: rx51: use snd_soc_dai_link_component for codec_conf ASoC: soc-core: remove legacy style of codec_conf include/sound/soc.h | 4 ++-- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- sound/soc/samsung/bells.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 12 +++++------- sound/soc/ti/rx51.c | 12 ++++++------ 15 files changed, 31 insertions(+), 33 deletions(-) -- 2.7.4

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[alsa-devel] WARNING in azx_rirb_get_response
by syzbot 16 Dec '19

16 Dec '19

Hello, syzbot found the following crash on: HEAD commit: 37d4e84f Merge tag 'ceph-for-5.5-rc2' of git://github.com/.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=10bbc9a6e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=79f79de2a27d3e3d dashboard link: https://syzkaller.appspot.com/bug?extid=b3028ac3933f5c466389 compiler: gcc (GCC) 9.0.0 20181231 (experimental) userspace arch: i386 Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+b3028ac3933f5c466389(a)syzkaller.appspotmail.com snd_hda_intel 0000:00:06.0: No response from codec, disabling MSI: last cmd=0x004a0000 ------------[ cut here ]------------ snd_hda_intel 0000:00:06.0: azx_get_response timeout, switching to single_cmd mode: last cmd=0x004a0000 WARNING: CPU: 2 PID: 8020 at sound/pci/hda/hda_controller.c:886 azx_rirb_get_response+0x5d6/0x720 sound/pci/hda/hda_controller.c:886 Kernel panic - not syncing: panic_on_warn set ... CPU: 2 PID: 8020 Comm: syz-executor.1 Not tainted 5.5.0-rc1-syzkaller #0 Hardware name: QEMU Standard PC (Q35 + ICH9, 2009), BIOS rel-1.12.0-59-gc9ba5276e321-prebuilt.qemu.org 04/01/2014 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x197/0x210 lib/dump_stack.c:118 panic+0x2e3/0x75c kernel/panic.c:221 __warn.cold+0x2f/0x3e kernel/panic.c:582 report_bug+0x289/0x300 lib/bug.c:195 fixup_bug arch/x86/kernel/traps.c:174 [inline] fixup_bug arch/x86/kernel/traps.c:169 [inline] do_error_trap+0x11b/0x200 arch/x86/kernel/traps.c:267 do_invalid_op+0x37/0x50 arch/x86/kernel/traps.c:286 invalid_op+0x23/0x30 arch/x86/entry/entry_64.S:1027 RIP: 0010:azx_rirb_get_response+0x5d6/0x720 sound/pci/hda/hda_controller.c:886 Code: f6 00 00 00 49 8b 5d 00 e8 27 75 7a fb 4c 89 ef e8 ef 21 5a fe 44 89 f9 48 89 da 48 c7 c7 80 07 ce 88 48 89 c6 e8 d9 1e 4b fb <0f> 0b 48 b8 00 00 00 00 00 fc ff df 48 8b 4d 88 48 89 ca 48 c1 ea RSP: 0018:ffffc9000236f708 EFLAGS: 00010286 RAX: 0000000000000000 RBX: ffff888078f64240 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffffffff815e88f6 RDI: fffff5200046ded3 RBP: ffffc9000236f780 R08: ffff8880673d0140 R09: ffffed1005a46621 R10: ffffed1005a46620 R11: ffff88802d233107 R12: ffff888029ef0000 R13: ffff88802759a0b0 R14: dffffc0000000000 R15: 00000000004a0000 azx_get_response sound/pci/hda/hda_controller.c:994 [inline] azx_get_response+0x14c/0x180 sound/pci/hda/hda_controller.c:984 snd_hdac_bus_exec_verb_unlocked+0x4b5/0x950 sound/hda/hdac_bus.c:115 codec_exec_verb+0x13d/0x340 sound/pci/hda/hda_codec.c:52 snd_hdac_exec_verb+0x65/0x110 sound/hda/hdac_device.c:257 codec_read+0xa2/0x100 sound/hda/hdac_device.c:986 snd_hdac_codec_read+0x34/0x40 sound/hda/hdac_device.c:1015 snd_hda_codec_read include/sound/hda_codec.h:318 [inline] update_pcm_format.part.0+0x36/0xc0 sound/pci/hda/hda_codec.c:1057 update_pcm_format sound/pci/hda/hda_codec.c:1056 [inline] snd_hda_codec_setup_stream.part.0+0x355/0xa40 sound/pci/hda/hda_codec.c:1102 snd_hda_codec_setup_stream+0x47/0x60 sound/pci/hda/hda_codec.c:1086 capture_pcm_prepare+0x63/0x100 sound/pci/hda/hda_generic.c:5376 snd_hda_codec_prepare+0xa5/0x420 sound/pci/hda/hda_codec.c:3160 azx_pcm_prepare+0x53f/0x950 sound/pci/hda/hda_controller.c:195 snd_pcm_do_prepare+0x61/0xa0 sound/core/pcm_native.c:1729 snd_pcm_action_single+0x7a/0x130 sound/core/pcm_native.c:1111 snd_pcm_action_nonatomic+0x9c/0xb0 sound/core/pcm_native.c:1227 snd_pcm_prepare+0x121/0x1b0 sound/core/pcm_native.c:1776 snd_pcm_kernel_ioctl+0x117/0x1f0 sound/core/pcm_native.c:3068 snd_pcm_oss_prepare+0x49/0x150 sound/core/oss/pcm_oss.c:1125 snd_pcm_oss_make_ready+0x12b/0x170 sound/core/oss/pcm_oss.c:1153 snd_pcm_oss_sync.isra.0+0x1c4/0x7e0 sound/core/oss/pcm_oss.c:1707 snd_pcm_oss_release+0x214/0x290 sound/core/oss/pcm_oss.c:2545 __fput+0x2ff/0x890 fs/file_table.c:280 ____fput+0x16/0x20 fs/file_table.c:313 task_work_run+0x145/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x316/0x380 arch/x86/entry/common.c:164 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:278 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:352 [inline] do_fast_syscall_32+0xbbd/0xe16 arch/x86/entry/common.c:408 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7faea39 Code: 00 00 00 89 d3 5b 5e 5f 5d c3 b8 80 96 98 00 eb c4 8b 04 24 c3 8b 1c 24 c3 8b 34 24 c3 8b 3c 24 c3 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000ffadef9c EFLAGS: 00000293 ORIG_RAX: 0000000000000006 RAX: 0000000000000000 RBX: 0000000000000004 RCX: 0000000000000000 RDX: 000000000812b000 RSI: 0000000000000005 RDI: 000000000815b660 RBP: 000000000815b660 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[alsa-devel] [PATCH][resend v2] ASoC: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 16 Dec '19

16 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - Add Rob to "To:" instead of "Cc:" .../devicetree/bindings/sound/simple-card.txt | 351 --------------- .../devicetree/bindings/sound/simple-card.yaml | 472 +++++++++++++++++++++ 2 files changed, 472 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6..0000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 0000000..ac3975f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,472 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle-array + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle-array + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + hp-det-gpio: + description: | + Reference to GPIO that signals when headphones are attached. + $ref: /schemas/types.yaml#/definitions/phandle + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + mic-det-gpio: + description: | + Reference to GPIO that signals when a microphone is attached. + $ref: /schemas/types.yaml#/definitions/phandle + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + oneOf: + - items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + clocks: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/phandle-array + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": true + "#size-cells": true + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + simple-audio-card,routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + "^simple-audio-card,cpu.*": + description: CPU node + $ref: "#/definitions/dai" + "^simple-audio-card,codec.*": + description: CPU node + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,hp-det-gpio$": + $ref: "#/definitions/hp-det-gpio" + "^simple-audio-card,mic-det-gpio$": + $ref: "#/definitions/mic-det-gpio" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" + + "^simple-audio-card,dai-link": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + hp-det-gpio: + $ref: "#/definitions/hp-det-gpio" + mic-det-gpio: + $ref: "#/definitions/mic-det-gpio" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + + patternProperties: + "^cpu.*": + $ref: "#/definitions/dai" + "^codec.*": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.7.4

1 0

[alsa-devel] [PATCH][resend v2] ASoC: renesas, rsnd: switch to yaml base Documentation
by Kuninori Morimoto 16 Dec '19

16 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. It is still keeping detail explanations at .txt Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - Add Rob to "To:" instead of "Cc:" .../devicetree/bindings/sound/renesas,rsnd.txt | 518 -------------------- .../devicetree/bindings/sound/renesas,rsnd.yaml | 527 +++++++++++++++++++++ 2 files changed, 527 insertions(+), 518 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 797fd03..b731f16 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -253,521 +253,3 @@ This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. see "Example: simple sound card for TDM" - -============================================= -Required properties: -============================================= - -- compatible : "renesas,rcar_sound-<soctype>", fallbacks - "renesas,rcar_sound-gen1" if generation1, and - "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) - Examples with soctypes are: - - "renesas,rcar_sound-r8a7743" (RZ/G1M) - - "renesas,rcar_sound-r8a7744" (RZ/G1N) - - "renesas,rcar_sound-r8a7745" (RZ/G1E) - - "renesas,rcar_sound-r8a77470" (RZ/G1C) - - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - - "renesas,rcar_sound-r8a7778" (R-Car M1A) - - "renesas,rcar_sound-r8a7779" (R-Car H1) - - "renesas,rcar_sound-r8a7790" (R-Car H2) - - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - - "renesas,rcar_sound-r8a7794" (R-Car E2) - - "renesas,rcar_sound-r8a7795" (R-Car H3) - - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - - "renesas,rcar_sound-r8a77990" (R-Car E3) - - "renesas,rcar_sound-r8a77995" (R-Car D3) -- reg : Should contain the register physical address. - required register is - SRU/ADG/SSI if generation1 - SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3 - Select extended AUDIO-DMAC-periperi address if SoC has it, - otherwise select normal AUDIO-DMAC-periperi address. -- reg-names : Should contain the register names. - scu/adg/ssi if generation1 - scu/adg/ssiu/ssi/audmapp if generation2/generation3 -- rcar_sound,ssi : Should contain SSI feature. - The number of SSI subnode should be same as HW. - see below for detail. -- rcar_sound,ssiu : Should contain SSIU feature. - The number of SSIU subnode should be same as HW. - see below for detail. -- rcar_sound,src : Should contain SRC feature. - The number of SRC subnode should be same as HW. - see below for detail. -- rcar_sound,ctu : Should contain CTU feature. - The number of CTU subnode should be same as HW. - see below for detail. -- rcar_sound,mix : Should contain MIX feature. - The number of MIX subnode should be same as HW. - see below for detail. -- rcar_sound,dvc : Should contain DVC feature. - The number of DVC subnode should be same as HW. - see below for detail. -- rcar_sound,dai : DAI contents. - The number of DAI subnode should be same as HW. - see below for detail. -- #sound-dai-cells : it must be 0 if your system is using single DAI - it must be 1 if your system is using multi DAI -- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. -- clock-names : List of necessary clock names. - "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", - "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" - -Optional properties: -- #clock-cells : it must be 0 if your system has audio_clkout - it must be 1 if your system has audio_clkout0/1/2/3 -- clock-frequency : for all audio_clkout0/1/2/3 -- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn - is asynchronizes with lr-clock. -- resets : References to SSI resets. -- reset-names : List of valid reset names. - "ssi-all", "ssi.X" - -SSI subnode properties: -- interrupts : Should contain SSI interrupt for PIO transfer -- shared-pin : if shared clock pin -- pio-transfer : use PIO transfer mode -- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case -- dma : Should contain Audio DMAC entry -- dma-names : SSI case "rx" (=playback), "tx" (=capture) - Deprecated: see SSIU subnode properties - SSIU case "rxu" (=playback), "txu" (=capture) - -SSIU subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -SRC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -DVC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "tx" (=playback/capture) - -DAI subnode properties: -- playback : list of playback modules -- capture : list of capture modules - - -============================================= -Example: -============================================= - -rcar_sound: sound@ec500000 { - #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; - reg = <0 0xec500000 0 0x1000>, /* SCU */ - <0 0xec5a0000 0 0x100>, /* ADG */ - <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ - reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; - - clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, - <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, - <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, - <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, - <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, - <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, - <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, - <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, - <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, - <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, - <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, - <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, - <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; - clock-names = "ssi-all", - "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", - "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", - "src.9", "src.8", "src.7", "src.6", "src.5", - "src.4", "src.3", "src.2", "src.1", "src.0", - "dvc.0", "dvc.1", - "clk_a", "clk_b", "clk_c", "clk_i"; - - rcar_sound,dvc { - dvc0: dvc-0 { - dmas = <&audma0 0xbc>; - dma-names = "tx"; - }; - dvc1: dvc-1 { - dmas = <&audma0 0xbe>; - dma-names = "tx"; - }; - }; - - rcar_sound,mix { - mix0: mix-0 { }; - mix1: mix-1 { }; - }; - - rcar_sound,ctu { - ctu00: ctu-0 { }; - ctu01: ctu-1 { }; - ctu02: ctu-2 { }; - ctu03: ctu-3 { }; - ctu10: ctu-4 { }; - ctu11: ctu-5 { }; - ctu12: ctu-6 { }; - ctu13: ctu-7 { }; - }; - - rcar_sound,src { - src0: src-0 { - interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x85>, <&audma1 0x9a>; - dma-names = "rx", "tx"; - }; - src1: src-1 { - interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x87>, <&audma1 0x9c>; - dma-names = "rx", "tx"; - }; - src2: src-2 { - interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x89>, <&audma1 0x9e>; - dma-names = "rx", "tx"; - }; - src3: src-3 { - interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8b>, <&audma1 0xa0>; - dma-names = "rx", "tx"; - }; - src4: src-4 { - interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8d>, <&audma1 0xb0>; - dma-names = "rx", "tx"; - }; - src5: src-5 { - interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8f>, <&audma1 0xb2>; - dma-names = "rx", "tx"; - }; - src6: src-6 { - interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x91>, <&audma1 0xb4>; - dma-names = "rx", "tx"; - }; - src7: src-7 { - interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x93>, <&audma1 0xb6>; - dma-names = "rx", "tx"; - }; - src8: src-8 { - interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x95>, <&audma1 0xb8>; - dma-names = "rx", "tx"; - }; - src9: src-9 { - interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x97>, <&audma1 0xba>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssiu { - ssiu00: ssiu-0 { - dmas = <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx"; - }; - ssiu01: ssiu-1 { - dmas = <&audma0 0x35>, <&audma1 0x36>; - dma-names = "rx", "tx"; - }; - - ... - - ssiu95: ssiu-49 { - dmas = <&audma0 0xA5>, <&audma1 0xA6>; - dma-names = "rx", "tx"; - }; - ssiu96: ssiu-50 { - dmas = <&audma0 0xA7>, <&audma1 0xA8>; - dma-names = "rx", "tx"; - }; - ssiu97: ssiu-51 { - dmas = <&audma0 0xA9>, <&audma1 0xAA>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssi { - ssi0: ssi-0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>; - dma-names = "rx", "tx"; - }; - ssi1: ssi-1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>; - dma-names = "rx", "tx"; - }; - - ... - - ssi8: ssi-8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>; - dma-names = "rx", "tx"; - }; - ssi9: ssi-9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,dai { - dai0 { - playback = <&ssi5 &src5>; - capture = <&ssi6>; - }; - dai1 { - playback = <&ssi3>; - }; - dai2 { - capture = <&ssi4>; - }; - dai3 { - playback = <&ssi7>; - }; - dai4 { - capture = <&ssi8>; - }; - }; -}; - -============================================= -Example: simple sound card -============================================= - - rsnd_ak4643: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; - }; - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; -}; - -&ssi1 { - shared-pin; -}; - -============================================= -Example: simple sound card for Asynchronous mode -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * SRC Asynchronous mode setting - * Playback: - * All input data will be converted to 48kHz - * Capture: - * Inputed 48kHz data will be converted to - * system specified Hz - */ - simple-audio-card,convert-rate = <48000>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for channel convert -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * CTU setting - * All input data will be converted to 2ch - * as output data - */ - simple-audio-card,convert-channels = <2>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for MIXer -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - simple-audio-card,cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - simple-audio-card,cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - simple-audio-card,codec { - ... - }; -}; - -&rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; - }; - dai1 { - playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; - }; - }; -}; - -============================================= -Example: simple sound card for TDM -============================================= - -rsnd_tdm: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - dai-tdm-slot-num = <6>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&xxx>; - }; -}; - -============================================= -Example: simple sound card for TDM Split -============================================= - -sound_card: sound { - compatible = "audio-graph-scu-card"; - prefix = "xxxx"; - routing = "xxxx Playback", "DAI0 Playback", - "xxxx Playback", "DAI1 Playback", - "xxxx Playback", "DAI2 Playback", - "xxxx Playback", "DAI3 Playback"; - convert-channels = <8>; /* TDM Split */ - - dais = <&rsnd_port0 /* playback ch1/ch2 */ - &rsnd_port1 /* playback ch3/ch4 */ - &rsnd_port2 /* playback ch5/ch6 */ - &rsnd_port3 /* playback ch7/ch8 */ - >; -}; - -audio-codec { - ... - port { - codec_0: endpoint@1 { - remote-endpoint = <&rsnd_ep0>; - }; - codec_1: endpoint@2 { - remote-endpoint = <&rsnd_ep1>; - }; - codec_2: endpoint@3 { - remote-endpoint = <&rsnd_ep2>; - }; - codec_3: endpoint@4 { - remote-endpoint = <&rsnd_ep3>; - }; - }; -}; - -&rcar_sound { - ... - ports { - rsnd_port0: port@0 { - rsnd_ep0: endpoint { - remote-endpoint = <&codec_0>; - ... - playback = <&ssiu30 &ssi3>; - }; - }; - rsnd_port1: port@1 { - rsnd_ep1: endpoint { - remote-endpoint = <&codec_1>; - ... - playback = <&ssiu31 &ssi3>; - }; - }; - rsnd_port2: port@2 { - rsnd_ep2: endpoint { - remote-endpoint = <&codec_2>; - ... - playback = <&ssiu32 &ssi3>; - }; - }; - rsnd_port3: port@3 { - rsnd_ep3: endpoint { - remote-endpoint = <&codec_3>; - ... - playback = <&ssiu33 &ssi3>; - }; - }; - }; -}; - -============================================= -Example: simple sound card for Multi channel -============================================= - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 0000000..5fb036c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,527 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + + reg-names: + description: | + Should contain the register names. + scu/adg/ssi if generation1 + scu/adg/ssiu/ssi/audmapp if generation2/generation3 + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + allOf: + - items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + $ref: /schemas/types.yaml#/definitions/uint32-array + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + description: References to SSI resets + $ref: /schemas/types.yaml#/definitions/phandle-array + + reset-names: + description: List of valid reset names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + + clocks: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + + clock-names: + description: List of necessary clock names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + - src.9 + - src.8 + - src.7 + - src.6 + - src.5 + - src.4 + - src.3 + - src.2 + - src.1 + - src.0 + - mix.1 + - mix.0 + - ctu.1 + - ctu.0 + - dvc.1 + - dvc.0 + - clk_a + - clk_b + - clk_c + - clk_i + + # For OF-graph + port: + description: OF-Graph subnode + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + endpoint: + type: object + properties: + remote-endpoint: + $ref: /schemas/types.yaml#/definitions/phandle-array + dai-format: + $ref: "simple-card.yaml#/definitions/format" + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - remote-endpoint + + # For multi OF-graph + ports: + description: multi OF-Graph subnode + type: object + patternProperties: + "port.*": + $ref: "#properties/port" + +patternProperties: + "^rcar_sound,dvc$": + description: DVC subnode. + type: object + patternProperties: + "dvc.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + const: "tx" + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,mix$": + description: MIX subnode. + type: object + patternProperties: + "mix.*": + type: object + additionalProperties: false + + "^rcar_sound,ctu$": + description: CTU subnode. + type: object + patternProperties: + "ctu.*": + type: object + additionalProperties: false + + "^rcar_sound,src$": + description: SRC subnode. + type: object + patternProperties: + "src.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + if: + properties: + status: + const: "ok" + then: + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssiu$": + description: SSIU subnode. + type: object + patternProperties: + "ssiu.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssi$": + description: SSI subnode. + type: object + patternProperties: + "ssi.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + "^rcar_sound,dai$": + description: DAI subnode. + type: object + patternProperties: + "dai.*": + type: object + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0 0xec500000 0 0x1000>, /* SCU */ + <0 0xec5a0000 0 0x100>, /* ADG */ + <0 0xec540000 0 0x1000>, /* SSIU */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* DVC0, DVC1 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CLKA, CLKB */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* OF-Graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + + /* Multi OF-Graph */ + ports { + #address-cells = <1>; + #size-cells = <0>; + rsnd_port0: port@0 { + reg = <0>; + rsnd_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + rsnd_port1: port@1 { + reg = <1>; + rsnd_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint2>; + + dai-format = "i2s"; + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + }; + }; + + + codec { + ports { + port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + port@1 { + codec_endpoint1: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + port@2 { + codec_endpoint2: endpoint { + remote-endpoint = <&rsnd_endpoint1>; + }; + }; + }; + }; -- 2.7.4

1 0

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Maybe, double, unsigned int, or unsigned int64? Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Maybe, double? Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bits signed integers for sample rate conversion.
by GitHub issues - opened 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was opened from crocket: Sample rate conversion should be done with the highest resolution possible. Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] [PATCH] ALSA: hda: Modify stream stripe mask only when needed
by Takashi Iwai 15 Dec '19

15 Dec '19

The recent in HD-audio stream management for changing the stripe control seems causing a regression on some platforms. The stripe control is currently used only by HDMI codec, and applying the stripe mask unconditionally may lead to scratchy and static noises as seen on some MacBooks. For addressing the regression, this patch changes the stream management code to apply the stripe mask conditionally only when the codec driver requested. Fixes: 9b6f7e7a296e ("ALSA: hda: program stripe bits for controller") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204477 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_stream.c | 19 ++++++++++++------- sound/pci/hda/patch_hdmi.c | 5 +++++ 3 files changed, 18 insertions(+), 7 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b260c5fd2337..e05b95e83d5a 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -493,6 +493,7 @@ struct hdac_stream { bool prepared:1; bool no_period_wakeup:1; bool locked:1; + bool stripe:1; /* apply stripe control */ /* timestamp */ unsigned long start_wallclk; /* start + minimum wallclk */ diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index d8fe7ff0cd58..f9707fb05efe 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -96,12 +96,14 @@ void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start) 1 << azx_dev->index, 1 << azx_dev->index); /* set stripe control */ - if (azx_dev->substream) - stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream); - else - stripe_ctl = 0; - snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, - stripe_ctl); + if (azx_dev->stripe) { + if (azx_dev->substream) + stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream); + else + stripe_ctl = 0; + snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, + stripe_ctl); + } /* set DMA start and interrupt mask */ snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_DMA_START | SD_INT_MASK); @@ -118,7 +120,10 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ - snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); + if (azx_dev->stripe) { + snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); + azx_dev->stripe = 0; + } azx_dev->running = false; } EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0032bba8cc9d..ed4e98a1057a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -32,6 +32,7 @@ #include <sound/hda_codec.h> #include "hda_local.h" #include "hda_jack.h" +#include "hda_controller.h" static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); @@ -1249,6 +1250,10 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_pin->cvt_nid = per_cvt->cvt_nid; hinfo->nid = per_cvt->cvt_nid; + /* flip stripe flag for the assigned stream if supported */ + if (get_wcaps(codec, per_cvt->cvt_nid) & AC_WCAP_STRIPE) + azx_stream(get_azx_dev(substream))->stripe = 1; + snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, -- 2.16.4

2 4

[alsa-devel] Code Example for Playing Sound in the Background
by GitHub issues - edited 14 Dec '19

14 Dec '19

alsa-project/alsa-lib issue #17 was edited from leimao: Hello, I wrote a simple C++ player based on alsa-lib which allows me to play `wav` files. I wonder if there is any convenient way to modify my code so that I can play the the `wav` file in the background and the main program is executing something else, say, a `while` loop. _Edited and Updated:_ I tried to use `std::thread` and `std::async`, I was able to play the sound in the background. I looked at the header file of alsa-lib, it seems that alsa-lib natively supports playing sound asynchronously. I wonder how I could do it easily. Thank you very much. Best, Lei Issue URL : https://github.com/alsa-project/alsa-lib/issues/17 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] Code Example for Playing Sound in the Background
by GitHub issues - edited 14 Dec '19

14 Dec '19

alsa-project/alsa-lib issue #17 was edited from leimao: Hello, I wrote a simple C++ player based on alsa-lib which allows me to play `wav` files. I wonder if there is any convenient way to modify my code so that I can play the the `wav` file in the background and the main program is executing something else, say, a `while` loop. Edited and Updated: I tried to use `std::thread` and `std::async`, I was able to play the sound in the background. I looked at the header file of alsa-lib, it seems that alsa-lib natively supports playing sound asynchronously. I wonder how I could do it easily. Thank you very much. Best, Lei Issue URL : https://github.com/alsa-project/alsa-lib/issues/17 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] Code Example for Playing Sound in the Background
by GitHub issues - opened 14 Dec '19

14 Dec '19

alsa-project/alsa-lib issue #17 was opened from leimao: Hello, I wrote a simple C++ player based on alsa-lib which allows me to play `wav` files. I wonder if there is any convenient way to modify my code so that I can play the the `wav` file in the background and the main program is executing something else, say, a `while` loop. I tried to use `std::thread` but I did not succeed. Thank you very much. Best, Lei Issue URL : https://github.com/alsa-project/alsa-lib/issues/17 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH] ALSA: hda: Fix regression by strip mask fix
by Takashi Iwai 14 Dec '19

14 Dec '19

The commit e38e486d66e2 ("ALSA: hda: Modify stream stripe mask only when needed") tried to address the regression by the unconditional application of the stripe mask, but this caused yet another regression for the previously working devices. Namely, the patch clears the azx_dev->stripe flag at snd_hdac_stream_clear(), but this may be called multiple times before restarting the stream, so this ended up with clearance of the flag for the whole time. This patch fixes the regression by moving the azx_dev->stripe flag clearance at the counter-part, the close callback of HDMI codec driver instead. Fixes: e38e486d66e2 ("ALSA: hda: Modify stream stripe mask only when needed") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205855 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204477 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_stream.c | 4 +--- sound/pci/hda/patch_hdmi.c | 2 ++ 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index f9707fb05efe..682ed39f79b0 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -120,10 +120,8 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ - if (azx_dev->stripe) { + if (azx_dev->stripe) snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); - azx_dev->stripe = 0; - } azx_dev->running = false; } EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78647ee02339..630b1f5c276d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2021,6 +2021,8 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; + azx_stream(get_azx_dev(substream))->stripe = 0; + mutex_lock(&spec->pcm_lock); snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: control: remove useless assignment in .info callback of PCM chmap element
by Takashi Sakamoto 14 Dec '19

14 Dec '19

Control elements for PCM chmap return information to userspace abount the maximum number of available PCM channels as the number of values in the element. In current implementation the number is once initialized to zero, then assigned to. This is useless and this commit fixes it. Fixes: 2d3391ec0ecc ("ALSA: PCM: channel mapping API implementation") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/pcm_lib.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e0c1f2..6f0a00fd3ae5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2308,7 +2308,6 @@ static int pcm_chmap_ctl_info(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 0; uinfo->count = info->max_channels; uinfo->value.integer.min = 0; uinfo->value.integer.max = SNDRV_CHMAP_LAST; -- 2.20.1

