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

[alsa-devel] [PATCH v2] ASoC: Intel: Power down links before turning off display audio power
by Pierre-Louis Bossart 13 Nov '18

13 Nov '18

On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. This patch was updated from an earlier version submitted upstream [1] which conflicted with the changes merged for HDaudio codec support with the Intel DSP. [1] https://patchwork.kernel.org/patch/10540213/ Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: fix copy/paste in error handling. sound/soc/codecs/hdac_hdmi.c | 11 +++++------ sound/soc/intel/skylake/skl.c | 12 ++++++------ 2 files changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e9854889a95..e63d6e33df48 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(dev, "Cannot turn on display power on i915\n"); - return err; - } hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { @@ -2201,7 +2196,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) + dev_err(dev, "Cannot turn off display power on i915\n"); + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index ebab04b51bc1..ac0b4ff21acc 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -811,6 +811,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -820,12 +826,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &bus->hlink_list, list) - snd_hdac_ext_bus_link_put(bus, hlink); - /* configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); -- 2.17.1

2 1

[alsa-devel] [PATCH 02/11] ASoC: AMD: add ACP3.0 PCI driver
by Vijendar Mukunda 13 Nov '18

13 Nov '18

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP 3.0 is a PCI audio device. This patch adds PCI driver to bind to this device and get PCI resources. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Signed-off-by: Sanju R Mehta <sanju.mehta(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 13 ++++++ sound/soc/amd/raven/pci-acp3x.c | 97 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 110 insertions(+) create mode 100644 sound/soc/amd/raven/acp3x.h create mode 100644 sound/soc/amd/raven/pci-acp3x.c diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h new file mode 100644 index 0000000..e9b4df0 --- /dev/null +++ b/sound/soc/amd/raven/acp3x.h @@ -0,0 +1,13 @@ +#include "chip_offset_byte.h" + +#define ACP3x_PHY_BASE_ADDRESS 0x1240000 + +static inline u32 rv_readl(void __iomem *base_addr) +{ + return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); +} + +static inline void rv_writel(u32 val, void __iomem *base_addr) +{ + writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); +} diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c new file mode 100644 index 0000000..27588ed --- /dev/null +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -0,0 +1,97 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2016 Advanced Micro Devices, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include <linux/pci.h> +#include <linux/module.h> +#include <linux/io.h> + +#include "acp3x.h" + +struct acp3x_dev_data { + void __iomem *acp3x_base; +}; + +static int snd_acp3x_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int ret; + u32 addr; + struct acp3x_dev_data *adata; + + if (pci_enable_device(pci)) { + dev_err(&pci->dev, "pci_enable_device failed\n"); + return -ENODEV; + } + + ret = pci_request_regions(pci, "AMD ACP3x audio"); + if (ret < 0) { + dev_err(&pci->dev, "pci_request_regions failed\n"); + goto disable_pci; + } + + adata = devm_kzalloc(&pci->dev, sizeof(struct acp3x_dev_data), + GFP_KERNEL); + if (!adata) { + ret = -ENOMEM; + goto release_regions; + } + + addr = pci_resource_start(pci, 0); + adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); + if (!adata->acp3x_base) { + ret = -ENOMEM; + goto release_regions; + } + pci_set_master(pci); + pci_set_drvdata(pci, adata); + return 0; + +release_regions: + pci_release_regions(pci); +disable_pci: + pci_disable_device(pci); + + return ret; +} + +static void snd_acp3x_remove(struct pci_dev *pci) +{ + struct acp3x_dev_data *adata = pci_get_drvdata(pci); + + iounmap(adata->acp3x_base); + pci_release_regions(pci); + pci_disable_device(pci); +} + +static const struct pci_device_id snd_acp3x_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_AMD, 0x15e2), + .class = PCI_CLASS_MULTIMEDIA_OTHER << 8, + .class_mask = 0xffffff }, + { 0, }, +}; +MODULE_DEVICE_TABLE(pci, snd_acp3x_ids); + +static struct pci_driver acp3x_driver = { + .name = KBUILD_MODNAME, + .id_table = snd_acp3x_ids, + .probe = snd_acp3x_probe, + .remove = snd_acp3x_remove, +}; + +module_pci_driver(acp3x_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP3x PCI driver"); +MODULE_LICENSE("GPL v2"); -- 2.7.4

2 1

[alsa-devel] [PATCH 03/11] ASoC: amd: create ACP3x PCM platform device
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP 3x IP has I2S controller device as one of IP blocks. Create a platform device for it, so that the PCM platform driver can be bound to this device. Pass PCI resources like MMIO, irq to the platform device. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <vishnuvardhanrao.ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 4 +++ sound/soc/amd/raven/pci-acp3x.c | 70 ++++++++++++++++++++++++++++++++++++++++- 2 files changed, 73 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index e9b4df0..83b1ed8 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -1,6 +1,10 @@ #include "chip_offset_byte.h" #define ACP3x_PHY_BASE_ADDRESS 0x1240000 +#define ACP3x_I2S_MODE 0 +#define ACP3x_REG_START 0x1240000 +#define ACP3x_REG_END 0x1250200 +#define I2S_MODE 0x04 static inline u32 rv_readl(void __iomem *base_addr) { diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 27588ed..0700162 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -16,19 +16,26 @@ #include <linux/pci.h> #include <linux/module.h> #include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> #include "acp3x.h" struct acp3x_dev_data { void __iomem *acp3x_base; + bool acp3x_audio_mode; + struct resource *res; + struct platform_device *pdev; }; static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr; + u32 addr, val; struct acp3x_dev_data *adata; + struct platform_device_info pdevinfo; + unsigned int irqflags; if (pci_enable_device(pci)) { dev_err(&pci->dev, "pci_enable_device failed\n"); @@ -48,6 +55,15 @@ static int snd_acp3x_probe(struct pci_dev *pci, goto release_regions; } + /* check for msi interrupt support */ + ret = pci_enable_msi(pci); + if (ret) + /* msi is not enabled */ + irqflags = IRQF_SHARED; + else + /* msi is enabled */ + irqflags = 0; + addr = pci_resource_start(pci, 0); adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); if (!adata->acp3x_base) { @@ -56,8 +72,57 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); + + val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); + switch (val) { + case I2S_MODE: + adata->res = devm_kzalloc(&pci->dev, + sizeof(struct resource) * 2, + GFP_KERNEL); + if (!adata->res) { + ret = -ENOMEM; + goto unmap_mmio; + } + + adata->res[0].name = "acp3x_i2s_iomem"; + adata->res[0].flags = IORESOURCE_MEM; + 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->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 (!adata->pdev) { + dev_err(&pci->dev, "cannot register %s device\n", + pdevinfo.name); + ret = -ENODEV; + goto unmap_mmio; + } + break; + default: + dev_err(&pci->dev, "Inavlid ACP audio mode : %d\n", val); + ret = -ENODEV; + goto unmap_mmio; + } return 0; +unmap_mmio: + pci_disable_msi(pci); + iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); disable_pci: @@ -70,7 +135,10 @@ static void snd_acp3x_remove(struct pci_dev *pci) { struct acp3x_dev_data *adata = pci_get_drvdata(pci); + platform_device_unregister(adata->pdev); iounmap(adata->acp3x_base); + + pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); } -- 2.7.4

2 1

[alsa-devel] [PATCH 04/11] ASoC: amd: add ACP3x PCM platform driver
by Vijendar Mukunda 13 Nov '18

13 Nov '18

PCM platform driver binds to the platform device created by ACP3x PCI device. PCM driver registers ALSA DMA and CPU DAI components with ASoC framework. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 263 ++++++++++++++++++++++++++++++++++++ sound/soc/amd/raven/acp3x.h | 7 + 2 files changed, 270 insertions(+) create mode 100644 sound/soc/amd/raven/acp3x-pcm-dma.c diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c new file mode 100644 index 0000000..3e806f7 --- /dev/null +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -0,0 +1,263 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2016 Advanced Micro Devices, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#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 "acp3x.h" + +#define DRV_NAME "acp3x-i2s-audio" + +struct i2s_dev_data { + void __iomem *acp3x_base; + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; +}; + +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 struct snd_pcm_ops acp3x_dma_ops = { + .open = NULL, + .close = NULL, + .ioctl = NULL, + .hw_params = NULL, + .hw_free = NULL, + .pointer = NULL, + .mmap = NULL, +}; + +struct snd_soc_dai_ops acp3x_dai_i2s_ops = { + .hw_params = NULL, + .trigger = NULL, + .set_fmt = NULL, +}; + +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, + .ops = &acp3x_dma_ops, + .pcm_new = acp3x_dma_new, +}; + +static int acp3x_audio_probe(struct platform_device *pdev) +{ + int status; + struct resource *res; + struct i2s_dev_data *adata; + unsigned int irqflags; + + if (!pdev->dev.platform_data) { + dev_err(&pdev->dev, "platform_data not retrieved\n"); + return -ENODEV; + } + irqflags = *((unsigned int *)(pdev->dev.platform_data)); + + adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), + GFP_KERNEL); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + return -ENODEV; + } + + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + + 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); + if (status) { + dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); + goto dev_err; + } + + 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; +} + +static int acp3x_audio_remove(struct platform_device *pdev) +{ + int ret; + struct i2s_dev_data *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"); + + return 0; +} + +static struct platform_driver acp3x_dma_driver = { + .probe = acp3x_audio_probe, + .remove = acp3x_audio_remove, + .driver = { + .name = "acp3x_rv_i2s", + }, +}; + +module_platform_driver(acp3x_dma_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_AUTHOR("Vijendar.Mukunda(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.h b/sound/soc/amd/raven/acp3x.h index 83b1ed8..f3d3e51 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -5,6 +5,13 @@ #define ACP3x_REG_START 0x1240000 #define ACP3x_REG_END 0x1250200 #define I2S_MODE 0x04 +#define BT_TX_THRESHOLD 26 +#define BT_RX_THRESHOLD 25 +#define ACP3x_POWER_ON 0x00 +#define ACP3x_POWER_ON_IN_PROGRESS 0x01 +#define ACP3x_POWER_OFF 0x02 +#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 static inline u32 rv_readl(void __iomem *base_addr) { -- 2.7.4

2 2

[alsa-devel] [PATCH 05/11] ASoC: amd: Interrupt handler changes for ACP3x DMA driver
by Vijendar Mukunda 13 Nov '18

13 Nov '18

Whenever audio data equal to the I2S FIFO watermark level are produced/consumed, interrupt is generated. Acknowledge the interrupt. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 46 +++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 3e806f7..94f915a 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -27,6 +27,7 @@ #define DRV_NAME "acp3x-i2s-audio" struct i2s_dev_data { + unsigned int i2s_irq; void __iomem *acp3x_base; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; @@ -132,6 +133,38 @@ static int acp3x_deinit(void __iomem *acp3x_base) return 0; } +static irqreturn_t i2s_irq_handler(int irq, void *dev_id) +{ + u16 play_flag, cap_flag; + u32 val; + struct i2s_dev_data *rv_i2s_data = dev_id; + + if (!rv_i2s_data) + return IRQ_NONE; + + play_flag = 0; + cap_flag = 0; + val = rv_readl(rv_i2s_data->acp3x_base + mmACP_EXTERNAL_INTR_STAT); + if ((val & BIT(BT_TX_THRESHOLD)) && rv_i2s_data->play_stream) { + rv_writel(BIT(BT_TX_THRESHOLD), rv_i2s_data->acp3x_base + + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->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 + + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->capture_stream); + cap_flag = 1; + } + + if (play_flag | cap_flag) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + static struct snd_pcm_ops acp3x_dma_ops = { .open = NULL, .close = NULL, @@ -205,6 +238,13 @@ static int acp3x_audio_probe(struct platform_device *pdev) adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + return -ENODEV; + } + + adata->i2s_irq = res->start; adata->play_stream = NULL; adata->capture_stream = NULL; @@ -220,6 +260,12 @@ static int acp3x_audio_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); 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"); + goto dev_err; + } return 0; dev_err: -- 2.7.4

2 1

[alsa-devel] [PATCH 06/11] ASoC: amd: add acp3x pcm driver dma ops
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP3x has a DMA controller to access system memory. This controller transfers data from/to system memory to/from the ACP internal FIFO. The patch adds PCM driver DMA operations. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 262 +++++++++++++++++++++++++++++++++++- sound/soc/amd/raven/acp3x.h | 20 +++ 2 files changed, 275 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 94f915a..06c1985 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -33,6 +33,54 @@ struct i2s_dev_data { struct snd_pcm_substream *capture_stream; }; +struct i2s_stream_instance { + u16 num_pages; + u16 channels; + u32 xfer_resolution; + struct page *pg; + void __iomem *acp3x_base; +}; + +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_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, + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +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, + .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, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + static int acp3x_power_on(void __iomem *acp3x_base, bool on) { u16 val, mask; @@ -165,14 +213,214 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) return IRQ_NONE; } +static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) +{ + u16 page_idx; + u64 addr; + u32 low, high, val, acp_fifo_addr; + struct page *pg = rtd->pg; + + /* 8 scratch registers used to map one 64 bit address */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + val = 0; + else + val = rtd->num_pages * 8; + + /* Group Enable */ + rv_writel(ACP_SRAM_PTE_OFFSET | BIT(31), rtd->acp3x_base + + mmACPAXI2AXI_ATU_BASE_ADDR_GRP_1); + rv_writel(PAGE_SIZE_4K_ENABLE, rtd->acp3x_base + + mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_1); + + for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) { + /* Load the low address of page int ACP SRAM through SRBM */ + addr = page_to_phys(pg); + low = lower_32_bits(addr); + high = upper_32_bits(addr); + + rv_writel(low, rtd->acp3x_base + mmACP_SCRATCH_REG_0 + val); + high |= BIT(31); + rv_writel(high, rtd->acp3x_base + mmACP_SCRATCH_REG_0 + val + + 4); + /* Move to next physically contiguos page */ + val += 8; + pg++; + } + + 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); + } 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); + } + + /* Enable watermark/period interrupt to host */ + rv_writel(BIT(BT_TX_THRESHOLD) | BIT(BT_RX_THRESHOLD), + rtd->acp3x_base + mmACP_EXTERNAL_INTR_CNTL); +} + +static int acp3x_dma_open(struct snd_pcm_substream *substream) +{ + int ret = 0; + + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + + struct i2s_stream_instance *i2s_data = kzalloc(sizeof(struct i2s_stream_instance), + GFP_KERNEL); + if (!i2s_data) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp3x_pcm_hardware_playback; + else + runtime->hw = acp3x_pcm_hardware_capture; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(component->dev, "set integer constraint failed\n"); + return ret; + } + + if (!adata->play_stream && !adata->capture_stream) + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + adata->play_stream = substream; + else + adata->capture_stream = substream; + + i2s_data->acp3x_base = adata->acp3x_base; + runtime->private_data = i2s_data; + return 0; +} + +static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int status; + u64 size; + struct snd_dma_buffer *dma_buffer; + struct page *pg; + struct snd_pcm_runtime *runtime = substream->runtime; + struct i2s_stream_instance *rtd = runtime->private_data; + + if (!rtd) + return -EINVAL; + + dma_buffer = &substream->dma_buffer; + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + pg = virt_to_page(substream->dma_buffer.area); + if (pg) { + rtd->pg = pg; + rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); + config_acp3x_dma(rtd, substream->stream); + status = 0; + } else { + status = -ENOMEM; + } + return status; +} + +static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream) +{ + u32 pos = 0; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + pos = rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINKPOSITIONCNTR); + else + pos = rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINKPOSITIONCNTR); + + if (pos >= MAX_BUFFER) + pos = 0; + + return bytes_to_frames(substream->runtime, pos); +} + +static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, + MAX_BUFFER); +} + +static int acp3x_dma_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int acp3x_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return snd_pcm_lib_default_mmap(substream, vma); +} + +static int acp3x_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + adata->play_stream = NULL; + else + adata->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) + rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); + return 0; +} + static struct snd_pcm_ops acp3x_dma_ops = { - .open = NULL, - .close = NULL, - .ioctl = NULL, - .hw_params = NULL, - .hw_free = NULL, - .pointer = NULL, - .mmap = NULL, + .open = acp3x_dma_open, + .close = acp3x_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = acp3x_dma_hw_params, + .hw_free = acp3x_dma_hw_free, + .pointer = acp3x_dma_pointer, + .mmap = acp3x_dma_mmap, }; struct snd_soc_dai_ops acp3x_dai_i2s_ops = { diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index f3d3e51..9f33e34 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -13,6 +13,26 @@ #define ACP3x_POWER_OFF_IN_PROGRESS 0x03 #define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 +#define ACP_SRAM_PTE_OFFSET 0x02050000 +#define PAGE_SIZE_4K_ENABLE 0x2 +#define MEM_WINDOW_START 0x4000000 +#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 +#define CAPTURE_FIFO_ADDR_OFFSET 0x500 + +#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 CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 4096 + +#define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define MIN_BUFFER MAX_BUFFER +#define FIFO_SIZE 0x100 +#define DMA_SIZE 0x40 + static inline u32 rv_readl(void __iomem *base_addr) { return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); -- 2.7.4

2 1

[alsa-devel] [PATCH 07/11] ASoC: amd: add acp3x i2s ops
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP3x has a i2s controller block for playback and capture. This patch adds ACP3x i2s DAI operations. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 89 +++++++++++++++++++++++++++++++++++-- 1 file changed, 86 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 06c1985..ed2c163 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -423,10 +423,93 @@ static struct snd_pcm_ops acp3x_dma_ops = { .mmap = acp3x_dma_mmap, }; +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: + 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; +} + struct snd_soc_dai_ops acp3x_dai_i2s_ops = { - .hw_params = NULL, - .trigger = NULL, - .set_fmt = NULL, + .hw_params = acp3x_dai_i2s_hwparams, + .trigger = acp3x_dai_i2s_trigger, }; static struct snd_soc_dai_driver acp3x_i2s_dai_driver = { -- 2.7.4

2 1

[alsa-devel] [PATCH 08/11] ASoC: amd: add acp3x tdm mode support
by Vijendar Mukunda 13 Nov '18

13 Nov '18

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: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/amd/raven/acp3x.h | 7 +++++ 2 files changed, 69 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index ed2c163..15a5f3b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -27,7 +27,9 @@ #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; @@ -423,6 +425,64 @@ static struct snd_pcm_ops acp3x_dma_ops = { .mmap = acp3x_dma_mmap, }; + +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) @@ -510,6 +570,8 @@ static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, 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 = { diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 9f33e34..95fef85 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -32,6 +32,13 @@ #define MIN_BUFFER MAX_BUFFER #define FIFO_SIZE 0x100 #define DMA_SIZE 0x40 +#define FRM_LEN 0x100 + +#define SLOT_WIDTH_8 0x08 +#define SLOT_WIDTH_16 0x10 +#define SLOT_WIDTH_24 0x18 +#define SLOT_WIDTH_32 0x20 + static inline u32 rv_readl(void __iomem *base_addr) { -- 2.7.4

2 1

[alsa-devel] [PATCH 09/11] ASoC: amd: add acp3x runtime pm ops
by Vijendar Mukunda 13 Nov '18

13 Nov '18

Added runtime PM operations for ACP3x PCM platform device. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 39 +++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 15a5f3b..710c5c5 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/err.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -660,6 +661,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) goto dev_err; } + pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); return 0; dev_err: status = acp3x_deinit(adata->acp3x_base); @@ -682,14 +686,49 @@ static int acp3x_audio_remove(struct platform_device *pdev) else dev_info(&pdev->dev, "ACP de-initialized\n"); + pm_runtime_disable(&pdev->dev); return 0; } +static int acp3x_pcm_runtime_suspend(struct device *dev) +{ + int status; + struct i2s_dev_data *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"); + + rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + + return 0; +} + +static int acp3x_pcm_runtime_resume(struct device *dev) +{ + int status; + struct i2s_dev_data *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; +} + +static const struct dev_pm_ops acp3x_pm_ops = { + .runtime_suspend = acp3x_pcm_runtime_suspend, + .runtime_resume = acp3x_pcm_runtime_resume, +}; + static struct platform_driver acp3x_dma_driver = { .probe = acp3x_audio_probe, .remove = acp3x_audio_remove, .driver = { .name = "acp3x_rv_i2s", + .pm = &acp3x_pm_ops, }, }; -- 2.7.4

2 1

[alsa-devel] [PATCH 10/11] ASoC: amd: add acp3x system resume pm op
by Vijendar Mukunda 13 Nov '18

13 Nov '18

When system wide suspend happens, ACP will be powered off. When system resumes, all the runtime configuration data for ACP needs to be programmed again. Added 'resume'pm call back to ACP pm ops. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 46 +++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 710c5c5..2e61cef 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -690,6 +690,51 @@ static int acp3x_audio_remove(struct platform_device *pdev) return 0; } +static int acp3x_resume(struct device *dev) +{ + int status; + u32 val; + struct i2s_dev_data *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 = + 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); + } + } + + 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); + } + } + + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + return 0; +} + + static int acp3x_pcm_runtime_suspend(struct device *dev) { int status; @@ -721,6 +766,7 @@ static int acp3x_pcm_runtime_resume(struct device *dev) static const struct dev_pm_ops acp3x_pm_ops = { .runtime_suspend = acp3x_pcm_runtime_suspend, .runtime_resume = acp3x_pcm_runtime_resume, + .resume = acp3x_resume, }; static struct platform_driver acp3x_dma_driver = { -- 2.7.4