2 1

[alsa-devel] [PATCH 0/2] Add the broadmobi BM818
by Angus Ainslie (Purism) 14 Dec '19

14 Dec '19

The broadmobi uses slightly different parameters from the option modems so add the paramters and document them. Angus Ainslie (Purism) (2): sound: codecs: gtm601: add Broadmobi bm818 sound profile ASoC: gtm601: add the broadmobi interface .../devicetree/bindings/sound/gtm601.txt | 10 +++++-- sound/soc/codecs/gtm601.c | 29 +++++++++++++++++-- 2 files changed, 35 insertions(+), 4 deletions(-) -- 2.17.1

4 6

[alsa-devel] [PATCH] ALSA: hda: Comment about snd_hdac_bus_update_rirb() and spinlock
by Takashi Iwai 13 Dec '19

13 Dec '19

The call of snd_hdac_bus_update_rirb() needs the bus->reg_lock spinlock protection for concurrency. Comment about it more explicitly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_controller.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 01787081552d..b856184af37f 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -181,6 +181,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_send_cmd); * @bus: HD-audio core bus * * Usually called from interrupt handler. + * The caller needs bus->reg_lock spinlock before calling this. */ void snd_hdac_bus_update_rirb(struct hdac_bus *bus) { -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: pcm: Avoid possible info leaks from PCM stream buffers
by Takashi Iwai 13 Dec '19

13 Dec '19

The current PCM code doesn't initialize explicitly the buffers allocated for PCM streams, hence it might leak some uninitialized kernel data or previous stream contents by mmapping or reading the buffer before actually starting the stream. Since this is a common problem, this patch simply adds the clearance of the buffer data at hw_params callback. Although this does only zero-clear no matter which format is used, which doesn't mean the silence for some formats, but it should be OK because the intention is just to clear the previous data on the buffer. Reported-by: Lionel Koenig <lionel.koenig(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a revised patch of the previous one 20191210130219.27919-1-tiwai(a)suse.de sound/core/pcm_native.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1fe581167b7b..d083225344a0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -739,6 +739,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) runtime->boundary *= 2; + /* clear the buffer for avoiding possible kernel info leaks */ + if (runtime->dma_area && !substream->ops->copy_user) + memset(runtime->dma_area, 0, runtime->dma_bytes); + snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); -- 2.16.4

2 1

[alsa-devel] [PATCH 0/2] ALSA: hda: Use RIRB waitq commonly
by Takashi Iwai 13 Dec '19

13 Dec '19

Hi, this is a follow-up patch set for adapting the RIRB waitq to HD-audio core, and a cleanup with it. Takashi === Takashi Iwai (2): ALSA: hda: Use waitqueue for RIRB in HDA-core helper, too ALSA: hda: Unify get_response handling include/sound/hda_codec.h | 1 - include/sound/hdaudio.h | 1 + sound/hda/hdac_controller.c | 25 +++++++++++++++++++-- sound/pci/hda/hda_controller.c | 49 ++++-------------------------------------- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- 8 files changed, 32 insertions(+), 52 deletions(-) -- 2.16.4

2 4

[alsa-devel] [PATCH] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 13 Dec '19

13 Dec '19

The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper --- [PATCH v2] : 1. remove unnecessary space from commit message 2. add Signed-off-by info Signed-off-by: Jeff Chang <richtek.jeff.chang(a)gmail.com> --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 1063 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 95 ++++ 4 files changed, 1174 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..806cd03 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,19 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + select CRC32 + select CRYPTO_SHA256 + select CRYTO_RSA + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..5ba8ec5 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,1063 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +#if GENERIC_DEBUGFS +static int mt6660_dbg_io_read(void *drvdata, u16 reg, void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int ret = 0, i = 0; + unsigned int data; + u8 *_val = val; + + if (size != 1 && size != 2 && size != 4) { + dev_err(chip->dev, "%s size not match\n", __func__); + return -EINVAL; + } + ret = regmap_read(chip->regmap, reg, &data); + if (reg < 0) + return ret; + + for (i = 0; i < size; i++) + _val[size-i-1] = (data >> (8*i))&0xff; + + return 0; +} + +static int mt6660_dbg_io_write(void *drvdata, u16 reg, + const void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int reg_size = mt6660_get_reg_size(reg); + int i = 0; + unsigned int regval = 0; + u8 *_val = (u8 *)val; + + if (size != reg_size) { + dev_err(chip->dev, + "%s size not match reg_size(%d), size(%d)\n", + __func__, reg_size, size); + return -EINVAL; + } + + for (i = 0; i < size; i++) { + regval <<= 8; + regval |= _val[i]; + } + + return regmap_write(chip->regmap, reg, regval); +} +#endif /* GENERIC_DEBUGFS */ + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + + return ret; +} + +static bool mt6660_volatile_reg(struct device *dev, unsigned int reg) +{ + return true; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, + .volatile_reg = mt6660_volatile_reg, +}; + +static unsigned int mt6660_component_io_read( + struct snd_soc_component *component, unsigned int reg) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret; + + ret = regmap_read(chip->regmap, reg, &val); + if (ret < 0) /* ret success -> >= 0, fail -> < - */ + return ret; + pr_err("%s val = 0x%x\n", __func__, val); + return val; +} + +static int mt6660_component_io_write(struct snd_soc_component *component, + unsigned int reg, unsigned int data) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + pr_err("%s data = 0x%x\n", __func__, data); + return regmap_write(chip->regmap, reg, data); +} + +static const int mt6660_dump_table[] = { + MT6660_REG_DEVID, + MT6660_REG_SYSTEM_CTRL, + MT6660_REG_IRQ_STATUS1, + MT6660_REG_ADDA_CLOCK, + MT6660_REG_SERIAL_CFG1, + MT6660_REG_DATAO_SEL, + MT6660_REG_TDM_CFG3, + MT6660_REG_HPF_CTRL, + MT6660_REG_HPF1_COEF, + MT6660_REG_HPF2_COEF, + MT6660_REG_PATH_BYPASS, + MT6660_REG_WDT_CTRL, + MT6660_REG_HCLIP_CTRL, + MT6660_REG_VOL_CTRL, + MT6660_REG_SPS_CTRL, + MT6660_REG_SIGMAX, + MT6660_REG_CALI_T0, + MT6660_REG_BST_CTRL, + MT6660_REG_PROTECTION_CFG, + MT6660_REG_DA_GAIN, + MT6660_REG_AUDIO_IN2_SEL, + MT6660_REG_SIG_GAIN, + MT6660_REG_PLL_CFG1, + MT6660_REG_DRE_CTRL, + MT6660_REG_DRE_THDMODE, + MT6660_REG_DRE_CORASE, + MT6660_REG_PWM_CTRL, + MT6660_REG_DC_PROTECT_CTRL, + MT6660_REG_ADC_USB_MODE, + MT6660_REG_INTERNAL_CFG, + MT6660_REG_RESV0, + MT6660_REG_RESV1, + MT6660_REG_RESV2, + MT6660_REG_RESV3, + MT6660_REG_RESV7, + MT6660_REG_RESV10, + MT6660_REG_RESV11, + MT6660_REG_RESV16, + MT6660_REG_RESV17, + MT6660_REG_RESV19, + MT6660_REG_RESV21, + MT6660_REG_RESV23, + MT6660_REG_RESV31, + MT6660_REG_RESV40, +}; + +#if GENERIC_DEBUGFS +#ifdef CONFIG_DEBUG_FS +/* reg/size/data/bustype */ +#define PREALLOC_RBUFFER_SIZE (32) +#define PREALLOC_WBUFFER_SIZE (1000) + +static int data_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + int i, ret; + + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* read transfer */ + if (!di->io_read) + return -EPERM; + ret = di->io_read(di->io_drvdata, d->reg, d->data_buffer, d->size); + if (ret < 0) + return ret; + pdata = d->data_buffer; + seq_printf(s, "0x"); + for (i = 0; i < d->size; i++) + seq_printf(s, "%02x,", *(pdata + i)); + seq_printf(s, "\n"); + return 0; +} + +static int data_debug_open(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_open(file, data_debug_show, inode->i_private); + return simple_open(inode, file); +} + +static ssize_t data_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + char buf[PREALLOC_WBUFFER_SIZE + 1], *token, *cur; + int val_cnt = 0, ret; + + if (cnt > PREALLOC_WBUFFER_SIZE) + return -ENOMEM; + if (copy_from_user(buf, user_buf, cnt)) + return -EFAULT; + buf[cnt] = 0; + /* buffer size check */ + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* data parsing */ + cur = buf; + pdata = d->data_buffer; + while ((token = strsep(&cur, ",\n")) != NULL) { + if (!*token) + break; + if (val_cnt++ >= d->size) + break; + if (kstrtou8(token, 16, pdata++)) + return -EINVAL; + } + if (val_cnt != d->size) + return -EINVAL; + /* write transfer */ + if (!di->io_write) + return -EPERM; + ret = di->io_write(di->io_drvdata, d->reg, d->data_buffer, d->size); + return (ret < 0) ? ret : cnt; +} + +static int data_debug_release(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_release(inode, file); + return 0; +} + +static const struct file_operations data_debug_fops = { + .open = data_debug_open, + .read = seq_read, + .write = data_debug_write, + .llseek = seq_lseek, + .release = data_debug_release, +}; + +static int type_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + + seq_printf(s, "%s,%s\n", di->typestr, di->devname); + return 0; +} + +static int type_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, type_debug_show, inode->i_private); +} + +static const struct file_operations type_debug_fops = { + .open = type_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static int dump_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct mt6660_chip *chip = + container_of(di, struct mt6660_chip, dbg_info); + int i = 0, j = 0, ret = 0; + unsigned int val; + int size = 0; + + if (!chip) { + pr_err("%s chip is null\n", __func__); + return -ENODEV; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_dump_table); i++) { + ret = regmap_read(chip->regmap, mt6660_dump_table[i], &val); + size = mt6660_get_reg_size(mt6660_dump_table[i]); + seq_printf(s, "reg0x%02x:0x", mt6660_dump_table[i]); + for (j = size - 1; j >= 0; j--) + seq_printf(s, "%x,", (val >> 8*j)&0xff); + seq_printf(s, "\n"); + } + return 0; +} + +static int dump_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, dump_debug_show, inode->i_private); +} + +static const struct file_operations dump_debug_fops = { + .open = dump_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static ssize_t lock_debug_read(struct file *file, + char __user *user_buf, size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + char buf[10]; + + snprintf(buf, sizeof(buf), "%d\n", mutex_is_locked(&d->io_lock)); + return simple_read_from_buffer(user_buf, cnt, loff, buf, strlen(buf)); +} + +static ssize_t lock_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + u32 lock; + int ret; + + ret = kstrtou32_from_user(user_buf, cnt, 0, &lock); + if (ret < 0) + return ret; + lock ? mutex_lock(&d->io_lock) : mutex_unlock(&d->io_lock); + return cnt; +} + +static const struct file_operations lock_debug_fops = { + .open = simple_open, + .read = lock_debug_read, + .write = lock_debug_write, +}; + +static int generic_debugfs_init(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + /* valid check */ + if (!di->dirname || !di->devname || !di->typestr) + return -EINVAL; + d->data_buffer_size = PREALLOC_RBUFFER_SIZE; + d->data_buffer = kzalloc(PREALLOC_RBUFFER_SIZE, GFP_KERNEL); + if (!d->data_buffer) + return -ENOMEM; + /* create debugfs */ + d->rt_root = debugfs_lookup("ext_dev_io", NULL); + if (!d->rt_root) { + d->rt_root = debugfs_create_dir("ext_dev_io", NULL); + if (!d->rt_root) + return -ENODEV; + d->rt_dir_create = true; + } + d->ic_root = debugfs_create_dir(di->dirname, d->rt_root); + if (!d->ic_root) + goto err_cleanup_rt; + if (!debugfs_create_u16("reg", 0644, d->ic_root, &d->reg)) + goto err_cleanup_ic; + if (!debugfs_create_u16("size", 0644, d->ic_root, &d->size)) + goto err_cleanup_ic; + if (!debugfs_create_file("data", 0644, + d->ic_root, di, &data_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("type", 0444, + d->ic_root, di, &type_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("lock", 0644, + d->ic_root, di, &lock_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("dumps", 0444, + d->ic_root, di, &dump_debug_fops)) + goto err_cleanup_ic; + mutex_init(&d->io_lock); + return 0; +err_cleanup_ic: + debugfs_remove_recursive(d->ic_root); +err_cleanup_rt: + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); + return -ENODEV; +} + +static void generic_debugfs_exit(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + mutex_destroy(&d->io_lock); + debugfs_remove_recursive(d->ic_root); + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); +} +#else +static inline int generic_debugfs_init(struct dbg_info *di) +{ + return 0; +} + +static inline void generic_debugfs_exit(struct dbg_info *di) {} +#endif /* CONFIG_DEBUG_FS */ +#endif /* GENERIC_DEBUGFS */ + +/* + * MT6660 Generic Setting make this chip work normally. + * it is tuned by Richtek RDs. + */ +static const struct codec_reg_val generic_reg_inits[] = { + { MT6660_REG_WDT_CTRL, 0x80, 0x00 }, + { MT6660_REG_SPS_CTRL, 0x01, 0x00 }, + { MT6660_REG_AUDIO_IN2_SEL, 0x1c, 0x04 }, + { MT6660_REG_RESV11, 0x0c, 0x00 }, + { MT6660_REG_RESV31, 0x03, 0x03 }, + { MT6660_REG_RESV40, 0x01, 0x00 }, + { MT6660_REG_RESV0, 0x44, 0x04 }, + { MT6660_REG_RESV19, 0xff, 0x82 }, + { MT6660_REG_RESV17, 0x7777, 0x7273 }, + { MT6660_REG_RESV16, 0x07, 0x03 }, + { MT6660_REG_DRE_CORASE, 0xe0, 0x20 }, + { MT6660_REG_ADDA_CLOCK, 0xff, 0x70 }, + { MT6660_REG_RESV21, 0xff, 0x20 }, + { MT6660_REG_DRE_THDMODE, 0xff, 0x40 }, + { MT6660_REG_RESV23, 0xffff, 0x17f8 }, + { MT6660_REG_PWM_CTRL, 0xff, 0x15 }, + { MT6660_REG_ADC_USB_MODE, 0xff, 0x00 }, + { MT6660_REG_PROTECTION_CFG, 0xff, 0x1d }, + { MT6660_REG_HPF1_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_HPF2_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_SIG_GAIN, 0xff, 0x58 }, + { MT6660_REG_RESV6, 0xff, 0xce }, + { MT6660_REG_SIGMAX, 0xffff, 0x7fff }, + { MT6660_REG_DA_GAIN, 0xffff, 0x0116 }, + { MT6660_REG_TDM_CFG3, 0x1800, 0x0800 }, + { MT6660_REG_DRE_CTRL, 0x1f, 0x07 }, +}; + +static int mt6660_component_init_setting(struct snd_soc_component *component) +{ + int i, len, ret; + const struct codec_reg_val *init_table; + + pr_info("%s start\n", __func__); + init_table = generic_reg_inits; + len = ARRAY_SIZE(generic_reg_inits); + + for (i = 0; i < len; i++) { + ret = snd_soc_component_update_bits(component, + init_table[i].addr, + init_table[i].mask, init_table[i].data); + if (ret < 0) + return ret; + } + pr_info("%s end\n", __func__); + return 0; +} + +static int mt6660_component_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + ret = regmap_read(chip->regmap, MT6660_REG_IRQ_STATUS1, &val); + dev_info(component->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + break; + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + /* Fall Through */ + case SND_SOC_BIAS_STANDBY: + default: + break; + } + dapm->bias_level = level; + dev_dbg(component->dev, "c bias_level = %d\n", level); + return 0; +} + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_PRE_PMD: + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "OUTP", NULL, "ClassD"}, + { "OUTN", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + int put_ret = 0; + + pm_runtime_get_sync(component->dev); + put_ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (put_ret < 0) + return put_ret; + pm_runtime_put(component->dev); + return put_ret; +} + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, mt6660_component_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT Switch", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Hard Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DC_Protect Switch", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("audio input selection", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF AUD IN Switch", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("AUD LOOP BACK Switch", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Mute Switch", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Bypass CS Comp Switch", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("T0 SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + pr_info("%s\n", __func__); + //snd_soc_component_init_regmap(component, chip->regmap); + + /* init chip setting */ + ret = mt6660_component_init_setting(component); + if (ret < 0) + dev_err(chip->dev, "chip i2c init setting fail\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + pr_info("%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + .read = mt6660_component_io_read, + .write = mt6660_component_io_write, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .set_bias_level = mt6660_component_set_bias_level, + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + pr_info("%s start\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + +#if GENERIC_DEBUGFS + /* debugfs interface */ + chip->dbg_info.dirname = devm_kasprintf(&client->dev, + GFP_KERNEL, "MT6660.%s", + dev_name(&client->dev)); + chip->dbg_info.devname = dev_name(&client->dev); + chip->dbg_info.typestr = devm_kasprintf(&client->dev, + GFP_KERNEL, "I2C,MT6660"); + chip->dbg_info.io_drvdata = chip; + chip->dbg_info.io_read = mt6660_dbg_io_read; + chip->dbg_info.io_write = mt6660_dbg_io_write; + + ret = generic_debugfs_init(&chip->dbg_info); + if (ret < 0) { + dev_err(&client->dev, "generic dbg init fail\n"); + return -EINVAL; + } +#endif /* GENERIC_DEBUGFS */ + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + pr_info("%s end, ret = %d\n", __func__, ret); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); +#if GENERIC_DEBUGFS + generic_debugfs_exit(&chip->dbg_info); +#endif /* GENERIC_DEBUGFS */ + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..9e62dcc --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,95 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +#define GENERIC_DEBUGFS 0 + +#if GENERIC_DEBUGFS +struct dbg_internal { + struct dentry *rt_root; + struct dentry *ic_root; + bool rt_dir_create; + struct mutex io_lock; + u16 reg; + u16 size; + u16 data_buffer_size; + void *data_buffer; +}; + +struct dbg_info { + const char *dirname; + const char *devname; + const char *typestr; + void *io_drvdata; + int (*io_read)(void *drvdata, u16 reg, void *val, u16 size); + int (*io_write)(void *drvdata, u16 reg, const void *val, u16 size); + struct dbg_internal internal; +}; +#endif /* GENERIC_DEBUGFS */ + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mutex io_lock; +#if GENERIC_DEBUGFS + struct dbg_info dbg_info; +#endif /* GENERIC_DEBUGFS */ + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

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[alsa-devel] [PATCH 0/3] ALSA: hda/ca0132 - Small fixes
by Takashi Iwai 13 Dec '19

13 Dec '19

Hi, this is a series of small fixes for CA0132 just for covering some corner cases. Takashi === Takashi Iwai (3): ALSA: hda/c0a132 - Keep power on during processing DSP response ALSA: hda/c0a132 - Avoid endless loop ALSA: hda/ca0132 - Fix work handling in delayed HP detection sound/pci/hda/patch_ca0132.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) -- 2.16.4

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Re: [alsa-devel] [PATCH v3 03/15] soc: tegra: Add Tegra PMC clock registrations into PMC driver
by Dmitry Osipenko 13 Dec '19

13 Dec '19

12.12.2019 06:54, Sowjanya Komatineni пишет: > > On 12/11/19 7:45 PM, Sowjanya Komatineni wrote: >> >> On 12/11/19 5:39 PM, Dmitry Osipenko wrote: >>> 11.12.2019 21:50, Sowjanya Komatineni пишет: >>>> On 12/10/19 5:06 PM, Sowjanya Komatineni wrote: >>>>> On 12/10/19 9:41 AM, Dmitry Osipenko wrote: >>>>>> 10.12.2019 19:53, Sowjanya Komatineni пишет: >>>>>>> On 12/9/19 3:03 PM, Sowjanya Komatineni wrote: >>>>>>>> On 12/9/19 12:46 PM, Sowjanya Komatineni wrote: >>>>>>>>> On 12/9/19 12:12 PM, Dmitry Osipenko wrote: >>>>>>>>>> 08.12.2019 00:36, Sowjanya Komatineni пишет: >>>>>>>>>>> On 12/7/19 11:59 AM, Sowjanya Komatineni wrote: >>>>>>>>>>>> On 12/7/19 8:00 AM, Dmitry Osipenko wrote: >>>>>>>>>>>>> 07.12.2019 18:53, Dmitry Osipenko пишет: >>>>>>>>>>>>>> 07.12.2019 18:47, Dmitry Osipenko пишет: >>>>>>>>>>>>>>> 07.12.2019 17:28, Dmitry Osipenko пишет: >>>>>>>>>>>>>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>>>>>>>>>>>>> Tegra210 and prior Tegra PMC has clk_out_1, clk_out_2, >>>>>>>>>>>>>>>>> clk_out_3 >>>>>>>>>>>>>>>>> with >>>>>>>>>>>>>>>>> mux and gate for each of these clocks. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Currently these PMC clocks are registered by Tegra clock >>>>>>>>>>>>>>>>> driver >>>>>>>>>>>>>>>>> using >>>>>>>>>>>>>>>>> clk_register_mux and clk_register_gate by passing PMC base >>>>>>>>>>>>>>>>> address >>>>>>>>>>>>>>>>> and register offsets and PMC programming for these clocks >>>>>>>>>>>>>>>>> happens >>>>>>>>>>>>>>>>> through direct PMC access by the clock driver. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> With this, when PMC is in secure mode any direct PMC >>>>>>>>>>>>>>>>> access >>>>>>>>>>>>>>>>> from the >>>>>>>>>>>>>>>>> non-secure world does not go through and these clocks will >>>>>>>>>>>>>>>>> not be >>>>>>>>>>>>>>>>> functional. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> This patch adds these clocks registration with PMC as a >>>>>>>>>>>>>>>>> clock >>>>>>>>>>>>>>>>> provider >>>>>>>>>>>>>>>>> for these clocks. clk_ops callback implementations for >>>>>>>>>>>>>>>>> these >>>>>>>>>>>>>>>>> clocks >>>>>>>>>>>>>>>>> uses tegra_pmc_readl and tegra_pmc_writel which >>>>>>>>>>>>>>>>> supports PMC >>>>>>>>>>>>>>>>> programming >>>>>>>>>>>>>>>>> in secure mode and non-secure mode. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Signed-off-by: Sowjanya Komatineni >>>>>>>>>>>>>>>>> <skomatineni(a)nvidia.com> >>>>>>>>>>>>>>>>> --- >>>>>>>>>>>>>>> [snip] >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> + >>>>>>>>>>>>>>>>> +static const struct clk_ops pmc_clk_gate_ops = { >>>>>>>>>>>>>>>>> + .is_enabled = pmc_clk_is_enabled, >>>>>>>>>>>>>>>>> + .enable = pmc_clk_enable, >>>>>>>>>>>>>>>>> + .disable = pmc_clk_disable, >>>>>>>>>>>>>>>>> +}; >>>>>>>>>>>>>>>> What's the benefit of separating GATE from the MUX? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> I think it could be a single clock. >>>>>>>>>>>>>>> According to TRM: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 1. GATE and MUX are separate entities. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 2. GATE is the parent of MUX (see PMC's CLK_OUT paths >>>>>>>>>>>>>>> diagram >>>>>>>>>>>>>>> in TRM). >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 3. PMC doesn't gate EXTPERIPH clock but could "force-enable" >>>>>>>>>>>>>>> it, >>>>>>>>>>>>>>> correct? >>>>>>>>>>> Was following existing clk-tegra-pmc as I am not sure of >>>>>>>>>>> reason for >>>>>>>>>>> having these clocks registered as separate mux and gate clocks. >>>>>>>>>>> >>>>>>>>>>> Yes, PMC clocks can be registered as single clock and can use >>>>>>>>>>> clk_ops >>>>>>>>>>> for set/get parent and enable/disable. >>>>>>>>>>> >>>>>>>>>>> enable/disable of PMC clocks is for force-enable to force the >>>>>>>>>>> clock to >>>>>>>>>>> run regardless of ACCEPT_REQ or INVERT_REQ. >>>>>>>>>>> >>>>>>>>>>>>>> 4. clk_m_div2/4 are internal PMC OSC dividers and thus these >>>>>>>>>>>>>> clocks >>>>>>>>>>>>>> should belong to PMC. >>>>>>>>>>>>> Also, it should be "osc" and not "clk_m". >>>>>>>>>>>> I followed the same parents as it were in existing >>>>>>>>>>>> clk-tegra-pmc >>>>>>>>>>>> driver. >>>>>>>>>>>> >>>>>>>>>>>> Yeah they are wrong and they should be from osc and not clk_m. >>>>>>>>>>>> >>>>>>>>>>>> Will fix in next version. >>>>>>>>>>>> >>>>>>>> Reg clk_m_div2/3, they are dividers at OSC pad and not really >>>>>>>> internal >>>>>>>> to PMC block. >>>>>>>> >>>>>>>> current clock driver creates clk_m_div clocks which should >>>>>>>> actually be >>>>>>>> osc_div2/osc_div4 clocks with osc as parent. >>>>>>>> >>>>>>>> There are no clk_m_div2 and clk_m_div4 from clk_m >>>>>>>> >>>>>>>> Will fix this in next version. >>>>>>>> >>>>>>>>>> Could you please describe the full EXTPERIPH clock topology and >>>>>>>>>> how the >>>>>>>>>> pinmux configuration is related to it all? >>>>>>>>>> >>>>>>>>>> What is internal to the Tegra chip and what are the external >>>>>>>>>> outputs? >>>>>>>>>> >>>>>>>>>> Is it possible to bypass PMC on T30+ for the EXTPERIPH clocks? >>>>>>>>> PMC CLK1/2/3 possible sources are OSC_DIV1, OSC_DIV2, OSC_DIV4, >>>>>>>>> EXTPERIPH from CAR. >>>>>>>>> >>>>>>>>> OSC_DIV1/2/4 are with internal dividers at the OSC Pads >>>>>>>>> >>>>>>>>> EXTPERIPH is from CAR and it has reset and enable controls >>>>>>>>> along with >>>>>>>>> clock source selections to choose one of the PLLA_OUT0, CLK_S, >>>>>>>>> PLLP_OUT0, CLK_M, PLLE_OUT0 >>>>>>>>> >>>>>>>>> So, PMC CLK1/2/4 possible parents are OSC_DIV1, OSC_DIV2, >>>>>>>>> OSC_DIV4, >>>>>>>>> EXTERN. >>>>>>>>> >>>>>>>>> >>>>>>>>> CLK1/2/3 also has Pinmux to route EXTPERIPH output on to these >>>>>>>>> pins. >>>>>>>>> >>>>>>>>> >>>>>>>>> When EXTERN output clock is selected for these PMC clocks thru >>>>>>>>> CLKx_SRC_SEL, output clock is from driver by EXTPERIPH from CAR >>>>>>>>> via >>>>>>>>> Pinmux logic or driven as per CLKx_SRC_SEL bypassing pinmux >>>>>>>>> based on >>>>>>>>> CLKx_ACCEPT_REQ bit. >>>>>>>>> >>>>>>>>> >>>>>>>>> PMC Clock control register has bit CLKx_ACCEPT_REQ >>>>>>>>> When CLKx_ACCEPT_REQ = 0, output clock driver is from by EXTPERIPH >>>>>>>>> through the pinmux >>>>>>>>> When CLKx_ACCEPT_REQ = 1, output clock is based on CLKx_SRC_SEL >>>>>>>>> bits >>>>>>>>> (OSC_DIV1/2/4 and EXTPERIPH clock bypassing the pinmux) >>>>>>>>> >>>>>>>>> FORCE_EN bit in PMC CLock control register forces the clock to run >>>>>>>>> regardless of this. >>>>>>> PMC clock gate is based on the state of CLKx_ACCEPT_REQ and FORCE_EN >>>>>>> like explained above. >>>>>>> >>>>>>> CLKx_ACCEPT_REQ is 0 default and FORCE_EN acts as gate to >>>>>>> enable/disable >>>>>>> EXTPERIPH clock output to PMC CLK_OUT_1/2/3. >>>>>> [and to enable OSC as well] >>>>>> >>>>>>> So I believe we need to register as MUX and Gate rather than as a >>>>>>> single >>>>>>> clock. Please confirm. >>>>>> 1. The force-enabling is applied to both OSC and EXTERN sources of >>>>>> PMC_CLK_OUT_x by PMC at once. >>>>>> >>>>>> 2. Both of PMC's force-enabling and OSC/EXTERN selection is internal >>>>>> to PMC. >>>>>> >>>>>> Should be better to define it as a single "pmc_clk_out_x". I don't >>>>>> see >>>>>> any good reasons for differentiating PMC's Gate from the MUX, it's a >>>>>> single hardware unit from a point of view of the rest of the system. >>>>>> >>>>>> Peter, do you have any objections? >>>>> We added fallback option for audio mclk and also added check for >>>>> assigned-clock-parents dt property in audio driver and if its not then >>>>> we do parent init configuration in audio driver. >>>>> >>>>> Current clock driver creates 2 separate clocks clk_out_1_mux and >>>>> clk_out_1 for each pmc clock in clock driver and uses extern1 as >>>>> parent to clk_out_1_mux and clk_out_1_mux is parent to clk_out_1. >>>>> >>>>> With change of registering each pmc clock as a single clock, when we >>>>> do parent init assignment in audio driver when >>>>> assigned-clock-properties are not used in DT (as we removed parent >>>>> inits for extern and clk_outs from clock driver), we should still try >>>>> to get clock based on clk_out_1_mux as parent assignment of extern1 is >>>>> for clk_out_1_mux as per existing clock tree. >>>>> >>>>> clk_out_1_mux clock retrieve will fail with this change of single >>>>> clock when any new platform device tree doesn't specify >>>>> assigned-clock-parents properties and tegra_asoc_utils_init fails. >>> You made the PMC/CaR changes before the audio changes, the clk_out_1_mux >>> won't exist for the audio driver patches. >>> >>> If you care about bisect-ability of the patches, then the clock and >>> audio changes need to be done in a single patch. But I don't think that >>> it's worthwhile. >>> >>>>> With single clock, extern1 is the parent for clk_out_1 and with >>>>> separate clocks for mux and gate, extern1 is the parent for >>>>> clk_out_1_mux. >>>> If we move to single clock now, it need one more additional fallback >>>> implementation in audio driver during parent configuration as >>>> clk_out_1_mux will not be there with single clock change and >>>> old/current >>>> kernel has it as it uses separate clocks for pmc mux and gate. >>> Why additional fallback? Additional to what? >>> >>>> Also, with single clock for both PMC mux and gate now, new DT should >>>> use >>>> extern1 as parent to CLK_OUT_1 as CLK_OUT_1_MUX will not be there old >>>> PMC dt-bindings has separate clocks for MUX (CLK_OUT_1_MUX) and gate >>>> (CLK_OUT_1) >>>> >>>> DT bindings will not be compatible b/w old and new changes if we >>>> move to >>>> Single PMC clock now. >>> Sorry, I don't understand what you're meaning by the "new changes". >>> >>>> Should we go with same separate clocks to have it compatible to avoid >>>> all this? >>>> >> The reason we added mclk fallback and also for doing parent >> configuration based on presence of assigned-clock-parents property is >> to have old dt compatible with new kernel and also to have new dt >> compatible with old kernel. >> >> So the point I was mentioning is to have new DT to work with old >> kernel, setting extern1 as parent to clk_out_1 (with single pmc clock) >> through assigned-clock-parents in DT will fail as old kernel has mux >> and gate as separate clocks and parent configuration is for mux clock >> (clk_out_1_mux) >> > Sorry never mind, with old kernel clock driver does all parent > configuration so should be ok. So no additional fallbacks are needed > except to the one we already added. > > OK, So its just that changes are slightly more to switch to single clock > compared to using separate clocks as gate clk_ops (which are needed > anyway for blink control) of clock enable and disable can't be used for > clk_out_1 enable/disable and need additional clk_enable and disable > callbacks. > > Will make changes to use single clock.. Please wait for the Peter's reply.

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[alsa-devel] WARNING: CPU: 3 PID: 1 at __flush_work.isra.47+0x22c/0x248
by Vicente Bergas 12 Dec '19

12 Dec '19

Hi, since v5.5-rc1 four equal consecutive traces appeared that seem related to rockchip sound. As i wasn't sure to whom sent the report just added everybody from ./scripts/get_maintainer.pl sound/soc/rockchip/rk3399_gru_sound.c which is the file containg one of the functions in the trace. By the way, sound works fine. After all traces, there is this message that could also be related: [ 0.625354] da7219 8-001a: Using default DAI clk names: da7219-dai-wclk, da7219-dai-bclk Regards, Vicente. [ 0.607955] ------------[ cut here ]------------ [ 0.607967] WARNING: CPU: 3 PID: 1 at __flush_work.isra.47+0x22c/0x248 [ 0.607972] CPU: 3 PID: 1 Comm: swapper/0 Not tainted 5.5.0-rc1 #1 [ 0.607973] Hardware name: Google Kevin (DT) [ 0.607977] pstate: 00000005 (nzcv daif -PAN -UAO) [ 0.607980] pc : __flush_work.isra.47+0x22c/0x248 [ 0.607982] lr : flush_delayed_work+0x34/0x58 [ 0.607984] sp : ffff80001004b950 [ 0.607985] x29: ffff80001004b950 x28: ffff8000109f47c8 [ 0.607989] x27: ffff800010b29890 x26: ffff800010acfdf0 [ 0.607992] x25: 0000000000000003 x24: 0000000000000001 [ 0.607994] x23: ffff0000f2779f00 x22: ffff800010c974f0 [ 0.607997] x21: 0000000000000000 x20: ffff800010c7ca48 [ 0.608000] x19: ffff0000f277f698 x18: 0000000000000014 [ 0.608003] x17: 000000007dad679e x16: 00000000be8a3c7e [ 0.608006] x15: 0000000072509968 x14: 0000000000000000 [ 0.608008] x13: 0000000000000000 x12: 0000000000000000 [ 0.608011] x11: 0000000000000008 x10: 0101010101010101 [ 0.608014] x9 : 0000000000000000 x8 : 7f7f7f7f7f7f7f7f [ 0.608017] x7 : ffff0000f3975800 x6 : 0080808080808080 [ 0.608019] x5 : dead000000000100 x4 : dead000000000122 [ 0.608022] x3 : 0000000000000000 x2 : 0000000000000000 [ 0.608025] x1 : 0000000000000000 x0 : ffff0000f277f698 [ 0.608028] Call trace: [ 0.608031] __flush_work.isra.47+0x22c/0x248 [ 0.608034] flush_delayed_work+0x34/0x58 [ 0.608040] soc_free_pcm_runtime.part.19+0x40/0x60 [ 0.608043] snd_soc_remove_dai_link+0x54/0x60 [ 0.608047] soc_cleanup_card_resources+0x160/0x298 [ 0.608050] snd_soc_bind_card+0x248/0x978 [ 0.608053] snd_soc_register_card+0xf0/0x108 [ 0.608057] devm_snd_soc_register_card+0x40/0x90 [ 0.608061] rockchip_sound_probe+0x210/0x2e8 [ 0.608066] platform_drv_probe+0x50/0xa0 [ 0.608070] really_probe+0xd8/0x2f0 [ 0.608073] driver_probe_device+0x54/0xe8 [ 0.608075] device_driver_attach+0x6c/0x78 [ 0.608078] __driver_attach+0x68/0xe8 [ 0.608082] bus_for_each_dev+0x60/0x98 [ 0.608085] driver_attach+0x20/0x28 [ 0.608088] bus_add_driver+0x170/0x1d0 [ 0.608091] driver_register+0x60/0x110 [ 0.608093] __platform_driver_register+0x44/0x50 [ 0.608098] rockchip_sound_driver_init+0x18/0x20 [ 0.608102] do_one_initcall+0x70/0x148 [ 0.608105] kernel_init_freeable+0x1b4/0x254 [ 0.608109] kernel_init+0x10/0xfc [ 0.608113] ret_from_fork+0x10/0x18 [ 0.608117] ---[ end trace fc1c70d7fb870a47 ]---

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[alsa-devel] [PATCH v5 00/11] soundwire: update ASoC interfaces
by Pierre-Louis Bossart 12 Dec '19

12 Dec '19

We need new fields in existing structures to a) deal with race conditions on codec probe/enumeration b) allow for multi-step ACPI scan/probe/startup on Intel plaforms c) deal with MSI issues using a single handler/threads for all audio interrupts d) deal with access to registers shared across multiple links on Intel platforms These structures for a) will be used by the SOF driver as well as codec drivers. The b) c) and d) cases are only for the Intel-specific implementation. To avoid conflicts between ASoC and Soundwire trees, these 11 patches are provided out-of-order, before the functionality enabled in these header files is added in follow-up patch series which can be applied separately in the ASoC and Soundwire trees. As discussed earlier, Vinod would need to provide an immutable tag for Mark Brown, and the integration on the ASoC side of SOF changes and new codecs drivers can proceed in parallel with SoundWire core changes. I had multiple offline discussions with Vinod/Mark/Takashi on how to proceed withe volume of SoundWire changes. Now that v5.5-rc1 is out we should go ahead with these interface changes. The next patchset "[PATCH v3 00/15] soundwire: intel: implement new ASoC interfaces" can still be reviewed but will not apply as is due to a one-line conflict. An update will be provided when Vinod applies this series to avoid noise on mailing lists. An update for the series "[PATCH v3 00/22] soundwire: code hardening and suspend-resume support" is ready but will be provided when both the interfaces changes and the implementation changes are merged. Comments and feedback welcome ~Pierre Changes since v4: One line change to remove EXPORT_SYMBOL(sdw_intel_init) Fix build issue with randconfig reported by kbuild (missing include of linux/irqreturn.h) Changes since v3: Reordered code and added kernel doc comments Added prototypes and fields needed to deal with SoundWire interrupts in a single handler/thread, following what is already done on the DSP side. Added mutex to control access to registers shared across links Added initial definitions for clock stop support on Intel platforms. Depending on power and latency requirements, different "quirks" can be supported. Changes since v2: Added new field to deal with a race condition leading to a timeout when the codec goes through a pm_runtime suspend/resume transition while the Master remains active. Clarified commit messages with detailed explanations what those race conditions are and why the changes were introduced. Reordered fields for Intel routines Added kernel-doc definitions for structures Modified the patch subjects to make the mapping between interface definition and implementation straightforward. Changes since v1 (no feedback received since October 23) additional initialization_complete utility to help codec drivers with their resume operation, waiting for the enumeration to complete is not always enough. Changes since v3: Reordered code and added kernel doc comments Added prototypes and fields needed to deal with SoundWire interrupts in a single handler/thread, following what is already done on the DSP side. Added mutex to control access to registers shared across links Added initial definitions for clock stop support on Intel platforms. Depending on power and latency requirements, different "quirks" can be supported. Changes since v2: Added new field to deal with a race condition leading to a timeout when the codec goes through a pm_runtime suspend/resume transition while the Master remains active. Clarified commit messages with detailed explanations what those race conditions are and why the changes were introduced. Reordered fields for Intel routines Added kernel-doc definitions for structures Modified the patch subjects to make the mapping between interface definition and implementation straightforward. Changes since v1 (no feedback received since October 23) additional initialization_complete utility to help codec drivers with their resume operation, waiting for the enumeration to complete is not always enough. Bard Liao (2): soundwire: intel: update headers for interrupts soundwire: intel: add link_list to handle interrupts with a single thread Pierre-Louis Bossart (7): soundwire: sdw_slave: add probe_complete structure and new fields soundwire: sdw_slave: add enumeration_complete structure soundwire: sdw_slave: add initialization_complete definition soundwire: sdw_slave: track unattach_request to handle all init sequences soundwire: intel: update interfaces between ASoC and SoundWire soundwire: intel: add mutex for shared SHIM register access soundwire: intel: add clock stop quirks Rander Wang (2): soundwire: intel: update stream callbacks for hwparams/free stream operations soundwire: intel: add prototype for WAKEEN interrupt processing drivers/soundwire/intel.c | 20 ++-- drivers/soundwire/intel.h | 13 ++- drivers/soundwire/intel_init.c | 32 ++---- include/linux/soundwire/sdw.h | 19 ++++ include/linux/soundwire/sdw_intel.h | 156 ++++++++++++++++++++++++++-- 5 files changed, 196 insertions(+), 44 deletions(-) -- 2.20.1

2 13

[alsa-devel] [RFC] topology: sof: import sof 1.4.1 topology files
by GitHub pull_request - opened 12 Dec '19

12 Dec '19

alsa-project/alsa-topology-conf pull request #1 was opened from juimonen: Import topology files from sof v1.4.1. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-topology-conf/pull/1 Patch URL : https://github.com/alsa-project/alsa-topology-conf/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-topology-conf

1 0

[alsa-devel] [PATCH] ALSA: hda/tegra: Remove unused "bus" variable
by Thierry Reding 12 Dec '19

12 Dec '19

From: Thierry Reding <treding(a)nvidia.com> After commit f36da9406e66 ("ALSA: hda: Support PCM sync_stop"), the local "bus" variable is no longer used. Remove it. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index fc2e0a294bc1..269f242fcbfd 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -166,7 +166,6 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); - struct hdac_bus *bus = azx_bus(chip); if (chip && chip->running) { azx_stop_chip(chip); -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: Add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 12 Dec '19

12 Dec '19

The MT6660 is a boosted BTL class-D amplifier with V/I sensing. A built-in DC-DC step-up converter is used to provide efficient power for class-D amplifier with multi-level class-G operation. The digital audio interface supports I2S, left-justified, right-justified, TDM and DSP A/B format for audio in with a data out used for chip information like voltage sense and current sense, which are able to be monitored via DATAO through proper register setting. --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 1063 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 95 ++++ 4 files changed, 1174 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..806cd03 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,19 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + select CRC32 + select CRYPTO_SHA256 + select CRYTO_RSA + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..92a2329 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,1063 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +#if GENERIC_DEBUGFS +static int mt6660_dbg_io_read(void *drvdata, u16 reg, void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int ret = 0, i = 0; + unsigned int data; + u8 *_val = val; + + if (size != 1 && size != 2 && size != 4) { + dev_err(chip->dev, "%s size not match\n", __func__); + return -EINVAL; + } + ret = regmap_read(chip->regmap, reg, &data); + if (reg < 0) + return ret; + + for (i = 0; i < size; i++) + _val[size-i-1] = (data >> (8*i))&0xff; + + return 0; +} + +static int mt6660_dbg_io_write(void *drvdata, u16 reg, + const void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int reg_size = mt6660_get_reg_size(reg); + int i = 0; + unsigned int regval = 0; + u8 *_val = (u8 *)val; + + if (size != reg_size) { + dev_err(chip->dev, + "%s size not match reg_size(%d), size(%d)\n", + __func__, reg_size, size); + return -EINVAL; + } + + for (i = 0; i < size; i++) { + regval <<= 8; + regval |= _val[i]; + } + + return regmap_write(chip->regmap, reg, regval); +} +#endif /* GENERIC_DEBUGFS */ + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + + return ret; +} + +static bool mt6660_volatile_reg(struct device *dev, unsigned int reg) +{ + return true; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, + .volatile_reg = mt6660_volatile_reg, +}; + +static unsigned int mt6660_component_io_read( + struct snd_soc_component *component, unsigned int reg) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret; + + ret = regmap_read(chip->regmap, reg, &val); + if (ret < 0) /* ret success -> >= 0, fail -> < - */ + return ret; + pr_err("%s val = 0x%x\n", __func__, val); + return val; +} + +static int mt6660_component_io_write(struct snd_soc_component *component, + unsigned int reg, unsigned int data) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + pr_err("%s data = 0x%x\n", __func__, data); + return regmap_write(chip->regmap, reg, data); +} + +static const int mt6660_dump_table[] = { + MT6660_REG_DEVID, + MT6660_REG_SYSTEM_CTRL, + MT6660_REG_IRQ_STATUS1, + MT6660_REG_ADDA_CLOCK, + MT6660_REG_SERIAL_CFG1, + MT6660_REG_DATAO_SEL, + MT6660_REG_TDM_CFG3, + MT6660_REG_HPF_CTRL, + MT6660_REG_HPF1_COEF, + MT6660_REG_HPF2_COEF, + MT6660_REG_PATH_BYPASS, + MT6660_REG_WDT_CTRL, + MT6660_REG_HCLIP_CTRL, + MT6660_REG_VOL_CTRL, + MT6660_REG_SPS_CTRL, + MT6660_REG_SIGMAX, + MT6660_REG_CALI_T0, + MT6660_REG_BST_CTRL, + MT6660_REG_PROTECTION_CFG, + MT6660_REG_DA_GAIN, + MT6660_REG_AUDIO_IN2_SEL, + MT6660_REG_SIG_GAIN, + MT6660_REG_PLL_CFG1, + MT6660_REG_DRE_CTRL, + MT6660_REG_DRE_THDMODE, + MT6660_REG_DRE_CORASE, + MT6660_REG_PWM_CTRL, + MT6660_REG_DC_PROTECT_CTRL, + MT6660_REG_ADC_USB_MODE, + MT6660_REG_INTERNAL_CFG, + MT6660_REG_RESV0, + MT6660_REG_RESV1, + MT6660_REG_RESV2, + MT6660_REG_RESV3, + MT6660_REG_RESV7, + MT6660_REG_RESV10, + MT6660_REG_RESV11, + MT6660_REG_RESV16, + MT6660_REG_RESV17, + MT6660_REG_RESV19, + MT6660_REG_RESV21, + MT6660_REG_RESV23, + MT6660_REG_RESV31, + MT6660_REG_RESV40, +}; + +#if GENERIC_DEBUGFS +#ifdef CONFIG_DEBUG_FS +/* reg/size/data/bustype */ +#define PREALLOC_RBUFFER_SIZE (32) +#define PREALLOC_WBUFFER_SIZE (1000) + +static int data_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + int i, ret; + + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* read transfer */ + if (!di->io_read) + return -EPERM; + ret = di->io_read(di->io_drvdata, d->reg, d->data_buffer, d->size); + if (ret < 0) + return ret; + pdata = d->data_buffer; + seq_printf(s, "0x"); + for (i = 0; i < d->size; i++) + seq_printf(s, "%02x,", *(pdata + i)); + seq_printf(s, "\n"); + return 0; +} + +static int data_debug_open(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_open(file, data_debug_show, inode->i_private); + return simple_open(inode, file); +} + +static ssize_t data_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + char buf[PREALLOC_WBUFFER_SIZE + 1], *token, *cur; + int val_cnt = 0, ret; + + if (cnt > PREALLOC_WBUFFER_SIZE) + return -ENOMEM; + if (copy_from_user(buf, user_buf, cnt)) + return -EFAULT; + buf[cnt] = 0; + /* buffer size check */ + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* data parsing */ + cur = buf; + pdata = d->data_buffer; + while ((token = strsep(&cur, ",\n")) != NULL) { + if (!*token) + break; + if (val_cnt++ >= d->size) + break; + if (kstrtou8(token, 16, pdata++)) + return -EINVAL; + } + if (val_cnt != d->size) + return -EINVAL; + /* write transfer */ + if (!di->io_write) + return -EPERM; + ret = di->io_write(di->io_drvdata, d->reg, d->data_buffer, d->size); + return (ret < 0) ? ret : cnt; +} + +static int data_debug_release(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_release(inode, file); + return 0; +} + +static const struct file_operations data_debug_fops = { + .open = data_debug_open, + .read = seq_read, + .write = data_debug_write, + .llseek = seq_lseek, + .release = data_debug_release, +}; + +static int type_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + + seq_printf(s, "%s,%s\n", di->typestr, di->devname); + return 0; +} + +static int type_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, type_debug_show, inode->i_private); +} + +static const struct file_operations type_debug_fops = { + .open = type_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static int dump_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct mt6660_chip *chip = + container_of(di, struct mt6660_chip, dbg_info); + int i = 0, j = 0, ret = 0; + unsigned int val; + int size = 0; + + if (!chip) { + pr_err("%s chip is null\n", __func__); + return -ENODEV; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_dump_table); i++) { + ret = regmap_read(chip->regmap, mt6660_dump_table[i], &val); + size = mt6660_get_reg_size(mt6660_dump_table[i]); + seq_printf(s, "reg0x%02x:0x", mt6660_dump_table[i]); + for (j = size - 1; j >= 0; j--) + seq_printf(s, "%x,", (val >> 8*j)&0xff); + seq_printf(s, "\n"); + } + return 0; +} + +static int dump_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, dump_debug_show, inode->i_private); +} + +static const struct file_operations dump_debug_fops = { + .open = dump_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static ssize_t lock_debug_read(struct file *file, + char __user *user_buf, size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + char buf[10]; + + snprintf(buf, sizeof(buf), "%d\n", mutex_is_locked(&d->io_lock)); + return simple_read_from_buffer(user_buf, cnt, loff, buf, strlen(buf)); +} + +static ssize_t lock_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + u32 lock; + int ret; + + ret = kstrtou32_from_user(user_buf, cnt, 0, &lock); + if (ret < 0) + return ret; + lock ? mutex_lock(&d->io_lock) : mutex_unlock(&d->io_lock); + return cnt; +} + +static const struct file_operations lock_debug_fops = { + .open = simple_open, + .read = lock_debug_read, + .write = lock_debug_write, +}; + +static int generic_debugfs_init(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + /* valid check */ + if (!di->dirname || !di->devname || !di->typestr) + return -EINVAL; + d->data_buffer_size = PREALLOC_RBUFFER_SIZE; + d->data_buffer = kzalloc(PREALLOC_RBUFFER_SIZE, GFP_KERNEL); + if (!d->data_buffer) + return -ENOMEM; + /* create debugfs */ + d->rt_root = debugfs_lookup("ext_dev_io", NULL); + if (!d->rt_root) { + d->rt_root = debugfs_create_dir("ext_dev_io", NULL); + if (!d->rt_root) + return -ENODEV; + d->rt_dir_create = true; + } + d->ic_root = debugfs_create_dir(di->dirname, d->rt_root); + if (!d->ic_root) + goto err_cleanup_rt; + if (!debugfs_create_u16("reg", 0644, d->ic_root, &d->reg)) + goto err_cleanup_ic; + if (!debugfs_create_u16("size", 0644, d->ic_root, &d->size)) + goto err_cleanup_ic; + if (!debugfs_create_file("data", 0644, + d->ic_root, di, &data_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("type", 0444, + d->ic_root, di, &type_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("lock", 0644, + d->ic_root, di, &lock_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("dumps", 0444, + d->ic_root, di, &dump_debug_fops)) + goto err_cleanup_ic; + mutex_init(&d->io_lock); + return 0; +err_cleanup_ic: + debugfs_remove_recursive(d->ic_root); +err_cleanup_rt: + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); + return -ENODEV; +} + +static void generic_debugfs_exit(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + mutex_destroy(&d->io_lock); + debugfs_remove_recursive(d->ic_root); + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); +} +#else +static inline int generic_debugfs_init(struct dbg_info *di) +{ + return 0; +} + +static inline void generic_debugfs_exit(struct dbg_info *di) {} +#endif /* CONFIG_DEBUG_FS */ +#endif /* GENERIC_DEBUGFS */ + +/* + * MT6660 Generic Setting make this chip work normally. + * it is tuned by Richtek RDs. + */ +static const struct codec_reg_val generic_reg_inits[] = { + { MT6660_REG_WDT_CTRL, 0x80, 0x00 }, + { MT6660_REG_SPS_CTRL, 0x01, 0x00 }, + { MT6660_REG_AUDIO_IN2_SEL, 0x1c, 0x04 }, + { MT6660_REG_RESV11, 0x0c, 0x00 }, + { MT6660_REG_RESV31, 0x03, 0x03 }, + { MT6660_REG_RESV40, 0x01, 0x00 }, + { MT6660_REG_RESV0, 0x44, 0x04 }, + { MT6660_REG_RESV19, 0xff, 0x82 }, + { MT6660_REG_RESV17, 0x7777, 0x7273 }, + { MT6660_REG_RESV16, 0x07, 0x03 }, + { MT6660_REG_DRE_CORASE, 0xe0, 0x20 }, + { MT6660_REG_ADDA_CLOCK, 0xff, 0x70 }, + { MT6660_REG_RESV21, 0xff, 0x20 }, + { MT6660_REG_DRE_THDMODE, 0xff, 0x40 }, + { MT6660_REG_RESV23, 0xffff, 0x17f8 }, + { MT6660_REG_PWM_CTRL, 0xff, 0x15 }, + { MT6660_REG_ADC_USB_MODE, 0xff, 0x00 }, + { MT6660_REG_PROTECTION_CFG, 0xff, 0x1d }, + { MT6660_REG_HPF1_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_HPF2_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_SIG_GAIN, 0xff, 0x58 }, + { MT6660_REG_RESV6, 0xff, 0xce }, + { MT6660_REG_SIGMAX, 0xffff, 0x7fff }, + { MT6660_REG_DA_GAIN, 0xffff, 0x0116 }, + { MT6660_REG_TDM_CFG3, 0x1800, 0x0800 }, + { MT6660_REG_DRE_CTRL, 0x1f, 0x07 }, +}; + +static int mt6660_component_init_setting(struct snd_soc_component *component) +{ + int i, len, ret; + const struct codec_reg_val *init_table; + + pr_info("%s start\n", __func__); + init_table = generic_reg_inits; + len = ARRAY_SIZE(generic_reg_inits); + + for (i = 0; i < len; i++) { + ret = snd_soc_component_update_bits(component, + init_table[i].addr, + init_table[i].mask, init_table[i].data); + if (ret < 0) + return ret; + } + pr_info("%s end\n", __func__); + return 0; +} + +static int mt6660_component_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + ret = regmap_read(chip->regmap, MT6660_REG_IRQ_STATUS1, &val); + dev_info(component->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + break; + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + /* Fall Through */ + case SND_SOC_BIAS_STANDBY: + default: + break; + } + dapm->bias_level = level; + dev_dbg(component->dev, "c bias_level = %d\n", level); + return 0; +} + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_PRE_PMD: + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "OUTP", NULL, "ClassD"}, + { "OUTN", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + int put_ret = 0; + + pm_runtime_get_sync(component->dev); + put_ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (put_ret < 0) + return put_ret; + pm_runtime_put(component->dev); + return put_ret; +} + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, mt6660_component_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT Switch", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Hard_Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DC_Protect Switch", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("audio input selection", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF_AUD_IN Switch", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("AUD_LOOP_BACK Switch", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Mute Switch", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Bypass CS Comp Switch", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("T0_SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip_Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + pr_info("%s\n", __func__); + //snd_soc_component_init_regmap(component, chip->regmap); + + /* init chip setting */ + ret = mt6660_component_init_setting(component); + if (ret < 0) + dev_err(chip->dev, "chip i2c init setting fail\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + pr_info("%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + .read = mt6660_component_io_read, + .write = mt6660_component_io_write, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .set_bias_level = mt6660_component_set_bias_level, + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + pr_info("%s start\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + +#if GENERIC_DEBUGFS + /* debugfs interface */ + chip->dbg_info.dirname = devm_kasprintf(&client->dev, + GFP_KERNEL, "MT6660.%s", + dev_name(&client->dev)); + chip->dbg_info.devname = dev_name(&client->dev); + chip->dbg_info.typestr = devm_kasprintf(&client->dev, + GFP_KERNEL, "I2C,MT6660"); + chip->dbg_info.io_drvdata = chip; + chip->dbg_info.io_read = mt6660_dbg_io_read; + chip->dbg_info.io_write = mt6660_dbg_io_write; + + ret = generic_debugfs_init(&chip->dbg_info); + if (ret < 0) { + dev_err(&client->dev, "generic dbg init fail\n"); + return -EINVAL; + } +#endif /* GENERIC_DEBUGFS */ + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + pr_info("%s end, ret = %d\n", __func__, ret); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); +#if GENERIC_DEBUGFS + generic_debugfs_exit(&chip->dbg_info); +#endif /* GENERIC_DEBUGFS */ + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..9e62dcc --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,95 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +#define GENERIC_DEBUGFS 0 + +#if GENERIC_DEBUGFS +struct dbg_internal { + struct dentry *rt_root; + struct dentry *ic_root; + bool rt_dir_create; + struct mutex io_lock; + u16 reg; + u16 size; + u16 data_buffer_size; + void *data_buffer; +}; + +struct dbg_info { + const char *dirname; + const char *devname; + const char *typestr; + void *io_drvdata; + int (*io_read)(void *drvdata, u16 reg, void *val, u16 size); + int (*io_write)(void *drvdata, u16 reg, const void *val, u16 size); + struct dbg_internal internal; +}; +#endif /* GENERIC_DEBUGFS */ + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mutex io_lock; +#if GENERIC_DEBUGFS + struct dbg_info dbg_info; +#endif /* GENERIC_DEBUGFS */ + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