2 1

[alsa-devel] [PATCH 11/11] ASoC: amd: enable acp3x drivers build
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP3x drivers can be built by selecting necessary kernel config option. The patch enables build support of the same. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 5 +++++ sound/soc/amd/Makefile | 1 + sound/soc/amd/raven/Makefile | 4 ++++ 3 files changed, 10 insertions(+) create mode 100644 sound/soc/amd/raven/Makefile diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 58c1dcb..5f3b560 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -19,3 +19,8 @@ config SND_SOC_AMD_CZ_RT5645_MACH depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for rt5645. + +config SND_SOC_AMD_ACP3x + tristate "AMD Audio Coprocessor-v3.x support" + help + This option enables ACP v3.x I2S support on AMD platform diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index 79b0622..8e1c571 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -5,3 +5,4 @@ snd-soc-acp-rt5645-mach-objs := acp-rt5645.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ diff --git a/sound/soc/amd/raven/Makefile b/sound/soc/amd/raven/Makefile new file mode 100644 index 0000000..8c03e6f --- /dev/null +++ b/sound/soc/amd/raven/Makefile @@ -0,0 +1,4 @@ +snd-pci-acp3x-objs := pci-acp3x.o +snd-acp3x-pcm-dma-objs := acp3x-pcm-dma.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-pci-acp3x.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-pcm-dma.o -- 2.7.4

2 2

[alsa-devel] [PATCH v3 0/8] ASoC: omap-mcbsp: Cleanup and split core/sidetone support
by Peter Ujfalusi 13 Nov '18

13 Nov '18

Hi, Changes since v2: - Split the dma_data addr and maxburst cleanup patches - Added Jarkko's Acked-by tot he series. Changes since v1: - Two patch added to address the code move triggered few warnings from build. One of them is around mcbsp->pdata checks (not valid complaint), the other is that we set -EINVAL to unsigned variables (they were not used in that case) While preparing for merging the davinci and omap directories (I will send the series for that shortly) I have taken some time to do cleanup on the McBSP driver. The mcbsp.c/h files were the result when we moved code from arch/arm/plat-omap to sound/soc/omap/ a long time ago and it contained code for McBSP core and the OMAP3 sidetone functionality. With this series I tried to split the core and sidetone code to separate files. There were two reasons for this: 1. to have the OMAP specific DAI driver files prefixed with omap-* under the new sound/soc/ti/ directory 2. easier to find functions as all sidetone code is in separate file. Regards, Peter --- Peter Ujfalusi (8): ASoC: omap-mcbsp: Clean up dma_data addr initialization code ASoC: omap-mcbsp: Skip dma_data.maxburst initialization ASoC: omap-mcbsp: Clean up the interrupt handlers ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1104 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 852 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1278 insertions(+), 1330 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 11