3 2

[alsa-devel] [PATCH for-5.6 0/4] staging: ALSA PCM API updates
by Takashi Iwai 12 Dec '19

12 Dec '19

Hi, this is a patch set to adapt the latest ALSA PCM API to staging drivers. Basically these are merely cleanups, as shown in diffstat, and there should be no functional changes. As the corresponding ALSA PCM API change is found in 5.5-rc1, please apply these on 5.5-rc1 or later. Or if you prefer, I can merge them through sound tree, too. Let me know. thanks, Takashi === Takashi Iwai (4): staging: most: Use managed buffer allocation staging: bcm2835-audio: Use managed buffer allocation staging: most: Drop superfluous ioctl PCM ops staging: bcm2835-audio: Drop superfluous ioctl PCM ops drivers/staging/most/sound/sound.c | 45 +--------------------- .../vc04_services/bcm2835-audio/bcm2835-pcm.c | 19 +-------- 2 files changed, 2 insertions(+), 62 deletions(-) -- 2.16.4

3 7

[alsa-devel] [PATCH] ASoC: add MediaTek MT6660 Speaker Amp Driver
by Jeff Chang 12 Dec '19

12 Dec '19

--- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 1063 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 95 ++++ 4 files changed, 1174 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..806cd03 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,19 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + select CRC32 + select CRYPTO_SHA256 + select CRYTO_RSA + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..92a2329 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,1063 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +#if GENERIC_DEBUGFS +static int mt6660_dbg_io_read(void *drvdata, u16 reg, void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int ret = 0, i = 0; + unsigned int data; + u8 *_val = val; + + if (size != 1 && size != 2 && size != 4) { + dev_err(chip->dev, "%s size not match\n", __func__); + return -EINVAL; + } + ret = regmap_read(chip->regmap, reg, &data); + if (reg < 0) + return ret; + + for (i = 0; i < size; i++) + _val[size-i-1] = (data >> (8*i))&0xff; + + return 0; +} + +static int mt6660_dbg_io_write(void *drvdata, u16 reg, + const void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int reg_size = mt6660_get_reg_size(reg); + int i = 0; + unsigned int regval = 0; + u8 *_val = (u8 *)val; + + if (size != reg_size) { + dev_err(chip->dev, + "%s size not match reg_size(%d), size(%d)\n", + __func__, reg_size, size); + return -EINVAL; + } + + for (i = 0; i < size; i++) { + regval <<= 8; + regval |= _val[i]; + } + + return regmap_write(chip->regmap, reg, regval); +} +#endif /* GENERIC_DEBUGFS */ + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + + return ret; +} + +static bool mt6660_volatile_reg(struct device *dev, unsigned int reg) +{ + return true; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, + .volatile_reg = mt6660_volatile_reg, +}; + +static unsigned int mt6660_component_io_read( + struct snd_soc_component *component, unsigned int reg) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret; + + ret = regmap_read(chip->regmap, reg, &val); + if (ret < 0) /* ret success -> >= 0, fail -> < - */ + return ret; + pr_err("%s val = 0x%x\n", __func__, val); + return val; +} + +static int mt6660_component_io_write(struct snd_soc_component *component, + unsigned int reg, unsigned int data) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + pr_err("%s data = 0x%x\n", __func__, data); + return regmap_write(chip->regmap, reg, data); +} + +static const int mt6660_dump_table[] = { + MT6660_REG_DEVID, + MT6660_REG_SYSTEM_CTRL, + MT6660_REG_IRQ_STATUS1, + MT6660_REG_ADDA_CLOCK, + MT6660_REG_SERIAL_CFG1, + MT6660_REG_DATAO_SEL, + MT6660_REG_TDM_CFG3, + MT6660_REG_HPF_CTRL, + MT6660_REG_HPF1_COEF, + MT6660_REG_HPF2_COEF, + MT6660_REG_PATH_BYPASS, + MT6660_REG_WDT_CTRL, + MT6660_REG_HCLIP_CTRL, + MT6660_REG_VOL_CTRL, + MT6660_REG_SPS_CTRL, + MT6660_REG_SIGMAX, + MT6660_REG_CALI_T0, + MT6660_REG_BST_CTRL, + MT6660_REG_PROTECTION_CFG, + MT6660_REG_DA_GAIN, + MT6660_REG_AUDIO_IN2_SEL, + MT6660_REG_SIG_GAIN, + MT6660_REG_PLL_CFG1, + MT6660_REG_DRE_CTRL, + MT6660_REG_DRE_THDMODE, + MT6660_REG_DRE_CORASE, + MT6660_REG_PWM_CTRL, + MT6660_REG_DC_PROTECT_CTRL, + MT6660_REG_ADC_USB_MODE, + MT6660_REG_INTERNAL_CFG, + MT6660_REG_RESV0, + MT6660_REG_RESV1, + MT6660_REG_RESV2, + MT6660_REG_RESV3, + MT6660_REG_RESV7, + MT6660_REG_RESV10, + MT6660_REG_RESV11, + MT6660_REG_RESV16, + MT6660_REG_RESV17, + MT6660_REG_RESV19, + MT6660_REG_RESV21, + MT6660_REG_RESV23, + MT6660_REG_RESV31, + MT6660_REG_RESV40, +}; + +#if GENERIC_DEBUGFS +#ifdef CONFIG_DEBUG_FS +/* reg/size/data/bustype */ +#define PREALLOC_RBUFFER_SIZE (32) +#define PREALLOC_WBUFFER_SIZE (1000) + +static int data_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + int i, ret; + + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* read transfer */ + if (!di->io_read) + return -EPERM; + ret = di->io_read(di->io_drvdata, d->reg, d->data_buffer, d->size); + if (ret < 0) + return ret; + pdata = d->data_buffer; + seq_printf(s, "0x"); + for (i = 0; i < d->size; i++) + seq_printf(s, "%02x,", *(pdata + i)); + seq_printf(s, "\n"); + return 0; +} + +static int data_debug_open(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_open(file, data_debug_show, inode->i_private); + return simple_open(inode, file); +} + +static ssize_t data_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + char buf[PREALLOC_WBUFFER_SIZE + 1], *token, *cur; + int val_cnt = 0, ret; + + if (cnt > PREALLOC_WBUFFER_SIZE) + return -ENOMEM; + if (copy_from_user(buf, user_buf, cnt)) + return -EFAULT; + buf[cnt] = 0; + /* buffer size check */ + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* data parsing */ + cur = buf; + pdata = d->data_buffer; + while ((token = strsep(&cur, ",\n")) != NULL) { + if (!*token) + break; + if (val_cnt++ >= d->size) + break; + if (kstrtou8(token, 16, pdata++)) + return -EINVAL; + } + if (val_cnt != d->size) + return -EINVAL; + /* write transfer */ + if (!di->io_write) + return -EPERM; + ret = di->io_write(di->io_drvdata, d->reg, d->data_buffer, d->size); + return (ret < 0) ? ret : cnt; +} + +static int data_debug_release(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_release(inode, file); + return 0; +} + +static const struct file_operations data_debug_fops = { + .open = data_debug_open, + .read = seq_read, + .write = data_debug_write, + .llseek = seq_lseek, + .release = data_debug_release, +}; + +static int type_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + + seq_printf(s, "%s,%s\n", di->typestr, di->devname); + return 0; +} + +static int type_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, type_debug_show, inode->i_private); +} + +static const struct file_operations type_debug_fops = { + .open = type_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static int dump_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct mt6660_chip *chip = + container_of(di, struct mt6660_chip, dbg_info); + int i = 0, j = 0, ret = 0; + unsigned int val; + int size = 0; + + if (!chip) { + pr_err("%s chip is null\n", __func__); + return -ENODEV; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_dump_table); i++) { + ret = regmap_read(chip->regmap, mt6660_dump_table[i], &val); + size = mt6660_get_reg_size(mt6660_dump_table[i]); + seq_printf(s, "reg0x%02x:0x", mt6660_dump_table[i]); + for (j = size - 1; j >= 0; j--) + seq_printf(s, "%x,", (val >> 8*j)&0xff); + seq_printf(s, "\n"); + } + return 0; +} + +static int dump_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, dump_debug_show, inode->i_private); +} + +static const struct file_operations dump_debug_fops = { + .open = dump_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static ssize_t lock_debug_read(struct file *file, + char __user *user_buf, size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + char buf[10]; + + snprintf(buf, sizeof(buf), "%d\n", mutex_is_locked(&d->io_lock)); + return simple_read_from_buffer(user_buf, cnt, loff, buf, strlen(buf)); +} + +static ssize_t lock_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + u32 lock; + int ret; + + ret = kstrtou32_from_user(user_buf, cnt, 0, &lock); + if (ret < 0) + return ret; + lock ? mutex_lock(&d->io_lock) : mutex_unlock(&d->io_lock); + return cnt; +} + +static const struct file_operations lock_debug_fops = { + .open = simple_open, + .read = lock_debug_read, + .write = lock_debug_write, +}; + +static int generic_debugfs_init(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + /* valid check */ + if (!di->dirname || !di->devname || !di->typestr) + return -EINVAL; + d->data_buffer_size = PREALLOC_RBUFFER_SIZE; + d->data_buffer = kzalloc(PREALLOC_RBUFFER_SIZE, GFP_KERNEL); + if (!d->data_buffer) + return -ENOMEM; + /* create debugfs */ + d->rt_root = debugfs_lookup("ext_dev_io", NULL); + if (!d->rt_root) { + d->rt_root = debugfs_create_dir("ext_dev_io", NULL); + if (!d->rt_root) + return -ENODEV; + d->rt_dir_create = true; + } + d->ic_root = debugfs_create_dir(di->dirname, d->rt_root); + if (!d->ic_root) + goto err_cleanup_rt; + if (!debugfs_create_u16("reg", 0644, d->ic_root, &d->reg)) + goto err_cleanup_ic; + if (!debugfs_create_u16("size", 0644, d->ic_root, &d->size)) + goto err_cleanup_ic; + if (!debugfs_create_file("data", 0644, + d->ic_root, di, &data_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("type", 0444, + d->ic_root, di, &type_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("lock", 0644, + d->ic_root, di, &lock_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("dumps", 0444, + d->ic_root, di, &dump_debug_fops)) + goto err_cleanup_ic; + mutex_init(&d->io_lock); + return 0; +err_cleanup_ic: + debugfs_remove_recursive(d->ic_root); +err_cleanup_rt: + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); + return -ENODEV; +} + +static void generic_debugfs_exit(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + mutex_destroy(&d->io_lock); + debugfs_remove_recursive(d->ic_root); + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); +} +#else +static inline int generic_debugfs_init(struct dbg_info *di) +{ + return 0; +} + +static inline void generic_debugfs_exit(struct dbg_info *di) {} +#endif /* CONFIG_DEBUG_FS */ +#endif /* GENERIC_DEBUGFS */ + +/* + * MT6660 Generic Setting make this chip work normally. + * it is tuned by Richtek RDs. + */ +static const struct codec_reg_val generic_reg_inits[] = { + { MT6660_REG_WDT_CTRL, 0x80, 0x00 }, + { MT6660_REG_SPS_CTRL, 0x01, 0x00 }, + { MT6660_REG_AUDIO_IN2_SEL, 0x1c, 0x04 }, + { MT6660_REG_RESV11, 0x0c, 0x00 }, + { MT6660_REG_RESV31, 0x03, 0x03 }, + { MT6660_REG_RESV40, 0x01, 0x00 }, + { MT6660_REG_RESV0, 0x44, 0x04 }, + { MT6660_REG_RESV19, 0xff, 0x82 }, + { MT6660_REG_RESV17, 0x7777, 0x7273 }, + { MT6660_REG_RESV16, 0x07, 0x03 }, + { MT6660_REG_DRE_CORASE, 0xe0, 0x20 }, + { MT6660_REG_ADDA_CLOCK, 0xff, 0x70 }, + { MT6660_REG_RESV21, 0xff, 0x20 }, + { MT6660_REG_DRE_THDMODE, 0xff, 0x40 }, + { MT6660_REG_RESV23, 0xffff, 0x17f8 }, + { MT6660_REG_PWM_CTRL, 0xff, 0x15 }, + { MT6660_REG_ADC_USB_MODE, 0xff, 0x00 }, + { MT6660_REG_PROTECTION_CFG, 0xff, 0x1d }, + { MT6660_REG_HPF1_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_HPF2_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_SIG_GAIN, 0xff, 0x58 }, + { MT6660_REG_RESV6, 0xff, 0xce }, + { MT6660_REG_SIGMAX, 0xffff, 0x7fff }, + { MT6660_REG_DA_GAIN, 0xffff, 0x0116 }, + { MT6660_REG_TDM_CFG3, 0x1800, 0x0800 }, + { MT6660_REG_DRE_CTRL, 0x1f, 0x07 }, +}; + +static int mt6660_component_init_setting(struct snd_soc_component *component) +{ + int i, len, ret; + const struct codec_reg_val *init_table; + + pr_info("%s start\n", __func__); + init_table = generic_reg_inits; + len = ARRAY_SIZE(generic_reg_inits); + + for (i = 0; i < len; i++) { + ret = snd_soc_component_update_bits(component, + init_table[i].addr, + init_table[i].mask, init_table[i].data); + if (ret < 0) + return ret; + } + pr_info("%s end\n", __func__); + return 0; +} + +static int mt6660_component_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + ret = regmap_read(chip->regmap, MT6660_REG_IRQ_STATUS1, &val); + dev_info(component->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + break; + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + /* Fall Through */ + case SND_SOC_BIAS_STANDBY: + default: + break; + } + dapm->bias_level = level; + dev_dbg(component->dev, "c bias_level = %d\n", level); + return 0; +} + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_PRE_PMD: + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "OUTP", NULL, "ClassD"}, + { "OUTN", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + int put_ret = 0; + + pm_runtime_get_sync(component->dev); + put_ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (put_ret < 0) + return put_ret; + pm_runtime_put(component->dev); + return put_ret; +} + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, mt6660_component_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT Switch", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Hard_Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DC_Protect Switch", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("audio input selection", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF_AUD_IN Switch", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("AUD_LOOP_BACK Switch", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Mute Switch", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Bypass CS Comp Switch", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("T0_SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip_Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + pr_info("%s\n", __func__); + //snd_soc_component_init_regmap(component, chip->regmap); + + /* init chip setting */ + ret = mt6660_component_init_setting(component); + if (ret < 0) + dev_err(chip->dev, "chip i2c init setting fail\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + pr_info("%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + .read = mt6660_component_io_read, + .write = mt6660_component_io_write, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .set_bias_level = mt6660_component_set_bias_level, + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + pr_info("%s start\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + +#if GENERIC_DEBUGFS + /* debugfs interface */ + chip->dbg_info.dirname = devm_kasprintf(&client->dev, + GFP_KERNEL, "MT6660.%s", + dev_name(&client->dev)); + chip->dbg_info.devname = dev_name(&client->dev); + chip->dbg_info.typestr = devm_kasprintf(&client->dev, + GFP_KERNEL, "I2C,MT6660"); + chip->dbg_info.io_drvdata = chip; + chip->dbg_info.io_read = mt6660_dbg_io_read; + chip->dbg_info.io_write = mt6660_dbg_io_write; + + ret = generic_debugfs_init(&chip->dbg_info); + if (ret < 0) { + dev_err(&client->dev, "generic dbg init fail\n"); + return -EINVAL; + } +#endif /* GENERIC_DEBUGFS */ + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + pr_info("%s end, ret = %d\n", __func__, ret); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); +#if GENERIC_DEBUGFS + generic_debugfs_exit(&chip->dbg_info); +#endif /* GENERIC_DEBUGFS */ + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..9e62dcc --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,95 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +#define GENERIC_DEBUGFS 0 + +#if GENERIC_DEBUGFS +struct dbg_internal { + struct dentry *rt_root; + struct dentry *ic_root; + bool rt_dir_create; + struct mutex io_lock; + u16 reg; + u16 size; + u16 data_buffer_size; + void *data_buffer; +}; + +struct dbg_info { + const char *dirname; + const char *devname; + const char *typestr; + void *io_drvdata; + int (*io_read)(void *drvdata, u16 reg, void *val, u16 size); + int (*io_write)(void *drvdata, u16 reg, const void *val, u16 size); + struct dbg_internal internal; +}; +#endif /* GENERIC_DEBUGFS */ + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mutex io_lock; +#if GENERIC_DEBUGFS + struct dbg_info dbg_info; +#endif /* GENERIC_DEBUGFS */ + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda: tegra: Fix unused variable compile warning
by Takashi Iwai 12 Dec '19

12 Dec '19

Forgot to remove the variable declaration as well in the last commit. sound/pci/hda/hda_tegra.c: In function 'hda_tegra_runtime_suspend': sound/pci/hda/hda_tegra.c:169:19: warning: unused variable 'bus' [-Wunused-variable] Fixes: f36da9406e66 ("ALSA: hda: Support PCM sync_stop") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_tegra.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index fc2e0a294bc1..269f242fcbfd 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -166,7 +166,6 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); - struct hdac_bus *bus = azx_bus(chip); if (chip && chip->running) { azx_stop_chip(chip); -- 2.16.4

1 0

[alsa-devel] [PATCH] ASoC: add MediaTek MT6660 Speaker Amp Driver.
by Jeff Chang 12 Dec '19

12 Dec '19

--- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 1065 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 95 ++++ 4 files changed, 1176 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89..806cd03 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1465,6 +1466,19 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "Mediatek MT6660 Speaker Amplifier" + depends on I2C + select CRC32 + select CRYPTO_SHA256 + select CRYTO_RSA + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373..2b6814c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -119,6 +119,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -403,6 +404,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..9e9d496 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,1065 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +#if GENERIC_DEBUGFS +static int mt6660_dbg_io_read(void *drvdata, u16 reg, void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int ret = 0, i = 0; + unsigned int data; + u8 *_val = val; + + if (size != 1 && size != 2 && size != 4) { + dev_err(chip->dev, "%s size not match\n", __func__); + return -EINVAL; + } + ret = regmap_read(chip->regmap, reg, &data); + if (reg < 0) + return ret; + + for (i = 0; i < size; i++) + _val[size-i-1] = (data >> (8*i))&0xff; + + return 0; +} + +static int mt6660_dbg_io_write(void *drvdata, u16 reg, + const void *val, u16 size) +{ + struct mt6660_chip *chip = (struct mt6660_chip *)drvdata; + int reg_size = mt6660_get_reg_size(reg); + int i = 0; + unsigned int regval = 0; + u8 *_val = (u8 *)val; + + if (size != reg_size) { + dev_err(chip->dev, + "%s size not match reg_size(%d), size(%d)\n", + __func__, reg_size, size); + return -EINVAL; + } + + for (i = 0; i < size; i++) { + regval <<= 8; + regval |= _val[i]; + } + + return regmap_write(chip->regmap, reg, regval); +} +#endif /* GENERIC_DEBUGFS */ + +static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; + + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); +} + +static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct mt6660_chip *chip = context; + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + *val = reg_data; + + return ret; +} + +static bool mt6660_volatile_reg(struct device *dev, unsigned int reg) +{ + return true; +} + +static struct regmap_config mt6660_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .reg_write = mt6660_reg_write, + .reg_read = mt6660_reg_read, + .volatile_reg = mt6660_volatile_reg, +}; + +static unsigned int mt6660_component_io_read( + struct snd_soc_component *component, unsigned int reg) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret; + + ret = regmap_read(chip->regmap, reg, &val); + if (ret < 0) /* ret success -> >= 0, fail -> < - */ + return ret; + pr_err("%s val = 0x%x\n", __func__, val); + return val; +} + +static int mt6660_component_io_write(struct snd_soc_component *component, + unsigned int reg, unsigned int data) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + pr_err("%s data = 0x%x\n", __func__, data); + return regmap_write(chip->regmap, reg, data); +} + +static const int mt6660_dump_table[] = { + MT6660_REG_DEVID, + MT6660_REG_SYSTEM_CTRL, + MT6660_REG_IRQ_STATUS1, + MT6660_REG_ADDA_CLOCK, + MT6660_REG_SERIAL_CFG1, + MT6660_REG_DATAO_SEL, + MT6660_REG_TDM_CFG3, + MT6660_REG_HPF_CTRL, + MT6660_REG_HPF1_COEF, + MT6660_REG_HPF2_COEF, + MT6660_REG_PATH_BYPASS, + MT6660_REG_WDT_CTRL, + MT6660_REG_HCLIP_CTRL, + MT6660_REG_VOL_CTRL, + MT6660_REG_SPS_CTRL, + MT6660_REG_SIGMAX, + MT6660_REG_CALI_T0, + MT6660_REG_BST_CTRL, + MT6660_REG_PROTECTION_CFG, + MT6660_REG_DA_GAIN, + MT6660_REG_AUDIO_IN2_SEL, + MT6660_REG_SIG_GAIN, + MT6660_REG_PLL_CFG1, + MT6660_REG_DRE_CTRL, + MT6660_REG_DRE_THDMODE, + MT6660_REG_DRE_CORASE, + MT6660_REG_PWM_CTRL, + MT6660_REG_DC_PROTECT_CTRL, + MT6660_REG_ADC_USB_MODE, + MT6660_REG_INTERNAL_CFG, + MT6660_REG_RESV0, + MT6660_REG_RESV1, + MT6660_REG_RESV2, + MT6660_REG_RESV3, + MT6660_REG_RESV7, + MT6660_REG_RESV10, + MT6660_REG_RESV11, + MT6660_REG_RESV16, + MT6660_REG_RESV17, + MT6660_REG_RESV19, + MT6660_REG_RESV21, + MT6660_REG_RESV23, + MT6660_REG_RESV31, + MT6660_REG_RESV40, +}; + +#if GENERIC_DEBUGFS +#ifdef CONFIG_DEBUG_FS +/* reg/size/data/bustype */ +#define PREALLOC_RBUFFER_SIZE (32) +#define PREALLOC_WBUFFER_SIZE (1000) + +static int data_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + int i, ret; + + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* read transfer */ + if (!di->io_read) + return -EPERM; + ret = di->io_read(di->io_drvdata, d->reg, d->data_buffer, d->size); + if (ret < 0) + return ret; + pdata = d->data_buffer; + seq_printf(s, "0x"); + for (i = 0; i < d->size; i++) + seq_printf(s, "%02x,", *(pdata + i)); + seq_printf(s, "\n"); + return 0; +} + +static int data_debug_open(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_open(file, data_debug_show, inode->i_private); + return simple_open(inode, file); +} + +static ssize_t data_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + void *buffer; + u8 *pdata; + char buf[PREALLOC_WBUFFER_SIZE + 1], *token, *cur; + int val_cnt = 0, ret; + + if (cnt > PREALLOC_WBUFFER_SIZE) + return -ENOMEM; + if (copy_from_user(buf, user_buf, cnt)) + return -EFAULT; + buf[cnt] = 0; + /* buffer size check */ + if (d->data_buffer_size < d->size) { + buffer = kzalloc(d->size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + kfree(d->data_buffer); + d->data_buffer = buffer; + d->data_buffer_size = d->size; + } + /* data parsing */ + cur = buf; + pdata = d->data_buffer; + while ((token = strsep(&cur, ",\n")) != NULL) { + if (!*token) + break; + if (val_cnt++ >= d->size) + break; + if (kstrtou8(token, 16, pdata++)) + return -EINVAL; + } + if (val_cnt != d->size) + return -EINVAL; + /* write transfer */ + if (!di->io_write) + return -EPERM; + ret = di->io_write(di->io_drvdata, d->reg, d->data_buffer, d->size); + return (ret < 0) ? ret : cnt; +} + +static int data_debug_release(struct inode *inode, struct file *file) +{ + if (file->f_mode & FMODE_READ) + return single_release(inode, file); + return 0; +} + +static const struct file_operations data_debug_fops = { + .open = data_debug_open, + .read = seq_read, + .write = data_debug_write, + .llseek = seq_lseek, + .release = data_debug_release, +}; + +static int type_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + + seq_printf(s, "%s,%s\n", di->typestr, di->devname); + return 0; +} + +static int type_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, type_debug_show, inode->i_private); +} + +static const struct file_operations type_debug_fops = { + .open = type_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static int dump_debug_show(struct seq_file *s, void *data) +{ + struct dbg_info *di = s->private; + struct mt6660_chip *chip = + container_of(di, struct mt6660_chip, dbg_info); + int i = 0, j = 0, ret = 0; + unsigned int val; + int size = 0; + + if (!chip) { + pr_err("%s chip is null\n", __func__); + return -ENODEV; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_dump_table); i++) { + ret = regmap_read(chip->regmap, mt6660_dump_table[i], &val); + size = mt6660_get_reg_size(mt6660_dump_table[i]); + seq_printf(s, "reg0x%02x:0x", mt6660_dump_table[i]); + for (j = size - 1; j >= 0; j--) + seq_printf(s, "%x,", (val >> 8*j)&0xff); + seq_printf(s, "\n"); + } + return 0; +} + +static int dump_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, dump_debug_show, inode->i_private); +} + +static const struct file_operations dump_debug_fops = { + .open = dump_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static ssize_t lock_debug_read(struct file *file, + char __user *user_buf, size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + char buf[10]; + + snprintf(buf, sizeof(buf), "%d\n", mutex_is_locked(&d->io_lock)); + return simple_read_from_buffer(user_buf, cnt, loff, buf, strlen(buf)); +} + +static ssize_t lock_debug_write(struct file *file, + const char __user *user_buf, + size_t cnt, loff_t *loff) +{ + struct dbg_info *di = file->private_data; + struct dbg_internal *d = &di->internal; + u32 lock; + int ret; + + ret = kstrtou32_from_user(user_buf, cnt, 0, &lock); + if (ret < 0) + return ret; + lock ? mutex_lock(&d->io_lock) : mutex_unlock(&d->io_lock); + return cnt; +} + +static const struct file_operations lock_debug_fops = { + .open = simple_open, + .read = lock_debug_read, + .write = lock_debug_write, +}; + +static int generic_debugfs_init(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + /* valid check */ + if (!di->dirname || !di->devname || !di->typestr) + return -EINVAL; + d->data_buffer_size = PREALLOC_RBUFFER_SIZE; + d->data_buffer = kzalloc(PREALLOC_RBUFFER_SIZE, GFP_KERNEL); + if (!d->data_buffer) + return -ENOMEM; + /* create debugfs */ + d->rt_root = debugfs_lookup("ext_dev_io", NULL); + if (!d->rt_root) { + d->rt_root = debugfs_create_dir("ext_dev_io", NULL); + if (!d->rt_root) + return -ENODEV; + d->rt_dir_create = true; + } + d->ic_root = debugfs_create_dir(di->dirname, d->rt_root); + if (!d->ic_root) + goto err_cleanup_rt; + if (!debugfs_create_u16("reg", 0644, d->ic_root, &d->reg)) + goto err_cleanup_ic; + if (!debugfs_create_u16("size", 0644, d->ic_root, &d->size)) + goto err_cleanup_ic; + if (!debugfs_create_file("data", 0644, + d->ic_root, di, &data_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("type", 0444, + d->ic_root, di, &type_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("lock", 0644, + d->ic_root, di, &lock_debug_fops)) + goto err_cleanup_ic; + if (!debugfs_create_file("dumps", 0444, + d->ic_root, di, &dump_debug_fops)) + goto err_cleanup_ic; + mutex_init(&d->io_lock); + return 0; +err_cleanup_ic: + debugfs_remove_recursive(d->ic_root); +err_cleanup_rt: + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); + return -ENODEV; +} + +static void generic_debugfs_exit(struct dbg_info *di) +{ + struct dbg_internal *d = &di->internal; + + mutex_destroy(&d->io_lock); + debugfs_remove_recursive(d->ic_root); + if (d->rt_dir_create) + debugfs_remove_recursive(d->rt_root); + kfree(d->data_buffer); +} +#else +static inline int generic_debugfs_init(struct dbg_info *di) +{ + return 0; +} + +static inline void generic_debugfs_exit(struct dbg_info *di) {} +#endif /* CONFIG_DEBUG_FS */ +#endif /* GENERIC_DEBUGFS */ + +/* + * MT6660 Generic Setting make this chip work normally. + * it is tuned by Richtek RDs. + */ +static const struct codec_reg_val generic_reg_inits[] = { + { MT6660_REG_WDT_CTRL, 0x80, 0x00 }, + { MT6660_REG_SPS_CTRL, 0x01, 0x00 }, + { MT6660_REG_AUDIO_IN2_SEL, 0x1c, 0x04 }, + { MT6660_REG_RESV11, 0x0c, 0x00 }, + { MT6660_REG_RESV31, 0x03, 0x03 }, + { MT6660_REG_RESV40, 0x01, 0x00 }, + { MT6660_REG_RESV0, 0x44, 0x04 }, + { MT6660_REG_RESV19, 0xff, 0x82 }, + { MT6660_REG_RESV17, 0x7777, 0x7273 }, + { MT6660_REG_RESV16, 0x07, 0x03 }, + { MT6660_REG_DRE_CORASE, 0xe0, 0x20 }, + { MT6660_REG_ADDA_CLOCK, 0xff, 0x70 }, + { MT6660_REG_RESV21, 0xff, 0x20 }, + { MT6660_REG_DRE_THDMODE, 0xff, 0x40 }, + { MT6660_REG_RESV23, 0xffff, 0x17f8 }, + { MT6660_REG_PWM_CTRL, 0xff, 0x15 }, + { MT6660_REG_ADC_USB_MODE, 0xff, 0x00 }, + { MT6660_REG_PROTECTION_CFG, 0xff, 0x1d }, + { MT6660_REG_HPF1_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_HPF2_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_SIG_GAIN, 0xff, 0x58 }, + { MT6660_REG_RESV6, 0xff, 0xce }, + { MT6660_REG_SIGMAX, 0xffff, 0x7fff }, + { MT6660_REG_DA_GAIN, 0xffff, 0x0116 }, + { MT6660_REG_TDM_CFG3, 0x1800, 0x0800 }, + { MT6660_REG_DRE_CTRL, 0x1f, 0x07 }, +}; + +static int mt6660_component_init_setting(struct snd_soc_component *component) +{ + int i, len, ret; + const struct codec_reg_val *init_table; + + pr_info("%s start\n", __func__); + init_table = generic_reg_inits; + len = ARRAY_SIZE(generic_reg_inits); + + for (i = 0; i < len; i++) { + pr_err("fuck 1\n"); + ret = snd_soc_component_update_bits(component, + init_table[i].addr, + init_table[i].mask, init_table[i].data); + pr_err("fuck 2\n"); + if (ret < 0) + return ret; + } + pr_info("%s end\n", __func__); + return 0; +} + +static int mt6660_component_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + unsigned int val; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + ret = regmap_read(chip->regmap, MT6660_REG_IRQ_STATUS1, &val); + dev_info(component->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + break; + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + /* Fall Through */ + case SND_SOC_BIAS_STANDBY: + default: + break; + } + dapm->bias_level = level; + dev_dbg(component->dev, "c bias_level = %d\n", level); + return 0; +} + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_PRE_PMD: + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUTP"), + SND_SOC_DAPM_OUTPUT("OUTN"), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "OUTP", NULL, "ClassD"}, + { "OUTN", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + int put_ret = 0; + + pm_runtime_get_sync(component->dev); + put_ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (put_ret < 0) + return put_ret; + pm_runtime_put(component->dev); + return put_ret; +} + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Digital Volume", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, mt6660_component_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT Switch", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Hard_Clip Switch", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Clip Switch", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DRE Switch", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DC_Protect Switch", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("audio input selection", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Left Channel Selection", + MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Data Output Right Channel Selection", + MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF_AUD_IN Switch", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("AUD_LOOP_BACK Switch", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Mute Switch", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Bypass CS Comp Switch", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("T0_SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip_Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + pr_info("%s\n", __func__); + //snd_soc_component_init_regmap(component, chip->regmap); + + /* init chip setting */ + ret = mt6660_component_init_setting(component); + if (ret < 0) + dev_err(chip->dev, "chip i2c init setting fail\n"); + + return ret; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + pr_info("%s\n", __func__); + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + .read = mt6660_component_io_read, + .write = mt6660_component_io_write, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .set_bias_level = mt6660_component_set_bias_level, + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int _mt6660_chip_id_check(struct mt6660_chip *chip) +{ + u8 id[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + return ret; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + int ret; + + /* turn on main pll first, then trigger reset */ + ret = regmap_write(chip->regmap, 0x03, 0x00); + if (ret < 0) + return ret; + ret = regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return regmap_write(chip->regmap, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +static int mt6660_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + pr_info("%s start\n", __func__); + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + chip->regmap = devm_regmap_init(&client->dev, + NULL, chip, &mt6660_regmap_config); + if (IS_ERR(chip->regmap)) { + ret = PTR_ERR(chip->regmap); + dev_err(&client->dev, "failed to initialise regmap: %d\n", ret); + return ret; + } + +#if GENERIC_DEBUGFS + /* debugfs interface */ + chip->dbg_info.dirname = devm_kasprintf(&client->dev, + GFP_KERNEL, "MT6660.%s", + dev_name(&client->dev)); + chip->dbg_info.devname = dev_name(&client->dev); + chip->dbg_info.typestr = devm_kasprintf(&client->dev, + GFP_KERNEL, "I2C,MT6660"); + chip->dbg_info.io_drvdata = chip; + chip->dbg_info.io_read = mt6660_dbg_io_read; + chip->dbg_info.io_write = mt6660_dbg_io_write; + + ret = generic_debugfs_init(&chip->dbg_info); + if (ret < 0) { + dev_err(&client->dev, "generic dbg init fail\n"); + return -EINVAL; + } +#endif /* GENERIC_DEBUGFS */ + + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + /* chip devid check */ + ret = _mt6660_chip_id_check(chip); + if (ret < 0) { + dev_err(chip->dev, "chip id check fail\n"); + goto probe_fail; + } + /* chip revision get */ + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); + + ret = devm_snd_soc_register_component(chip->dev, + &mt6660_component_driver, + &mt6660_codec_dai, 1); + pr_info("%s end, ret = %d\n", __func__, ret); + return ret; +probe_fail: + _mt6660_chip_power_on(chip, 0); + mutex_destroy(&chip->io_lock); + return ret; +} + +static int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + pm_runtime_disable(chip->dev); + pm_runtime_set_suspended(chip->dev); +#if GENERIC_DEBUGFS + generic_debugfs_exit(&chip->dbg_info); +#endif /* GENERIC_DEBUGFS */ + mutex_destroy(&chip->io_lock); + return 0; +} + +static int __maybe_unused mt6660_i2c_runtime_suspend(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "enter low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x01); +} + +static int __maybe_unused mt6660_i2c_runtime_resume(struct device *dev) +{ + struct mt6660_chip *chip = dev_get_drvdata(dev); + + dev_dbg(dev, "exit low power mode\n"); + return regmap_update_bits(chip->regmap, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); +} + +static const struct dev_pm_ops mt6660_dev_pm_ops = { + SET_RUNTIME_PM_OPS(mt6660_i2c_runtime_suspend, + mt6660_i2c_runtime_resume, NULL) +}; + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .of_match_table = of_match_ptr(mt6660_of_id), + .pm = &mt6660_dev_pm_ops, + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.7_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..9e62dcc --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,95 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> +#include <linux/regmap.h> + +#define GENERIC_DEBUGFS 0 + +#if GENERIC_DEBUGFS +struct dbg_internal { + struct dentry *rt_root; + struct dentry *ic_root; + bool rt_dir_create; + struct mutex io_lock; + u16 reg; + u16 size; + u16 data_buffer_size; + void *data_buffer; +}; + +struct dbg_info { + const char *dirname; + const char *devname; + const char *typestr; + void *io_drvdata; + int (*io_read)(void *drvdata, u16 reg, void *val, u16 size); + int (*io_write)(void *drvdata, u16 reg, const void *val, u16 size); + struct dbg_internal internal; +}; +#endif /* GENERIC_DEBUGFS */ + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct platform_device *param_dev; + struct mutex io_lock; +#if GENERIC_DEBUGFS + struct dbg_info dbg_info; +#endif /* GENERIC_DEBUGFS */ + struct regmap *regmap; + u16 chip_rev; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

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[alsa-devel] [PATCH 00/15] ASoC: soc-core cleanup step7
by Kuninori Morimoto 12 Dec '19

12 Dec '19

Hi Mark ASoC has codec_conf, and we are using original dev_name, of_node for it to finding codec. But, we already have snd_soc_dai_link_component method. We can reuse it for codec_conf, too. This patches are for it. Kuninori Morimoto (15): ASoC: soc-core: support snd_soc_dai_link_component for codec_conf ASoC: fsl: imx-audmix: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_da7219_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_rt5663_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_rt5663_rt5514_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: skl_nau88l25_ssm4567: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8173-rt5650-rt5514: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8173-rt5650-rt5676: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8183-da7219-max98357: use snd_soc_dai_link_component for codec_conf ASoC: samsung: bells: use snd_soc_dai_link_component for codec_conf ASoC: samsung: lowland: use snd_soc_dai_link_component for codec_conf ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for codec_conf ASoC: samsung: speyside: use snd_soc_dai_link_component for codec_conf ASoC: ti: rx51: use snd_soc_dai_link_component for codec_conf ASoC: soc-core: remove legacy style of codec_conf include/sound/soc.h | 4 ++-- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- sound/soc/samsung/bells.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 12 +++++------- sound/soc/ti/rx51.c | 12 ++++++------ 15 files changed, 31 insertions(+), 33 deletions(-) -- 2.7.4

2 17

[alsa-devel] [PATCH v3 00/15] Move PMC clocks into Tegra PMC driver
by Sowjanya Komatineni 12 Dec '19

12 Dec '19

Tegra PMC has clk_out_1, clk_out_2, clk_out_3 and blink controls which are currently registered by Tegra clock driver using clk_regiser_mux and clk_register_gate which performs direct Tegra PMC register access. When Tegra PMC is in secure mode, any access from non-secure world will not go through. This patch series adds these Tegra PMC clocks and blink controls to Tegra PMC driver with PMC as clock provider and removed them from Tegra clock driver. This also adds PMC specific clock id's to use in device tree and removed clock ids of PMC clock from Tegra clock driver. clk_out_1 is dedicated for audio mclk and current ASoC driver does not setting extern1 as parent for clk_out_1 and enabling clk_out_1 and currently this is taken care by Tegra clock driver during clock inits and there is no need to enable this during clock init. So, this series also includes patch that updates ASoC driver to take care of configuring parent of mclk and enabling mclk using both extern1 and clk_out_1 and updates all device trees to use clk_out_1 from pmc as mclk incase if device tree don't specify assigned-clock-parents. This series also includes a patch for mclk fallback to use extern1 when retrieving mclk fails with new device tree which uses pmc provider to have this backward compatible of new DT with old kernels. This series also includes a patch to remove clock ids for these clocks from clock dt-binding as these clocks are not used in any of the existing device tree except in tegra210-smaug.dts and this series includes a patch to update clock provider from tegra_car to pmc in tegra210-smaug.dts for clk_out_2. [v3]: Changes between v2 and v3 are - Removes set parent of clk_out_1_mux to extern1 and enabling extern1 from the clock driver. - Doesn't enable clk_out_1 and blink by default in pmc driver - Updates ASoC driver to take care of audio mclk parent configuration incase if device tree don't specify assigned clock parent properties and enables mclk using both clk_out_1 and extern1. - updates all device trees using extern1 as mclk in sound node to use clk_out_1 from pmc. - patch for YAML format pmc dt-binding - Includes v2 feedback [v2]: Changes between v1 and v2 are - v2 includes patches for adding clk_out_1, clk_out_2, clk_out_3, blink controls to Tegra PMC driver and removing clk-tegra-pmc. - feedback related to pmc clocks in Tegra PMC driver from v1 - Removed patches for WB0 PLLM overrides and PLLE IDDQ PMC programming by the clock driver using helper functions from Tegra PMC. Note: To use helper functions from PMC driver, PMC early init need to happen prior to using helper functions and these helper functions are for PLLM Override and PLLE IDDQ programming in PMC during PLLM/PLLE clock registration which happen in clock_init prior to Tegra PMC probe. Moving PLLM/PLLE clocks registration to happen after Tegra PMC impacts other clocks EMC, MC and corresponding tegra_emc_init and tegra_mc_init. This implementation of configuring PMC registers thru helper functions in clock driver needs proper changes across PMC, Clock, EMC and MC inits to have it work across all Tegra platforms. Currently PLLM Override is not enabled in the bootloader so proper patches for this fix will be taken care separately. [v1]: v1 includes patches for below fixes. - adding clk_out_1, clk_out_2, clk_out_3, blink controls to Tegra PMC driver and removing clk-tegra-pmc. - updated clock provider from tegra_car to pmc in the device tree tegra210-smaug.dts that uses clk_out_2. - Added helper functions in PMC driver for WB0 PLLM overrides and PLLE IDDQ programming to use by clock driver and updated clock driver to use these helper functions and removed direct PMC access from clock driver and all pmc base address references in clock driver. Sowjanya Komatineni (15): dt-bindings: soc: tegra-pmc: Add Tegra PMC clock bindings dt-bindings: tegra: Convert Tegra PMC bindings to YAML soc: tegra: Add Tegra PMC clock registrations into PMC driver dt-bindings: soc: tegra-pmc: Add id for Tegra PMC blink control soc: pmc: Add blink output clock registration to Tegra PMC clk: tegra: Remove tegra_pmc_clk_init along with clk ids dt-bindings: clock: tegra: Remove pmc clock ids from clock dt-bindings ASoC: tegra: Add audio mclk control through clk_out_1 and extern1 ASoC: tegra: Add fallback for audio mclk clk: tegra: Remove extern1 and cdev1 from clocks inittable ARM: dts: tegra: Add clock-cells property to pmc arm64: tegra: Add clock-cells property to Tegra PMC node ARM: tegra: Update sound node clocks in device tree arm64: tegra: smaug: Change clk_out_2 provider to pmc ASoC: nau8825: change Tegra clk_out_2 provider from tegra_car to pmc .../bindings/arm/tegra/nvidia,tegra20-pmc.txt | 45 ++- .../bindings/arm/tegra/nvidia,tegra20-pmc.yaml | 291 ++++++++++++++++++ .../devicetree/bindings/sound/nau8825.txt | 2 +- arch/arm/boot/dts/tegra114-dalmore.dts | 7 +- arch/arm/boot/dts/tegra114.dtsi | 4 +- arch/arm/boot/dts/tegra124-apalis-v1.2.dtsi | 7 +- arch/arm/boot/dts/tegra124-apalis.dtsi | 7 +- arch/arm/boot/dts/tegra124-jetson-tk1.dts | 7 +- arch/arm/boot/dts/tegra124-nyan.dtsi | 7 +- arch/arm/boot/dts/tegra124-venice2.dts | 7 +- arch/arm/boot/dts/tegra124.dtsi | 4 +- arch/arm/boot/dts/tegra20.dtsi | 4 +- arch/arm/boot/dts/tegra30-apalis-v1.1.dtsi | 7 +- arch/arm/boot/dts/tegra30-apalis.dtsi | 7 +- arch/arm/boot/dts/tegra30-beaver.dts | 7 +- arch/arm/boot/dts/tegra30-cardhu.dtsi | 7 +- arch/arm/boot/dts/tegra30-colibri.dtsi | 7 +- arch/arm/boot/dts/tegra30.dtsi | 4 +- arch/arm64/boot/dts/nvidia/tegra132.dtsi | 4 +- arch/arm64/boot/dts/nvidia/tegra210-smaug.dts | 2 +- arch/arm64/boot/dts/nvidia/tegra210.dtsi | 2 + drivers/clk/tegra/Makefile | 1 - drivers/clk/tegra/clk-id.h | 7 - drivers/clk/tegra/clk-tegra-pmc.c | 122 -------- drivers/clk/tegra/clk-tegra114.c | 12 - drivers/clk/tegra/clk-tegra124.c | 28 +- drivers/clk/tegra/clk-tegra20.c | 5 - drivers/clk/tegra/clk-tegra210.c | 12 - drivers/clk/tegra/clk-tegra30.c | 13 - drivers/clk/tegra/clk.h | 1 - drivers/soc/tegra/pmc.c | 340 +++++++++++++++++++++ include/dt-bindings/clock/tegra114-car.h | 14 +- include/dt-bindings/clock/tegra124-car-common.h | 14 +- include/dt-bindings/clock/tegra20-car.h | 2 +- include/dt-bindings/clock/tegra210-car.h | 14 +- include/dt-bindings/clock/tegra30-car.h | 14 +- include/dt-bindings/soc/tegra-pmc.h | 19 ++ sound/soc/tegra/tegra_asoc_utils.c | 76 ++++- sound/soc/tegra/tegra_asoc_utils.h | 1 + 39 files changed, 890 insertions(+), 244 deletions(-) create mode 100644 Documentation/devicetree/bindings/arm/tegra/nvidia,tegra20-pmc.yaml delete mode 100644 drivers/clk/tegra/clk-tegra-pmc.c create mode 100644 include/dt-bindings/soc/tegra-pmc.h -- 2.7.4

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Re: [alsa-devel] [PATCH v3 03/15] soc: tegra: Add Tegra PMC clock registrations into PMC driver
by Dmitry Osipenko 12 Dec '19

12 Dec '19

11.12.2019 21:50, Sowjanya Komatineni пишет: > > On 12/10/19 5:06 PM, Sowjanya Komatineni wrote: >> >> On 12/10/19 9:41 AM, Dmitry Osipenko wrote: >>> 10.12.2019 19:53, Sowjanya Komatineni пишет: >>>> On 12/9/19 3:03 PM, Sowjanya Komatineni wrote: >>>>> On 12/9/19 12:46 PM, Sowjanya Komatineni wrote: >>>>>> On 12/9/19 12:12 PM, Dmitry Osipenko wrote: >>>>>>> 08.12.2019 00:36, Sowjanya Komatineni пишет: >>>>>>>> On 12/7/19 11:59 AM, Sowjanya Komatineni wrote: >>>>>>>>> On 12/7/19 8:00 AM, Dmitry Osipenko wrote: >>>>>>>>>> 07.12.2019 18:53, Dmitry Osipenko пишет: >>>>>>>>>>> 07.12.2019 18:47, Dmitry Osipenko пишет: >>>>>>>>>>>> 07.12.2019 17:28, Dmitry Osipenko пишет: >>>>>>>>>>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>>>>>>>>>> Tegra210 and prior Tegra PMC has clk_out_1, clk_out_2, >>>>>>>>>>>>>> clk_out_3 >>>>>>>>>>>>>> with >>>>>>>>>>>>>> mux and gate for each of these clocks. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Currently these PMC clocks are registered by Tegra clock >>>>>>>>>>>>>> driver >>>>>>>>>>>>>> using >>>>>>>>>>>>>> clk_register_mux and clk_register_gate by passing PMC base >>>>>>>>>>>>>> address >>>>>>>>>>>>>> and register offsets and PMC programming for these clocks >>>>>>>>>>>>>> happens >>>>>>>>>>>>>> through direct PMC access by the clock driver. >>>>>>>>>>>>>> >>>>>>>>>>>>>> With this, when PMC is in secure mode any direct PMC access >>>>>>>>>>>>>> from the >>>>>>>>>>>>>> non-secure world does not go through and these clocks will >>>>>>>>>>>>>> not be >>>>>>>>>>>>>> functional. >>>>>>>>>>>>>> >>>>>>>>>>>>>> This patch adds these clocks registration with PMC as a clock >>>>>>>>>>>>>> provider >>>>>>>>>>>>>> for these clocks. clk_ops callback implementations for these >>>>>>>>>>>>>> clocks >>>>>>>>>>>>>> uses tegra_pmc_readl and tegra_pmc_writel which supports PMC >>>>>>>>>>>>>> programming >>>>>>>>>>>>>> in secure mode and non-secure mode. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>>>>>>>>>>> --- >>>>>>>>>>>> [snip] >>>>>>>>>>>> >>>>>>>>>>>>>> + >>>>>>>>>>>>>> +static const struct clk_ops pmc_clk_gate_ops = { >>>>>>>>>>>>>> + .is_enabled = pmc_clk_is_enabled, >>>>>>>>>>>>>> + .enable = pmc_clk_enable, >>>>>>>>>>>>>> + .disable = pmc_clk_disable, >>>>>>>>>>>>>> +}; >>>>>>>>>>>>> What's the benefit of separating GATE from the MUX? >>>>>>>>>>>>> >>>>>>>>>>>>> I think it could be a single clock. >>>>>>>>>>>> According to TRM: >>>>>>>>>>>> >>>>>>>>>>>> 1. GATE and MUX are separate entities. >>>>>>>>>>>> >>>>>>>>>>>> 2. GATE is the parent of MUX (see PMC's CLK_OUT paths diagram >>>>>>>>>>>> in TRM). >>>>>>>>>>>> >>>>>>>>>>>> 3. PMC doesn't gate EXTPERIPH clock but could "force-enable" >>>>>>>>>>>> it, >>>>>>>>>>>> correct? >>>>>>>> Was following existing clk-tegra-pmc as I am not sure of reason for >>>>>>>> having these clocks registered as separate mux and gate clocks. >>>>>>>> >>>>>>>> Yes, PMC clocks can be registered as single clock and can use >>>>>>>> clk_ops >>>>>>>> for set/get parent and enable/disable. >>>>>>>> >>>>>>>> enable/disable of PMC clocks is for force-enable to force the >>>>>>>> clock to >>>>>>>> run regardless of ACCEPT_REQ or INVERT_REQ. >>>>>>>> >>>>>>>>>>> 4. clk_m_div2/4 are internal PMC OSC dividers and thus these >>>>>>>>>>> clocks >>>>>>>>>>> should belong to PMC. >>>>>>>>>> Also, it should be "osc" and not "clk_m". >>>>>>>>> I followed the same parents as it were in existing clk-tegra-pmc >>>>>>>>> driver. >>>>>>>>> >>>>>>>>> Yeah they are wrong and they should be from osc and not clk_m. >>>>>>>>> >>>>>>>>> Will fix in next version. >>>>>>>>> >>>>> Reg clk_m_div2/3, they are dividers at OSC pad and not really internal >>>>> to PMC block. >>>>> >>>>> current clock driver creates clk_m_div clocks which should actually be >>>>> osc_div2/osc_div4 clocks with osc as parent. >>>>> >>>>> There are no clk_m_div2 and clk_m_div4 from clk_m >>>>> >>>>> Will fix this in next version. >>>>> >>>>>>> Could you please describe the full EXTPERIPH clock topology and >>>>>>> how the >>>>>>> pinmux configuration is related to it all? >>>>>>> >>>>>>> What is internal to the Tegra chip and what are the external >>>>>>> outputs? >>>>>>> >>>>>>> Is it possible to bypass PMC on T30+ for the EXTPERIPH clocks? >>>>>> PMC CLK1/2/3 possible sources are OSC_DIV1, OSC_DIV2, OSC_DIV4, >>>>>> EXTPERIPH from CAR. >>>>>> >>>>>> OSC_DIV1/2/4 are with internal dividers at the OSC Pads >>>>>> >>>>>> EXTPERIPH is from CAR and it has reset and enable controls along with >>>>>> clock source selections to choose one of the PLLA_OUT0, CLK_S, >>>>>> PLLP_OUT0, CLK_M, PLLE_OUT0 >>>>>> >>>>>> So, PMC CLK1/2/4 possible parents are OSC_DIV1, OSC_DIV2, OSC_DIV4, >>>>>> EXTERN. >>>>>> >>>>>> >>>>>> CLK1/2/3 also has Pinmux to route EXTPERIPH output on to these pins. >>>>>> >>>>>> >>>>>> When EXTERN output clock is selected for these PMC clocks thru >>>>>> CLKx_SRC_SEL, output clock is from driver by EXTPERIPH from CAR via >>>>>> Pinmux logic or driven as per CLKx_SRC_SEL bypassing pinmux based on >>>>>> CLKx_ACCEPT_REQ bit. >>>>>> >>>>>> >>>>>> PMC Clock control register has bit CLKx_ACCEPT_REQ >>>>>> When CLKx_ACCEPT_REQ = 0, output clock driver is from by EXTPERIPH >>>>>> through the pinmux >>>>>> When CLKx_ACCEPT_REQ = 1, output clock is based on CLKx_SRC_SEL bits >>>>>> (OSC_DIV1/2/4 and EXTPERIPH clock bypassing the pinmux) >>>>>> >>>>>> FORCE_EN bit in PMC CLock control register forces the clock to run >>>>>> regardless of this. >>>> PMC clock gate is based on the state of CLKx_ACCEPT_REQ and FORCE_EN >>>> like explained above. >>>> >>>> CLKx_ACCEPT_REQ is 0 default and FORCE_EN acts as gate to >>>> enable/disable >>>> EXTPERIPH clock output to PMC CLK_OUT_1/2/3. >>> [and to enable OSC as well] >>> >>>> So I believe we need to register as MUX and Gate rather than as a >>>> single >>>> clock. Please confirm. >>> 1. The force-enabling is applied to both OSC and EXTERN sources of >>> PMC_CLK_OUT_x by PMC at once. >>> >>> 2. Both of PMC's force-enabling and OSC/EXTERN selection is internal >>> to PMC. >>> >>> Should be better to define it as a single "pmc_clk_out_x". I don't see >>> any good reasons for differentiating PMC's Gate from the MUX, it's a >>> single hardware unit from a point of view of the rest of the system. >>> >>> Peter, do you have any objections? >> >> We added fallback option for audio mclk and also added check for >> assigned-clock-parents dt property in audio driver and if its not then >> we do parent init configuration in audio driver. >> >> Current clock driver creates 2 separate clocks clk_out_1_mux and >> clk_out_1 for each pmc clock in clock driver and uses extern1 as >> parent to clk_out_1_mux and clk_out_1_mux is parent to clk_out_1. >> >> With change of registering each pmc clock as a single clock, when we >> do parent init assignment in audio driver when >> assigned-clock-properties are not used in DT (as we removed parent >> inits for extern and clk_outs from clock driver), we should still try >> to get clock based on clk_out_1_mux as parent assignment of extern1 is >> for clk_out_1_mux as per existing clock tree. >> >> clk_out_1_mux clock retrieve will fail with this change of single >> clock when any new platform device tree doesn't specify >> assigned-clock-parents properties and tegra_asoc_utils_init fails. You made the PMC/CaR changes before the audio changes, the clk_out_1_mux won't exist for the audio driver patches. If you care about bisect-ability of the patches, then the clock and audio changes need to be done in a single patch. But I don't think that it's worthwhile. >> With single clock, extern1 is the parent for clk_out_1 and with >> separate clocks for mux and gate, extern1 is the parent for >> clk_out_1_mux. > > If we move to single clock now, it need one more additional fallback > implementation in audio driver during parent configuration as > clk_out_1_mux will not be there with single clock change and old/current > kernel has it as it uses separate clocks for pmc mux and gate. Why additional fallback? Additional to what? > Also, with single clock for both PMC mux and gate now, new DT should use > extern1 as parent to CLK_OUT_1 as CLK_OUT_1_MUX will not be there old > PMC dt-bindings has separate clocks for MUX (CLK_OUT_1_MUX) and gate > (CLK_OUT_1) > > DT bindings will not be compatible b/w old and new changes if we move to > Single PMC clock now. Sorry, I don't understand what you're meaning by the "new changes". > Should we go with same separate clocks to have it compatible to avoid > all this? >