[alsa-devel] Applied "ASoC: omap-mcbsp: Skip dma_data.maxburst initialization" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Skip dma_data.maxburst initialization 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 c9ece9c29e263a4bda60b141b7b9dece4256b7dd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:54 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Skip dma_data.maxburst initialization It is configured runtime so no need to initialize it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index d6784f2887e4..8c3cedfcb878 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -1035,11 +1035,8 @@ int omap_mcbsp_init(struct platform_device *pdev) mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, SNDRV_PCM_STREAM_PLAYBACK); - mcbsp->dma_data[0].maxburst = 4; - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, SNDRV_PCM_STREAM_CAPTURE); - mcbsp->dma_data[1].maxburst = 4; mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Clean up the interrupt handlers" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Clean up the interrupt handlers 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 d63a7625a6dff8f75604b952de600d24b58b8b54 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:55 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Clean up the interrupt handlers The parameter name of dev_id is leftover from the old times when we passed numeric ID as data for the interrupt handlers. The mcbsp_rx and mcbsp_tx is misleading as they are pointers to the mcbsp struct. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 8c3cedfcb878..fca0b0e9186b 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -110,9 +110,9 @@ static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) dev_dbg(mcbsp->dev, "***********************\n"); } -static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id) +static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) { - struct omap_mcbsp *mcbsp = dev_id; + struct omap_mcbsp *mcbsp = data; u16 irqst; irqst = MCBSP_READ(mcbsp, IRQST); @@ -151,37 +151,37 @@ static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id) return IRQ_HANDLED; } -static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) +static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data) { - struct omap_mcbsp *mcbsp_tx = dev_id; + struct omap_mcbsp *mcbsp = data; u16 irqst_spcr2; - irqst_spcr2 = MCBSP_READ(mcbsp_tx, SPCR2); - dev_dbg(mcbsp_tx->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); + irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2); + dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); if (irqst_spcr2 & XSYNC_ERR) { - dev_err(mcbsp_tx->dev, "TX Frame Sync Error! : 0x%x\n", + dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n", irqst_spcr2); /* Writing zero to XSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp_tx, SPCR2, MCBSP_READ_CACHE(mcbsp_tx, SPCR2)); + MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2)); } return IRQ_HANDLED; } -static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) +static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data) { - struct omap_mcbsp *mcbsp_rx = dev_id; + struct omap_mcbsp *mcbsp = data; u16 irqst_spcr1; - irqst_spcr1 = MCBSP_READ(mcbsp_rx, SPCR1); - dev_dbg(mcbsp_rx->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); + irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1); + dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); if (irqst_spcr1 & RSYNC_ERR) { - dev_err(mcbsp_rx->dev, "RX Frame Sync Error! : 0x%x\n", + dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n", irqst_spcr1); /* Writing zero to RSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp_rx, SPCR1, MCBSP_READ_CACHE(mcbsp_rx, SPCR1)); + MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1)); } return IRQ_HANDLED; -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters 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 59d177f65f50df07f5f0b461c968d2fab7597af1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:56 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters We either start/stop TX or RX, never both. Move the tx/rx direction selection within the functions. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 8 ++++++-- sound/soc/omap/mcbsp.h | 4 ++-- sound/soc/omap/omap-mcbsp.c | 9 ++++----- 3 files changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index fca0b0e9186b..7ff22561f00f 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -637,8 +637,10 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) * If no transmitter or receiver is active prior calling, then sample-rate * generator and frame sync are started. */ -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) +void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) { + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; int enable_srg = 0; u16 w; @@ -694,8 +696,10 @@ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) omap_mcbsp_dump_reg(mcbsp); } -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) +void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) { + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; int idle; u16 w; diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 46ae1269a698..92472c6ef358 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -339,8 +339,8 @@ u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); int omap_mcbsp_request(struct omap_mcbsp *mcbsp); void omap_mcbsp_free(struct omap_mcbsp *mcbsp); -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx); -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx); +void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream); +void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream); /* McBSP functional clock source changing function */ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index caa24c463b3e..a18b7ecc3a2e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -201,27 +201,26 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: mcbsp->active++; - omap_mcbsp_start(mcbsp, play, !play); + omap_mcbsp_start(mcbsp, substream->stream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - omap_mcbsp_stop(mcbsp, play, !play); + omap_mcbsp_stop(mcbsp, substream->stream); mcbsp->active--; break; default: - err = -EINVAL; + return -EINVAL; } - return err; + return 0; } static snd_pcm_sframes_t omap_mcbsp_dai_delay( -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay 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 be51c576e8495fd2cffbababad6de7e0a0a562ba Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:57 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay Check if the McBSP have FIFO in the omap_mcbsp_set_threshold() and omap_mcbsp_dai_delay() delay function to skip calling the lower layer if it is not needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 12 ------------ sound/soc/omap/omap-mcbsp.c | 8 ++++++++ 2 files changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 7ff22561f00f..b19168f5c110 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -450,9 +450,6 @@ int omap_st_is_enabled(struct omap_mcbsp *mcbsp) */ void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) { - if (mcbsp->pdata->buffer_size == 0) - return; - if (threshold && threshold <= mcbsp->max_tx_thres) MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); } @@ -464,9 +461,6 @@ void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) */ void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) { - if (mcbsp->pdata->buffer_size == 0) - return; - if (threshold && threshold <= mcbsp->max_rx_thres) MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); } @@ -478,9 +472,6 @@ u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) { u16 buffstat; - if (mcbsp->pdata->buffer_size == 0) - return 0; - /* Returns the number of free locations in the buffer */ buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); @@ -496,9 +487,6 @@ u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) { u16 buffstat, threshold; - if (mcbsp->pdata->buffer_size == 0) - return 0; - /* Returns the number of used locations in the buffer */ buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); /* RX threshold */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index a18b7ecc3a2e..69a6b8ad6d42 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -71,6 +71,10 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); int words; + /* No need to proceed further if McBSP does not have FIFO */ + if (mcbsp->pdata->buffer_size == 0) + return; + /* * Configure McBSP threshold based on either: * packet_size, when the sDMA is in packet mode, or based on the @@ -233,6 +237,10 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( u16 fifo_use; snd_pcm_sframes_t delay; + /* No need to proceed further if McBSP does not have FIFO */ + if (mcbsp->pdata->buffer_size == 0) + return 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_use = omap_mcbsp_get_tx_delay(mcbsp); else -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split 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 9c34d023dc35a4b3d005d94db512742dfcdc8f82 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:58 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split The mcbsp.c was copied a while back from arch/arm/plat-omap/mcbsp.c and it contained a mix of McBSP and McBSP sidetone functions. Create new file structure with the following split: omap-mcbsp.c - McBSP related functions omap-mcbsp-st.c - McBSP sidetone functionality omap-mcbsp-priv.h - Private header for internal use omap-mcbsp.h - Header for user drivers I have tried to do the code move with minimal code change, cleanup patches can be based on the new structure. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1094 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 838 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1269 insertions(+), 1315 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index af50cdd6eea9..d005338dd13c 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -2,7 +2,7 @@ # OMAP Platform Support snd-soc-sdma-objs := sdma-pcm.o snd-soc-omap-dmic-objs := omap-dmic.o -snd-soc-omap-mcbsp-objs := omap-mcbsp.o mcbsp.o +snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-audio-objs := omap-hdmi-audio.o diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c deleted file mode 100644 index b19168f5c110..000000000000 --- a/sound/soc/omap/mcbsp.c +++ /dev/null @@ -1,1094 +0,0 @@ -/* - * sound/soc/omap/mcbsp.c - * - * Copyright (C) 2004 Nokia Corporation - * Author: Samuel Ortiz <samuel.ortiz(a)nokia.com> - * - * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> - * Peter Ujfalusi <peter.ujfalusi(a)ti.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Multichannel mode not supported. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/err.h> -#include <linux/clk.h> -#include <linux/delay.h> -#include <linux/io.h> -#include <linux/slab.h> -#include <linux/pm_runtime.h> - -#include <linux/platform_data/asoc-ti-mcbsp.h> - -#include "mcbsp.h" - -static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) -{ - void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; - - if (mcbsp->pdata->reg_size == 2) { - ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; - writew_relaxed((u16)val, addr); - } else { - ((u32 *)mcbsp->reg_cache)[reg] = val; - writel_relaxed(val, addr); - } -} - -static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache) -{ - void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; - - if (mcbsp->pdata->reg_size == 2) { - return !from_cache ? readw_relaxed(addr) : - ((u16 *)mcbsp->reg_cache)[reg]; - } else { - return !from_cache ? readl_relaxed(addr) : - ((u32 *)mcbsp->reg_cache)[reg]; - } -} - -static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) -{ - writel_relaxed(val, mcbsp->st_data->io_base_st + reg); -} - -static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) -{ - return readl_relaxed(mcbsp->st_data->io_base_st + reg); -} - -#define MCBSP_READ(mcbsp, reg) \ - omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0) -#define MCBSP_WRITE(mcbsp, reg, val) \ - omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val) -#define MCBSP_READ_CACHE(mcbsp, reg) \ - omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1) - -#define MCBSP_ST_READ(mcbsp, reg) \ - omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg) -#define MCBSP_ST_WRITE(mcbsp, reg, val) \ - omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) - -static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) -{ - dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); - dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", - MCBSP_READ(mcbsp, DRR2)); - dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", - MCBSP_READ(mcbsp, DRR1)); - dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", - MCBSP_READ(mcbsp, DXR2)); - dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", - MCBSP_READ(mcbsp, DXR1)); - dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", - MCBSP_READ(mcbsp, SPCR2)); - dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", - MCBSP_READ(mcbsp, SPCR1)); - dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n", - MCBSP_READ(mcbsp, RCR2)); - dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n", - MCBSP_READ(mcbsp, RCR1)); - dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n", - MCBSP_READ(mcbsp, XCR2)); - dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n", - MCBSP_READ(mcbsp, XCR1)); - dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", - MCBSP_READ(mcbsp, SRGR2)); - dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", - MCBSP_READ(mcbsp, SRGR1)); - dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n", - MCBSP_READ(mcbsp, PCR0)); - dev_dbg(mcbsp->dev, "***********************\n"); -} - -static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) -{ - struct omap_mcbsp *mcbsp = data; - u16 irqst; - - irqst = MCBSP_READ(mcbsp, IRQST); - dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); - - if (irqst & RSYNCERREN) - dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); - if (irqst & RFSREN) - dev_dbg(mcbsp->dev, "RX Frame Sync\n"); - if (irqst & REOFEN) - dev_dbg(mcbsp->dev, "RX End Of Frame\n"); - if (irqst & RRDYEN) - dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); - if (irqst & RUNDFLEN) - dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); - if (irqst & ROVFLEN) - dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); - - if (irqst & XSYNCERREN) - dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); - if (irqst & XFSXEN) - dev_dbg(mcbsp->dev, "TX Frame Sync\n"); - if (irqst & XEOFEN) - dev_dbg(mcbsp->dev, "TX End Of Frame\n"); - if (irqst & XRDYEN) - dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); - if (irqst & XUNDFLEN) - dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); - if (irqst & XOVFLEN) - dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); - if (irqst & XEMPTYEOFEN) - dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); - - MCBSP_WRITE(mcbsp, IRQST, irqst); - - return IRQ_HANDLED; -} - -static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data) -{ - struct omap_mcbsp *mcbsp = data; - u16 irqst_spcr2; - - irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2); - dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); - - if (irqst_spcr2 & XSYNC_ERR) { - dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n", - irqst_spcr2); - /* Writing zero to XSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2)); - } - - return IRQ_HANDLED; -} - -static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data) -{ - struct omap_mcbsp *mcbsp = data; - u16 irqst_spcr1; - - irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1); - dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); - - if (irqst_spcr1 & RSYNC_ERR) { - dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n", - irqst_spcr1); - /* Writing zero to RSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1)); - } - - return IRQ_HANDLED; -} - -/* - * omap_mcbsp_config simply write a config to the - * appropriate McBSP. - * You either call this function or set the McBSP registers - * by yourself before calling omap_mcbsp_start(). - */ -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config) -{ - dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", - mcbsp->id, mcbsp->phys_base); - - /* We write the given config */ - MCBSP_WRITE(mcbsp, SPCR2, config->spcr2); - MCBSP_WRITE(mcbsp, SPCR1, config->spcr1); - MCBSP_WRITE(mcbsp, RCR2, config->rcr2); - MCBSP_WRITE(mcbsp, RCR1, config->rcr1); - MCBSP_WRITE(mcbsp, XCR2, config->xcr2); - MCBSP_WRITE(mcbsp, XCR1, config->xcr1); - MCBSP_WRITE(mcbsp, SRGR2, config->srgr2); - MCBSP_WRITE(mcbsp, SRGR1, config->srgr1); - MCBSP_WRITE(mcbsp, MCR2, config->mcr2); - MCBSP_WRITE(mcbsp, MCR1, config->mcr1); - MCBSP_WRITE(mcbsp, PCR0, config->pcr0); - if (mcbsp->pdata->has_ccr) { - MCBSP_WRITE(mcbsp, XCCR, config->xccr); - MCBSP_WRITE(mcbsp, RCCR, config->rccr); - } - /* Enable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); - - /* Enable TX/RX sync error interrupts by default */ - if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | - RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); -} - -/** - * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register - * @mcbsp: omap_mcbsp struct for the McBSP instance - * @stream: Stream direction (playback/capture) - * - * Returns the address of mcbsp data transmit register or data receive register - * to be used by DMA for transferring/receiving data - */ -static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, - unsigned int stream) -{ - int data_reg; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (mcbsp->pdata->reg_size == 2) - data_reg = OMAP_MCBSP_REG_DXR1; - else - data_reg = OMAP_MCBSP_REG_DXR; - } else { - if (mcbsp->pdata->reg_size == 2) - data_reg = OMAP_MCBSP_REG_DRR1; - else - data_reg = OMAP_MCBSP_REG_DRR; - } - - return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; -} - -static void omap_st_on(struct omap_mcbsp *mcbsp) -{ - unsigned int w; - - if (mcbsp->pdata->force_ick_on) - mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, true); - - /* Disable Sidetone clock auto-gating for normal operation */ - w = MCBSP_ST_READ(mcbsp, SYSCONFIG); - MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); - - /* Enable McBSP Sidetone */ - w = MCBSP_READ(mcbsp, SSELCR); - MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); - - /* Enable Sidetone from Sidetone Core */ - w = MCBSP_ST_READ(mcbsp, SSELCR); - MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN); -} - -static void omap_st_off(struct omap_mcbsp *mcbsp) -{ - unsigned int w; - - w = MCBSP_ST_READ(mcbsp, SSELCR); - MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN)); - - w = MCBSP_READ(mcbsp, SSELCR); - MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); - - /* Enable Sidetone clock auto-gating to reduce power consumption */ - w = MCBSP_ST_READ(mcbsp, SYSCONFIG); - MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); - - if (mcbsp->pdata->force_ick_on) - mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, false); -} - -static void omap_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) -{ - u16 val, i; - - val = MCBSP_ST_READ(mcbsp, SSELCR); - - if (val & ST_COEFFWREN) - MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); - - MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN); - - for (i = 0; i < 128; i++) - MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]); - - i = 0; - - val = MCBSP_ST_READ(mcbsp, SSELCR); - while (!(val & ST_COEFFWRDONE) && (++i < 1000)) - val = MCBSP_ST_READ(mcbsp, SSELCR); - - MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); - - if (i == 1000) - dev_err(mcbsp->dev, "McBSP FIR load error!\n"); -} - -static void omap_st_chgain(struct omap_mcbsp *mcbsp) -{ - u16 w; - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - w = MCBSP_ST_READ(mcbsp, SSELCR); - - MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | \ - ST_CH1GAIN(st_data->ch1gain)); -} - -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENOENT; - - spin_lock_irq(&mcbsp->lock); - if (channel == 0) - st_data->ch0gain = chgain; - else if (channel == 1) - st_data->ch1gain = chgain; - else - ret = -EINVAL; - - if (st_data->enabled) - omap_st_chgain(mcbsp); - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENOENT; - - spin_lock_irq(&mcbsp->lock); - if (channel == 0) - *chgain = st_data->ch0gain; - else if (channel == 1) - *chgain = st_data->ch1gain; - else - ret = -EINVAL; - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -static int omap_st_start(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (st_data->enabled && !st_data->running) { - omap_st_fir_write(mcbsp, st_data->taps); - omap_st_chgain(mcbsp); - - if (!mcbsp->free) { - omap_st_on(mcbsp); - st_data->running = 1; - } - } - - return 0; -} - -int omap_st_enable(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (!st_data) - return -ENODEV; - - spin_lock_irq(&mcbsp->lock); - st_data->enabled = 1; - omap_st_start(mcbsp); - spin_unlock_irq(&mcbsp->lock); - - return 0; -} - -static int omap_st_stop(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (st_data->running) { - if (!mcbsp->free) { - omap_st_off(mcbsp); - st_data->running = 0; - } - } - - return 0; -} - -int omap_st_disable(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENODEV; - - spin_lock_irq(&mcbsp->lock); - omap_st_stop(mcbsp); - st_data->enabled = 0; - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -int omap_st_is_enabled(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (!st_data) - return -ENODEV; - - return st_data->enabled; -} - -/* - * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. - * The threshold parameter is 1 based, and it is converted (threshold - 1) - * for the THRSH2 register. - */ -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) -{ - if (threshold && threshold <= mcbsp->max_tx_thres) - MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); -} - -/* - * omap_mcbsp_set_rx_threshold configures the receive threshold in words. - * The threshold parameter is 1 based, and it is converted (threshold - 1) - * for the THRSH1 register. - */ -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) -{ - if (threshold && threshold <= mcbsp->max_rx_thres) - MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); -} - -/* - * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO - */ -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) -{ - u16 buffstat; - - /* Returns the number of free locations in the buffer */ - buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); - - /* Number of slots are different in McBSP ports */ - return mcbsp->pdata->buffer_size - buffstat; -} - -/* - * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO - * to reach the threshold value (when the DMA will be triggered to read it) - */ -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) -{ - u16 buffstat, threshold; - - /* Returns the number of used locations in the buffer */ - buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); - /* RX threshold */ - threshold = MCBSP_READ(mcbsp, THRSH1); - - /* Return the number of location till we reach the threshold limit */ - if (threshold <= buffstat) - return 0; - else - return threshold - buffstat; -} - -int omap_mcbsp_request(struct omap_mcbsp *mcbsp) -{ - void *reg_cache; - int err; - - reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); - if (!reg_cache) { - return -ENOMEM; - } - - spin_lock(&mcbsp->lock); - if (!mcbsp->free) { - dev_err(mcbsp->dev, "McBSP%d is currently in use\n", - mcbsp->id); - err = -EBUSY; - goto err_kfree; - } - - mcbsp->free = false; - mcbsp->reg_cache = reg_cache; - spin_unlock(&mcbsp->lock); - - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) - mcbsp->pdata->ops->request(mcbsp->id - 1); - - /* - * Make sure that transmitter, receiver and sample-rate generator are - * not running before activating IRQs. - */ - MCBSP_WRITE(mcbsp, SPCR1, 0); - MCBSP_WRITE(mcbsp, SPCR2, 0); - - if (mcbsp->irq) { - err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, - "McBSP", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request IRQ\n"); - goto err_clk_disable; - } - } else { - err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, - "McBSP TX", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); - goto err_clk_disable; - } - - err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, - "McBSP RX", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); - goto err_free_irq; - } - } - - return 0; -err_free_irq: - free_irq(mcbsp->tx_irq, (void *)mcbsp); -err_clk_disable: - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); - - /* Disable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - - spin_lock(&mcbsp->lock); - mcbsp->free = true; - mcbsp->reg_cache = NULL; -err_kfree: - spin_unlock(&mcbsp->lock); - kfree(reg_cache); - - return err; -} - -void omap_mcbsp_free(struct omap_mcbsp *mcbsp) -{ - void *reg_cache; - - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); - - /* Disable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - - /* Disable interrupt requests */ - if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, 0); - - if (mcbsp->irq) { - free_irq(mcbsp->irq, (void *)mcbsp); - } else { - free_irq(mcbsp->rx_irq, (void *)mcbsp); - free_irq(mcbsp->tx_irq, (void *)mcbsp); - } - - reg_cache = mcbsp->reg_cache; - - /* - * Select CLKS source from internal source unconditionally before - * marking the McBSP port as free. - * If the external clock source via MCBSP_CLKS pin has been selected the - * system will refuse to enter idle if the CLKS pin source is not reset - * back to internal source. - */ - if (!mcbsp_omap1()) - omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); - - spin_lock(&mcbsp->lock); - if (mcbsp->free) - dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); - else - mcbsp->free = true; - mcbsp->reg_cache = NULL; - spin_unlock(&mcbsp->lock); - - kfree(reg_cache); -} - -/* - * Here we start the McBSP, by enabling transmitter, receiver or both. - * If no transmitter or receiver is active prior calling, then sample-rate - * generator and frame sync are started. - */ -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) -{ - int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); - int rx = !tx; - int enable_srg = 0; - u16 w; - - if (mcbsp->st_data) - omap_st_start(mcbsp); - - /* Only enable SRG, if McBSP is master */ - w = MCBSP_READ_CACHE(mcbsp, PCR0); - if (w & (FSXM | FSRM | CLKXM | CLKRM)) - enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | - MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); - - if (enable_srg) { - /* Start the sample generator */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6)); - } - - /* Enable transmitter and receiver */ - tx &= 1; - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | tx); - - rx &= 1; - w = MCBSP_READ_CACHE(mcbsp, SPCR1); - MCBSP_WRITE(mcbsp, SPCR1, w | rx); - - /* - * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec - * REVISIT: 100us may give enough time for two CLKSRG, however - * due to some unknown PM related, clock gating etc. reason it - * is now at 500us. - */ - udelay(500); - - if (enable_srg) { - /* Start frame sync */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7)); - } - - if (mcbsp->pdata->has_ccr) { - /* Release the transmitter and receiver */ - w = MCBSP_READ_CACHE(mcbsp, XCCR); - w &= ~(tx ? XDISABLE : 0); - MCBSP_WRITE(mcbsp, XCCR, w); - w = MCBSP_READ_CACHE(mcbsp, RCCR); - w &= ~(rx ? RDISABLE : 0); - MCBSP_WRITE(mcbsp, RCCR, w); - } - - /* Dump McBSP Regs */ - omap_mcbsp_dump_reg(mcbsp); -} - -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) -{ - int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); - int rx = !tx; - int idle; - u16 w; - - /* Reset transmitter */ - tx &= 1; - if (mcbsp->pdata->has_ccr) { - w = MCBSP_READ_CACHE(mcbsp, XCCR); - w |= (tx ? XDISABLE : 0); - MCBSP_WRITE(mcbsp, XCCR, w); - } - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w & ~tx); - - /* Reset receiver */ - rx &= 1; - if (mcbsp->pdata->has_ccr) { - w = MCBSP_READ_CACHE(mcbsp, RCCR); - w |= (rx ? RDISABLE : 0); - MCBSP_WRITE(mcbsp, RCCR, w); - } - w = MCBSP_READ_CACHE(mcbsp, SPCR1); - MCBSP_WRITE(mcbsp, SPCR1, w & ~rx); - - idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | - MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); - - if (idle) { - /* Reset the sample rate generator */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6)); - } - - if (mcbsp->st_data) - omap_st_stop(mcbsp); -} - -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) -{ - struct clk *fck_src; - const char *src; - int r; - - if (fck_src_id == MCBSP_CLKS_PAD_SRC) - src = "pad_fck"; - else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) - src = "prcm_fck"; - else - return -EINVAL; - - fck_src = clk_get(mcbsp->dev, src); - if (IS_ERR(fck_src)) { - dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src); - return -EINVAL; - } - - pm_runtime_put_sync(mcbsp->dev); - - r = clk_set_parent(mcbsp->fclk, fck_src); - if (r) { - dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", - src); - clk_put(fck_src); - return r; - } - - pm_runtime_get_sync(mcbsp->dev); - - clk_put(fck_src); - - return 0; - -} - -#define max_thres(m) (mcbsp->pdata->buffer_size) -#define valid_threshold(m, val) ((val) <= max_thres(m)) -#define THRESHOLD_PROP_BUILDER(prop) \ -static ssize_t prop##_show(struct device *dev, \ - struct device_attribute *attr, char *buf) \ -{ \ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ - \ - return sprintf(buf, "%u\n", mcbsp->prop); \ -} \ - \ -static ssize_t prop##_store(struct device *dev, \ - struct device_attribute *attr, \ - const char *buf, size_t size) \ -{ \ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ - unsigned long val; \ - int status; \ - \ - status = kstrtoul(buf, 0, &val); \ - if (status) \ - return status; \ - \ - if (!valid_threshold(mcbsp, val)) \ - return -EDOM; \ - \ - mcbsp->prop = val; \ - return size; \ -} \ - \ -static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store); - -THRESHOLD_PROP_BUILDER(max_tx_thres); -THRESHOLD_PROP_BUILDER(max_rx_thres); - -static const char *dma_op_modes[] = { - "element", "threshold", -}; - -static ssize_t dma_op_mode_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - int dma_op_mode, i = 0; - ssize_t len = 0; - const char * const *s; - - dma_op_mode = mcbsp->dma_op_mode; - - for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { - if (dma_op_mode == i) - len += sprintf(buf + len, "[%s] ", *s); - else - len += sprintf(buf + len, "%s ", *s); - } - len += sprintf(buf + len, "\n"); - - return len; -} - -static ssize_t dma_op_mode_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t size) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - int i; - - i = sysfs_match_string(dma_op_modes, buf); - if (i < 0) - return i; - - spin_lock_irq(&mcbsp->lock); - if (!mcbsp->free) { - size = -EBUSY; - goto unlock; - } - mcbsp->dma_op_mode = i; - -unlock: - spin_unlock_irq(&mcbsp->lock); - - return size; -} - -static DEVICE_ATTR_RW(dma_op_mode); - -static const struct attribute *additional_attrs[] = { - &dev_attr_max_tx_thres.attr, - &dev_attr_max_rx_thres.attr, - &dev_attr_dma_op_mode.attr, - NULL, -}; - -static const struct attribute_group additional_attr_group = { - .attrs = (struct attribute **)additional_attrs, -}; - -static ssize_t st_taps_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - ssize_t status = 0; - int i; - - spin_lock_irq(&mcbsp->lock); - for (i = 0; i < st_data->nr_taps; i++) - status += sprintf(&buf[status], (i ? ", %d" : "%d"), - st_data->taps[i]); - if (i) - status += sprintf(&buf[status], "\n"); - spin_unlock_irq(&mcbsp->lock); - - return status; -} - -static ssize_t st_taps_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t size) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int val, tmp, status, i = 0; - - spin_lock_irq(&mcbsp->lock); - memset(st_data->taps, 0, sizeof(st_data->taps)); - st_data->nr_taps = 0; - - do { - status = sscanf(buf, "%d%n", &val, &tmp); - if (status < 0 || status == 0) { - size = -EINVAL; - goto out; - } - if (val < -32768 || val > 32767) { - size = -EINVAL; - goto out; - } - st_data->taps[i++] = val; - buf += tmp; - if (*buf != ',') - break; - buf++; - } while (1); - - st_data->nr_taps = i; - -out: - spin_unlock_irq(&mcbsp->lock); - - return size; -} - -static DEVICE_ATTR_RW(st_taps); - -static const struct attribute *sidetone_attrs[] = { - &dev_attr_st_taps.attr, - NULL, -}; - -static const struct attribute_group sidetone_attr_group = { - .attrs = (struct attribute **)sidetone_attrs, -}; - -static int omap_st_add(struct omap_mcbsp *mcbsp, struct resource *res) -{ - struct omap_mcbsp_st_data *st_data; - int err; - - st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); - if (!st_data) - return -ENOMEM; - - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); - if (IS_ERR(st_data->mcbsp_iclk)) { - dev_warn(mcbsp->dev, - "Failed to get ick, sidetone might be broken\n"); - st_data->mcbsp_iclk = NULL; - } - - st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, - resource_size(res)); - if (!st_data->io_base_st) - return -ENOMEM; - - err = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); - if (err) - return err; - - mcbsp->st_data = st_data; - return 0; -} - -/* - * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. - * 730 has only 2 McBSP, and both of them are MPU peripherals. - */ -int omap_mcbsp_init(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - struct resource *res; - int ret = 0; - - spin_lock_init(&mcbsp->lock); - mcbsp->free = true; - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); - if (!res) - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - - mcbsp->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(mcbsp->io_base)) - return PTR_ERR(mcbsp->io_base); - - mcbsp->phys_base = res->start; - mcbsp->reg_cache_size = resource_size(res); - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); - if (!res) - mcbsp->phys_dma_base = mcbsp->phys_base; - else - mcbsp->phys_dma_base = res->start; - - /* - * OMAP1, 2 uses two interrupt lines: TX, RX - * OMAP2430, OMAP3 SoC have combined IRQ line as well. - * OMAP4 and newer SoC only have the combined IRQ line. - * Use the combined IRQ if available since it gives better debugging - * possibilities. - */ - mcbsp->irq = platform_get_irq_byname(pdev, "common"); - if (mcbsp->irq == -ENXIO) { - mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); - - if (mcbsp->tx_irq == -ENXIO) { - mcbsp->irq = platform_get_irq(pdev, 0); - mcbsp->tx_irq = 0; - } else { - mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); - mcbsp->irq = 0; - } - } - - if (!pdev->dev.of_node) { - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); - if (!res) { - dev_err(&pdev->dev, "invalid tx DMA channel\n"); - return -ENODEV; - } - mcbsp->dma_req[0] = res->start; - mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); - if (!res) { - dev_err(&pdev->dev, "invalid rx DMA channel\n"); - return -ENODEV; - } - mcbsp->dma_req[1] = res->start; - mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; - } else { - mcbsp->dma_data[0].filter_data = "tx"; - mcbsp->dma_data[1].filter_data = "rx"; - } - - mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, - SNDRV_PCM_STREAM_PLAYBACK); - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, - SNDRV_PCM_STREAM_CAPTURE); - - mcbsp->fclk = clk_get(&pdev->dev, "fck"); - if (IS_ERR(mcbsp->fclk)) { - ret = PTR_ERR(mcbsp->fclk); - dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); - return ret; - } - - mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; - if (mcbsp->pdata->buffer_size) { - /* - * Initially configure the maximum thresholds to a safe value. - * The McBSP FIFO usage with these values should not go under - * 16 locations. - * If the whole FIFO without safety buffer is used, than there - * is a possibility that the DMA will be not able to push the - * new data on time, causing channel shifts in runtime. - */ - mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; - mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); - if (ret) { - dev_err(mcbsp->dev, - "Unable to create additional controls\n"); - goto err_thres; - } - } else { - mcbsp->max_tx_thres = -EINVAL; - mcbsp->max_rx_thres = -EINVAL; - } - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone"); - if (res) { - ret = omap_st_add(mcbsp, res); - if (ret) { - dev_err(mcbsp->dev, - "Unable to create sidetone controls\n"); - goto err_st; - } - } - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); -err_thres: - clk_put(mcbsp->fclk); - return ret; -} - -void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) -{ - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/omap-mcbsp-priv.h similarity index 70% rename from sound/soc/omap/mcbsp.h rename to sound/soc/omap/omap-mcbsp-priv.h index 92472c6ef358..7865cda4bf0a 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/omap-mcbsp-priv.h @@ -1,28 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * sound/soc/omap/mcbsp.h - * * OMAP Multi-Channel Buffered Serial Port * * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * Peter Ujfalusi <peter.ujfalusi(a)ti.