1 0

[alsa-devel] [PATCH for-5.6 00/23] ASoC: PCM API updates (part II)
by Takashi Iwai 11 Dec '19

11 Dec '19

Hi, this is the second part of PCM API updates for ASoC. This one just removes the superfluous PCM ioctl ops (and relevant cleanups). Applicable on top of the previous patch set. thanks, Takashi === Takashi Iwai (23): ASoC: amd: Drop superfluous ioctl PCM ops ASoC: atmel: Drop superfluous ioctl PCM ops ASoC: au1x: Drop superfluous ioctl PCM ops ASoC: bcm: Drop superfluous ioctl PCM ops ASoC: dwc: Drop superfluous ioctl PCM ops ASoC: fsl: Drop superfluous ioctl PCM ops ASoC: intel: Drop superfluous ioctl PCM ops ASoC: kirkwood: Drop superfluous ioctl PCM ops ASoC: mediatek: Drop superfluous ioctl PCM ops ASoC: meson: Drop superfluous ioctl PCM ops ASoC: pxa: Drop superfluous ioctl PCM ops ASoC: qcom: Drop superfluous ioctl PCM ops ASoC: samsung: Drop superfluous ioctl PCM ops ASoC: sh: Drop superfluous ioctl PCM ops ASoC: generic-dmaengine: Drop superfluous ioctl PCM ops ASoC: SOF: Drop superfluous ioctl PCM ops ASoC: sprd: Drop superfluous ioctl PCM ops ASoC: txx9: Drop superfluous ioctl PCM ops ASoC: uniphier: Drop superfluous ioctl PCM ops ASoC: xilinx: Drop superfluous ioctl PCM ops ASoC: xtensa: Drop superfluous ioctl PCM ops ASoC: utils: Drop superfluous ioctl PCM ops ASoC: Drop snd_soc_pcm_lib_ioctl() include/sound/soc.h | 5 ----- sound/soc/amd/acp-pcm-dma.c | 1 - sound/soc/amd/raven/acp3x-pcm-dma.c | 1 - sound/soc/atmel/atmel-pcm-pdc.c | 1 - sound/soc/au1x/dbdma2.c | 1 - sound/soc/au1x/dma.c | 1 - sound/soc/bcm/cygnus-pcm.c | 1 - sound/soc/dwc/dwc-pcm.c | 1 - sound/soc/fsl/fsl_asrc_dma.c | 1 - sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 1 - sound/soc/fsl/mpc5200_dma.c | 1 - sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 - sound/soc/intel/baytrail/sst-baytrail-pcm.c | 1 - sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/kirkwood/kirkwood-dma.c | 1 - sound/soc/mediatek/common/mtk-afe-platform-driver.c | 1 - sound/soc/mediatek/common/mtk-btcvsd.c | 1 - sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 1 - sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 1 - sound/soc/meson/axg-frddr.c | 3 --- sound/soc/meson/axg-toddr.c | 3 --- sound/soc/pxa/mmp-pcm.c | 1 - sound/soc/pxa/pxa-ssp.c | 1 - sound/soc/pxa/pxa2xx-ac97.c | 1 - sound/soc/pxa/pxa2xx-i2s.c | 1 - sound/soc/pxa/pxa2xx-pcm.c | 1 - sound/soc/qcom/lpass-platform.c | 1 - sound/soc/qcom/qdsp6/q6asm-dai.c | 1 - sound/soc/samsung/idma.c | 1 - sound/soc/sh/dma-sh7760.c | 1 - sound/soc/sh/fsi.c | 1 - sound/soc/sh/rcar/core.c | 1 - sound/soc/sh/siu_pcm.c | 1 - sound/soc/soc-core.c | 12 ------------ sound/soc/soc-generic-dmaengine-pcm.c | 2 -- sound/soc/soc-utils.c | 1 - sound/soc/sof/pcm.c | 1 - sound/soc/sprd/sprd-pcm-dma.c | 1 - sound/soc/txx9/txx9aclc.c | 1 - sound/soc/uniphier/aio-dma.c | 1 - sound/soc/xilinx/xlnx_formatter_pcm.c | 1 - sound/soc/xtensa/xtfpga-i2s.c | 1 - 44 files changed, 64 deletions(-) -- 2.16.4

7 58

[alsa-devel] [PATCH for-5.6 00/23] ASoC: PCM API updates (part I)
by Takashi Iwai 11 Dec '19

11 Dec '19

Hi, this is a patch set to adapt the recent PCM API updates to ASoC drivers. This is the first half of the whole changes, just for updating the buffer allocation code with the new API. There should be no functional changes. The ASoC core changes are already found in 5.5-rc1, so please merge these onto 5.5-rc1 or later branch. Or I can merge through my tree, too, just let me know. thanks, Takashi === Takashi Iwai (23): ASoC: amd: Use managed buffer allocation ASoC: au1x: Use managed buffer allocation ASoC: dwc: Use managed buffer allocation ASoC: mediatek: Use managed buffer allocation ASoC: meson: Use managed buffer allocation ASoC: dma-sh7760: Use managed buffer allocation ASoC: fsi: Use managed buffer allocation ASoC: rcar: Use managed buffer allocation ASoC: siu_pcm: Use managed buffer allocation ASoC: generic-dmaengine-pcm: Use managed buffer allocation ASoC: SOF: Use managed buffer allocation ASoC: stm32: Use managed buffer allocation ASoC: txx9: Use managed buffer allocation ASoC: uniphier: Use managed buffer allocation ASoC: xilinx: Use managed buffer allocation ASoC: xtensa: Use managed buffer allocation ASoC: cros_ec_codec: Use managed buffer allocation ASoC: rt5514-spi: Use managed buffer allocation ASoC: rt5677-spi: Use managed buffer allocation ASoC: intel: atom: Use managed buffer allocation ASoC: intel: baytrail: Use managed buffer allocation ASoC: intel: haswell: Use managed buffer allocation ASoC: intel: skylake: Use managed buffer allocation sound/soc/amd/acp-pcm-dma.c | 58 ++++++++-------------- sound/soc/amd/raven/acp3x-pcm-dma.c | 30 +++-------- sound/soc/au1x/dbdma2.c | 14 +----- sound/soc/au1x/dma.c | 21 +++----- sound/soc/codecs/cros_ec_codec.c | 8 +-- sound/soc/codecs/rt5514-spi.c | 10 ++-- sound/soc/codecs/rt5677-spi.c | 10 ++-- sound/soc/dwc/dwc-pcm.c | 24 +-------- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 25 +--------- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 19 +------ sound/soc/intel/haswell/sst-haswell-pcm.c | 17 +------ sound/soc/intel/skylake/skl-pcm.c | 26 +++------- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 14 +----- sound/soc/mediatek/common/mtk-afe-fe-dai.h | 2 - .../soc/mediatek/common/mtk-afe-platform-driver.c | 12 +---- .../soc/mediatek/common/mtk-afe-platform-driver.h | 2 - sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 - sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 1 - sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 1 - sound/soc/meson/axg-fifo.c | 13 ++--- sound/soc/sh/dma-sh7760.c | 14 +----- sound/soc/sh/fsi.c | 18 +------ sound/soc/sh/rcar/core.c | 23 +++------ sound/soc/sh/siu_pcm.c | 39 +-------------- sound/soc/soc-generic-dmaengine-pcm.c | 12 +---- sound/soc/sof/pcm.c | 34 ++++--------- sound/soc/stm/stm32_adfsdm.c | 29 +---------- sound/soc/txx9/txx9aclc.c | 14 +----- sound/soc/uniphier/aio-dma.c | 30 +---------- sound/soc/xilinx/xlnx_formatter_pcm.c | 13 +---- sound/soc/xtensa/xtfpga-i2s.c | 9 ++-- 31 files changed, 93 insertions(+), 451 deletions(-) -- 2.16.4

3 47

[alsa-devel] [PATCH] ASoC: soc-core: merge soc_set_name_prefix() and soc_set_of_name_prefix()
by Kuninori Morimoto 11 Dec '19

11 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_set_name_prefix() is calling soc_set_of_name_prefix(). We don't need to separate these operation. This patch merges these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 750469a..f8090bd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1206,22 +1206,12 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, return ret; } -static void soc_set_of_name_prefix(struct snd_soc_component *component) -{ - struct device_node *of_node = soc_component_to_node(component); - const char *str; - int ret; - - ret = of_property_read_string(of_node, "sound-name-prefix", &str); - if (!ret) - component->name_prefix = str; -} - static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { struct device_node *of_node = soc_component_to_node(component); - int i; + const char *str; + int ret, i; for (i = 0; i < card->num_configs; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; @@ -1238,7 +1228,11 @@ static void soc_set_name_prefix(struct snd_soc_card *card, * If there is no configuration table or no match in the table, * check if a prefix is provided in the node */ - soc_set_of_name_prefix(component); + ret = of_property_read_string(of_node, "sound-name-prefix", &str); + if (ret < 0) + return; + + component->name_prefix = str; } static void soc_remove_component(struct snd_soc_component *component, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: soc-core: tidyup for CONFIG_DMI
by Kuninori Morimoto 11 Dec '19

11 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-core.c has 2 #ifdef CONFIG_DMI, but we can merge these. OTOH, soc.h has dmi_longname, but it is needed if CONFIG_DMI was defined. In other words, It is not needed if CONFIG_DMI was not defined. This patch tidyup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 32 +++++++++++++++----------------- 2 files changed, 17 insertions(+), 17 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c920f17..a6a3a7d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -986,7 +986,9 @@ struct snd_soc_card { const char *long_name; const char *driver_name; const char *components; +#ifdef CONFIG_DMI char dmi_longname[80]; +#endif /* CONFIG_DMI */ char topology_shortname[32]; struct device *dev; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f8090bd..51a404f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -73,23 +73,6 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); -#ifdef CONFIG_DMI -/* - * If a DMI filed contain strings in this blacklist (e.g. - * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken - * as invalid and dropped when setting the card long name from DMI info. - */ -static const char * const dmi_blacklist[] = { - "To be filled by OEM", - "TBD by OEM", - "Default String", - "Board Manufacturer", - "Board Vendor Name", - "Board Product Name", - NULL, /* terminator */ -}; -#endif - static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) { @@ -1608,6 +1591,21 @@ EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); #ifdef CONFIG_DMI /* + * If a DMI filed contain strings in this blacklist (e.g. + * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken + * as invalid and dropped when setting the card long name from DMI info. + */ +static const char * const dmi_blacklist[] = { + "To be filled by OEM", + "TBD by OEM", + "Default String", + "Board Manufacturer", + "Board Vendor Name", + "Board Product Name", + NULL, /* terminator */ +}; + +/* * Trim special characters, and replace '-' with '_' since '-' is used to * separate different DMI fields in the card long name. Only number and * alphabet characters and a few separator characters are kept. -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: soc-core: tidyup for CONFIG_DEBUG_FS
by Kuninori Morimoto 11 Dec '19

11 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-core.c has 2 #ifdef CONFIG_DEBUG_FS, but we can merge these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 51a404f..3eb874c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -45,11 +45,6 @@ #define NAME_SIZE 32 -#ifdef CONFIG_DEBUG_FS -struct dentry *snd_soc_debugfs_root; -EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); -#endif - static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); static LIST_HEAD(unbind_card_list); @@ -133,6 +128,9 @@ static const struct attribute_group *soc_dev_attr_groups[] = { }; #ifdef CONFIG_DEBUG_FS +struct dentry *snd_soc_debugfs_root; +EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); + static void soc_init_component_debugfs(struct snd_soc_component *component) { if (!component->card->debugfs_card_root) -- 2.7.4

2 1

[alsa-devel] [PATCH v4 00/11] soundwire: update ASoC interfaces
by Pierre-Louis Bossart 11 Dec '19

11 Dec '19

We need new fields in existing structures to a) deal with race conditions on codec probe/enumeration b) allow for multi-step ACPI scan/probe/startup on Intel plaforms c) deal with MSI issues using a single handler/threads for all audio interrupts d) deal with access to registers shared across multiple links on Intel platforms These structures for a) will be used by the SOF driver as well as codec drivers. The b) c) and d) cases are only for the Intel-specific implementation. To avoid conflicts between ASoC and Soundwire trees, these 11 patches are provided out-of-order, before the functionality enabled in these header files is added in follow-up patch series which can be applied separately in the ASoC and Soundwire trees. As discussed earlier, Vinod would need to provide an immutable tag for Mark Brown, and the integration on the ASoC side of SOF changes and new codecs drivers can proceed in parallel with SoundWire core changes. I had multiple offline discussions with Vinod/Mark/Takashi on how to proceed withe volume of SoundWire changes. Now that v5.5-rc1 is out we should go ahead with these interface changes. The next patchset has not changed, the series "[PATCH v3 00/15] soundwire: intel: implement new ASoC interfaces" can still be reviewed. An update for the series "[PATCH v3 00/22] soundwire: code hardening and suspend-resume support" will be provided later this week (one last minute issue to fix) Comments and feedback welcome ~Pierre Changes since v3: Reordered code and added kernel doc comments Added prototypes and fields needed to deal with SoundWire interrupts in a single handler/thread, following what is already done on the DSP side. Added mutex to control access to registers shared across links Added initial definitions for clock stop support on Intel platforms. Depending on power and latency requirements, different "quirks" can be supported. Changes since v2: Added new field to deal with a race condition leading to a timeout when the codec goes through a pm_runtime suspend/resume transition while the Master remains active. Clarified commit messages with detailed explanations what those race conditions are and why the changes were introduced. Reordered fields for Intel routines Added kernel-doc definitions for structures Modified the patch subjects to make the mapping between interface definition and implementation straightforward. Changes since v1 (no feedback received since October 23) additional initialization_complete utility to help codec drivers with their resume operation, waiting for the enumeration to complete is not always enough. Bard Liao (2): soundwire: intel: update headers for interrupts soundwire: intel: add link_list to handle interrupts with a single thread Pierre-Louis Bossart (7): soundwire: sdw_slave: add probe_complete structure and new fields soundwire: sdw_slave: add enumeration_complete structure soundwire: sdw_slave: add initialization_complete definition soundwire: sdw_slave: track unattach_request to handle all init sequences soundwire: intel: update interfaces between ASoC and SoundWire soundwire: intel: add mutex for shared SHIM register access soundwire: intel: add clock stop quirks Rander Wang (2): soundwire: intel: update stream callbacks for hwparams/free stream operations soundwire: intel: add prototype for WAKEEN interrupt processing drivers/soundwire/intel.c | 20 ++-- drivers/soundwire/intel.h | 13 ++- drivers/soundwire/intel_init.c | 31 ++---- include/linux/soundwire/sdw.h | 19 ++++ include/linux/soundwire/sdw_intel.h | 154 ++++++++++++++++++++++++++-- 5 files changed, 194 insertions(+), 43 deletions(-) base-commit: e42617b825f8073569da76dc4510bfa019b1c35a -- 2.20.1

2 13

Re: [alsa-devel] [v6, 5/6] ASoC: sgtl5000: Fix of unmute outputs on probe
by Alison Wang 11 Dec '19

11 Dec '19

Hi, Oleksandr, I have a question about this patch. Could you give me some ideas? Thanks. In this patch, snd_soc_component_update_bits is used instead of snd_soc_component_write. Although EN_HP_ZCD and EN_ADC_ZCD are enabled in ANA_CTRL register, MUTE_LO, MUTE_HP and MUTE_ADC bits are remained as the default value. It causes LO, HP and ADC are all muted after driver probe. BTW, in the original code, snd_soc_component_write is used and MUTE_LO, MUTE_HP and MUTE_ADC are all set as zero (unmute). I saw the above phenomenon on the latest linux-next. For LO and HP are muted, no sound can be heard. Best Regards, Alison Wang > Subject: [v6,5/6] ASoC: sgtl5000: Fix of unmute outputs on probe > > To enable "zero cross detect" for ADC/HP, change HP_ZCD_EN/ADC_ZCD_EN > bits only instead of writing the whole CHIP_ANA_CTRL register. > > Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> > Reviewed-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> > Reviewed-by: Igor Opaniuk <igor.opaniuk(a)toradex.com> > Reviewed-by: Fabio Estevam <festevam(a)gmail.com> > > --- > > Changes in v6: None > Changes in v5: None > Changes in v4: None > Changes in v3: None > Changes in v2: > - Fix patch formatting > > sound/soc/codecs/sgtl5000.c | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > > > diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index > b65232521ea8..23f4ae2f0723 100644 > --- a/sound/soc/codecs/sgtl5000.c > +++ b/sound/soc/codecs/sgtl5000.c > @@ -1453,6 +1453,7 @@ static int sgtl5000_probe(struct > snd_soc_component *component) > int ret; > u16 reg; > struct sgtl5000_priv *sgtl5000 = > snd_soc_component_get_drvdata(component); > + unsigned int zcd_mask = SGTL5000_HP_ZCD_EN | > SGTL5000_ADC_ZCD_EN; > > /* power up sgtl5000 */ > ret = sgtl5000_set_power_regs(component); > @@ -1480,9 +1481,8 @@ static int sgtl5000_probe(struct > snd_soc_component *component) > 0x1f); > snd_soc_component_write(component, SGTL5000_CHIP_PAD_STRENGTH, > reg); > > - snd_soc_component_write(component, SGTL5000_CHIP_ANA_CTRL, > - SGTL5000_HP_ZCD_EN | > - SGTL5000_ADC_ZCD_EN); > + snd_soc_component_update_bits(component, > SGTL5000_CHIP_ANA_CTRL, > + zcd_mask, zcd_mask); > > snd_soc_component_update_bits(component, > SGTL5000_CHIP_MIC_CTRL, > SGTL5000_BIAS_R_MASK, >

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Line-out jack doesn't work on a Dell AIO
by Hui Wang 11 Dec '19

11 Dec '19

After applying the fixup ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, the Line-out jack works well. And instead of adding a new set of pin definition in the pin_fixup_tbl, we put a more generic matching entry in the fallback_pin_fixup_tbl. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d6e34b3b3aa..dbfafee97931 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7643,11 +7643,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x90a70130}, {0x1b, 0x90170110}, {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, - {0x12, 0xb7a60130}, - {0x13, 0xb8a61140}, - {0x16, 0x90170110}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, @@ -7841,6 +7836,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + {0x19, 0x40000000}, + {0x1a, 0x40000000}), {} }; -- 2.17.1

2 1

[alsa-devel] [alsa-lib][PATCH] ucm: setup conf_format after getting ALSA_CONFIG_UCM_VAR successfully
by libin.yang＠linux.intel.com 11 Dec '19

11 Dec '19

From: Libin Yang <libin.yang(a)linux.intel.com> Set the conf_format to 1 after getting ALSA_CONFIG_UCM_VAR successfully. Otherwise, the conf_format is not set in this scenario. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- src/ucm/parser.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index ba50027..d61124a 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -89,6 +89,8 @@ static void configuration_filename(snd_use_case_mgr_t *uc_mgr, env = getenv(ALSA_CONFIG_UCM2_VAR); if (env == NULL) { env = getenv(ALSA_CONFIG_UCM_VAR); + if (env) + uc_mgr->conf_format = 1; } else { uc_mgr->conf_format = 2; } -- 2.7.4

1 0

[alsa-devel] [PATCH 00/23] ALSA: Drop superfluous ioctl PCM ops (for 5.6)
by Takashi Iwai 11 Dec '19

11 Dec '19

Hi, this is another series of patches targeted to 5.6 kernel. This one is applied on top of the previous series for managed allocation, and again specific to non-ASoC drivers in sound/*. For ASoC and other subsystems, another patchset will be submitted later. The changes are pretty simple, just removing the snd_pcm_lib_ioctl entries that became superfluous. Takashi === Takashi Iwai (23): ALSA: aoa: Drop superfluous ioctl PCM ops ALSA: arm: Drop superfluous ioctl PCM ops ALSA: atmel: Drop superfluous ioctl PCM ops ALSA: drivers: Drop superfluous ioctl PCM ops ALSA: firewire: Drop superfluous ioctl PCM ops ALSA: es1688: Drop superfluous ioctl PCM ops ALSA: isa: Drop superfluous ioctl PCM ops ALSA: mips: Drop superfluous ioctl PCM ops ALSA: parisc: Drop superfluous ioctl PCM ops ALSA: intel8x0: Drop superfluous ioctl PCM ops ALSA: via82xx: Drop superfluous ioctl PCM ops ALSA: atiixp: Drop superfluous ioctl PCM ops ALSA: asihpi: Drop superfluous ioctl PCM ops ALSA: trident: Drop superfluous ioctl PCM ops ALSA: pci: Drop superfluous ioctl PCM ops ALSA: pcmcia: Drop superfluous ioctl PCM ops ALSA: ppc: Drop superfluous ioctl PCM ops ALSA: sh: Drop superfluous ioctl PCM ops ALSA: sparc: Drop superfluous ioctl PCM ops ALSA: spi: Drop superfluous ioctl PCM ops ALSA: usb: Drop superfluous ioctl PCM ops ALSA: x86: Drop superfluous ioctl PCM ops ALSA: xen: Drop superfluous ioctl PCM ops sound/aoa/soundbus/i2sbus/pcm.c | 2 -- sound/arm/aaci.c | 2 -- sound/arm/pxa2xx-ac97.c | 1 - sound/atmel/ac97c.c | 2 -- sound/drivers/aloop.c | 1 - sound/drivers/dummy.c | 2 -- sound/drivers/ml403-ac97cr.c | 2 -- sound/drivers/pcsp/pcsp_lib.c | 1 - sound/drivers/vx/vx_pcm.c | 2 -- sound/firewire/bebob/bebob_pcm.c | 2 -- sound/firewire/dice/dice-pcm.c | 2 -- sound/firewire/digi00x/digi00x-pcm.c | 2 -- sound/firewire/fireface/ff-pcm.c | 2 -- sound/firewire/fireworks/fireworks_pcm.c | 2 -- sound/firewire/isight.c | 1 - sound/firewire/motu/motu-pcm.c | 2 -- sound/firewire/oxfw/oxfw-pcm.c | 2 -- sound/firewire/tascam/tascam-pcm.c | 2 -- sound/isa/ad1816a/ad1816a_lib.c | 2 -- sound/isa/es1688/es1688_lib.c | 8 -------- sound/isa/es18xx.c | 2 -- sound/isa/gus/gus_pcm.c | 2 -- sound/isa/msnd/msnd.c | 2 -- sound/isa/sb/emu8000_pcm.c | 1 - sound/isa/sb/sb16_main.c | 2 -- sound/isa/sb/sb8_main.c | 2 -- sound/isa/wss/wss_lib.c | 2 -- sound/mips/hal2.c | 2 -- sound/mips/sgio2audio.c | 3 --- sound/parisc/harmony.c | 2 -- sound/pci/ad1889.c | 2 -- sound/pci/ali5451/ali5451.c | 4 ---- sound/pci/als300.c | 2 -- sound/pci/als4000.c | 2 -- sound/pci/asihpi/asihpi.c | 17 ----------------- sound/pci/atiixp.c | 3 --- sound/pci/atiixp_modem.c | 2 -- sound/pci/au88x0/au88x0_pcm.c | 1 - sound/pci/aw2/aw2-alsa.c | 2 -- sound/pci/azt3328.c | 3 --- sound/pci/bt87x.c | 1 - sound/pci/ca0106/ca0106_main.c | 8 -------- sound/pci/cmipci.c | 5 ----- sound/pci/cs4281.c | 2 -- sound/pci/cs46xx/cs46xx_lib.c | 10 ---------- sound/pci/cs5535audio/cs5535audio_pcm.c | 2 -- sound/pci/ctxfi/ctpcm.c | 2 -- sound/pci/echoaudio/echoaudio.c | 4 ---- sound/pci/emu10k1/emu10k1x.c | 2 -- sound/pci/emu10k1/emupcm.c | 6 ------ sound/pci/emu10k1/p16v.c | 2 -- sound/pci/ens1370.c | 3 --- sound/pci/es1938.c | 2 -- sound/pci/es1968.c | 2 -- sound/pci/fm801.c | 2 -- sound/pci/hda/hda_controller.c | 1 - sound/pci/ice1712/ice1712.c | 5 ----- sound/pci/ice1712/ice1724.c | 5 ----- sound/pci/intel8x0.c | 12 ------------ sound/pci/intel8x0m.c | 2 -- sound/pci/lola/lola_pcm.c | 1 - sound/pci/lx6464es/lx6464es.c | 2 -- sound/pci/maestro3.c | 2 -- sound/pci/mixart/mixart.c | 2 -- sound/pci/nm256/nm256.c | 2 -- sound/pci/oxygen/oxygen_pcm.c | 6 ------ sound/pci/pcxhr/pcxhr.c | 1 - sound/pci/riptide/riptide.c | 2 -- sound/pci/rme32.c | 8 -------- sound/pci/rme96.c | 4 ---- sound/pci/sis7019.c | 2 -- sound/pci/sonicvibes.c | 2 -- sound/pci/trident/trident_main.c | 31 ------------------------------- sound/pci/via82xx.c | 5 ----- sound/pci/via82xx_modem.c | 2 -- sound/pci/ymfpci/ymfpci_main.c | 5 ----- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 - sound/ppc/pmac.c | 2 -- sound/ppc/snd_ps3.c | 1 - sound/sh/aica.c | 1 - sound/sh/sh_dac_audio.c | 1 - sound/sparc/amd7930.c | 2 -- sound/sparc/cs4231.c | 2 -- sound/sparc/dbri.c | 1 - sound/spi/at73c213.c | 1 - sound/usb/6fire/pcm.c | 1 - sound/usb/caiaq/audio.c | 1 - sound/usb/hiface/pcm.c | 1 - sound/usb/line6/capture.c | 1 - sound/usb/line6/playback.c | 1 - sound/usb/misc/ua101.c | 2 -- sound/usb/pcm.c | 2 -- sound/usb/usx2y/usbusx2yaudio.c | 1 - sound/usb/usx2y/usx2yhwdeppcm.c | 1 - sound/x86/intel_hdmi_audio.c | 1 - sound/xen/xen_snd_front_alsa.c | 2 -- 96 files changed, 281 deletions(-) -- 2.16.4

2 26

[alsa-devel] [PATCH AUTOSEL 4.4 56/71] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 11 Dec '19

11 Dec '19

From: Ben Zhang <benzh(a)chromium.org> [ Upstream commit eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 ] The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 69d987a9935c9..90f8173123f6c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -295,6 +295,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_PLL1_CTRL2: -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 33/71] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ca58f55108fee41d87c9123f85ad4863e5de7f45 ] This is an alternative fix attemp for the issue reported in the commit caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was reverted later due to regressions. Instead of tweaking the hardware disablement order and the enforced irq flushing, do calling cancel_work_sync() of the unsol work early enough, and explicitly ignore the unsol events during the shutdown by checking the bus->shutdown flag. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 7ea201c05e5da..d0d6dfbfcfdf8 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -42,6 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ef8955abd9186..3e3277100f08a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1310,8 +1310,11 @@ static int azx_free(struct azx *chip) static int azx_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; + struct hdac_bus *bus = azx_bus(chip); chip->bus.shutdown = 1; + cancel_work_sync(&bus->unsol_work); + return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 69/91] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 11 Dec '19

11 Dec '19

From: Ben Zhang <benzh(a)chromium.org> [ Upstream commit eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 ] The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 65ac4518ad060..49ab26e69f2fc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -305,6 +305,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_ASRC_22: -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 50/91] ALSA: timer: Limit max amount of slave instances
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit fdea53fe5de532969a332d6e5e727f2ad8bf084d ] The fuzzer tries to open the timer instances as much as possible, and this may cause a system hiccup easily. We've already introduced the cap for the max number of available instances for the h/w timers, and we should put such a limit also to the slave timers, too. This patch introduces the limit to the multiple opened slave timers. The upper limit is hard-coded to 1000 for now, which should suffice for any practical usages up to now. Link: https://lore.kernel.org/r/20191106154257.5853-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index e944d27f79c34..f8a4b2a2f8f6b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -87,6 +87,9 @@ static LIST_HEAD(snd_timer_slave_list); /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); +#define MAX_SLAVE_INSTANCES 1000 +static int num_slaves; + static DEFINE_MUTEX(register_mutex); static int snd_timer_free(struct snd_timer *timer); @@ -265,6 +268,10 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EINVAL; goto unlock; } + if (num_slaves >= MAX_SLAVE_INSTANCES) { + err = -EBUSY; + goto unlock; + } timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { err = -ENOMEM; @@ -274,6 +281,7 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = tid->device; timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); + num_slaves++; err = snd_timer_check_slave(timeri); if (err < 0) { snd_timer_close_locked(timeri, &card_dev_to_put); @@ -363,6 +371,8 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri, struct snd_timer_instance *slave, *tmp; list_del(&timeri->open_list); + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + num_slaves--; /* force to stop the timer */ snd_timer_stop(timeri); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 43/91] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ca58f55108fee41d87c9123f85ad4863e5de7f45 ] This is an alternative fix attemp for the issue reported in the commit caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was reverted later due to regressions. Instead of tweaking the hardware disablement order and the enforced irq flushing, do calling cancel_work_sync() of the unsol work early enough, and explicitly ignore the unsol events during the shutdown by checking the bus->shutdown flag. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 7ea201c05e5da..d0d6dfbfcfdf8 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -42,6 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f2f1d9fd848c8..3234e9ca02cec 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1275,8 +1275,11 @@ static int azx_free(struct azx *chip) static int azx_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; + struct hdac_bus *bus = azx_bus(chip); chip->bus.shutdown = 1; + cancel_work_sync(&bus->unsol_work); + return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 104/130] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Sasha Levin 11 Dec '19

11 Dec '19

From: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> [ Upstream commit e2db787bdcb4f2722ecf410168f0583764634e45 ] On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20190923162940.199580-1-yuhsuan@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 41cb1fefbd427..405196283688c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -422,6 +422,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 103/130] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 11 Dec '19

11 Dec '19

From: Ben Zhang <benzh(a)chromium.org> [ Upstream commit eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 ] The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 1cd20b88a3a9c..82ee8f4b965be 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -297,6 +297,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_ASRC_22: -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 080/130] ALSA: timer: Limit max amount of slave instances
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit fdea53fe5de532969a332d6e5e727f2ad8bf084d ] The fuzzer tries to open the timer instances as much as possible, and this may cause a system hiccup easily. We've already introduced the cap for the max number of available instances for the h/w timers, and we should put such a limit also to the slave timers, too. This patch introduces the limit to the multiple opened slave timers. The upper limit is hard-coded to 1000 for now, which should suffice for any practical usages up to now. Link: https://lore.kernel.org/r/20191106154257.5853-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index c60dfd52e8a6d..22589a073423b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -88,6 +88,9 @@ static LIST_HEAD(snd_timer_slave_list); /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); +#define MAX_SLAVE_INSTANCES 1000 +static int num_slaves; + static DEFINE_MUTEX(register_mutex); static int snd_timer_free(struct snd_timer *timer); @@ -266,6 +269,10 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EINVAL; goto unlock; } + if (num_slaves >= MAX_SLAVE_INSTANCES) { + err = -EBUSY; + goto unlock; + } timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { err = -ENOMEM; @@ -275,6 +282,7 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = tid->device; timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); + num_slaves++; err = snd_timer_check_slave(timeri); if (err < 0) { snd_timer_close_locked(timeri, &card_dev_to_put); @@ -364,6 +372,8 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri, struct snd_timer_instance *slave, *tmp; list_del(&timeri->open_list); + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + num_slaves--; /* force to stop the timer */ snd_timer_stop(timeri); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 060/130] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ca58f55108fee41d87c9123f85ad4863e5de7f45 ] This is an alternative fix attemp for the issue reported in the commit caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was reverted later due to regressions. Instead of tweaking the hardware disablement order and the enforced irq flushing, do calling cancel_work_sync() of the unsol work early enough, and explicitly ignore the unsol events during the shutdown by checking the bus->shutdown flag. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 8db1890605f60..c175b2cf63f77 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -42,6 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 96e9b3944b925..890793ad85ca1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1450,8 +1450,11 @@ static int azx_free(struct azx *chip) static int azx_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; + struct hdac_bus *bus = azx_bus(chip); chip->bus.shutdown = 1; + cancel_work_sync(&bus->unsol_work); + return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 161/177] ASoC: Intel: bytcr_rt5640: Update quirk for Acer Switch 10 SW5-012 2-in-1
by Sasha Levin 10 Dec '19

10 Dec '19

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 0bb887709eb16bdc4b5baddd8337abf3de72917f ] When the Acer Switch 10 SW5-012 quirk was added we did not have jack-detection support yet; and the builtin microphone selection of the original quirk is wrong too. Fix the microphone-input quirk and add jack-detection info so that the internal-microphone and headphone/set jack on the Switch 10 work properly. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191119145138.59162-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b6dc524830b21..6acd5dd599dc4 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -414,10 +414,12 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), DMI_MATCH(DMI_PRODUCT_NAME, "Aspire SW5-012"), }, - .driver_data = (void *)(BYT_RT5640_IN1_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), - + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { .matches = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 160/177] ASoC: wm5100: add missed pm_runtime_disable
by Sasha Levin 10 Dec '19

10 Dec '19

From: Chuhong Yuan <hslester96(a)gmail.com> [ Upstream commit b1176bbb70866f24099cd2720283c7219fb4a81c ] The driver forgets to call pm_runtime_disable in remove and probe failure. Add the calls to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20191118073707.28298-1-hslester96@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index ba89d9d711f76..b793701aafcda 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2620,6 +2620,7 @@ static int wm5100_i2c_probe(struct i2c_client *i2c, return ret; err_reset: + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); @@ -2643,6 +2644,7 @@ static int wm5100_i2c_remove(struct i2c_client *i2c) { struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 158/177] ASoC: wm2200: add missed operations in remove and probe failure
by Sasha Levin 10 Dec '19

10 Dec '19

From: Chuhong Yuan <hslester96(a)gmail.com> [ Upstream commit 2dab09be49a1e7a4dd13cb47d3a1441a2ef33a87 ] This driver misses calls to pm_runtime_disable and regulator_bulk_disable in remove and a call to free_irq in probe failure. Add the calls to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> Link: https://lore.kernel.org/r/20191118073633.28237-1-hslester96@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm2200.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index deff651615042..0a3b746fb909b 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2413,6 +2413,8 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); + if (i2c->irq) + free_irq(i2c->irq, wm2200); err_reset: if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); @@ -2429,12 +2431,15 @@ static int wm2200_i2c_remove(struct i2c_client *i2c) { struct wm2200_priv *wm2200 = i2c_get_clientdata(i2c); + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 151/177] ASoC: wm8904: fix regcache handling
by Sasha Levin 10 Dec '19

10 Dec '19

From: Michael Walle <michael(a)walle.cc> [ Upstream commit e9149b8c00d25dbaef1aa174fc604bed207e576d ] The current code assumes that the power is turned off in SND_SOC_BIAS_OFF. If there are no actual regulator the codec isn't turned off and the registers are not reset to their default values but the regcache is still marked as dirty. Thus a value might not be written to the hardware if it is set to the default value. Do a software reset before turning off the power to make sure the registers are always reset to their default states. Signed-off-by: Michael Walle <michael(a)walle.cc> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20191112223629.21867-1-michael@walle.cc Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 1965635ec07c7..d14e851b91605 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1902,6 +1902,7 @@ static int wm8904_set_bias_level(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); + snd_soc_component_write(component, WM8904_SW_RESET_AND_ID, 0); regcache_cache_only(wm8904->regmap, true); regcache_mark_dirty(wm8904->regmap); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 138/177] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Sasha Levin 10 Dec '19

10 Dec '19

From: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> [ Upstream commit e2db787bdcb4f2722ecf410168f0583764634e45 ] On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20190923162940.199580-1-yuhsuan@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index b8a03f58ac8cc..f36e33a147287 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -423,6 +423,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 136/177] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 10 Dec '19

10 Dec '19

From: Ben Zhang <benzh(a)chromium.org> [ Upstream commit eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 ] The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 9b7a1833d3316..71b7b881df396 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -297,6 +297,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_ASRC_22: -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 130/177] soundwire: intel: fix PDI/stream mapping for Bulk
by Sasha Levin 10 Dec '19

10 Dec '19

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit c134f914e9f55b7817e2bae625ec0e5f1379f7cd ] The previous formula is incorrect for PDI0/1, the mapping is not linear but has a discontinuity between PDI1 and PDI2. This change has no effect on PCM PDIs (same mapping). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191022232948.17156-1-pierre-louis.bossart@linux… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 29bc99c4a7b66..e49d3c810677a 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -352,7 +352,10 @@ intel_pdi_shim_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; int pdi_conf = 0; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* * Program stream parameters to stream SHIM register @@ -381,7 +384,10 @@ intel_pdi_alh_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; unsigned int conf; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* Program Stream config ALH register */ conf = intel_readl(alh, SDW_ALH_STRMZCFG(pdi->intel_alh_id)); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 106/177] ALSA: timer: Limit max amount of slave instances
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit fdea53fe5de532969a332d6e5e727f2ad8bf084d ] The fuzzer tries to open the timer instances as much as possible, and this may cause a system hiccup easily. We've already introduced the cap for the max number of available instances for the h/w timers, and we should put such a limit also to the slave timers, too. This patch introduces the limit to the multiple opened slave timers. The upper limit is hard-coded to 1000 for now, which should suffice for any practical usages up to now. Link: https://lore.kernel.org/r/20191106154257.5853-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index 86a31e69fc7d9..b5dc51030316a 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -88,6 +88,9 @@ static LIST_HEAD(snd_timer_slave_list); /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); +#define MAX_SLAVE_INSTANCES 1000 +static int num_slaves; + static DEFINE_MUTEX(register_mutex); static int snd_timer_free(struct snd_timer *timer); @@ -266,6 +269,10 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EINVAL; goto unlock; } + if (num_slaves >= MAX_SLAVE_INSTANCES) { + err = -EBUSY; + goto unlock; + } timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { err = -ENOMEM; @@ -275,6 +282,7 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = tid->device; timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); + num_slaves++; err = snd_timer_check_slave(timeri); if (err < 0) { snd_timer_close_locked(timeri, &card_dev_to_put); @@ -364,6 +372,8 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri, struct snd_timer_instance *slave, *tmp; list_del(&timeri->open_list); + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + num_slaves--; /* force to stop the timer */ snd_timer_stop(timeri); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 086/177] ALSA: hda/hdmi - implement mst_no_extra_pcms flag
by Sasha Levin 10 Dec '19

10 Dec '19

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 2a2edfbbfee47947dd05f5860c66c0e80ee5e09d ] To support the DP-MST multiple streams via single connector feature, the HDMI driver was extended with the concept of backup PCMs. See commit 9152085defb6 ("ALSA: hda - add DP MST audio support"). This implementation works fine with snd_hda_intel.c as PCM topology is fully managed within the single driver. When the HDA codec driver is used from ASoC components, the concept of backup PCMs no longer fits. For ASoC topologies, the physical HDMI converters are presented as backend DAIs and these should match with hardware capabilities. The ASoC topology may define arbitrary PCMs (i.e. frontend DAIs) and have processing elements before eventual routing to the HDMI BE DAIs. With backup PCMs, the link between FE and BE DAIs would become dynamic and change when monitors are (un)plugged. This would lead to modifying the topology every time hotplug events happen, which is not currently possible in ASoC and there does not seem to be any obvious benefits from this design. To overcome above problems and enable the HDMI driver to be used from ASoC, this patch adds a new mode (mst_no_extra_pcms flags) to patch_hdmi.c. In this mode, the codec driver does not assume the backup PCMs to be created. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191029134017.18901-2-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 19 ++++++++++++++----- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2003403ce1c82..199927694aeff 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -262,6 +262,7 @@ struct hda_codec { unsigned int force_pin_prefix:1; /* Add location prefix */ unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ + unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c827a2a89cc3d..9d5e3c8d62b93 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2063,15 +2063,24 @@ static bool is_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx) static int generic_hdmi_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - int idx; + int idx, pcm_num; /* * for non-mst mode, pcm number is the same as before - * for DP MST mode, pcm number is (nid number + dev_num - 1) - * dev_num is the device entry number in a pin - * + * for DP MST mode without extra PCM, pcm number is same + * for DP MST mode with extra PCMs, pcm number is + * (nid number + dev_num - 1) + * dev_num is the device entry number in a pin */ - for (idx = 0; idx < spec->num_nids + spec->dev_num - 1; idx++) { + + if (codec->mst_no_extra_pcms) + pcm_num = spec->num_nids; + else + pcm_num = spec->num_nids + spec->dev_num - 1; + + codec_dbg(codec, "hdmi: pcm_num set to %d\n", pcm_num); + + for (idx = 0; idx < pcm_num; idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 082/177] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ca58f55108fee41d87c9123f85ad4863e5de7f45 ] This is an alternative fix attemp for the issue reported in the commit caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was reverted later due to regressions. Instead of tweaking the hardware disablement order and the enforced irq flushing, do calling cancel_work_sync() of the unsol work early enough, and explicitly ignore the unsol events during the shutdown by checking the bus->shutdown flag. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 8db1890605f60..c175b2cf63f77 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -42,6 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0b24c5ce2fd6a..d63fea5d1c929 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1492,8 +1492,11 @@ static int azx_free(struct azx *chip) static int azx_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; + struct hdac_bus *bus = azx_bus(chip); chip->bus.shutdown = 1; + cancel_work_sync(&bus->unsol_work); + return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 322/350] ASoC: soc-pcm: check symmetry before hw_params
by Sasha Levin 10 Dec '19

10 Dec '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 5cca59516de5df9de6bdecb328dd55fb5bcccb41 ] This reverts commit 957ce0c6b8a1f (ASoC: soc-pcm: check symmetry after hw_params). That commit cause soc_pcm_params_symmetry can't take effect. cpu_dai->rate, cpu_dai->channels and cpu_dai->sample_bits are updated in the middle of soc_pcm_hw_params, so move soc_pcm_params_symmetry to the end of soc_pcm_hw_params is not a good solution, for judgement of symmetry in the function is always true. FIXME: According to the comments of that commit, I think the case described in the commit should disable symmetric_rates in Back-End, rather than changing the position of soc_pcm_params_symmetry. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1573555602-5403-1-git-send-email-shengjiu.wang@nx… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cdce96a3051bf..a6e96cf1d8ff9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -877,6 +877,11 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto out; + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -958,9 +963,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } component = NULL; - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto component_err; out: mutex_unlock(&rtd->card->pcm_mutex); return ret; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 311/350] ASoC: Intel: bytcr_rt5640: Update quirk for Acer Switch 10 SW5-012 2-in-1
by Sasha Levin 10 Dec '19

10 Dec '19

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 0bb887709eb16bdc4b5baddd8337abf3de72917f ] When the Acer Switch 10 SW5-012 quirk was added we did not have jack-detection support yet; and the builtin microphone selection of the original quirk is wrong too. Fix the microphone-input quirk and add jack-detection info so that the internal-microphone and headphone/set jack on the Switch 10 work properly. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191119145138.59162-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9c1aa4ec9cba4..dd2b5ad086591 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -405,10 +405,12 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), DMI_MATCH(DMI_PRODUCT_NAME, "Aspire SW5-012"), }, - .driver_data = (void *)(BYT_RT5640_IN1_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), - + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { .matches = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 304/350] ASoC: wm5100: add missed pm_runtime_disable
by Sasha Levin 10 Dec '19

10 Dec '19

From: Chuhong Yuan <hslester96(a)gmail.com> [ Upstream commit b1176bbb70866f24099cd2720283c7219fb4a81c ] The driver forgets to call pm_runtime_disable in remove and probe failure. Add the calls to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20191118073707.28298-1-hslester96@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4af0e519e6235..91cc63c5a51fa 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2617,6 +2617,7 @@ static int wm5100_i2c_probe(struct i2c_client *i2c, return ret; err_reset: + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); @@ -2640,6 +2641,7 @@ static int wm5100_i2c_remove(struct i2c_client *i2c) { struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 302/350] ASoC: wm2200: add missed operations in remove and probe failure
by Sasha Levin 10 Dec '19

10 Dec '19

From: Chuhong Yuan <hslester96(a)gmail.com> [ Upstream commit 2dab09be49a1e7a4dd13cb47d3a1441a2ef33a87 ] This driver misses calls to pm_runtime_disable and regulator_bulk_disable in remove and a call to free_irq in probe failure. Add the calls to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> Link: https://lore.kernel.org/r/20191118073633.28237-1-hslester96@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm2200.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index cf64e109c6587..7b087d94141bd 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2410,6 +2410,8 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); + if (i2c->irq) + free_irq(i2c->irq, wm2200); err_reset: if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); @@ -2426,12 +2428,15 @@ static int wm2200_i2c_remove(struct i2c_client *i2c) { struct wm2200_priv *wm2200 = i2c_get_clientdata(i2c); + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 292/350] ASoC: wm8904: fix regcache handling
by Sasha Levin 10 Dec '19

10 Dec '19

From: Michael Walle <michael(a)walle.cc> [ Upstream commit e9149b8c00d25dbaef1aa174fc604bed207e576d ] The current code assumes that the power is turned off in SND_SOC_BIAS_OFF. If there are no actual regulator the codec isn't turned off and the registers are not reset to their default values but the regcache is still marked as dirty. Thus a value might not be written to the hardware if it is set to the default value. Do a software reset before turning off the power to make sure the registers are always reset to their default states. Signed-off-by: Michael Walle <michael(a)walle.cc> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20191112223629.21867-1-michael@walle.cc Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index bcb3c9d5abf0c..9e8c564f6e9c4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1917,6 +1917,7 @@ static int wm8904_set_bias_level(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); + snd_soc_component_write(component, WM8904_SW_RESET_AND_ID, 0); regcache_cache_only(wm8904->regmap, true); regcache_mark_dirty(wm8904->regmap); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 269/350] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Sasha Levin 10 Dec '19

10 Dec '19

From: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> [ Upstream commit e2db787bdcb4f2722ecf410168f0583764634e45 ] On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20190923162940.199580-1-yuhsuan@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a0..67b276a65a8d2 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -400,6 +400,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 265/350] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 10 Dec '19

10 Dec '19

From: Ben Zhang <benzh(a)chromium.org> [ Upstream commit eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 ] The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 315a3d39bc091..8bc9450da79cf 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -298,6 +298,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_ASRC_22: -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 253/350] soundwire: intel: fix PDI/stream mapping for Bulk
by Sasha Levin 10 Dec '19

10 Dec '19

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit c134f914e9f55b7817e2bae625ec0e5f1379f7cd ] The previous formula is incorrect for PDI0/1, the mapping is not linear but has a discontinuity between PDI1 and PDI2. This change has no effect on PCM PDIs (same mapping). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191022232948.17156-1-pierre-louis.bossart@linux… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 13c54eac0cc3e..d1839707128a8 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -479,7 +479,10 @@ intel_pdi_shim_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; int pdi_conf = 0; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* * Program stream parameters to stream SHIM register @@ -508,7 +511,10 @@ intel_pdi_alh_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; unsigned int conf; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* Program Stream config ALH register */ conf = intel_readl(alh, SDW_ALH_STRMZCFG(pdi->intel_alh_id)); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 214/350] ALSA: timer: Limit max amount of slave instances
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit fdea53fe5de532969a332d6e5e727f2ad8bf084d ] The fuzzer tries to open the timer instances as much as possible, and this may cause a system hiccup easily. We've already introduced the cap for the max number of available instances for the h/w timers, and we should put such a limit also to the slave timers, too. This patch introduces the limit to the multiple opened slave timers. The upper limit is hard-coded to 1000 for now, which should suffice for any practical usages up to now. Link: https://lore.kernel.org/r/20191106154257.5853-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index 59ae21b0bb936..013f0e69ff0f7 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -74,6 +74,9 @@ static LIST_HEAD(snd_timer_slave_list); /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); +#define MAX_SLAVE_INSTANCES 1000 +static int num_slaves; + static DEFINE_MUTEX(register_mutex); static int snd_timer_free(struct snd_timer *timer); @@ -252,6 +255,10 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EINVAL; goto unlock; } + if (num_slaves >= MAX_SLAVE_INSTANCES) { + err = -EBUSY; + goto unlock; + } timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { err = -ENOMEM; @@ -261,6 +268,7 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = tid->device; timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); + num_slaves++; err = snd_timer_check_slave(timeri); if (err < 0) { snd_timer_close_locked(timeri, &card_dev_to_put); @@ -356,6 +364,8 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri, } list_del(&timeri->open_list); + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + num_slaves--; /* force to stop the timer */ snd_timer_stop(timeri); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 208/350] ASoC: SOF: topology: set trigger order for FE DAI link
by Sasha Levin 10 Dec '19

10 Dec '19

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit 5eee2b3f60065a2530d13f28e771be48b989eb4c ] Set trigger order for FE DAI links to SND_SOC_DPCM_TRIGGER_POST to trigger the BE DAI's before the FE DAI's. This prevents the xruns seen on playback pipelines using the link DMA. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191104224812.3393-3-ranjani.sridharan@linux.int… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/topology.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 4452594c2e17a..fa299e0781561 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2828,6 +2828,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, if (!link->no_pcm) { link->nonatomic = true; + /* set trigger order */ + link->trigger[0] = SND_SOC_DPCM_TRIGGER_POST; + link->trigger[1] = SND_SOC_DPCM_TRIGGER_POST; + /* nothing more to do for FE dai links */ return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 195/350] ALSA: pcm: Fix missing check of the new non-cached buffer type
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 6111fd2370eecae9f11bfdc08ba097e0b51fcfd3 ] The check for the mmap support via hw_support_mmap() function misses the case where the device is with SNDRV_DMA_TYPE_DEV_UC, which should have been treated equally as SNDRV_DMA_TYPE_DEV. Let's fix it. Note that this bug doesn't hit any practical problem, because SNDRV_DMA_TYPE_DEV_UC is used only for x86-specific drivers (snd-hda-intel and snd-intel8x0) for the specific platforms that need the non-cached buffers. And, on such platforms, hw_support_mmap() already returns true in anyway. That's the reason I didn't put Cc-to-stable mark here. This is only for any theoretical future extension. Fixes: 425da159707b ("ALSA: pcm: use dma_can_mmap() to check if a device supports dma_mmap_*") Fixes: 42e748a0b325 ("ALSA: memalloc: Add non-cached buffer type") Link: https://lore.kernel.org/r/20191104101115.27311-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 91c6ad58729fe..c3a139436ac26 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -222,7 +222,8 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) return false; if (substream->ops->mmap || - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV) + (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && + substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) return true; return dma_can_mmap(substream->dma_buffer.dev.dev); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 191/350] ALSA: bebob: expand sleep just after breaking connections for protocol version 1
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d3eabe939aee3ffd5b133766a932629a9746298c ] As long as I investigated, some devices with BeBoB protocol version 1 can be freezed during several hundreds milliseconds after breaking connections. When accessing during the freezed time, any transaction is corrupted. In the worst case, the device is going to reboot. I can see this issue in: * Roland FA-66 * M-Audio FireWire Solo This commit expands sleep just after breaking connections to avoid the freezed time as much as possible. I note that the freeze/reboot behaviour is similar to below models: * Focusrite Saffire Pro 10 I/O * Focusrite Saffire Pro 26 I/O The above models certainly reboot after breaking connections. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20191101131323.17300-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/bebob/bebob_stream.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 6c1497d9f52ba..ce07ea0d4e71d 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -415,15 +415,16 @@ static int make_both_connections(struct snd_bebob *bebob) return 0; } -static void -break_both_connections(struct snd_bebob *bebob) +static void break_both_connections(struct snd_bebob *bebob) { cmp_connection_break(&bebob->in_conn); cmp_connection_break(&bebob->out_conn); - /* These models seems to be in transition state for a longer time. */ - if (bebob->maudio_special_quirk != NULL) - msleep(200); + // These models seem to be in transition state for a longer time. When + // accessing in the state, any transactions is corrupted. In the worst + // case, the device is going to reboot. + if (bebob->version < 2) + msleep(600); } static int -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 179/350] ALSA: hda/hdmi - implement mst_no_extra_pcms flag
by Sasha Levin 10 Dec '19