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ -#ifndef __ASOC_MCBSP_H -#define __ASOC_MCBSP_H + +#ifndef __OMAP_MCBSP_PRIV_H__ +#define __OMAP_MCBSP_PRIV_H__ + +#include <linux/platform_data/asoc-ti-mcbsp.h> #ifdef CONFIG_ARCH_OMAP1 #define mcbsp_omap1() 1 @@ -30,8 +17,6 @@ #define mcbsp_omap1() 0 #endif -#include <sound/dmaengine_pcm.h> - /* McBSP register numbers. Register address offset = num * reg_step */ enum { /* Common registers */ @@ -85,15 +70,6 @@ enum { OMAP_MCBSP_REG_SSELCR, }; -/* OMAP3 sidetone control registers */ -#define OMAP_ST_REG_REV 0x00 -#define OMAP_ST_REG_SYSCONFIG 0x10 -#define OMAP_ST_REG_IRQSTATUS 0x18 -#define OMAP_ST_REG_IRQENABLE 0x1C -#define OMAP_ST_REG_SGAINCR 0x24 -#define OMAP_ST_REG_SFIRCR 0x28 -#define OMAP_ST_REG_SSELCR 0x2C - /************************** McBSP SPCR1 bit definitions ***********************/ #define RRST BIT(0) #define RRDY BIT(1) @@ -202,24 +178,6 @@ enum { #define SIDLEMODE(value) (((value) & 0x3) << 3) #define CLOCKACTIVITY(value) (((value) & 0x3) << 8) -/********************** McBSP SSELCR bit definitions ***********************/ -#define SIDETONEEN BIT(10) - -/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ -#define ST_AUTOIDLE BIT(0) - -/********************** McBSP Sidetone SGAINCR bit definitions *************/ -#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ -#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ - -/********************** McBSP Sidetone SFIRCR bit definitions **************/ -#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ - -/********************** McBSP Sidetone SSELCR bit definitions **************/ -#define ST_SIDETONEEN BIT(0) -#define ST_COEFFWREN BIT(1) -#define ST_COEFFWRDONE BIT(2) - /********************** McBSP DMA operating modes **************************/ #define MCBSP_DMA_MODE_ELEMENT 0 #define MCBSP_DMA_MODE_THRESHOLD 1 @@ -278,16 +236,7 @@ struct omap_mcbsp_reg_cfg { u16 rccr; }; -struct omap_mcbsp_st_data { - void __iomem *io_base_st; - struct clk *mcbsp_iclk; - bool running; - bool enabled; - s16 taps[128]; /* Sidetone filter coefficients */ - int nr_taps; /* Number of filter coefficients in use */ - s16 ch0gain; - s16 ch1gain; -}; +struct omap_mcbsp_st_data; struct omap_mcbsp { struct device *dev; @@ -330,29 +279,46 @@ struct omap_mcbsp { struct pm_qos_request pm_qos_req; }; -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config); -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp); -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); -int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); -int omap_mcbsp_request(struct omap_mcbsp *mcbsp); -void omap_mcbsp_free(struct omap_mcbsp *mcbsp); -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream); -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream); - -/* McBSP functional clock source changing function */ -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); +static inline void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; + writew_relaxed((u16)val, addr); + } else { + ((u32 *)mcbsp->reg_cache)[reg] = val; + writel_relaxed(val, addr); + } +} + +static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, + bool from_cache) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + return !from_cache ? readw_relaxed(addr) : + ((u16 *)mcbsp->reg_cache)[reg]; + } else { + return !from_cache ? readl_relaxed(addr) : + ((u32 *)mcbsp->reg_cache)[reg]; + } +} + +#define MCBSP_READ(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0) +#define MCBSP_WRITE(mcbsp, reg, val) \ + omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val) +#define MCBSP_READ_CACHE(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1) + /* Sidetone specific API */ -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain); -int omap_st_enable(struct omap_mcbsp *mcbsp); -int omap_st_disable(struct omap_mcbsp *mcbsp); -int omap_st_is_enabled(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_init(struct platform_device *pdev); +void omap_mcbsp_st_cleanup(struct platform_device *pdev); -int omap_mcbsp_init(struct platform_device *pdev); -void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); -#endif /* __ASOC_MCBSP_H */ +#endif /* __OMAP_MCBSP_PRIV_H__ */ diff --git a/sound/soc/omap/omap-mcbsp-st.c b/sound/soc/omap/omap-mcbsp-st.c new file mode 100644 index 000000000000..1a3fe854e856 --- /dev/null +++ b/sound/soc/omap/omap-mcbsp-st.c @@ -0,0 +1,516 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * McBSP Sidetone support + * + * Copyright (C) 2004 Nokia Corporation + * Author: Samuel Ortiz <samuel.ortiz(a)nokia.com> + * + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/slab.h> +#include <linux/pm_runtime.h> + +#include "omap-mcbsp.h" +#include "omap-mcbsp-priv.h" + +/* OMAP3 sidetone control registers */ +#define OMAP_ST_REG_REV 0x00 +#define OMAP_ST_REG_SYSCONFIG 0x10 +#define OMAP_ST_REG_IRQSTATUS 0x18 +#define OMAP_ST_REG_IRQENABLE 0x1C +#define OMAP_ST_REG_SGAINCR 0x24 +#define OMAP_ST_REG_SFIRCR 0x28 +#define OMAP_ST_REG_SSELCR 0x2C + +/********************** McBSP SSELCR bit definitions ***********************/ +#define SIDETONEEN BIT(10) + +/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ +#define ST_AUTOIDLE BIT(0) + +/********************** McBSP Sidetone SGAINCR bit definitions *************/ +#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ +#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ + +/********************** McBSP Sidetone SFIRCR bit definitions **************/ +#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ + +/********************** McBSP Sidetone SSELCR bit definitions **************/ +#define ST_SIDETONEEN BIT(0) +#define ST_COEFFWREN BIT(1) +#define ST_COEFFWRDONE BIT(2) + +struct omap_mcbsp_st_data { + void __iomem *io_base_st; + struct clk *mcbsp_iclk; + bool running; + bool enabled; + s16 taps[128]; /* Sidetone filter coefficients */ + int nr_taps; /* Number of filter coefficients in use */ + s16 ch0gain; + s16 ch1gain; +}; + +static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + writel_relaxed(val, mcbsp->st_data->io_base_st + reg); +} + +static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) +{ + return readl_relaxed(mcbsp->st_data->io_base_st + reg); +} + +#define MCBSP_ST_READ(mcbsp, reg) omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg) +#define MCBSP_ST_WRITE(mcbsp, reg, val) \ + omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) + +static void omap_mcbsp_st_on(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + if (mcbsp->pdata->force_ick_on) + mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, true); + + /* Disable Sidetone clock auto-gating for normal operation */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); + + /* Enable McBSP Sidetone */ + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); + + /* Enable Sidetone from Sidetone Core */ + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN); +} + +static void omap_mcbsp_st_off(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN)); + + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); + + /* Enable Sidetone clock auto-gating to reduce power consumption */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); + + if (mcbsp->pdata->force_ick_on) + mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, false); +} + +static void omap_mcbsp_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) +{ + u16 val, i; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + + if (val & ST_COEFFWREN) + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN); + + for (i = 0; i < 128; i++) + MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]); + + i = 0; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + while (!(val & ST_COEFFWRDONE) && (++i < 1000)) + val = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + if (i == 1000) + dev_err(mcbsp->dev, "McBSP FIR load error!\n"); +} + +static void omap_mcbsp_st_chgain(struct omap_mcbsp *mcbsp) +{ + u16 w; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | + ST_CH1GAIN(st_data->ch1gain)); +} + +static int omap_mcbsp_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, + s16 chgain) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + st_data->ch0gain = chgain; + else if (channel == 1) + st_data->ch1gain = chgain; + else + ret = -EINVAL; + + if (st_data->enabled) + omap_mcbsp_st_chgain(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, + s16 *chgain) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + *chgain = st_data->ch0gain; + else if (channel == 1) + *chgain = st_data->ch1gain; + else + ret = -EINVAL; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_enable(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + st_data->enabled = 1; + omap_mcbsp_st_start(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return 0; +} + +static int omap_mcbsp_st_disable(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + omap_mcbsp_st_stop(mcbsp); + st_data->enabled = 0; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_is_enabled(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + return st_data->enabled; +} + +static ssize_t st_taps_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + ssize_t status = 0; + int i; + + spin_lock_irq(&mcbsp->lock); + for (i = 0; i < st_data->nr_taps; i++) + status += sprintf(&buf[status], (i ? ", %d" : "%d"), + st_data->taps[i]); + if (i) + status += sprintf(&buf[status], "\n"); + spin_unlock_irq(&mcbsp->lock); + + return status; +} + +static ssize_t st_taps_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int val, tmp, status, i = 0; + + spin_lock_irq(&mcbsp->lock); + memset(st_data->taps, 0, sizeof(st_data->taps)); + st_data->nr_taps = 0; + + do { + status = sscanf(buf, "%d%n", &val, &tmp); + if (status < 0 || status == 0) { + size = -EINVAL; + goto out; + } + if (val < -32768 || val > 32767) { + size = -EINVAL; + goto out; + } + st_data->taps[i++] = val; + buf += tmp; + if (*buf != ',') + break; + buf++; + } while (1); + + st_data->nr_taps = i; + +out: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR_RW(st_taps); + +static const struct attribute *sidetone_attrs[] = { + &dev_attr_st_taps.attr, + NULL, +}; + +static const struct attribute_group sidetone_attr_group = { + .attrs = (struct attribute **)sidetone_attrs, +}; + +int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data->enabled && !st_data->running) { + omap_mcbsp_st_fir_write(mcbsp, st_data->taps); + omap_mcbsp_st_chgain(mcbsp); + + if (!mcbsp->free) { + omap_mcbsp_st_on(mcbsp); + st_data->running = 1; + } + } + + return 0; +} + +int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data->running) { + if (!mcbsp->free) { + omap_mcbsp_st_off(mcbsp); + st_data->running = 0; + } + } + + return 0; +} + +int omap_mcbsp_st_init(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + struct omap_mcbsp_st_data *st_data; + struct resource *res; + int ret; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone"); + if (!res) + return 0; + + st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); + if (!st_data) + return -ENOMEM; + + st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + if (IS_ERR(st_data->mcbsp_iclk)) { + dev_warn(mcbsp->dev, + "Failed to get ick, sidetone might be broken\n"); + st_data->mcbsp_iclk = NULL; + } + + st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, + resource_size(res)); + if (!st_data->io_base_st) + return -ENOMEM; + + ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + if (ret) + return ret; + + mcbsp->st_data = st_data; + + return 0; +} + +void omap_mcbsp_st_cleanup(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + + if (mcbsp->st_data) { + sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); + clk_put(mcbsp->st_data->mcbsp_iclk); + } +} + +static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = min; + uinfo->value.integer.max = max; + return 0; +} + +#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel) \ +static int \ +omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ + struct soc_mixer_control *mc = \ + (struct soc_mixer_control *)kc->private_value; \ + int max = mc->max; \ + int min = mc->min; \ + int val = uc->value.integer.value[0]; \ + \ + if (val < min || val > max) \ + return -EINVAL; \ + \ + /* OMAP McBSP implementation uses index values 0..4 */ \ + return omap_mcbsp_st_set_chgain(mcbsp, channel, val); \ +} \ + \ +static int \ +omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ + s16 chgain; \ + \ + if (omap_mcbsp_st_get_chgain(mcbsp, channel, &chgain)) \ + return -EAGAIN; \ + \ + uc->value.integer.value[0] = chgain; \ + return 0; \ +} + +OMAP_MCBSP_ST_CHANNEL_VOLUME(0) +OMAP_MCBSP_ST_CHANNEL_VOLUME(1) + +static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + u8 value = ucontrol->value.integer.value[0]; + + if (value == omap_mcbsp_st_is_enabled(mcbsp)) + return 0; + + if (value) + omap_mcbsp_st_enable(mcbsp); + else + omap_mcbsp_st_disable(mcbsp); + + return 1; +} + +static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + + ucontrol->value.integer.value[0] = omap_mcbsp_st_is_enabled(mcbsp); + return 0; +} + +#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = omap_mcbsp_st_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.min = xmin, .max = xmax} } + +#define OMAP_MCBSP_ST_CONTROLS(port) \ +static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \ +SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0, \ + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch0_volume, \ + omap_mcbsp_set_st_ch0_volume), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch1_volume, \ + omap_mcbsp_set_st_ch1_volume), \ +} + +OMAP_MCBSP_ST_CONTROLS(2); +OMAP_MCBSP_ST_CONTROLS(3); + +int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) +{ + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + + if (!mcbsp->st_data) { + dev_warn(mcbsp->dev, "No sidetone data for port\n"); + return 0; + } + + switch (port_id) { + case 2: /* McBSP 2 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp2_st_controls, + ARRAY_SIZE(omap_mcbsp2_st_controls)); + case 3: /* McBSP 3 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp3_st_controls, + ARRAY_SIZE(omap_mcbsp3_st_controls)); + default: + dev_err(mcbsp->dev, "Port %d not supported\n", port_id); + break; + } + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 69a6b8ad6d42..e48fad11a0cc 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -35,21 +35,12 @@ #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include <linux/platform_data/asoc-ti-mcbsp.h> -#include "mcbsp.h" +#include "omap-mcbsp-priv.h" #include "omap-mcbsp.h" #include "sdma-pcm.h" #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) -#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ - xhandler_get, xhandler_put) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = omap_mcbsp_st_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long) &(struct soc_mixer_control) \ - {.min = xmin, .max = xmax} } - enum { OMAP_MCBSP_WORD_8 = 0, OMAP_MCBSP_WORD_12, @@ -59,6 +50,702 @@ enum { OMAP_MCBSP_WORD_32, }; +static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) +{ + dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); + dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", MCBSP_READ(mcbsp, DRR2)); + dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", MCBSP_READ(mcbsp, DRR1)); + dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", MCBSP_READ(mcbsp, DXR2)); + dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", MCBSP_READ(mcbsp, DXR1)); + dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", MCBSP_READ(mcbsp, SPCR2)); + dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", MCBSP_READ(mcbsp, SPCR1)); + dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n", MCBSP_READ(mcbsp, RCR2)); + dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n", MCBSP_READ(mcbsp, RCR1)); + dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n", MCBSP_READ(mcbsp, XCR2)); + dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n", MCBSP_READ(mcbsp, XCR1)); + dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", MCBSP_READ(mcbsp, SRGR2)); + dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", MCBSP_READ(mcbsp, SRGR1)); + dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n", MCBSP_READ(mcbsp, PCR0)); + dev_dbg(mcbsp->dev, "***********************\n"); +} + +static int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) +{ + struct clk *fck_src; + const char *src; + int r; + + if (fck_src_id == MCBSP_CLKS_PAD_SRC) + src = "pad_fck"; + else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) + src = "prcm_fck"; + else + return -EINVAL; + + fck_src = clk_get(mcbsp->dev, src); + if (IS_ERR(fck_src)) { + dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src); + return -EINVAL; + } + + pm_runtime_put_sync(mcbsp->dev); + + r = clk_set_parent(mcbsp->fclk, fck_src); + if (r) { + dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", + src); + clk_put(fck_src); + return r; + } + + pm_runtime_get_sync(mcbsp->dev); + + clk_put(fck_src); + + return 0; +} + +static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst; + + irqst = MCBSP_READ(mcbsp, IRQST); + dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); + + if (irqst & RSYNCERREN) + dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); + if (irqst & RFSREN) + dev_dbg(mcbsp->dev, "RX Frame Sync\n"); + if (irqst & REOFEN) + dev_dbg(mcbsp->dev, "RX End Of Frame\n"); + if (irqst & RRDYEN) + dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); + if (irqst & RUNDFLEN) + dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); + if (irqst & ROVFLEN) + dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); + + if (irqst & XSYNCERREN) + dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); + if (irqst & XFSXEN) + dev_dbg(mcbsp->dev, "TX Frame Sync\n"); + if (irqst & XEOFEN) + dev_dbg(mcbsp->dev, "TX End Of Frame\n"); + if (irqst & XRDYEN) + dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); + if (irqst & XUNDFLEN) + dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); + if (irqst & XOVFLEN) + dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); + if (irqst & XEMPTYEOFEN) + dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); + + MCBSP_WRITE(mcbsp, IRQST, irqst); + + return IRQ_HANDLED; +} + +static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst_spcr2; + + irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2); + dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); + + if (irqst_spcr2 & XSYNC_ERR) { + dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n", + irqst_spcr2); + /* Writing zero to XSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2)); + } + + return IRQ_HANDLED; +} + +static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst_spcr1; + + irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1); + dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); + + if (irqst_spcr1 & RSYNC_ERR) { + dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n", + irqst_spcr1); + /* Writing zero to RSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1)); + } + + return IRQ_HANDLED; +} + +/* + * omap_mcbsp_config simply write a config to the + * appropriate McBSP. + * You either call this function or set the McBSP registers + * by yourself before calling omap_mcbsp_start(). + */ +static void omap_mcbsp_config(struct omap_mcbsp *mcbsp, + const struct omap_mcbsp_reg_cfg *config) +{ + dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", + mcbsp->id, mcbsp->phys_base); + + /* We write the given config */ + MCBSP_WRITE(mcbsp, SPCR2, config->spcr2); + MCBSP_WRITE(mcbsp, SPCR1, config->spcr1); + MCBSP_WRITE(mcbsp, RCR2, config->rcr2); + MCBSP_WRITE(mcbsp, RCR1, config->rcr1); + MCBSP_WRITE(mcbsp, XCR2, config->xcr2); + MCBSP_WRITE(mcbsp, XCR1, config->xcr1); + MCBSP_WRITE(mcbsp, SRGR2, config->srgr2); + MCBSP_WRITE(mcbsp, SRGR1, config->srgr1); + MCBSP_WRITE(mcbsp, MCR2, config->mcr2); + MCBSP_WRITE(mcbsp, MCR1, config->mcr1); + MCBSP_WRITE(mcbsp, PCR0, config->pcr0); + if (mcbsp->pdata->has_ccr) { + MCBSP_WRITE(mcbsp, XCCR, config->xccr); + MCBSP_WRITE(mcbsp, RCCR, config->rccr); + } + /* Enable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); + + /* Enable TX/RX sync error interrupts by default */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | + RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); +} + +/** + * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register + * @mcbsp: omap_mcbsp struct for the McBSP instance + * @stream: Stream direction (playback/capture) + * + * Returns the address of mcbsp data transmit register or data receive register + * to be used by DMA for transferring/receiving data + */ +static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, + unsigned int stream) +{ + int data_reg; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (mcbsp->pdata->reg_size == 2) + data_reg = OMAP_MCBSP_REG_DXR1; + else + data_reg = OMAP_MCBSP_REG_DXR; + } else { + if (mcbsp->pdata->reg_size == 2) + data_reg = OMAP_MCBSP_REG_DRR1; + else + data_reg = OMAP_MCBSP_REG_DRR; + } + + return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; +} + +/* + * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH2 register. + */ +static void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +{ + if (threshold && threshold <= mcbsp->max_tx_thres) + MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); +} + +/* + * omap_mcbsp_set_rx_threshold configures the receive threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH1 register. + */ +static void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +{ + if (threshold && threshold <= mcbsp->max_rx_thres) + MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); +} + +/* + * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO + */ +static u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) +{ + u16 buffstat; + + /* Returns the number of free locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); + + /* Number of slots are different in McBSP ports */ + return mcbsp->pdata->buffer_size - buffstat; +} + +/* + * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO + * to reach the threshold value (when the DMA will be triggered to read it) + */ +static u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) +{ + u16 buffstat, threshold; + + /* Returns the number of used locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); + /* RX threshold */ + threshold = MCBSP_READ(mcbsp, THRSH1); + + /* Return the number of location till we reach the threshold limit */ + if (threshold <= buffstat) + return 0; + else + return threshold - buffstat; +} + +static int omap_mcbsp_request(struct omap_mcbsp *mcbsp) +{ + void *reg_cache; + int err; + + reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); + if (!reg_cache) + return -ENOMEM; + + spin_lock(&mcbsp->lock); + if (!mcbsp->free) { + dev_err(mcbsp->dev, "McBSP%d is currently in use\n", mcbsp->id); + err = -EBUSY; + goto err_kfree; + } + + mcbsp->free = false; + mcbsp->reg_cache = reg_cache; + spin_unlock(&mcbsp->lock); + + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) + mcbsp->pdata->ops->request(mcbsp->id - 1); + + /* + * Make sure that transmitter, receiver and sample-rate generator are + * not running before activating IRQs. + */ + MCBSP_WRITE(mcbsp, SPCR1, 0); + MCBSP_WRITE(mcbsp, SPCR2, 0); + + if (mcbsp->irq) { + err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, + "McBSP", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request IRQ\n"); + goto err_clk_disable; + } + } else { + err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, + "McBSP TX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); + goto err_clk_disable; + } + + err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, + "McBSP RX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); + goto err_free_irq; + } + } + + return 0; +err_free_irq: + free_irq(mcbsp->tx_irq, (void *)mcbsp); +err_clk_disable: + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id - 1); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + spin_lock(&mcbsp->lock); + mcbsp->free = true; + mcbsp->reg_cache = NULL; +err_kfree: + spin_unlock(&mcbsp->lock); + kfree(reg_cache); + + return err; +} + +static void omap_mcbsp_free(struct omap_mcbsp *mcbsp) +{ + void *reg_cache; + + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id - 1); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + /* Disable interrupt requests */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, 0); + + if (mcbsp->irq) { + free_irq(mcbsp->irq, (void *)mcbsp); + } else { + free_irq(mcbsp->rx_irq, (void *)mcbsp); + free_irq(mcbsp->tx_irq, (void *)mcbsp); + } + + reg_cache = mcbsp->reg_cache; + + /* + * Select CLKS source from internal source unconditionally before + * marking the McBSP port as free. + * If the external clock source via MCBSP_CLKS pin has been selected the + * system will refuse to enter idle if the CLKS pin source is not reset + * back to internal source. + */ + if (!mcbsp_omap1()) + omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); + + spin_lock(&mcbsp->lock); + if (mcbsp->free) + dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); + else + mcbsp->free = true; + mcbsp->reg_cache = NULL; + spin_unlock(&mcbsp->lock); + + kfree(reg_cache); +} + +/* + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. + */ +static void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) +{ + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; + int enable_srg = 0; + u16 w; + + if (mcbsp->st_data) + omap_mcbsp_st_start(mcbsp); + + /* Only enable SRG, if McBSP is master */ + w = MCBSP_READ_CACHE(mcbsp, PCR0); + if (w & (FSXM | FSRM | CLKXM | CLKRM)) + enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (enable_srg) { + /* Start the sample generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6)); + } + + /* Enable transmitter and receiver */ + tx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | tx); + + rx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w | rx); + + /* + * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec + * REVISIT: 100us may give enough time for two CLKSRG, however + * due to some unknown PM related, clock gating etc. reason it + * is now at 500us. + */ + udelay(500); + + if (enable_srg) { + /* Start frame sync */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7)); + } + + if (mcbsp->pdata->has_ccr) { + /* Release the transmitter and receiver */ + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w &= ~(tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w &= ~(rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + + /* Dump McBSP Regs */ + omap_mcbsp_dump_reg(mcbsp); +} + +static void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) +{ + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; + int idle; + u16 w; + + /* Reset transmitter */ + tx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w |= (tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~tx); + + /* Reset receiver */ + rx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w |= (rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w & ~rx); + + idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6)); + } + + if (mcbsp->st_data) + omap_mcbsp_st_stop(mcbsp); +} + +#define max_thres(m) (mcbsp->pdata->buffer_size) +#define valid_threshold(m, val) ((val) <= max_thres(m)) +#define THRESHOLD_PROP_BUILDER(prop) \ +static ssize_t prop##_show(struct device *dev, \ + struct device_attribute *attr, char *buf) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + \ + return sprintf(buf, "%u\n", mcbsp->prop); \ +} \ + \ +static ssize_t prop##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t size) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + unsigned long val; \ + int status; \ + \ + status = kstrtoul(buf, 0, &val); \ + if (status) \ + return status; \ + \ + if (!valid_threshold(mcbsp, val)) \ + return -EDOM; \ + \ + mcbsp->prop = val; \ + return size; \ +} \ + \ +static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store) + +THRESHOLD_PROP_BUILDER(max_tx_thres); +THRESHOLD_PROP_BUILDER(max_rx_thres); + +static const char * const dma_op_modes[] = { + "element", "threshold", +}; + +static ssize_t dma_op_mode_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int dma_op_mode, i = 0; + ssize_t len = 0; + const char * const *s; + + dma_op_mode = mcbsp->dma_op_mode; + + for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { + if (dma_op_mode == i) + len += sprintf(buf + len, "[%s] ", *s); + else + len += sprintf(buf + len, "%s ", *s); + } + len += sprintf(buf + len, "\n"); + + return len; +} + +static ssize_t dma_op_mode_store(struct device *dev, + struct device_attribute *attr, const char *buf, + size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int i; + + i = sysfs_match_string(dma_op_modes, buf); + if (i < 0) + return i; + + spin_lock_irq(&mcbsp->lock); + if (!mcbsp->free) { + size = -EBUSY; + goto unlock; + } + mcbsp->dma_op_mode = i; + +unlock: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR_RW(dma_op_mode); + +static const struct attribute *additional_attrs[] = { + &dev_attr_max_tx_thres.attr, + &dev_attr_max_rx_thres.attr, + &dev_attr_dma_op_mode.attr, + NULL, +}; + +static const struct attribute_group additional_attr_group = { + .attrs = (struct attribute **)additional_attrs, +}; + +/* + * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. + * 730 has only 2 McBSP, and both of them are MPU peripherals. + */ +static int omap_mcbsp_init(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + struct resource *res; + int ret = 0; + + spin_lock_init(&mcbsp->lock); + mcbsp->free = true; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!res) + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + mcbsp->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(mcbsp->io_base)) + return PTR_ERR(mcbsp->io_base); + + mcbsp->phys_base = res->start; + mcbsp->reg_cache_size = resource_size(res); + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); + if (!res) + mcbsp->phys_dma_base = mcbsp->phys_base; + else + mcbsp->phys_dma_base = res->start; + + /* + * OMAP1, 2 uses two interrupt lines: TX, RX + * OMAP2430, OMAP3 SoC have combined IRQ line as well. + * OMAP4 and newer SoC only have the combined IRQ line. + * Use the combined IRQ if available since it gives better debugging + * possibilities. + */ + mcbsp->irq = platform_get_irq_byname(pdev, "common"); + if (mcbsp->irq == -ENXIO) { + mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); + + if (mcbsp->tx_irq == -ENXIO) { + mcbsp->irq = platform_get_irq(pdev, 0); + mcbsp->tx_irq = 0; + } else { + mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); + mcbsp->irq = 0; + } + } + + if (!pdev->dev.of_node) { + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); + if (!res) { + dev_err(&pdev->dev, "invalid tx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[0] = res->start; + mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); + if (!res) { + dev_err(&pdev->dev, "invalid rx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[1] = res->start; + mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; + } else { + mcbsp->dma_data[0].filter_data = "tx"; + mcbsp->dma_data[1].filter_data = "rx"; + } + + mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, + SNDRV_PCM_STREAM_PLAYBACK); + mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, + SNDRV_PCM_STREAM_CAPTURE); + + mcbsp->fclk = clk_get(&pdev->dev, "fck"); + if (IS_ERR(mcbsp->fclk)) { + ret = PTR_ERR(mcbsp->fclk); + dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); + return ret; + } + + mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; + if (mcbsp->pdata->buffer_size) { + /* + * Initially configure the maximum thresholds to a safe value. + * The McBSP FIFO usage with these values should not go under + * 16 locations. + * If the whole FIFO without safety buffer is used, than there + * is a possibility that the DMA will be not able to push the + * new data on time, causing channel shifts in runtime. + */ + mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; + mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; + + ret = sysfs_create_group(&mcbsp->dev->kobj, + &additional_attr_group); + if (ret) { + dev_err(mcbsp->dev, + "Unable to create additional controls\n"); + goto err_thres; + } + } else { + mcbsp->max_tx_thres = -EINVAL; + mcbsp->max_rx_thres = -EINVAL; + } + + ret = omap_mcbsp_st_init(pdev); + if (ret) + goto err_st; + + return 0; + +err_st: + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); +err_thres: + clk_put(mcbsp->fclk); + return ret; +} + /* * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs @@ -656,132 +1343,6 @@ static const struct snd_soc_component_driver omap_mcbsp_component = { .name = "omap-mcbsp", }; -static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; - int min = mc->min; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = min; - uinfo->value.integer.max = max; - return 0; -} - -#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel) \ -static int \ -omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ - struct snd_ctl_elem_value *uc) \ -{ \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ - struct soc_mixer_control *mc = \ - (struct soc_mixer_control *)kc->private_value; \ - int max = mc->max; \ - int min = mc->min; \ - int val = uc->value.integer.value[0]; \ - \ - if (val < min || val > max) \ - return -EINVAL; \ - \ - /* OMAP McBSP implementation uses index values 0..4 */ \ - return omap_st_set_chgain(mcbsp, channel, val); \ -} \ - \ -static int \ -omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ - struct snd_ctl_elem_value *uc) \ -{ \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ - s16 chgain; \ - \ - if (omap_st_get_chgain(mcbsp, channel, &chgain)) \ - return -EAGAIN; \ - \ - uc->value.integer.value[0] = chgain; \ - return 0; \ -} - -OMAP_MCBSP_ST_CHANNEL_VOLUME(0) -OMAP_MCBSP_ST_CHANNEL_VOLUME(1) - -static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - u8 value = ucontrol->value.integer.value[0]; - - if (value == omap_st_is_enabled(mcbsp)) - return 0; - - if (value) - omap_st_enable(mcbsp); - else - omap_st_disable(mcbsp); - - return 1; -} - -static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - - ucontrol->value.integer.value[0] = omap_st_is_enabled(mcbsp); - return 0; -} - -#define OMAP_MCBSP_ST_CONTROLS(port) \ -static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \ -SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0, \ - omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), \ -OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \ - -32768, 32767, \ - omap_mcbsp_get_st_ch0_volume, \ - omap_mcbsp_set_st_ch0_volume), \ -OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \ - -32768, 32767, \ - omap_mcbsp_get_st_ch1_volume, \ - omap_mcbsp_set_st_ch1_volume), \ -} - -OMAP_MCBSP_ST_CONTROLS(2); -OMAP_MCBSP_ST_CONTROLS(3); - -int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - - if (!mcbsp->st_data) { - dev_warn(mcbsp->dev, "No sidetone data for port\n"); - return 0; - } - - switch (port_id) { - case 2: /* McBSP 2 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp2_st_controls, - ARRAY_SIZE(omap_mcbsp2_st_controls)); - case 3: /* McBSP 3 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp3_st_controls, - ARRAY_SIZE(omap_mcbsp3_st_controls)); - default: - dev_err(mcbsp->dev, "Port %d not supported\n", port_id); - break; - } - - return -EINVAL; -} -EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); - static struct omap_mcbsp_platform_data omap2420_pdata = { .reg_step = 4, .reg_size = 2, @@ -893,7 +1454,10 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (pm_qos_request_active(&mcbsp->pm_qos_req)) pm_qos_remove_request(&mcbsp->pm_qos_req); - omap_mcbsp_cleanup(mcbsp); + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); + + omap_mcbsp_st_cleanup(pdev); clk_put(mcbsp->fclk); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 2e3369c27be3..7911d24898c9 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -22,8 +22,10 @@ * */ -#ifndef __OMAP_I2S_H__ -#define __OMAP_I2S_H__ +#ifndef __OMAP_MCBSP_H__ +#define __OMAP_MCBSP_H__ + +#include <sound/dmaengine_pcm.h> /* Source clocks for McBSP sample rate generator */ enum omap_mcbsp_clksrg_clk { @@ -41,4 +43,4 @@ enum omap_mcbsp_div { int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id); -#endif +#endif /* __OMAP_MCBSP_H__ */ -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata 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 465d85b3976a15c9e0e205ff50be64efa11b7af3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:59 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata The driver will not probe if the pdata is not provided or created. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/omap-mcbsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index e48fad11a0cc..8cab1151d3bb 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -324,7 +324,7 @@ static int omap_mcbsp_request(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = reg_cache; spin_unlock(&mcbsp->lock); - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) + if(mcbsp->pdata->ops && mcbsp->pdata->ops->request) mcbsp->pdata->ops->request(mcbsp->id - 1); /* @@ -361,7 +361,7 @@ static int omap_mcbsp_request(struct omap_mcbsp *mcbsp) err_free_irq: free_irq(mcbsp->tx_irq, (void *)mcbsp); err_clk_disable: - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + if(mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id - 1); /* Disable wakeup behavior */ @@ -382,7 +382,7 @@ static void omap_mcbsp_free(struct omap_mcbsp *mcbsp) { void *reg_cache; - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + if(mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id - 1); /* Disable wakeup behavior */ -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used 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 f15c19af929d380d0acb2a8d8277d688351bd5de Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:30:00 +0200 Subject: [PATCH] ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used Initializing to -EINVAL is not correct as the variables are unsigned and if buffer_size is 0 then they are not used anyway. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/omap-mcbsp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8cab1151d3bb..e0ef4774c710 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -727,9 +727,6 @@ static int omap_mcbsp_init(struct platform_device *pdev) "Unable to create additional controls\n"); goto err_thres; } - } else { - mcbsp->max_tx_thres = -EINVAL; - mcbsp->max_rx_thres = -EINVAL; } ret = omap_mcbsp_st_init(pdev); -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: rockchip: add missing slave_config setting for I2S
by Katsuhiro Suzuki 13 Nov '18

13 Nov '18

This patch adds missing prepare_sleve_config that is needed for setup the DMA slave channel for I2S. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index 9e7b5fa4cf59..4ac78d7a4b2d 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -33,6 +33,7 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { .pcm_hardware = &snd_rockchip_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .prealloc_buffer_size = 32 * 1024, }; -- 2.19.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Fix dma-unsafe read of scratch registers
by Richard Fitzgerald 13 Nov '18

13 Nov '18

Stack memory isn't DMA-safe so it isn't safe to use either regmap_raw_read or regmap_bulk_read to read into stack memory. The two functions to read the scratch registers were using stack memory and regmap_raw_read. It's not worth allocating memory just for this trivial read, and it isn't time-critical. A simple regmap_read for each register is sufficient. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a53dc174bbf0..66501b8dc46f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -765,38 +765,41 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) { - u16 scratch[4]; + unsigned int scratch[4]; + unsigned int addr = dsp->base + ADSP2_SCRATCH0; + unsigned int i; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, - scratch, sizeof(scratch)); - if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); - return; + for (i = 0; i < ARRAY_SIZE(scratch); ++i) { + ret = regmap_read(dsp->regmap, addr + i, &scratch[i]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret); + return; + } } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", - be16_to_cpu(scratch[0]), - be16_to_cpu(scratch[1]), - be16_to_cpu(scratch[2]), - be16_to_cpu(scratch[3])); + scratch[0], scratch[1], scratch[2], scratch[3]); } static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp) { - u32 scratch[2]; + unsigned int scratch[2]; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, - scratch, sizeof(scratch)); - + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, + &scratch[0]); if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret); return; } - scratch[0] = be32_to_cpu(scratch[0]); - scratch[1] = be32_to_cpu(scratch[1]); + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3, + &scratch[1]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret); + return; + } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", scratch[0] & 0xFFFF, -- 2.11.0

2 3

Re: [alsa-devel] Realtek ALC294 Intel Sunrise Point-LP HD Audio driver - sound does not work
by Takashi Iwai 13 Nov '18

13 Nov '18

On Tue, 13 Nov 2018 19:11:18 +0100, Tomasz PrzybyX2 wrote: > > Hello Takashi, > > > Does the headphone work? > > No. For making the headphone working, you don't need any jack detection, in general. You can disable the auto-mute mode. And, at best, avoid PulseAudio for the basic testing in this level. If both headphone and speaker don't work even with the statically enabled routing, I have no idea what's wrong. You really need to ping ASUS for a better help. It's vendor-specific things, after all. thanks, Takashi > > Internal speakers are muted > Headphones and external speakers connected via JACK output combo are > muted . Even with `options snd_hda_intel index=0 > model=dell-headset-multi` custom flag > > QUOTE from arch wiki > https://wiki.archlinux.org/index.php/Advanced_Linux_Sound_Architecture#Corr… > --- > Correctly detect microphone plugged in a 4-pin 3.5mm (TRRS) jack > On some modern laptops you may have a combined 3.5mm headset jack, > instead of two separated ones, which may not be correctly detected by > default. To make ALSA correctly detect plug-in status on your 3.5mm > jack, you could put the following line into your > /etc/modprobe.d/alsa-base.conf : > > # options snd_hda_intel index=0 model=[YOUR_MODEL_SETTING] > For complete list of options put in [YOUR_MODEL_SETTING], see > HD-Audio-Models.txt. Tested with Gigabyte Aero15 2017 (P65 Model), > with ALC255 and set model to dell-headset-multi. > > ---- > > See main bug report > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1784485 > > Others forum issue reports: > > https://forum.manjaro.org/t/alc294-analog-audio-card-detected-but-no-sound-… > > https://forum.ubuntuusers.de/topic/asus-zenbook-lautsprecher-ohne-funktion/ > > > One more report from me: > ``` > > [manjaro@manjaro ~]$ pactl list sinks > Sink #0 > State: SUSPENDED > Name: alsa_output.pci-0000_00_1f.3.analog-stereo > Description: Built-in Audio Analog Stereo > Driver: module-alsa-card.c > Sample Specification: s16le 2ch 44100Hz > Channel Map: front-left,front-right > Owner Module: 6 > Mute: no > Volume: front-left: 30129 / 46% / -20.25 dB, front-right: > 30129 / 46% / -20.25 dB > balance 0.00 > Base Volume: 65536 / 100% / 0.00 dB > Monitor Source: alsa_output.pci-0000_00_1f.3.analog-stereo.monitor > Latency: 0 usec, configured 0 usec > Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY > Properties: > alsa.resolution_bits = "16" > device.api = "alsa" > device.class = "sound" > alsa.class = "generic" > alsa.subclass = "generic-mix" > alsa.name = "ALC294 Analog" > alsa.id = "ALC294 Analog" > alsa.subdevice = "0" > alsa.subdevice_name = "subdevice #0" > alsa.device = "0" > alsa.card = "0" > alsa.card_name = "HDA Intel PCH" > alsa.long_card_name = "HDA Intel PCH at 0xef528000 irq 130" > alsa.driver_name = "snd_hda_intel" > device.bus_path = "pci-0000:00:1f.3" > sysfs.path = "/devices/pci0000:00/0000:00:1f.3/sound/card0" > device.bus = "pci" > device.vendor.id = "8086" > device.vendor.name = "Intel Corporation" > device.product.id = "9d71" > device.product.name = "Sunrise Point-LP HD Audio" > device.form_factor = "internal" > device.string = "front:0" > device.buffering.buffer_size = "352800" > device.buffering.fragment_size = "176400" > device.access_mode = "mmap+timer" > device.profile.name = "analog-stereo" > device.profile.description = "Analog Stereo" > device.description = "Built-in Audio Analog Stereo" > alsa.mixer_name = "Realtek ALC294" > alsa.components = "HDA:10ec0294,10431a00,00100004 > HDA:8086280b,80860101,00100000" > module-udev-detect.discovered = "1" > device.icon_name = "audio-card-pci" > Ports: > analog-output-speaker: Speakers (priority: 10000, not > available) > analog-output-headphones: Headphones (priority: 9000, > available) > Active Port: analog-output-headphones > Formats: > pcm > [manjaro@manjaro ~]$ > > ``` > > > This is common issue. > > Tomasz > > W dniu 13.11.2018 o 18:37, Takashi Iwai pisze: > > On Sat, 10 Nov 2018 14:42:01 +0100, > > Tomasz PrzybyX2 wrote: > >> > >> Hello Devs > >> > >> Lots of users with Intel Sunrise Point-LP HD Audio driver (ALC294) > >> (snd_hda_intel) have major issue. Not sound at all. Only crackling > >> from external speakers OR silence from internal notebook speakers. > >> Tried fix with extra boot flags does not work at all. > > > > Does the headphone work? If yes, it's possibly either a wrong pin > > config or some extra stuff needed to drive the speaker amp. > > In the latter case, only vendors know the details. > > > > > > Takashi > > > >> > >> Also affected Asus VivoBook R542UF > >> > >> ``` > >> Machine: > >> Type: Laptop System: ASUSTeK product: VivoBook 15_ASUS Laptop X542UF > >> v: 1.0 serial: <filter> > >> Mobo: ASUSTeK model: X542UF v: 1.0 serial: <filter> > >> UEFI: American Megatrends v: X542UF.303 date: 03/15/2018 > >> Audio: > >> Device-1: Intel Sunrise Point-LP HD Audio driver: snd_hda_intel > >> v: kernel bus ID: 00:1f.3 > >> Sound Server: ALSA Kernel: 4.19 > >> ``` > >> ``` > >> aplay -l > >> **** List of PLAYBACK Hardware Devices **** > >> card 0: PCH [HDA Intel PCH], device 0: ALC294 Analog [ALC294 Analog] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> ``` > >> Dmesg parts > >> ``` > >> [ 12.342528] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: > >> line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker > >> [ 12.342531] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 > >> (0x0/0x0/0x0/0x0/0x0) > >> [ 12.342533] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 > >> (0x21/0x0/0x0/0x0/0x0) > >> [ 12.342535] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 > >> [ 12.342536] snd_hda_codec_realtek hdaudioC0D0: inputs: > >> [ 12.342538] snd_hda_codec_realtek hdaudioC0D0: Mic=0x1b > >> ``` > >> ``` > >> [ 12.634171] input: HDA Intel PCH Headphone as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input18 > >> [ 12.634389] input: HDA Intel PCH HDMI/DP,pcm=3 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input19 > >> [ 12.634563] input: HDA Intel PCH HDMI/DP,pcm=7 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 > >> [ 12.634737] input: HDA Intel PCH HDMI/DP,pcm=8 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 > >> [ 12.634905] input: HDA Intel PCH HDMI/DP,pcm=9 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input22 > >> [ 12.635045] input: HDA Intel PCH HDMI/DP,pcm=10 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input23 > >> ``` > >> > >> Regards > >> > >> Tomasz > >> > >> _______________________________________________ > >> Alsa-devel mailing list > >> Alsa-devel(a)alsa-project.org > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

1 0

[alsa-devel] [sound:for-linus 5/5] sound/pci//hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'?
by kbuild test robot 13 Nov '18

13 Nov '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: d99501b8575dc1248bacf1b58d2241cb4b265d49 commit: d99501b8575dc1248bacf1b58d2241cb4b265d49 [5/5] ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap() config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d99501b8575dc1248bacf1b58d2241cb4b265d49 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci//hda/patch_ca0132.c: In function 'ca0132_free': >> sound/pci//hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(codec->bus->pci, spec->mem_base); ^~~~~~~~~~~ pcim_iounmap sound/pci//hda/patch_ca0132.c: In function 'patch_ca0132': sound/pci//hda/patch_ca0132.c:8799:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci//hda/patch_ca0132.c:8799:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +8416 sound/pci//hda/patch_ca0132.c 8386 8387 static void ca0132_free(struct hda_codec *codec) 8388 { 8389 struct ca0132_spec *spec = codec->spec; 8390 8391 cancel_delayed_work_sync(&spec->unsol_hp_work); 8392 snd_hda_power_up(codec); 8393 switch (spec->quirk) { 8394 case QUIRK_SBZ: 8395 sbz_exit_chip(codec); 8396 break; 8397 case QUIRK_ZXR: 8398 zxr_exit_chip(codec); 8399 break; 8400 case QUIRK_R3D: 8401 r3d_exit_chip(codec); 8402 break; 8403 case QUIRK_AE5: 8404 ae5_exit_chip(codec); 8405 break; 8406 case QUIRK_R3DI: 8407 r3di_gpio_shutdown(codec); 8408 break; 8409 } 8410 8411 snd_hda_sequence_write(codec, spec->base_exit_verbs); 8412 ca0132_exit_chip(codec); 8413 8414 snd_hda_power_down(codec); 8415 if (spec->mem_base) > 8416 pci_iounmap(codec->bus->pci, spec->mem_base); 8417 kfree(spec->spec_init_verbs); 8418 kfree(codec->spec); 8419 } 8420 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 17/17] ASoc: bcm2835-i2s: Switch to SPDX identifier
by Stefan Wahren 13 Nov '18

13 Nov '18

Adopt the SPDX license identifier headers to ease license compliance management. Cc: Florian Kauer <florian.kauer(a)koalo.de> Signed-off-by: Stefan Wahren <stefan.wahren(a)i2se.com> --- sound/soc/bcm/bcm2835-i2s.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index d5f73a8..0e4a0e9 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -1,3 +1,4 @@ +// SPDX-License-Identifier: GPL-2.0 /* * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC * @@ -20,15 +21,6 @@ * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver * Author: Timur Tabi <timur(a)freescale.com> * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. */ #include <linux/bitops.h> -- 2.7.4

2 1

[alsa-devel] snd_pcm_hw_params resets microphone boost
by sonntex 13 Nov '18

13 Nov '18

I have small program that captures audio from a microphone. As my microphone is very sensitive, I prefer to mute the microphone boost before capturing. I open alsamixer and set volume of the microphone boost to zero, save settings and exit. Then I execute my program. I see that it resets volume of the microphone boost in snd_pcm_hw_params. What should I do resolve this problem? Where can I find default value and change it?

2 1

[alsa-devel] Realtek ALC294 Intel Sunrise Point-LP HD Audio driver - sound does not work
by Tomasz Przybył 13 Nov '18

13 Nov '18

Hello Devs Lots of users with Intel Sunrise Point-LP HD Audio driver (ALC294) (snd_hda_intel) have major issue. Not sound at all. Only crackling from external speakers OR silence from internal notebook speakers. Tried fix with extra boot flags does not work at all. Also affected Asus VivoBook R542UF ``` Machine: Type: Laptop System: ASUSTeK product: VivoBook 15_ASUS Laptop X542UF v: 1.0 serial: <filter> Mobo: ASUSTeK model: X542UF v: 1.0 serial: <filter> UEFI: American Megatrends v: X542UF.303 date: 03/15/2018 Audio: Device-1: Intel Sunrise Point-LP HD Audio driver: snd_hda_intel v: kernel bus ID: 00:1f.3 Sound Server: ALSA Kernel: 4.19 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC294 Analog [ALC294 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Dmesg parts ``` [ 12.342528] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker [ 12.342531] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 12.342533] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 12.342535] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.342536] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.342538] snd_hda_codec_realtek hdaudioC0D0: Mic=0x1b ``` ``` [ 12.634171] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input18 [ 12.634389] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input19 [ 12.634563] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 [ 12.634737] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 [ 12.634905] input: HDA Intel PCH HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input22 [ 12.635045] input: HDA Intel PCH HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input23 ``` Regards Tomasz

2 1

[alsa-devel] [PATCH v5 2/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 13 Nov '18

13 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..135c0826859b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5764,6 +5765,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6594,6 +6611,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.17.2

2 1

[alsa-devel] using the softvol plugin with min_dB > 0
by Guillermo Rodriguez Garcia 13 Nov '18

13 Nov '18

Hi all, I have a board with a tlv320aic3106 codec and a digital mic. The tlv320aic3106 does not support hardware gain for the digital mic, so I am using the softvol plugin to provide this via software. I would like to setup softvol so that it applies a gain in a range that goes from approx +10dB to approx +40dB. The problem is that softvol only accepts negative values for min_dB, i.e. assumes that you *always* want to do attenuation, and that you might or might not want to have gain (max_dB can be zero or negative). Is there any reason for this, or is it just historical ? (I am aware that the original softvol plugin only supported attenuation, and that support for gain was added later) Is it possible to setup softvol to accept a positive min_dB value ? Best regards, Guillermo Rodriguez Garcia guille.rodriguez(a)gmail.com

2 1

[alsa-devel] Help with extplug transfer callback
by GitHub issues - opened 13 Nov '18

13 Nov '18

alsa-project/alsa-lib issue #2 was opened from dchabot: I've tried posting the following at alsa-users mailing list, but keep getting hit with "awaiting moderator approval" (I am registered to the list). I'm trying to create an extplug and having trouble deciphering how the source and destination buffer reading/writing is supposed to be done in the transfer callback. My goal is to create an extplug that takes a stereo data stream, divides that into 6 or 8 output streams, and applies FIR filters to each output stream. With the code I currently have (https://github.com/dchabot/extplug), trying to play a 1kHz tone results in a garbled, noisy mess. There is a tone in there, but it is a mess. I started from the skeleton code here - http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html. Note that the snd_pcm_extplug_create call here should be passing the "slave" variable, not "conf". That tripped me up for the better part of an hour. I based my initial transfer callback (snd_pcm_extplug_callback_t.transfer member) on that found in the speex plugin. Initially I didn't notice that speex operated with single in/out channels, so you can imagine how that went. Looking at another example of an extplug (https://github.com/dpapavas/alsaloudness), I noted that I should be iterating over each channel for src_areas and dst_areas. I just cannot see how this should be done properly. Is documentation for this critical step missing or simply hard to find. Seems kinda important to know how buffer operations should be done... At this point, all I want to do is copy input channels to output channels. I'm grappling with the fact that this is more complicated than I initially thought, or perhaps more difficult than it should be. My transfer callback currently looks like this: ``` /* copied from pcm-plugins/speex/pcm-speex.c */ static inline void* area_addr(const snd_pcm_channel_area_t *area, snd_pcm_uframes_t offset) { unsigned int bitofs = area->first + area->step * offset; return (char *) area->addr + bitofs / 8; } static snd_pcm_sframes_t transfer_callback(snd_pcm_extplug_t *ext, const snd_pcm_channel_area_t *dst_areas, snd_pcm_uframes_t dst_offset, const snd_pcm_channel_area_t *src_areas, snd_pcm_uframes_t src_offset, snd_pcm_uframes_t size) { int C = ext->channels; short *src, *dst; for(int i = 0; i < C; i++) { src = area_addr(&src_areas[i], src_offset); dst = area_addr(&dst_areas[i], dst_offset); memcpy(dst, src, size); } return size; } ``` Need a little help please. Issue URL : https://github.com/alsa-project/alsa-lib/issues/2 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH 1/4] README: update Vinod's email
by Vinod Koul 13 Nov '18

13 Nov '18

Update email address in README from old invalid id Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- README | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/README b/README index 689ca75e07fd..8ea6c26f5381 100644 --- a/README +++ b/README @@ -1,7 +1,7 @@ README for tinycompress ======================= - vinod.koul(a)linux.intel.com - ========================== + vkoul(a)kernel.org + ================ 1. WHAT tinycompress is a userspace library for anyone who wants to use the ALSA -- 2.14.5

1 3

[alsa-devel] [PATCH] tinycompress: Update headers to include AUDIOCODEC_BESPOKE
by Charles Keepax 13 Nov '18

13 Nov '18

commit dbb6b94339e8 ("ALSA: compress: Add SND_AUDIOCODEC_BESPOKE") added an additional CODEC type for bespoke CODECs. Update the local headers to include this. Reported-by: Andrew Ford <andrew.ford(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/compress_params.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h index 9802292..7286556 100644 --- a/include/sound/compress_params.h +++ b/include/sound/compress_params.h @@ -38,8 +38,9 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) +#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE #define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001) #define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001) -- 2.11.0