10 Dec '19

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 2a2edfbbfee47947dd05f5860c66c0e80ee5e09d ] To support the DP-MST multiple streams via single connector feature, the HDMI driver was extended with the concept of backup PCMs. See commit 9152085defb6 ("ALSA: hda - add DP MST audio support"). This implementation works fine with snd_hda_intel.c as PCM topology is fully managed within the single driver. When the HDA codec driver is used from ASoC components, the concept of backup PCMs no longer fits. For ASoC topologies, the physical HDMI converters are presented as backend DAIs and these should match with hardware capabilities. The ASoC topology may define arbitrary PCMs (i.e. frontend DAIs) and have processing elements before eventual routing to the HDMI BE DAIs. With backup PCMs, the link between FE and BE DAIs would become dynamic and change when monitors are (un)plugged. This would lead to modifying the topology every time hotplug events happen, which is not currently possible in ASoC and there does not seem to be any obvious benefits from this design. To overcome above problems and enable the HDMI driver to be used from ASoC, this patch adds a new mode (mst_no_extra_pcms flags) to patch_hdmi.c. In this mode, the codec driver does not assume the backup PCMs to be created. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191029134017.18901-2-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 19 ++++++++++++++----- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 9a0393cf024c2..ac18f428eda6c 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -254,6 +254,7 @@ struct hda_codec { unsigned int force_pin_prefix:1; /* Add location prefix */ unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ + unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d14f6684737d0..b8579cd218098 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2075,15 +2075,24 @@ static bool is_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx) static int generic_hdmi_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - int idx; + int idx, pcm_num; /* * for non-mst mode, pcm number is the same as before - * for DP MST mode, pcm number is (nid number + dev_num - 1) - * dev_num is the device entry number in a pin - * + * for DP MST mode without extra PCM, pcm number is same + * for DP MST mode with extra PCMs, pcm number is + * (nid number + dev_num - 1) + * dev_num is the device entry number in a pin */ - for (idx = 0; idx < spec->num_nids + spec->dev_num - 1; idx++) { + + if (codec->mst_no_extra_pcms) + pcm_num = spec->num_nids; + else + pcm_num = spec->num_nids + spec->dev_num - 1; + + codec_dbg(codec, "hdmi: pcm_num set to %d\n", pcm_num); + + for (idx = 0; idx < pcm_num; idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 170/350] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ca58f55108fee41d87c9123f85ad4863e5de7f45 ] This is an alternative fix attemp for the issue reported in the commit caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was reverted later due to regressions. Instead of tweaking the hardware disablement order and the enforced irq flushing, do calling cancel_work_sync() of the unsol work early enough, and explicitly ignore the unsol events during the shutdown by checking the bus->shutdown flag. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 8272b50b83493..6a85645663759 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -43,6 +43,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c52419376c748..86a416cdeb296 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1382,8 +1382,11 @@ static int azx_free(struct azx *chip) static int azx_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; + struct hdac_bus *bus = azx_bus(chip); chip->bus.shutdown = 1; + cancel_work_sync(&bus->unsol_work); + return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 138/350] ASoC: soc-pcm: fixup dpcm_prune_paths() loop continue
by Sasha Levin 10 Dec '19

10 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit bed646dc3f7bcec91178c278deaf969cce0700a8 ] dpcm_prune_paths() is checking widget at 2 parts. (A) is for CPU, (B) is for Codec. If we focus to (A) part, continue at (a) is for (1) loop. But, if we focus to (B) part, continue at (b) is for (2) loop, not for (1). This is bug. This patch fixup this issue. static int dpcm_prune_paths(...) { ... (1) for_each_dpcm_be(fe, stream, dpcm) { ... ^ widget = dai_get_widget(...); | (A) if (widget && widget_in_list(...)) | (a) continue; v ^ (2) for_each_rtd_codec_dai(...) { | widget = dai_get_widget(...); (B) | if (widget && widget_in_list(...)) v (b) continue; } ... Fixes: 2e5894d73789 ("ASoC: pcm: Add support for DAI multicodec") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87blui64mf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b600d3eaaf5cd..cdce96a3051bf 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1385,6 +1385,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget *widget; struct snd_soc_dai *dai; int prune = 0; + int do_prune; /* Destroy any old FE <--> BE connections */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1398,13 +1399,16 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* is there a valid CODEC DAI widget for this BE */ + do_prune = 1; for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) - continue; + do_prune = 0; } + if (!do_prune) + continue; dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", stream ? "capture" : "playback", -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 088/350] ASoC: SOF: enable sync_write in hdac_bus
by Sasha Levin 10 Dec '19

10 Dec '19

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit f3416e7144f5d4ba0fc5dcef6ebfff891266c46a ] Align SOF HDA implementation with snd-hda-intel driver and enable sync_write flag for all supported Intel platforms in SOF. When set, a sync is issued after each verb write. Sync after write has helped to overcome intermittent delays in system resume flow on Intel Coffee Lake systems, and most recently probe errors related to the HDMI codec on Ice Lake systems. Matches the snd-hda-intel driver change done in commit 2756d9143aa5 ("ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips"). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191008164443.1358-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 06e84679087bc..5a5163eef2ef4 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -268,6 +268,7 @@ static int hda_init(struct snd_sof_dev *sdev) bus->use_posbuf = 1; bus->bdl_pos_adj = 0; + bus->sync_write = 1; mutex_init(&hbus->prepare_mutex); hbus->pci = pci; -- 2.20.1

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Re: [alsa-devel] [PATCH v3 03/15] soc: tegra: Add Tegra PMC clock registrations into PMC driver
by Dmitry Osipenko 10 Dec '19

10 Dec '19

10.12.2019 22:18, Sowjanya Komatineni пишет: > > On 12/10/19 10:30 AM, Dmitry Osipenko wrote: >> 10.12.2019 20:48, Sowjanya Komatineni пишет: >>> On 12/10/19 9:41 AM, Dmitry Osipenko wrote: >>>> 09.12.2019 23:46, Sowjanya Komatineni пишет: >>>>> On 12/9/19 12:12 PM, Dmitry Osipenko wrote: >>>>>> 08.12.2019 00:36, Sowjanya Komatineni пишет: >>>>>>> On 12/7/19 11:59 AM, Sowjanya Komatineni wrote: >>>>>>>> On 12/7/19 8:00 AM, Dmitry Osipenko wrote: >>>>>>>>> 07.12.2019 18:53, Dmitry Osipenko пишет: >>>>>>>>>> 07.12.2019 18:47, Dmitry Osipenko пишет: >>>>>>>>>>> 07.12.2019 17:28, Dmitry Osipenko пишет: >>>>>>>>>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>>>>>>>>> Tegra210 and prior Tegra PMC has clk_out_1, clk_out_2, >>>>>>>>>>>>> clk_out_3 >>>>>>>>>>>>> with >>>>>>>>>>>>> mux and gate for each of these clocks. >>>>>>>>>>>>> >>>>>>>>>>>>> Currently these PMC clocks are registered by Tegra clock >>>>>>>>>>>>> driver >>>>>>>>>>>>> using >>>>>>>>>>>>> clk_register_mux and clk_register_gate by passing PMC base >>>>>>>>>>>>> address >>>>>>>>>>>>> and register offsets and PMC programming for these clocks >>>>>>>>>>>>> happens >>>>>>>>>>>>> through direct PMC access by the clock driver. >>>>>>>>>>>>> >>>>>>>>>>>>> With this, when PMC is in secure mode any direct PMC access >>>>>>>>>>>>> from the >>>>>>>>>>>>> non-secure world does not go through and these clocks will >>>>>>>>>>>>> not be >>>>>>>>>>>>> functional. >>>>>>>>>>>>> >>>>>>>>>>>>> This patch adds these clocks registration with PMC as a clock >>>>>>>>>>>>> provider >>>>>>>>>>>>> for these clocks. clk_ops callback implementations for these >>>>>>>>>>>>> clocks >>>>>>>>>>>>> uses tegra_pmc_readl and tegra_pmc_writel which supports PMC >>>>>>>>>>>>> programming >>>>>>>>>>>>> in secure mode and non-secure mode. >>>>>>>>>>>>> >>>>>>>>>>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>>>>>>>>>> --- >>>>>>>>>>> [snip] >>>>>>>>>>> >>>>>>>>>>>>> + >>>>>>>>>>>>> +static const struct clk_ops pmc_clk_gate_ops = { >>>>>>>>>>>>> + .is_enabled = pmc_clk_is_enabled, >>>>>>>>>>>>> + .enable = pmc_clk_enable, >>>>>>>>>>>>> + .disable = pmc_clk_disable, >>>>>>>>>>>>> +}; >>>>>>>>>>>> What's the benefit of separating GATE from the MUX? >>>>>>>>>>>> >>>>>>>>>>>> I think it could be a single clock. >>>>>>>>>>> According to TRM: >>>>>>>>>>> >>>>>>>>>>> 1. GATE and MUX are separate entities. >>>>>>>>>>> >>>>>>>>>>> 2. GATE is the parent of MUX (see PMC's CLK_OUT paths diagram in >>>>>>>>>>> TRM). >>>>>>>>>>> >>>>>>>>>>> 3. PMC doesn't gate EXTPERIPH clock but could "force-enable" it, >>>>>>>>>>> correct? >>>>>>> Was following existing clk-tegra-pmc as I am not sure of reason for >>>>>>> having these clocks registered as separate mux and gate clocks. >>>>>>> >>>>>>> Yes, PMC clocks can be registered as single clock and can use >>>>>>> clk_ops >>>>>>> for set/get parent and enable/disable. >>>>>>> >>>>>>> enable/disable of PMC clocks is for force-enable to force the >>>>>>> clock to >>>>>>> run regardless of ACCEPT_REQ or INVERT_REQ. >>>>>>> >>>>>>>>>> 4. clk_m_div2/4 are internal PMC OSC dividers and thus these >>>>>>>>>> clocks >>>>>>>>>> should belong to PMC. >>>>>>>>> Also, it should be "osc" and not "clk_m". >>>>>>>> I followed the same parents as it were in existing clk-tegra-pmc >>>>>>>> driver. >>>>>>>> >>>>>>>> Yeah they are wrong and they should be from osc and not clk_m. >>>>>>>> >>>>>>>> Will fix in next version. >>>>>>>> >>>>>> Could you please describe the full EXTPERIPH clock topology and >>>>>> how the >>>>>> pinmux configuration is related to it all? >>>>>> >>>>>> What is internal to the Tegra chip and what are the external outputs? >>>>>> >>>>>> Is it possible to bypass PMC on T30+ for the EXTPERIPH clocks? >>>>> PMC CLK1/2/3 possible sources are OSC_DIV1, OSC_DIV2, OSC_DIV4, >>>>> EXTPERIPH from CAR. >>>>> >>>>> OSC_DIV1/2/4 are with internal dividers at the OSC Pads >>>>> >>>>> EXTPERIPH is from CAR and it has reset and enable controls along with >>>>> clock source selections to choose one of the PLLA_OUT0, CLK_S, >>>>> PLLP_OUT0, CLK_M, PLLE_OUT0 >>>> Are you sure that EXTPERIPH has a reset? What will it reset? Why it's >>>> not documented in TRM? >>> Yes, Extperiph1/2/3 has RST part of CAR RST_DEVICES_V bits 24/25/26 >> Are these bits not documented in a public TRMs? I checked >> T30/114/124/210 TRMs and CLK_RST_CONTROLLER_RST_DEVICES_V_0 doesn't have >> those bits in the docs. >> > Yeah these bits are missing in all Tegra TRM docs. Will request for > having EXTPERIPH reset bits to be updated in TRM... Thanks

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Re: [alsa-devel] [PATCH v3 03/15] soc: tegra: Add Tegra PMC clock registrations into PMC driver
by Dmitry Osipenko 10 Dec '19

10 Dec '19

10.12.2019 20:48, Sowjanya Komatineni пишет: > > On 12/10/19 9:41 AM, Dmitry Osipenko wrote: >> 09.12.2019 23:46, Sowjanya Komatineni пишет: >>> On 12/9/19 12:12 PM, Dmitry Osipenko wrote: >>>> 08.12.2019 00:36, Sowjanya Komatineni пишет: >>>>> On 12/7/19 11:59 AM, Sowjanya Komatineni wrote: >>>>>> On 12/7/19 8:00 AM, Dmitry Osipenko wrote: >>>>>>> 07.12.2019 18:53, Dmitry Osipenko пишет: >>>>>>>> 07.12.2019 18:47, Dmitry Osipenko пишет: >>>>>>>>> 07.12.2019 17:28, Dmitry Osipenko пишет: >>>>>>>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>>>>>>> Tegra210 and prior Tegra PMC has clk_out_1, clk_out_2, clk_out_3 >>>>>>>>>>> with >>>>>>>>>>> mux and gate for each of these clocks. >>>>>>>>>>> >>>>>>>>>>> Currently these PMC clocks are registered by Tegra clock driver >>>>>>>>>>> using >>>>>>>>>>> clk_register_mux and clk_register_gate by passing PMC base address >>>>>>>>>>> and register offsets and PMC programming for these clocks happens >>>>>>>>>>> through direct PMC access by the clock driver. >>>>>>>>>>> >>>>>>>>>>> With this, when PMC is in secure mode any direct PMC access >>>>>>>>>>> from the >>>>>>>>>>> non-secure world does not go through and these clocks will not be >>>>>>>>>>> functional. >>>>>>>>>>> >>>>>>>>>>> This patch adds these clocks registration with PMC as a clock >>>>>>>>>>> provider >>>>>>>>>>> for these clocks. clk_ops callback implementations for these >>>>>>>>>>> clocks >>>>>>>>>>> uses tegra_pmc_readl and tegra_pmc_writel which supports PMC >>>>>>>>>>> programming >>>>>>>>>>> in secure mode and non-secure mode. >>>>>>>>>>> >>>>>>>>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>>>>>>>> --- >>>>>>>>> [snip] >>>>>>>>> >>>>>>>>>>> + >>>>>>>>>>> +static const struct clk_ops pmc_clk_gate_ops = { >>>>>>>>>>> + .is_enabled = pmc_clk_is_enabled, >>>>>>>>>>> + .enable = pmc_clk_enable, >>>>>>>>>>> + .disable = pmc_clk_disable, >>>>>>>>>>> +}; >>>>>>>>>> What's the benefit of separating GATE from the MUX? >>>>>>>>>> >>>>>>>>>> I think it could be a single clock. >>>>>>>>> According to TRM: >>>>>>>>> >>>>>>>>> 1. GATE and MUX are separate entities. >>>>>>>>> >>>>>>>>> 2. GATE is the parent of MUX (see PMC's CLK_OUT paths diagram in >>>>>>>>> TRM). >>>>>>>>> >>>>>>>>> 3. PMC doesn't gate EXTPERIPH clock but could "force-enable" it, >>>>>>>>> correct? >>>>> Was following existing clk-tegra-pmc as I am not sure of reason for >>>>> having these clocks registered as separate mux and gate clocks. >>>>> >>>>> Yes, PMC clocks can be registered as single clock and can use clk_ops >>>>> for set/get parent and enable/disable. >>>>> >>>>> enable/disable of PMC clocks is for force-enable to force the clock to >>>>> run regardless of ACCEPT_REQ or INVERT_REQ. >>>>> >>>>>>>> 4. clk_m_div2/4 are internal PMC OSC dividers and thus these clocks >>>>>>>> should belong to PMC. >>>>>>> Also, it should be "osc" and not "clk_m". >>>>>> I followed the same parents as it were in existing clk-tegra-pmc >>>>>> driver. >>>>>> >>>>>> Yeah they are wrong and they should be from osc and not clk_m. >>>>>> >>>>>> Will fix in next version. >>>>>> >>>> Could you please describe the full EXTPERIPH clock topology and how the >>>> pinmux configuration is related to it all? >>>> >>>> What is internal to the Tegra chip and what are the external outputs? >>>> >>>> Is it possible to bypass PMC on T30+ for the EXTPERIPH clocks? >>> PMC CLK1/2/3 possible sources are OSC_DIV1, OSC_DIV2, OSC_DIV4, >>> EXTPERIPH from CAR. >>> >>> OSC_DIV1/2/4 are with internal dividers at the OSC Pads >>> >>> EXTPERIPH is from CAR and it has reset and enable controls along with >>> clock source selections to choose one of the PLLA_OUT0, CLK_S, >>> PLLP_OUT0, CLK_M, PLLE_OUT0 >> Are you sure that EXTPERIPH has a reset? What will it reset? Why it's >> not documented in TRM? > Yes, Extperiph1/2/3 has RST part of CAR RST_DEVICES_V bits 24/25/26 Are these bits not documented in a public TRMs? I checked T30/114/124/210 TRMs and CLK_RST_CONTROLLER_RST_DEVICES_V_0 doesn't have those bits in the docs. >>> So, PMC CLK1/2/4 possible parents are OSC_DIV1, OSC_DIV2, OSC_DIV4, EXTERN. >>> >>> >>> CLK1/2/3 also has Pinmux to route EXTPERIPH output on to these pins. >> Could you please clarify what are "these" pins? Perhaps you meant the >> EXTERN pin of PMC? > By CLK1/2/3 pins, I am referring to CLK_OUT_1/2/3 pins from Tegra I see now what you meant, thanks. [snip}

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Re: [alsa-devel] [PATCH v3 03/15] soc: tegra: Add Tegra PMC clock registrations into PMC driver
by Dmitry Osipenko 10 Dec '19

10 Dec '19

09.12.2019 23:46, Sowjanya Komatineni пишет: > > On 12/9/19 12:12 PM, Dmitry Osipenko wrote: >> 08.12.2019 00:36, Sowjanya Komatineni пишет: >>> On 12/7/19 11:59 AM, Sowjanya Komatineni wrote: >>>> On 12/7/19 8:00 AM, Dmitry Osipenko wrote: >>>>> 07.12.2019 18:53, Dmitry Osipenko пишет: >>>>>> 07.12.2019 18:47, Dmitry Osipenko пишет: >>>>>>> 07.12.2019 17:28, Dmitry Osipenko пишет: >>>>>>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>>>>>> Tegra210 and prior Tegra PMC has clk_out_1, clk_out_2, clk_out_3 >>>>>>>>> with >>>>>>>>> mux and gate for each of these clocks. >>>>>>>>> >>>>>>>>> Currently these PMC clocks are registered by Tegra clock driver >>>>>>>>> using >>>>>>>>> clk_register_mux and clk_register_gate by passing PMC base address >>>>>>>>> and register offsets and PMC programming for these clocks happens >>>>>>>>> through direct PMC access by the clock driver. >>>>>>>>> >>>>>>>>> With this, when PMC is in secure mode any direct PMC access >>>>>>>>> from the >>>>>>>>> non-secure world does not go through and these clocks will not be >>>>>>>>> functional. >>>>>>>>> >>>>>>>>> This patch adds these clocks registration with PMC as a clock >>>>>>>>> provider >>>>>>>>> for these clocks. clk_ops callback implementations for these >>>>>>>>> clocks >>>>>>>>> uses tegra_pmc_readl and tegra_pmc_writel which supports PMC >>>>>>>>> programming >>>>>>>>> in secure mode and non-secure mode. >>>>>>>>> >>>>>>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>>>>>> --- >>>>>>> [snip] >>>>>>> >>>>>>>>> + >>>>>>>>> +static const struct clk_ops pmc_clk_gate_ops = { >>>>>>>>> + .is_enabled = pmc_clk_is_enabled, >>>>>>>>> + .enable = pmc_clk_enable, >>>>>>>>> + .disable = pmc_clk_disable, >>>>>>>>> +}; >>>>>>>> What's the benefit of separating GATE from the MUX? >>>>>>>> >>>>>>>> I think it could be a single clock. >>>>>>> According to TRM: >>>>>>> >>>>>>> 1. GATE and MUX are separate entities. >>>>>>> >>>>>>> 2. GATE is the parent of MUX (see PMC's CLK_OUT paths diagram in >>>>>>> TRM). >>>>>>> >>>>>>> 3. PMC doesn't gate EXTPERIPH clock but could "force-enable" it, >>>>>>> correct? >>> Was following existing clk-tegra-pmc as I am not sure of reason for >>> having these clocks registered as separate mux and gate clocks. >>> >>> Yes, PMC clocks can be registered as single clock and can use clk_ops >>> for set/get parent and enable/disable. >>> >>> enable/disable of PMC clocks is for force-enable to force the clock to >>> run regardless of ACCEPT_REQ or INVERT_REQ. >>> >>>>>> 4. clk_m_div2/4 are internal PMC OSC dividers and thus these clocks >>>>>> should belong to PMC. >>>>> Also, it should be "osc" and not "clk_m". >>>> I followed the same parents as it were in existing clk-tegra-pmc >>>> driver. >>>> >>>> Yeah they are wrong and they should be from osc and not clk_m. >>>> >>>> Will fix in next version. >>>> >> Could you please describe the full EXTPERIPH clock topology and how the >> pinmux configuration is related to it all? >> >> What is internal to the Tegra chip and what are the external outputs? >> >> Is it possible to bypass PMC on T30+ for the EXTPERIPH clocks? > > PMC CLK1/2/3 possible sources are OSC_DIV1, OSC_DIV2, OSC_DIV4, > EXTPERIPH from CAR. > > OSC_DIV1/2/4 are with internal dividers at the OSC Pads > > EXTPERIPH is from CAR and it has reset and enable controls along with > clock source selections to choose one of the PLLA_OUT0, CLK_S, > PLLP_OUT0, CLK_M, PLLE_OUT0 Are you sure that EXTPERIPH has a reset? What will it reset? Why it's not documented in TRM? > So, PMC CLK1/2/4 possible parents are OSC_DIV1, OSC_DIV2, OSC_DIV4, EXTERN. > > > CLK1/2/3 also has Pinmux to route EXTPERIPH output on to these pins. Could you please clarify what are "these" pins? Perhaps you meant the EXTERN pin of PMC? > When EXTERN output clock is selected for these PMC clocks thru > CLKx_SRC_SEL, output clock is from driver by EXTPERIPH from CAR via > Pinmux logic or driven as per CLKx_SRC_SEL bypassing pinmux based on > CLKx_ACCEPT_REQ bit. > > > PMC Clock control register has bit CLKx_ACCEPT_REQ > When CLKx_ACCEPT_REQ = 0, output clock driver is from by EXTPERIPH > through the pinmux > When CLKx_ACCEPT_REQ = 1, output clock is based on CLKx_SRC_SEL bits > (OSC_DIV1/2/4 and EXTPERIPH clock bypassing the pinmux) > > FORCE_EN bit in PMC CLock control register forces the clock to run > regardless of this. Okay.

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[alsa-devel] [PATCH] ALSA: pcm: Fill silence on buffers at hw_params
by Takashi Iwai 10 Dec '19

10 Dec '19

The current PCM code doesn't initialize explicitly the buffers allocated for PCM streams, hence it might leak some uninitialized kernel data by mmapping or reading the buffer before actually reading or writing. Since this is a common problem, let's initialize the data on the buffers each hw_params properly. For that, an existing helper function is exposed as snd_pcm_fill_silence_frames() and call it from snd_pcm_hw_params(). Reported-and-tested-by: Lionel Koenig <lionel.koenig(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 13 ++++--------- sound/core/pcm_local.h | 2 ++ sound/core/pcm_native.c | 3 +++ 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e0c1f2..3c63324b8bb7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -30,9 +30,6 @@ #define trace_applptr(substream, prev, curr) #endif -static int fill_silence_frames(struct snd_pcm_substream *substream, - snd_pcm_uframes_t off, snd_pcm_uframes_t frames); - /* * fill ring buffer with silence * runtime->silence_start: starting pointer to silence area @@ -100,7 +97,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram ofs = runtime->silence_start % runtime->buffer_size; while (frames > 0) { transfer = ofs + frames > runtime->buffer_size ? runtime->buffer_size - ofs : frames; - err = fill_silence_frames(substream, ofs, transfer); + err = snd_pcm_fill_silence_frames(substream, ofs, transfer); snd_BUG_ON(err < 0); runtime->silence_filled += transfer; frames -= transfer; @@ -1945,8 +1942,6 @@ static int fill_silence(struct snd_pcm_substream *substream, int channel, { struct snd_pcm_runtime *runtime = substream->runtime; - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - return 0; if (substream->ops->fill_silence) return substream->ops->fill_silence(substream, channel, hwoff, bytes); @@ -2030,10 +2025,10 @@ static int noninterleaved_copy(struct snd_pcm_substream *substream, } /* fill silence on the given buffer position; - * called from snd_pcm_playback_silence() + * called from snd_pcm_playback_silence() and snd_pcm_hw_params() */ -static int fill_silence_frames(struct snd_pcm_substream *substream, - snd_pcm_uframes_t off, snd_pcm_uframes_t frames) +int snd_pcm_fill_silence_frames(struct snd_pcm_substream *substream, + snd_pcm_uframes_t off, snd_pcm_uframes_t frames) { if (substream->runtime->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED || substream->runtime->access == SNDRV_PCM_ACCESS_MMAP_INTERLEAVED) diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 384efd002984..ac4f455b1fff 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -34,6 +34,8 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr); +int snd_pcm_fill_silence_frames(struct snd_pcm_substream *substream, + snd_pcm_uframes_t off, snd_pcm_uframes_t frames); static inline snd_pcm_uframes_t snd_pcm_avail(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1fe581167b7b..d6f270c639b4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -739,6 +739,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) runtime->boundary *= 2; + /* clear the buffer once for avoiding information leaks */ + snd_pcm_fill_silence_frames(substream, 0, runtime->period_size); + snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); -- 2.16.4

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