2 1

[alsa-devel] [PATCH - dmix v5 0/1] pcm: dmix: Add option to allow alignment of slave pointers
by twischer＠de.adit-jv.com 13 Nov '18

13 Nov '18

On 11/12/18 17:20, Takashi Iwai wrote: > Please provide your own sign-off when you submitting a patch, too. > > The subject line should be adjusted better to match with the change; > it's about adding an option to allow alignment of hwptr reporting. > > About the changes: > the term "hw_ptr_filter" might sound confusing. IMO, this is about > the (hw)ptr alignment, not about filtering. > Also, a too long option string can easily lead to a misspell :) > > After all, we do either "no" alignment, "roundup" or "rounddown", or > "auto" choice. > > > thanks, > > Takashi Subject changed. Sign-off added. Option changed to hw_ptr_alignment. Option values changed to no, roundup, rounddown, auto Best regards Timo

2 2

[alsa-devel] [PATCH] ALSA: oxfw: add support for APOGEE duet FireWire
by Takashi Sakamoto 13 Nov '18

13 Nov '18

This commit adds support for APOGEE duet FireWire, launched 2007, already discontinued. This model uses Oxford Semiconductor FW971 as its communication engine. Below is information on Configuration ROM of this unit. The unit supports some AV/C commands defined by Audio subunit specification and vendor dependent commands. $ ./hinawa-config-rom-printer /dev/fw1 { 'bus-info': { 'adj': False, 'bmc': False, 'chip_ID': 42949742248, 'cmc': False, 'cyc_clk_acc': 255, 'generation': 0, 'imc': False, 'isc': True, 'link_spd': 3, 'max_ROM': 0, 'max_rec': 64, 'name': '1394', 'node_vendor_ID': 987, 'pmc': False}, 'root-directory': [ ['VENDOR', 987], ['DESCRIPTOR', 'Apogee Electronics'], ['MODEL', 122333], ['DESCRIPTOR', 'Duet'], [ 'NODE_CAPABILITIES', { 'addressing': {'64': True, 'fix': True, 'prv': False}, 'misc': {'int': False, 'ms': False, 'spt': True}, 'state': { 'atn': False, 'ded': False, 'drq': True, 'elo': False, 'init': False, 'lst': True, 'off': False}, 'testing': {'bas': False, 'ext': False}}], [ 'UNIT', [ ['SPECIFIER_ID', 41005], ['VERSION', 65537], ['MODEL', 122333], ['DESCRIPTOR', 'Duet']]]]} Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/oxfw/oxfw.c | 8 ++++++++ 2 files changed, 9 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 8a146b039276..44cedb65bb88 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -41,6 +41,7 @@ config SND_OXFW * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer + * APOGEE duet FireWire To compile this driver as a module, choose M here: the module will be called snd-oxfw. diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index afb78d90384b..3d27f3378d5d 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -20,6 +20,7 @@ #define VENDOR_LACIE 0x00d04b #define VENDOR_TASCAM 0x00022e #define OUI_STANTON 0x001260 +#define OUI_APOGEE 0x0003db #define MODEL_SATELLITE 0x00200f @@ -397,6 +398,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = { .vendor_id = OUI_STANTON, .model_id = 0x002000, }, + // APOGEE, duet FireWire + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_APOGEE, + .model_id = 0x01dddd, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); -- 2.19.1

2 1

[alsa-devel] [PATCH - dmix v4 0/1] pcm: dmix: Align slave_hw_ptr to slave period
by twischer＠de.adit-jv.com 12 Nov '18

12 Nov '18

Hi Takashi, in the following we are providing the patch as requested by you: On 11/6/18 17:32, Takashi Iwai wrote: > I guess we may provide an option for choosing one of the following > modes: > > - current default mode; > no latency, but wakeup may be delayed, and may require 2*periods > > - appl_ptr round up to the next period; > the current workaround for periods <= 2. > This imposes the start latency. > > - round down to the current period start; > No latency, but will drop a few samples at start. > > ... and ideally, provide a mode "auto" for choosing the appropriate > one depending on the situation by a smart guess. Please provide your feedback. Best regards Timo

2 2

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()
by Takashi Iwai 12 Nov '18

12 Nov '18

We need to call pci_iounmap() instead of iounmap() for the regions obtained via pci_iomap() call for some archs that need special treatment. Fixes: aa31704fd81c ("ALSA: hda/ca0132: Add PCI region2 iomap for SBZ") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0a24037184c3..bdbbc51e8d18 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -8413,7 +8413,7 @@ static void ca0132_free(struct hda_codec *codec) snd_hda_power_down(codec); if (spec->mem_base) - iounmap(spec->mem_base); + pci_iounmap(codec->bus->pci, spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); } -- 2.19.1

1 0

[alsa-devel] [PATCH 0/3] constify snd_soc_dai_ops structures
by Julia Lawall 12 Nov '18

12 Nov '18

The snd_soc_dai_ops structures are only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structures const as well. Done with the help of Coccinelle. --- drivers/soundwire/intel.c | 4 ++-- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/hdac_hda.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 8 ++++---- 6 files changed, 10 insertions(+), 10 deletions(-)

3 6

[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirk entry for HP Pavilion 15
by Takashi Iwai 12 Nov '18

12 Nov '18

HP Pavilion 15 (103c:820d) with ALC295 codec requires the quirk for the mute LED control over mic3 pin. Added the corresponding quirk entry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201653 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..970bc44a378b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6481,6 +6481,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), -- 2.19.1

1 0

[alsa-devel] [PATCH v4 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 11 Nov '18

11 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables and disable this LED accordingly. This enables micmute LED for: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/huawei_wmi_helper.c | 47 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 14 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..7c91f914ffba --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI micmute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_LAPTOP) +#include <linux/platform_data/x86/huawei_wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + huawei_wmi_micmute_led_set_func(spec->micmute_led.led_value); +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = (huawei_wmi_micmute_led_set_func(false) < 0) + || (snd_hda_gen_add_micmute_led(codec, update_huawei_wmi_micmute_led) < 0); + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 135c0826859b..5431dee40f4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5374,6 +5374,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc_fixup_huawei_micmute_led */ +#include "huawei_wmi_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5765,6 +5769,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5779,7 +5787,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6611,6 +6621,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

3 2

[alsa-devel] [PATCH 00/17] bcm2835: Treewide switch to SPDX identifier
by Stefan Wahren 10 Nov '18

10 Nov '18

After fixing the module license of some bcm2835 drivers in "Fix MODULE_LICENSE of bcm2835 drivers", we can finally switch all outstanding bcm2835 / Raspberry Pi drivers to SPDX identifier and drop the unnecessary license text. I omit the staging area, because there are proposed a lot of changes for vc04_services. I hope it is okay to have the binding header integrated in the tree-wise patches instead of a separate one. This should avoid conflicts while merging. Stefan Wahren (17): ARM: dts: bcm2835-rpi-zero: Switch to SPDX identifier ARM: mach-bcm: Switch bcm2835 and platsmp to SPDX identifier hw_random: bcm2835-rng: Switch to SPDX identifier clk: bcm2835: Switch to SPDX identifier clocksource: bcm2835: Switch to SPDX identifier dma: bcm2835: Switch to SPDX identifier firmware: raspberrypi: Switch to SPDX identifier i2c: bcm2835: Switch to SPDX identifier irqchip: bcm283x: Switch to SPDX identifier mailbox: bcm2835: Switch to SPDX identifier mmc: bcm2835: Switch to SPDX identifier pinctrl: bcm2835: Switch to SPDX identifier pwm: bcm2835: Switch to SPDX identifier soc: bcm: Switch raspberrypi-power to SPDX identifier spi: bcm2835: Switch to SPDX identifier thermal: bcm2835: Switch to SPDX identifier ASoc: bcm2835-i2s: Switch to SPDX identifier arch/arm/boot/dts/bcm2835-rpi-zero-w.dts | 8 +------- arch/arm/boot/dts/bcm2835-rpi-zero.dts | 8 +------- arch/arm/mach-bcm/board_bcm2835.c | 11 +---------- arch/arm/mach-bcm/platsmp.c | 10 +--------- arch/arm/mach-bcm/platsmp.h | 6 +----- drivers/char/hw_random/bcm2835-rng.c | 7 ++----- drivers/clk/bcm/clk-bcm2835-aux.c | 11 +---------- drivers/clk/bcm/clk-bcm2835.c | 12 +----------- drivers/clocksource/bcm2835_timer.c | 15 +-------------- drivers/dma/bcm2835-dma.c | 11 +---------- drivers/firmware/raspberrypi.c | 5 +---- drivers/i2c/busses/i2c-bcm2835.c | 10 +--------- drivers/irqchip/irq-bcm2835.c | 11 +---------- drivers/irqchip/irq-bcm2836.c | 11 +---------- drivers/mailbox/bcm2835-mailbox.c | 8 +------- drivers/mmc/host/bcm2835.c | 13 +------------ drivers/pinctrl/bcm/pinctrl-bcm2835.c | 11 +---------- drivers/pwm/pwm-bcm2835.c | 5 +---- drivers/soc/bcm/raspberrypi-power.c | 5 +---- drivers/spi/spi-bcm2835.c | 11 +---------- drivers/spi/spi-bcm2835aux.c | 11 +---------- drivers/thermal/broadcom/bcm2835_thermal.c | 11 +---------- include/dt-bindings/clock/bcm2835-aux.h | 10 +--------- include/dt-bindings/clock/bcm2835.h | 10 +--------- include/dt-bindings/pinctrl/bcm2835.h | 8 +------- include/dt-bindings/power/raspberrypi-power.h | 5 +---- include/soc/bcm2835/raspberrypi-firmware.h | 5 +---- sound/soc/bcm/bcm2835-i2s.c | 10 +--------- 28 files changed, 29 insertions(+), 230 deletions(-) -- 2.7.4

1 0

[alsa-devel] [PATCH v4 2/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 10 Nov '18

10 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..135c0826859b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5764,6 +5765,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6594,6 +6611,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.17.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: Power down links before turning off display audio power
by Pierre-Louis Bossart 09 Nov '18

09 Nov '18

On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. This patch was updated from an earlier version submitted upstream [1] which conflicted with the changes merged for HDaudio codec support with the Intel DSP. [1] https://patchwork.kernel.org/patch/10540213/ Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/hdac_hdmi.c | 13 +++++++------ sound/soc/intel/skylake/skl.c | 12 ++++++------ 2 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e9854889a95..e0fe682cea04 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(dev, "Cannot turn on display power on i915\n"); - return err; - } hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { @@ -2201,7 +2196,13 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(dev, "Cannot turn off display power on i915\n"); + return err; + } + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 29225623b4b4..1586c97d9450 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -815,6 +815,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -824,12 +830,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &bus->hlink_list, list) - snd_hdac_ext_bus_link_put(bus, hlink); - /* configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); -- 2.17.1

1 1

[alsa-devel] [PATCH] ASoC: sun8i-codec: fix crash on module removal
by Vasily Khoruzhick 09 Nov '18

09 Nov '18

drvdata is actually sun8i_codec, not snd_soc_card, so it crashes when calling snd_soc_card_get_drvdata(). Drop card and scodec vars anyway since we don't need to disable/unprepare clocks - it's already done by calling runtime_suspend() Drop clk_disable_unprepare() calls for the same reason. Fixes: 36c684936fae7 ("ASoC: Add sun8i digital audio codec") Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/sunxi/sun8i-codec.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 522a72fde78d..c4196d4e5915 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -605,16 +605,10 @@ static int sun8i_codec_probe(struct platform_device *pdev) static int sun8i_codec_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct sun8i_codec *scodec = snd_soc_card_get_drvdata(card); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) sun8i_codec_runtime_suspend(&pdev->dev); - clk_disable_unprepare(scodec->clk_module); - clk_disable_unprepare(scodec->clk_bus); - return 0; } -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: use ring buffer for rsnd_mod_name()
by Kuninori Morimoto 09 Nov '18

09 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit c0ea089dbad4 ("ASoC: rsnd: rsnd_mod_name() handles both name and ID") merged "name" and "ID" on rsnd_mod_name() to handle sub-ID (= for CTU/BUSIF). Then, it decided to share static char to avoid pointless memory. But, it doesn't work correctry in below case, because last called name will be used. dev_xxx(dev, "%s is connected to %s\n", rsnd_mod_name(mod_a), /* ssiu[00] */ rsnd_mod_name(mod_b)); /* ssi[0] */ -> rcar_sound ec500000.sound: ssi[0] is connected to ssi[0] ~~~~~~ ~~~~~~ We still don't want to have pointless memory, so let's use ring buffer. 16byte x 5 is very enough for this purpose. dev_xxx(dev, "%s is connected to %s\n", rsnd_mod_name(mod_a), /* ssiu[00] */ rsnd_mod_name(mod_b)); /* ssi[0] */ -> rcar_sound ec500000.sound: ssiu[00] is connected to ssi[0] ~~~~~~~~ ~~~~~~ Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 12f559e..56469ac 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -137,10 +137,17 @@ struct dma_chan *rsnd_mod_dma_req(struct rsnd_dai_stream *io, return mod->ops->dma_req(io, mod); } +#define MOD_NAME_NUM 5 #define MOD_NAME_SIZE 16 char *rsnd_mod_name(struct rsnd_mod *mod) { - static char name[MOD_NAME_SIZE]; + static char names[MOD_NAME_NUM][MOD_NAME_SIZE]; + static int num; + char *name = names[num]; + + num++; + if (num >= MOD_NAME_NUM) + num = 0; /* * Let's use same char to avoid pointlessness memory -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Optimize for non-PCI configuration
by Takashi Iwai 09 Nov '18

09 Nov '18

All the recent support of Creative boards and onboard audio depend on PCI, but they can't be trimmed easily even if you build without CONFIG_PCI, since the quirk is detected dynamically and the code has many branches with the flag check like spec->quirk type or spec->use_alt_functions. This patch makes these checks static for CONFIG_PCI=n case so that the compiler optimizes out. The access to flags are replaced with macros that are replaced with a static value for CONFIG_PCI=n. The macros look slightly ugly for avoiding compiler warnings wrt unused variables, and some additional default-case handlings for another compiler warnings, but the rest are very straightforward changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 205 +++++++++++++++++++++-------------- 1 file changed, 123 insertions(+), 82 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0a24037184c3..9c0b94ba786e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1081,6 +1081,18 @@ enum { QUIRK_AE5, }; +#ifdef CONFIG_PCI +#define ca0132_quirk(spec) ((spec)->quirk) +#define ca0132_use_pci_mmio(spec) ((spec)->use_pci_mmio) +#define ca0132_use_alt_functions(spec) ((spec)->use_alt_functions) +#define ca0132_use_alt_controls(spec) ((spec)->use_alt_controls) +#else +#define ca0132_quirk(spec) ({ (void)(spec); QUIRK_NONE; }) +#define ca0132_use_alt_functions(spec) ({ (void)(spec); false; }) +#define ca0132_use_pci_mmio(spec) ({ (void)(spec); false; }) +#define ca0132_use_alt_controls(spec) ({ (void)(spec); false; }) +#endif + static const struct hda_pintbl alienware_pincfgs[] = { { 0x0b, 0x90170110 }, /* Builtin Speaker */ { 0x0c, 0x411111f0 }, /* N/A */ @@ -3100,7 +3112,7 @@ static void dspload_post_setup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "---- dspload_post_setup ------\n"); - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { /*set DSP speaker to 2.0 configuration*/ chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); @@ -3332,7 +3344,7 @@ static void ca0132_gpio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); @@ -3343,6 +3355,8 @@ static void ca0132_gpio_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B); break; + default: + break; } } @@ -3352,7 +3366,7 @@ static void ca0132_gpio_setup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, 0x07); @@ -3371,6 +3385,8 @@ static void ca0132_gpio_setup(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x0C); break; + default: + break; } } @@ -4171,7 +4187,7 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) switch (spec->cur_out_type) { case SPEAKER_OUT: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: ca0113_mmio_gpio_set(codec, 7, false); ca0113_mmio_gpio_set(codec, 4, true); @@ -4202,10 +4218,12 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) chipio_set_control_param(codec, 0x0d, 0xa4); chipio_write(codec, 0x18b03c, 0x00000012); break; + default: + break; } break; case HEADPHONE_OUT: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: ca0113_mmio_gpio_set(codec, 7, true); ca0113_mmio_gpio_set(codec, 4, true); @@ -4237,10 +4255,12 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) chipio_set_control_param(codec, 0x0d, 0xa1); chipio_write(codec, 0x18b03c, 0x00000012); break; + default: + break; } break; case SURROUND_OUT: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: ca0113_mmio_gpio_set(codec, 7, false); ca0113_mmio_gpio_set(codec, 4, true); @@ -4271,6 +4291,8 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) chipio_set_control_param(codec, 0x0d, 0xa4); chipio_write(codec, 0x18b03c, 0x00000012); break; + default: + break; } break; } @@ -4445,7 +4467,7 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) to_delayed_work(work), struct ca0132_spec, unsol_hp_work); struct hda_jack_tbl *jack; - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(spec->codec); else ca0132_select_out(spec->codec); @@ -4529,14 +4551,14 @@ static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); if (spec->in_enum_val == REAR_LINE_IN) tmp = FLOAT_ZERO; else { - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) tmp = FLOAT_THREE; else tmp = FLOAT_ONE; @@ -4548,7 +4570,7 @@ static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) codec_dbg(codec, "%s: on.", __func__); chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_16_000); if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID]) @@ -4644,7 +4666,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) switch (spec->cur_mic_type) { case REAR_MIC: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: ca0113_mmio_gpio_set(codec, 0, false); @@ -4668,14 +4690,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); dspio_set_uint_param(codec, 0x80, 0x00, tmp); chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x0000000C); @@ -4688,12 +4710,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x0000004C); break; + default: + break; } ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); break; case REAR_LINE_IN: ca0132_mic_boost_set(codec, 0); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: ca0113_mmio_gpio_set(codec, 0, false); @@ -4704,28 +4728,32 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case QUIRK_AE5: ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00); break; + default: + break; } chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ZERO; dspio_set_uint_param(codec, 0x80, 0x00, tmp); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: chipio_write(codec, 0x18B098, 0x00000000); chipio_write(codec, 0x18B09C, 0x00000000); break; + default: + break; } chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); break; case FRONT_MIC: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: ca0113_mmio_gpio_set(codec, 0, true); @@ -4747,7 +4775,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); dspio_set_uint_param(codec, 0x80, 0x00, tmp); @@ -4755,7 +4783,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x000000CC); @@ -4764,6 +4792,8 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x0000004C); break; + default: + break; } ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); break; @@ -4858,7 +4888,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) val = 0; /* If Voice Focus on SBZ, set to two channel. */ - if ((nid == VOICE_FOCUS) && (spec->use_pci_mmio) + if ((nid == VOICE_FOCUS) && ca0132_use_pci_mmio(spec) && (spec->cur_mic_type != REAR_LINE_IN)) { if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) { @@ -4877,7 +4907,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) * For SBZ noise reduction, there's an extra command * to module ID 0x47. No clue why. */ - if ((nid == NOISE_REDUCTION) && (spec->use_pci_mmio) + if ((nid == NOISE_REDUCTION) && ca0132_use_pci_mmio(spec) && (spec->cur_mic_type != REAR_LINE_IN)) { if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) { @@ -4893,7 +4923,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) } /* If rear line in disable effects. */ - if (spec->use_alt_functions && + if (ca0132_use_alt_functions(spec) && spec->in_enum_val == REAR_LINE_IN) val = 0; } @@ -4923,7 +4953,7 @@ static int ca0132_pe_switch_set(struct hda_codec *codec) codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n", spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); i = OUT_EFFECT_START_NID - EFFECT_START_NID; @@ -4983,7 +5013,7 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec) /* set correct vipsource */ oldval = stop_mic1(codec); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ret |= ca0132_alt_set_vipsource(codec, 1); else ret |= ca0132_set_vipsource(codec, 1); @@ -5052,7 +5082,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; if (!auto_jack) { - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); else ca0132_select_out(codec); @@ -5069,7 +5099,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, } if (nid == VNID_HP_ASEL) { - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); else ca0132_select_out(codec); @@ -5783,7 +5813,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, /* mic boost */ if (nid == spec->input_pins[0]) { spec->cur_mic_boost = *valp; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { if (spec->in_enum_val != REAR_LINE_IN) changed = ca0132_mic_boost_set(codec, *valp); } else { @@ -6079,7 +6109,7 @@ static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, /* If using alt_controls, add FX: prefix. But, don't add FX: * prefix to OutFX or InFX enable controls. */ - if ((spec->use_alt_controls) && (nid <= IN_EFFECT_END_NID)) + if (ca0132_use_alt_controls(spec) && (nid <= IN_EFFECT_END_NID)) sprintf(namestr, "FX: %s %s Switch", pfx, dirstr[dir]); else sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); @@ -6356,7 +6386,7 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } /* Setup vmaster with surround slaves for desktop ca0132 devices */ - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT, spec->tlv); snd_hda_add_vmaster(codec, "Master Playback Volume", @@ -6376,7 +6406,7 @@ static int ca0132_build_controls(struct hda_codec *codec) num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; for (i = 0; i < num_fx; i++) { /* Desktop cards break if Echo Cancellation is used. */ - if (spec->use_pci_mmio) { + if (ca0132_use_pci_mmio(spec)) { if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID + OUT_EFFECTS_COUNT)) continue; @@ -6393,7 +6423,7 @@ static int ca0132_build_controls(struct hda_codec *codec) * EQ presets, and Smart Volume presets. Also, change names to add FX * prefix, and change PlayEnhancement and CrystalVoice to match. */ - if (spec->use_alt_controls) { + if (ca0132_use_alt_controls(spec)) { err = ca0132_alt_add_svm_enum(codec); if (err < 0) return err; @@ -6447,7 +6477,7 @@ static int ca0132_build_controls(struct hda_codec *codec) * to select the new outputs and inputs, plus add the new mic boost * setting control. */ - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { err = ca0132_alt_add_output_enum(codec); if (err < 0) return err; @@ -6458,14 +6488,14 @@ static int ca0132_build_controls(struct hda_codec *codec) * ZxR only has microphone input, there is no front panel * header on the card, and aux-in is handled by the DBPro board. */ - if (spec->quirk != QUIRK_ZXR) { + if (ca0132_quirk(spec) != QUIRK_ZXR) { err = ca0132_alt_add_input_enum(codec); if (err < 0) return err; } } - if (spec->quirk == QUIRK_AE5) { + if (ca0132_quirk(spec) == QUIRK_AE5) { err = ae5_add_headphone_gain_enum(codec); if (err < 0) return err; @@ -6474,7 +6504,7 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } - if (spec->quirk == QUIRK_ZXR) { + if (ca0132_quirk(spec) == QUIRK_ZXR) { err = zxr_add_headphone_gain_switch(codec); if (err < 0) return err; @@ -6504,7 +6534,7 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_add_chmap_ctls(codec); return 0; @@ -6582,7 +6612,7 @@ static int ca0132_build_pcms(struct hda_codec *codec) info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); if (!info) return -ENOMEM; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { info->own_chmap = true; info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap = ca0132_alt_chmaps; @@ -6596,7 +6626,7 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; /* With the DSP enabled, desktops don't use this ADC. */ - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); if (!info) return -ENOMEM; @@ -6794,7 +6824,7 @@ static void ca0132_init_dmic(struct hda_codec *codec) * Bit 6: set to select Data2, clear for Data1 * Bit 7: set to enable DMic, clear for AMic */ - if (spec->quirk == QUIRK_ALIENWARE_M17XR4) + if (ca0132_quirk(spec) == QUIRK_ALIENWARE_M17XR4) val = 0x33; else val = 0x23; @@ -6876,7 +6906,7 @@ static void ca0132_alt_init_analog_mics(struct hda_codec *codec) /* Mic 1 Setup */ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) { + if (ca0132_quirk(spec) == QUIRK_R3DI) { chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ONE; } else @@ -6886,7 +6916,7 @@ static void ca0132_alt_init_analog_mics(struct hda_codec *codec) /* Mic 2 setup (not present on desktop cards) */ chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ZERO; dspio_set_uint_param(codec, 0x80, 0x01, tmp); @@ -6948,7 +6978,7 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) chipio_set_stream_channels(codec, 0x0C, 6); chipio_set_stream_control(codec, 0x0C, 1); /* No clue what these control */ - if (spec->quirk == QUIRK_SBZ) { + if (ca0132_quirk(spec) == QUIRK_SBZ) { chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0); chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1); chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2); @@ -6961,7 +6991,7 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) chipio_write_no_mutex(codec, 0x190054, 0x0001edc9); chipio_write_no_mutex(codec, 0x190058, 0x0001eaca); chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb); - } else if (spec->quirk == QUIRK_ZXR) { + } else if (ca0132_quirk(spec) == QUIRK_ZXR) { chipio_write_no_mutex(codec, 0x190038, 0x000140c2); chipio_write_no_mutex(codec, 0x19003c, 0x000141c3); chipio_write_no_mutex(codec, 0x190040, 0x000150c4); @@ -6991,7 +7021,7 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) * why this is, but multiple tests have confirmed it. */ for (i = 0; i < 2; i++) { - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: tmp = 0x00000003; @@ -7020,6 +7050,8 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) tmp = 0x00000000; dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); break; + default: + break; } msleep(100); } @@ -7042,7 +7074,7 @@ static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec) chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: chipio_write(codec, 0x18b098, 0x0000000c); chipio_write(codec, 0x18b09C, 0x0000000c); @@ -7051,6 +7083,8 @@ static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec) chipio_write(codec, 0x18b098, 0x0000000c); chipio_write(codec, 0x18b09c, 0x0000004c); break; + default: + break; } } @@ -7272,7 +7306,7 @@ static void r3d_setup_defaults(struct hda_codec *codec) /* Set speaker source? */ dspio_set_uint_param(codec, 0x32, 0x00, tmp); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); /* Setup effect defaults */ @@ -7419,7 +7453,7 @@ static void ca0132_init_flags(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1); chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1); chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1); @@ -7452,7 +7486,7 @@ static void ca0132_init_params(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); chipio_set_conn_rate(codec, 0x0B, SR_48_000); chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0); @@ -7489,7 +7523,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) * can use the default firmware, but I'll leave the option in case * it needs it again. */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: case QUIRK_AE5: @@ -7563,7 +7597,7 @@ static void ca0132_download_dsp(struct hda_codec *codec) } /* For codecs using alt functions, this is already done earlier */ - if (spec->dsp_state == DSP_DOWNLOADED && (!spec->use_alt_functions)) + if (spec->dsp_state == DSP_DOWNLOADED && !ca0132_use_alt_functions(spec)) ca0132_set_dsp_msr(codec, true); } @@ -7600,7 +7634,7 @@ static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_in(codec); else ca0132_select_mic(codec); @@ -7615,7 +7649,7 @@ static void ca0132_init_unsol(struct hda_codec *codec) snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP, ca0132_process_dsp_response); /* Front headphone jack detection */ - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) snd_hda_jack_detect_enable_callback(codec, spec->unsol_tag_front_hp, hp_callback); } @@ -7705,7 +7739,7 @@ static void ca0132_init_chip(struct hda_codec *codec) mutex_init(&spec->chipio_mutex); spec->cur_out_type = SPEAKER_OUT; - if (!spec->use_alt_functions) + if (!ca0132_use_alt_functions(spec)) spec->cur_mic_type = DIGITAL_MIC; else spec->cur_mic_type = REAR_MIC; @@ -7731,7 +7765,7 @@ static void ca0132_init_chip(struct hda_codec *codec) * Sets defaults for the effect slider controls, only for alternative * ca0132 codecs. Also sets x-bass crossover frequency to 80hz. */ - if (spec->use_alt_controls) { + if (ca0132_use_alt_controls(spec)) { spec->xbass_xover_freq = 8; for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++) spec->fx_ctl_val[i] = effect_slider_defaults[i]; @@ -7746,7 +7780,7 @@ static void ca0132_init_chip(struct hda_codec *codec) * the daughter board. So, there is no input enum control, and we need * to make sure that spec->in_enum_val is set properly. */ - if (spec->quirk == QUIRK_ZXR) + if (ca0132_quirk(spec) == QUIRK_ZXR) spec->in_enum_val = REAR_MIC; #ifdef ENABLE_TUNING_CONTROLS @@ -8087,27 +8121,27 @@ static void ca0132_mmio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0x400); else writel(0x00000000, spec->mem_base + 0x400); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0x408); else writel(0x00000000, spec->mem_base + 0x408); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0x40c); else writel(0x00000000, spec->mem_base + 0x40C); - if (spec->quirk == QUIRK_ZXR) + if (ca0132_quirk(spec) == QUIRK_ZXR) writel(0x00880640, spec->mem_base + 0x01C); else writel(0x00880680, spec->mem_base + 0x01C); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000080, spec->mem_base + 0xC0C); else writel(0x00000083, spec->mem_base + 0xC0C); @@ -8115,7 +8149,7 @@ static void ca0132_mmio_init(struct hda_codec *codec) writel(0x00000030, spec->mem_base + 0xC00); writel(0x00000000, spec->mem_base + 0xC04); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000000, spec->mem_base + 0xC0C); else writel(0x00000003, spec->mem_base + 0xC0C); @@ -8124,7 +8158,7 @@ static void ca0132_mmio_init(struct hda_codec *codec) writel(0x00000003, spec->mem_base + 0xC0C); writel(0x00000003, spec->mem_base + 0xC0C); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0xC08); else writel(0x000000C1, spec->mem_base + 0xC08); @@ -8135,7 +8169,7 @@ static void ca0132_mmio_init(struct hda_codec *codec) writel(0x000000C1, spec->mem_base + 0xC08); writel(0x00000080, spec->mem_base + 0xC04); - if (spec->quirk == QUIRK_AE5) { + if (ca0132_quirk(spec) == QUIRK_AE5) { writel(0x00000000, spec->mem_base + 0x42c); writel(0x00000000, spec->mem_base + 0x46c); writel(0x00000000, spec->mem_base + 0x4ac); @@ -8210,7 +8244,7 @@ static void ca0132_alt_init(struct hda_codec *codec) ca0132_alt_vol_setup(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: codec_dbg(codec, "SBZ alt_init"); ca0132_gpio_init(codec); @@ -8247,6 +8281,8 @@ static void ca0132_alt_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); break; + default: + break; } } @@ -8273,7 +8309,7 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_reload = true; spec->dsp_state = DSP_DOWNLOAD_INIT; } else { - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) sbz_dsp_startup_check(codec); return 0; } @@ -8283,12 +8319,12 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; - if (spec->use_pci_mmio) + if (ca0132_use_pci_mmio(spec)) ca0132_mmio_init(codec); snd_hda_power_up_pm(codec); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) ae5_register_set(codec); ca0132_init_unsol(codec); @@ -8297,14 +8333,14 @@ static int ca0132_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->base_init_verbs); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_init(codec); ca0132_download_dsp(codec); ca0132_refresh_widget_caps(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_R3DI: case QUIRK_R3D: r3d_setup_defaults(codec); @@ -8333,7 +8369,7 @@ static int ca0132_init(struct hda_codec *codec) init_input(codec, cfg->dig_in_pin, spec->dig_in); - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D); @@ -8341,11 +8377,11 @@ static int ca0132_init(struct hda_codec *codec) VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20); } - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) ca0132_gpio_setup(codec); snd_hda_sequence_write(codec, spec->spec_init_verbs); - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { ca0132_alt_select_out(codec); ca0132_alt_select_in(codec); } else { @@ -8390,7 +8426,7 @@ static void ca0132_free(struct hda_codec *codec) cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: sbz_exit_chip(codec); break; @@ -8406,6 +8442,8 @@ static void ca0132_free(struct hda_codec *codec) case QUIRK_R3DI: r3di_gpio_shutdown(codec); break; + default: + break; } snd_hda_sequence_write(codec, spec->base_exit_verbs); @@ -8460,12 +8498,12 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = 3; - if (!spec->use_alt_functions) + if (!ca0132_use_alt_functions(spec)) spec->multiout.max_channels = 2; else spec->multiout.max_channels = 6; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_ALIENWARE: codec_dbg(codec, "%s: QUIRK_ALIENWARE applied.\n", __func__); snd_hda_apply_pincfgs(codec, alienware_pincfgs); @@ -8490,9 +8528,11 @@ static void ca0132_config(struct hda_codec *codec) codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__); snd_hda_apply_pincfgs(codec, r3di_pincfgs); break; + default: + break; } - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_ALIENWARE: spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ @@ -8653,7 +8693,7 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) * Since desktop cards use pci_mmio, this can be used to determine * whether or not to use these verbs instead of a separate bool. */ - if (spec->use_pci_mmio) + if (ca0132_use_pci_mmio(spec)) spec->desktop_init_verbs = ca0132_init_verbs1; spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS, sizeof(struct hda_verb), @@ -8728,11 +8768,10 @@ static int patch_ca0132(struct hda_codec *codec) spec->quirk = quirk->value; else spec->quirk = QUIRK_NONE; - - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) sbz_detect_quirk(codec); - if (spec->quirk == QUIRK_ZXR_DBPRO) + if (ca0132_quirk(spec) == QUIRK_ZXR_DBPRO) codec->patch_ops = dbpro_patch_ops; else codec->patch_ops = ca0132_patch_ops; @@ -8745,7 +8784,7 @@ static int patch_ca0132(struct hda_codec *codec) spec->num_mixers = 1; /* Set which mixers each quirk uses. */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: spec->mixers[0] = desktop_mixer; snd_hda_codec_set_name(codec, "Sound Blaster Z"); @@ -8774,7 +8813,7 @@ static int patch_ca0132(struct hda_codec *codec) } /* Setup whether or not to use alt functions/controls/pci_mmio */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: case QUIRK_AE5: @@ -8795,6 +8834,7 @@ static int patch_ca0132(struct hda_codec *codec) break; } +#ifdef CONFIG_PCI if (spec->use_pci_mmio) { spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); if (spec->mem_base == NULL) { @@ -8802,6 +8842,7 @@ static int patch_ca0132(struct hda_codec *codec) spec->quirk = QUIRK_NONE; } } +#endif spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; -- 2.19.1

1 0

[alsa-devel] KMSAN: uninit-value in linear_transfer (2)
by syzbot 09 Nov '18

09 Nov '18

Hello, syzbot found the following crash on: HEAD commit: 7438a3b20295 kmsan: print user address when reporting info.. git tree: https://github.com/google/kmsan.git/master console output: https://syzkaller.appspot.com/x/log.txt?x=15b42133400000 kernel config: https://syzkaller.appspot.com/x/.config?x=8df5fc509a1b351b dashboard link: https://syzkaller.appspot.com/bug?extid=1cb36954e127c98dd037 compiler: clang version 8.0.0 (trunk 343298) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=12be9825400000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=13e2c425400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+1cb36954e127c98dd037(a)syzkaller.appspotmail.com ================================================================== BUG: KMSAN: uninit-value in __arch_swab32 arch/x86/include/uapi/asm/swab.h:10 [inline] BUG: KMSAN: uninit-value in __fswab32 include/uapi/linux/swab.h:59 [inline] BUG: KMSAN: uninit-value in do_convert sound/core/oss/linear.c:50 [inline] BUG: KMSAN: uninit-value in convert sound/core/oss/linear.c:81 [inline] BUG: KMSAN: uninit-value in linear_transfer+0x92d/0xca0 sound/core/oss/linear.c:110 CPU: 1 PID: 6835 Comm: syz-executor866 Not tainted 4.19.0+ #78 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x32d/0x480 lib/dump_stack.c:113 kmsan_report+0x19f/0x300 mm/kmsan/kmsan.c:911 __msan_warning+0x76/0xd0 mm/kmsan/kmsan_instr.c:415 __arch_swab32 arch/x86/include/uapi/asm/swab.h:10 [inline] __fswab32 include/uapi/linux/swab.h:59 [inline] do_convert sound/core/oss/linear.c:50 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x92d/0xca0 sound/core/oss/linear.c:110 snd_pcm_plug_read_transfer+0x3bf/0x590 sound/core/oss/pcm_plugin.c:659 snd_pcm_oss_read2 sound/core/oss/pcm_oss.c:1480 [inline] snd_pcm_oss_read1 sound/core/oss/pcm_oss.c:1518 [inline] snd_pcm_oss_read+0xe6d/0x1cb0 sound/core/oss/pcm_oss.c:2758 __vfs_read+0x1e2/0xb10 fs/read_write.c:416 vfs_read+0x380/0x6b0 fs/read_write.c:452 ksys_read fs/read_write.c:578 [inline] __do_sys_read fs/read_write.c:588 [inline] __se_sys_read+0x17a/0x370 fs/read_write.c:586 __x64_sys_read+0x4a/0x70 fs/read_write.c:586 do_syscall_64+0xcf/0x110 arch/x86/entry/common.c:291 entry_SYSCALL_64_after_hwframe+0x63/0xe7 RIP: 0033:0x446379 Code: e8 0c e8 ff ff 48 83 c4 18 c3 0f 1f 80 00 00 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 7b 08 fc ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007fcd5b97cda8 EFLAGS: 00000246 ORIG_RAX: 0000000000000000 RAX: ffffffffffffffda RBX: 00000000006dac38 RCX: 0000000000446379 RDX: 0000000000000184 RSI: 0000000020002180 RDI: 0000000000000003 RBP: 00000000006dac30 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000006dac3c R13: 7073642f7665642f R14: 00800000c0045002 R15: 0000000000000001 Uninit was stored to memory at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:252 [inline] kmsan_save_stack mm/kmsan/kmsan.c:267 [inline] kmsan_internal_chain_origin+0x136/0x240 mm/kmsan/kmsan.c:569 kmsan_memcpy_origins+0x13d/0x1b0 mm/kmsan/kmsan.c:393 __msan_memcpy+0x6f/0x80 mm/kmsan/kmsan_instr.c:242 do_convert sound/core/oss/linear.c:48 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x74c/0xca0 sound/core/oss/linear.c:110 snd_pcm_plug_read_transfer+0x3bf/0x590 sound/core/oss/pcm_plugin.c:659 snd_pcm_oss_read2 sound/core/oss/pcm_oss.c:1480 [inline] snd_pcm_oss_read1 sound/core/oss/pcm_oss.c:1518 [inline] snd_pcm_oss_read+0xe6d/0x1cb0 sound/core/oss/pcm_oss.c:2758 __vfs_read+0x1e2/0xb10 fs/read_write.c:416 vfs_read+0x380/0x6b0 fs/read_write.c:452 ksys_read fs/read_write.c:578 [inline] __do_sys_read fs/read_write.c:588 [inline] __se_sys_read+0x17a/0x370 fs/read_write.c:586 __x64_sys_read+0x4a/0x70 fs/read_write.c:586 do_syscall_64+0xcf/0x110 arch/x86/entry/common.c:291 entry_SYSCALL_64_after_hwframe+0x63/0xe7 Uninit was created at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:252 [inline] kmsan_internal_alloc_meta_for_pages+0x155/0x740 mm/kmsan/kmsan.c:689 kmsan_alloc_page+0x77/0xe0 mm/kmsan/kmsan_hooks.c:320 __alloc_pages_nodemask+0x12cc/0x6640 mm/page_alloc.c:4416 alloc_pages_current+0x584/0x7e0 mm/mempolicy.c:2093 alloc_pages include/linux/gfp.h:511 [inline] __vmalloc_area_node mm/vmalloc.c:1689 [inline] __vmalloc_node_range+0x879/0x12a0 mm/vmalloc.c:1752 __vmalloc_node mm/vmalloc.c:1797 [inline] __vmalloc_node_flags mm/vmalloc.c:1811 [inline] vmalloc+0xd8/0xf0 mm/vmalloc.c:1833 snd_pcm_plugin_alloc+0x255/0xc80 sound/core/oss/pcm_plugin.c:71 snd_pcm_plug_alloc+0x281/0x600 sound/core/oss/pcm_plugin.c:137 snd_pcm_oss_change_params_locked+0x5e40/0x6e30 sound/core/oss/pcm_oss.c:1039 snd_pcm_oss_change_params sound/core/oss/pcm_oss.c:1107 [inline] snd_pcm_oss_get_active_substream+0x4f7/0x5a0 sound/core/oss/pcm_oss.c:1124 snd_pcm_oss_get_rate sound/core/oss/pcm_oss.c:1774 [inline] snd_pcm_oss_set_rate sound/core/oss/pcm_oss.c:1766 [inline] snd_pcm_oss_ioctl+0x4adb/0x8860 sound/core/oss/pcm_oss.c:2614 do_vfs_ioctl+0xf77/0x2d30 fs/ioctl.c:46 ksys_ioctl fs/ioctl.c:702 [inline] __do_sys_ioctl fs/ioctl.c:709 [inline] __se_sys_ioctl+0x1da/0x270 fs/ioctl.c:707 __x64_sys_ioctl+0x4a/0x70 fs/ioctl.c:707 do_syscall_64+0xcf/0x110 arch/x86/entry/common.c:291 entry_SYSCALL_64_after_hwframe+0x63/0xe7 ================================================================== --- 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#bug-status-tracking for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

3 4

[alsa-devel] [PATCH v2 0/7] ASoC: omap-mcbsp: Cleanup and split core/sidetone support
by Peter Ujfalusi 08 Nov '18

08 Nov '18

Hi, Changes since v1: - Two patch added to address the code move triggered few warnings from build. One of them is around mcbsp->pdata checks (not valid complaint), the other is that we set -EINVAL to unsigned variables (they were not used in that case) While preparing for merging the davinci and omap directories (I will send the series for that shortly) I have taken some time to do cleanup on the McBSP driver. The mcbsp.c/h files were the result when we moved code from arch/arm/plat-omap to sound/soc/omap/ a long time ago and it contained code for McBSP core and the OMAP3 sidetone functionality. With this series I tried to split the core and sidetone code to separate files. There were two reasons for this: 1. to have the OMAP specific DAI driver files prefixed with omap-* under the new sound/soc/ti/ directory 2. easier to find functions as all sidetone code is in separate file. Regards, Peter --- Peter Ujfalusi (7): ASoC: omap-mcbsp: Clean up dma_data addr initialization code ASoC: omap-mcbsp: Clean up the interrupt handlers ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1104 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 852 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1278 insertions(+), 1330 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 10

[alsa-devel] High CPU usage in mpd
by GitHub issues - opened 08 Nov '18

08 Nov '18

alsa-project/alsa-utils issue #2 was opened from buttjake: I've ran into a strange bug in mpd where I get really high cpu usage. All the details are in the following two short threads. https://github.com/MusicPlayerDaemon/MPD/issues/391#issuecomment-435698235 https://bbs.archlinux.org/viewtopic.php?id=241722 ( buttcake is me ) I can also solve the bug by adding ``` audio_output { type "alsa" device "hw:0,0" } ``` to the mpd.conf Issue URL : https://github.com/alsa-project/alsa-utils/issues/2 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro
by Yangtao Li 07 Nov '18

07 Nov '18

Use DEFINE_SHOW_ATTRIBUTE macro to simplify the code. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> --- sound/soc/fsl/fsl_ssi_dbg.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 1255dfe19eef..6f6294149476 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -124,17 +124,7 @@ static int fsl_ssi_stats_show(struct seq_file *s, void *unused) return 0; } -static int fsl_ssi_stats_open(struct inode *inode, struct file *file) -{ - return single_open(file, fsl_ssi_stats_show, inode->i_private); -} - -static const struct file_operations fsl_ssi_stats_ops = { - .open = fsl_ssi_stats_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(fsl_ssi_stats); int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) { @@ -144,7 +134,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, - &fsl_ssi_stats_ops); + &fsl_ssi_stats_fops); if (!ssi_dbg->dbg_stats) { debugfs_remove(ssi_dbg->dbg_dir); return -ENOMEM; -- 2.17.0

2 1

[alsa-devel] UCM profile dirs without card config
by Takashi Iwai 07 Nov '18

07 Nov '18

Hi, some of the recently added UCM profiles are the files to be included by others, and they are not card targets. Unfortunately alsaucm doesn't know about it, and it aborts with error, e.g. ==== % alsaucm ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/bytcr/bytcr.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card bytcr alsaucm: unable to obtain card list: No such file or directory ==== The similar error is found in nau8824, rt5640 and rt5651. After putting a placeholder card config file (e.g. bytcr.conf), the error is gone. But certainly we don't want to allow user to choose this as the proper card. So, I guess we need to put some flag to each such directory indicating that it's no card config. For example, putting /usr/share/alsa/ucm/bytcr/.nocard file or such... Not sure whether it's wise to have a file starting with dot, though. Maybe clearer to be "nocard" or any other visible one? Comments / suggestions appreciated. thanks, Takashi

1 0

[alsa-devel] Applied "ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro" to the asoc tree
by Mark Brown 07 Nov '18

07 Nov '18

The patch ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro 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 d3ff06b03c106171cdc7a718c4de724ce6ae346d Mon Sep 17 00:00:00 2001 From: Yangtao Li <tiny.windzz(a)gmail.com> Date: Mon, 5 Nov 2018 09:37:30 -0500 Subject: [PATCH] ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro Use DEFINE_SHOW_ATTRIBUTE macro to simplify the code. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_ssi_dbg.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 1255dfe19eef..6f6294149476 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -124,17 +124,7 @@ static int fsl_ssi_stats_show(struct seq_file *s, void *unused) return 0; } -static int fsl_ssi_stats_open(struct inode *inode, struct file *file) -{ - return single_open(file, fsl_ssi_stats_show, inode->i_private); -} - -static const struct file_operations fsl_ssi_stats_ops = { - .open = fsl_ssi_stats_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(fsl_ssi_stats); int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) { @@ -144,7 +134,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, - &fsl_ssi_stats_ops); + &fsl_ssi_stats_fops); if (!ssi_dbg->dbg_stats) { debugfs_remove(ssi_dbg->dbg_dir); return -ENOMEM; -- 2.19.0.rc2

1 0

Re: [alsa-devel] Build regressions/improvements in v4.20-rc1
by Geert Uytterhoeven 07 Nov '18

07 Nov '18

On Mon, Nov 5, 2018 at 11:07 PM Geert Uytterhoeven <geert(a)linux-m68k.org> wrote: > Below is the list of build error/warning regressions/improvements in > v4.20-rc1[1] compared to v4.19[2]. > > Summarized: > - build errors: +3/-0 > - build warnings: +449/-2712 > > Happy fixing! ;-) > > Thanks to the linux-next team for providing the build service. > > [1] http://kisskb.ellerman.id.au/kisskb/branch/linus/head/651022382c7f8da46cb48… (all 240 configs) > [2] http://kisskb.ellerman.id.au/kisskb/branch/linus/head/84df9525b0c27f3ebc2eb… (all 240 configs) > > > *** ERRORS *** > > 3 error regressions: > + /kisskb/src/arch/s390/kernel/nospec-branch.c: error: macro "memcpy" passed 6 arguments, but takes just 3: => 130:61 s390-all{mod,yes}config > + /kisskb/src/sound/pci/hda/patch_ca0132.c: error: implicit declaration of function 'pci_iomap' [-Werror=implicit-function-declaration]: => 8799:3 sh4-all{mod,yes}config Looks like d9b84a15892c0233 ("ALSA: hda: Fix implicit definition of pci_iomap() on SH") is not sufficient? > + {standard input}: Error: Unrecognized opcode: `slbfee.': => 89, 37 powerpc-all{mod,yes}config Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

5 8

[alsa-devel] [PATCH 0/1] slimbus: simplify getting .driver_data
by Wolfram Sang 07 Nov '18

07 Nov '18

I got tired of fixing this in Renesas drivers manually, so I took the big hammer. Remove this cumbersome code pattern which got copy-pasted too much already: - struct platform_device *pdev = to_platform_device(dev); - struct ep93xx_keypad *keypad = platform_get_drvdata(pdev); + struct ep93xx_keypad *keypad = dev_get_drvdata(dev); A branch, tested by buildbot, can be found here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git coccinelle/get_drvdata I have been asked if it couldn't be done for dev_set_drvdata as well. I checked it and did not find one occasion where it could be simplified like this. Not much of a surprise because driver_data is usually set in probe() functions which access struct platform_device in many other ways. I am open for other comments, suggestions, too, of course. Here is the cocci-script I created: @@ struct device* d; identifier pdev; expression *ptr; @@ ( - struct platform_device *pdev = to_platform_device(d); | - struct platform_device *pdev; ... - pdev = to_platform_device(d); ) <... when != pdev - &pdev->dev + d ...> ptr = - platform_get_drvdata(pdev) + dev_get_drvdata(d) <... when != pdev - &pdev->dev + d ...> Kind regards, Wolfram Wolfram Sang (1): slimbus: qcom-ctrl: simplify getting .driver_data drivers/slimbus/qcom-ctrl.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) -- 2.19.0

2 2

[alsa-devel] [PATCH 0/2] ASoC: qdsp6: few routing fixes
by Srinivas Kandagatla 07 Nov '18

07 Nov '18

Jimmy and Rohit found a issue w.r.t backend dais being powered on all the time on there device. Looks like there was a bug in the driver since start. This patchset fixes it and also fixes another bug where asm routes are added each time at dai probe. Thanks, srini Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: Only add routing once. ASoC: qdsp6: q6afe-dai: Fix the dai widgets sound/soc/qcom/qdsp6/q6afe-dai.c | 208 +++++++++++++++---------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ----- sound/soc/qcom/qdsp6/q6routing.c | 19 +++ 3 files changed, 123 insertions(+), 137 deletions(-) -- 2.19.1

3 5

[alsa-devel] [PATCH] ASoC: ak5558: Remove redundant snd_soc_component_read32 calls
by Axel Lin 06 Nov '18

06 Nov '18

snd_soc_component_update_bits() will only update the mask bits, so remove the redundant snd_soc_component_read32(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ak5558.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 60f1f12c81ea..8179512129d3 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -130,16 +130,12 @@ static int ak5558_hw_params(struct snd_pcm_substream *substream, u8 bits; int pcm_width = max(params_physical_width(params), ak5558->slot_width); - /* set master/slave audio interface */ - bits = snd_soc_component_read32(component, AK5558_02_CONTROL1); - bits &= ~AK5558_BITS; - switch (pcm_width) { case 16: - bits |= AK5558_DIF_24BIT_MODE; + bits = AK5558_DIF_24BIT_MODE; break; case 32: - bits |= AK5558_DIF_32BIT_MODE; + bits = AK5558_DIF_32BIT_MODE; break; default: return -EINVAL; @@ -168,18 +164,15 @@ static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set master/slave audio interface */ - format = snd_soc_component_read32(component, AK5558_02_CONTROL1); - format &= ~AK5558_DIF; - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - format |= AK5558_DIF_I2S_MODE; + format = AK5558_DIF_I2S_MODE; break; case SND_SOC_DAIFMT_LEFT_J: - format |= AK5558_DIF_MSB_MODE; + format = AK5558_DIF_MSB_MODE; break; case SND_SOC_DAIFMT_DSP_B: - format |= AK5558_DIF_MSB_MODE; + format = AK5558_DIF_MSB_MODE; break; default: return -EINVAL; -- 2.17.1

3 2

[alsa-devel] Applied "ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new 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 223bc10b84970fd772c105b550beeef3ac3502be Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 6 Nov 2018 04:33:02 +0000 Subject: [PATCH] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. These information are needed for driver not for user. Let's remove these from snd_kcontrol_new. There is no detail explanation for these register on datasheet, but, judging from its behavior, it seems these register needs clock to read. Otherwise, I2C returns error. Therefor, amixer command will be failed under non working timing. Without this patch, user often fail amixer command. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3168a.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 5f4327f5b725..115f21658222 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), - SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), - SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), - SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), - SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +172,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), - SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), - SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), - SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: omap: Remove unused machine driver for AM3517-evm
by Peter Ujfalusi 06 Nov '18

06 Nov '18

The audio support is done via simple-audio-card via DT, the custom driver is no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/Kconfig | 9 --- sound/soc/omap/Makefile | 2 - sound/soc/omap/am3517evm.c | 141 ------------------------------------- 3 files changed, 152 deletions(-) delete mode 100644 sound/soc/omap/am3517evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6dccea6fdaeb..3d418482d242 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -78,15 +78,6 @@ config SND_OMAP_SOC_OSK5912 help Say Y if you want to add support for SoC audio on osk5912. -config SND_OMAP_SOC_AM3517EVM - tristate "SoC Audio support for OMAP3517 / AM3517 EVM" - depends on SND_SDMA_SOC && MACH_OMAP3517EVM && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC23_I2C - help - Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 - EVM. - config SND_OMAP_SOC_OMAP_TWL4030 tristate "SoC Audio support for TI SoC based boards with twl4030 codec" depends on TWL4030_CORE && SND_SDMA_SOC diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 8922aa5a840c..d005338dd13c 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -17,7 +17,6 @@ snd-soc-n810-objs := n810.o snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o -snd-soc-am3517evm-objs := am3517evm.o snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o snd-soc-omap-twl4030-objs := omap-twl4030.o snd-soc-omap3pandora-objs := omap3pandora.o @@ -26,7 +25,6 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c deleted file mode 100644 index d5651026ec10..000000000000 --- a/sound/soc/omap/am3517evm.c +++ /dev/null @@ -1,141 +0,0 @@ -/* - * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM - * - * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> - * - * Based on sound/soc/omap/beagle.c by Steve Sakoman - * - * Copyright (C) 2009 Texas Instruments Incorporated - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation version 2. - * - * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, - * whether express or implied; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - */ - -#include <linux/clk.h> -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <linux/platform_data/asoc-ti-mcbsp.h> - -#include "omap-mcbsp.h" - -#include "../codecs/tlv320aic23.h" - -#define CODEC_CLOCK 12000000 - -static int am3517evm_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; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret < 0) - printk(KERN_ERR "can't set codec system clock\n"); - - return ret; -} - -static const struct snd_soc_ops am3517evm_ops = { - .hw_params = am3517evm_hw_params, -}; - -/* am3517evm machine dapm widgets */ -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Line Out", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic In", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* Line Out connected to LLOUT, RLOUT */ - {"Line Out", NULL, "LOUT"}, - {"Line Out", NULL, "ROUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic In"}, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link am3517evm_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "omap-mcbsp.1", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "omap-mcbsp.1", - .codec_name = "tlv320aic23-codec.2-001a", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &am3517evm_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_am3517evm = { - .name = "am3517evm", - .owner = THIS_MODULE, - .dai_link = &am3517evm_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *am3517evm_snd_device; - -static int __init am3517evm_soc_init(void) -{ - int ret; - - if (!machine_is_omap3517evm()) - return -ENODEV; - pr_info("OMAP3517 / AM3517 EVM SoC init\n"); - - am3517evm_snd_device = platform_device_alloc("soc-audio", -1); - if (!am3517evm_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(am3517evm_snd_device, &snd_soc_am3517evm); - - ret = platform_device_add(am3517evm_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(am3517evm_snd_device); - - return ret; -} - -static void __exit am3517evm_soc_exit(void) -{ - platform_device_unregister(am3517evm_snd_device); -} - -module_init(am3517evm_soc_init); -module_exit(am3517evm_soc_exit); - -MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); -MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); -MODULE_LICENSE("GPL v2"); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: pxa: change ac97 dependencies
by Arnd Bergmann 06 Nov '18

06 Nov '18

Enabling both the old AC97_BUS code and the new AC97_BUS_COMPAT causes problems because both modules provide an exported snd_ac97_reset() function. I had tried to fix the problem of having both coexist earlier, but my patch only prevented them from being built-in. This is because of a special Kconfig feature that lets a symbol have a dependency on another one being disabled, but still allow both to be loadable modules. Changing the dependency to =n avoids that problem, now we can only build the new driver if the old one is completely disabled. If we could figure out a way to let rename one of the reset functions and have each driver link to exactly the old or the compat code, that would also work, but I could not find if that's possible. Fixes: bec5ecdf41d4 ("ASoC: pxa: avoid AC97_BUS build warning") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/pxa/Kconfig | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 943b44de1464..6075fc32e6b9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -79,7 +79,7 @@ config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -89,7 +89,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -99,7 +99,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -109,7 +109,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -120,7 +120,7 @@ config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -131,7 +131,7 @@ config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -161,7 +161,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -201,7 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help -- 2.18.0

2 1

[alsa-devel] Applied "ASoC: acpi: define common interface for machine driver configuration" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: acpi: define common interface for machine driver configuration 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 8679284b37e9e2d529d60d1acc4133af4aa5dd34 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:48 -0500 Subject: [PATCH] ASoC: acpi: define common interface for machine driver configuration The machine drivers may need information provided by the platform driver. Currently the information is passed using pdata specific to each plaform driver. This prevents other drivers, such as SOF, from reusing machine drivers directly. Add a new structure which contains the required fields. This proposal requires a bit more work on the platform side but this generic interface helps reuse code directly. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-acpi.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index e45b2330d16a..5154c6359609 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -37,6 +37,19 @@ snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines); +/** + * snd_soc_acpi_mach_params: interface for machine driver configuration + * + * @acpi_ipc_irq_index: used for BYT-CR detection + * @platform: string used for HDaudio codec support + * @codec_mask: used for HDAudio support + */ +struct snd_soc_acpi_mach_params { + u32 acpi_ipc_irq_index; + const char *platform; + u32 codec_mask; +}; + /** * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are * related to the hardware, except for the firmware and topology file names. @@ -68,6 +81,7 @@ struct snd_soc_acpi_mach { struct snd_soc_acpi_mach * (*machine_quirk)(void *arg); const void *quirk_data; void *pdata; + struct snd_soc_acpi_mach_params mach_params; const char *sof_fw_filename; const char *sof_tplg_filename; const char *asoc_plat_name; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: use standard interface for Hdaudio machine driver" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: use standard interface for Hdaudio machine driver 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 842bb5135f1016151235413726e4956210a79664 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:49 -0500 Subject: [PATCH] ASoC: Intel: use standard interface for Hdaudio machine driver Don't rely on internal Skylake-specific data structures, use generic interface to let other drivers use the same machine driver as is, e.g. SOF to support HDaudio codecs and HDMI outputs. Tested on LeafHill CRB board, no regression seen with this change. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 20 ++++++++++---------- sound/soc/intel/skylake/skl.c | 10 +++------- 2 files changed, 13 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index b415dd4c85f5..b6f287fa9505 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -12,8 +12,8 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/hdac_hdmi.h" -#include "../skylake/skl.h" #include "skl_hda_dsp_common.h" static const struct snd_soc_dapm_widget skl_hda_widgets[] = { @@ -101,17 +101,17 @@ static struct snd_soc_card hda_soc_card = { #define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) #define IDISP_CODEC_MASK 0x4 -static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) +static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) { struct snd_soc_card *card = &hda_soc_card; struct snd_soc_dai_link *dai_link; u32 codec_count, codec_mask; int i, num_links, num_route; - codec_mask = pdata->codec_mask; + codec_mask = mach_params->codec_mask; codec_count = hweight_long(codec_mask); - if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) { + if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) { num_links = IDISP_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { @@ -127,14 +127,14 @@ static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) card->num_dapm_routes = num_route; for_each_card_prelinks(card, i, dai_link) - dai_link->platform_name = pdata->platform; + dai_link->platform_name = mach_params->platform; return 0; } static int skl_hda_audio_probe(struct platform_device *pdev) { - struct skl_machine_pdata *pdata; + struct snd_soc_acpi_mach *mach; struct skl_hda_private *ctx; int ret; @@ -146,11 +146,11 @@ static int skl_hda_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - pdata = dev_get_drvdata(&pdev->dev); - if (!pdata) + mach = dev_get_drvdata(&pdev->dev); + if (!mach) return -EINVAL; - ret = skl_hda_fill_card_info(pdata); + ret = skl_hda_fill_card_info(&mach->mach_params); if (ret < 0) { dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); return ret; @@ -158,7 +158,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->pcm_count = hda_soc_card.num_links; ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ - ctx->platform_name = pdata->platform; + ctx->platform_name = mach->mach_params.platform; hda_soc_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&hda_soc_card, ctx); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 29225623b4b4..ebab04b51bc1 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -525,7 +525,6 @@ static int skl_machine_device_register(struct skl *skl) { struct snd_soc_acpi_mach *mach = skl->mach; struct hdac_bus *bus = skl_to_bus(skl); - struct skl_machine_pdata *pdata; struct platform_device *pdev; int ret; @@ -542,12 +541,9 @@ static int skl_machine_device_register(struct skl *skl) return -EIO; } - if (mach->pdata) { - pdata = (struct skl_machine_pdata *)mach->pdata; - pdata->platform = dev_name(bus->dev); - pdata->codec_mask = bus->codec_mask; - dev_set_drvdata(&pdev->dev, mach->pdata); - } + mach->mach_params.platform = dev_name(bus->dev); + mach->mach_params.codec_mask = bus->codec_mask; + dev_set_drvdata(&pdev->dev, mach); skl->i2s_dev = pdev; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: use standard interface for Atom machine drivers" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: use standard interface for Atom machine drivers 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 3ee1cd4f81e15f51638db80fb9f1371b3bdf05ba Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:50 -0500 Subject: [PATCH] ASoC: Intel: use standard interface for Atom machine drivers Don't rely on internal Atom/SST-specific data structures, use generic interface to let other drivers use the same machine drivers as is, e.g. SOF to support BYT-CR devices Tested-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_acpi.c | 4 ++++ sound/soc/intel/boards/bytcr_rt5640.c | 6 +----- sound/soc/intel/boards/bytcr_rt5651.c | 6 +----- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +----- 4 files changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index c90b04cc071d..ac542535b9d5 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -341,6 +341,10 @@ static int sst_acpi_probe(struct platform_device *pdev) byt_rvp_platform_data.res_info = &bytcr_res_info; } + /* update machine parameters */ + mach->mach_params.acpi_ipc_irq_index = + pdata->res_info->acpi_ipc_irq_index; + plat_dev = platform_device_register_data(dev, pdata->platform, -1, NULL, 0); if (IS_ERR(plat_dev)) { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8587bd3d1cc1..09591144ea7d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -29,7 +29,6 @@ #include <linux/input.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -1152,10 +1151,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (is_valleyview()) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index c44298130720..e528995668b7 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -32,7 +32,6 @@ #include <linux/slab.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -920,10 +919,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (x86_match_cpu(baytrail_cpu_ids)) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index f5a5ea6a093c..250a356a0cbf 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -27,7 +27,6 @@ #include <linux/dmi.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -585,10 +584,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (is_valleyview()) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: boards: fix Skylake typo" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: boards: fix Skylake typo 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 48bf41a2be51ef3f67b60f85bf75cecfb57884ba Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:51 -0500 Subject: [PATCH] ASoC: Intel: boards: fix Skylake typo s/skylaye/skylake Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++-- 4 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..3bc78e3a91ee 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -441,7 +441,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) } -static struct snd_soc_ops skylaye_refcap_ops = { +static struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; @@ -525,7 +525,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", @@ -736,7 +736,7 @@ static struct snd_soc_dai_link kabylake_max98927_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 99e1320c485f..6ea969c0a5fb 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -586,7 +586,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static struct snd_soc_ops skylaye_refcap_ops = { +static struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; @@ -655,7 +655,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d31482b8c9bb..552958ce736d 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -400,7 +400,7 @@ static int skylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static const struct snd_soc_ops skylaye_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = skylake_refcap_startup, }; @@ -447,7 +447,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [SKL_DPCM_AUDIO_DMIC_CP] = { .name = "Skl Audio DMIC cap", diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index e877bb60beb1..f985b30a1d0e 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -449,7 +449,7 @@ static int skylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static const struct snd_soc_ops skylaye_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = skylake_refcap_startup, }; @@ -496,7 +496,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [SKL_DPCM_AUDIO_DMIC_CP] = { .name = "Skl Audio DMIC cap", -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL 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 972b0d456e645ea8fd3fdc70f95f0e41c27c0870 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:52 -0500 Subject: [PATCH] ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL GFP_ATOMIC is not required on any Intel drivers, use GFP_KERNEL instead. A first cleanup was merged in April but missed a number occurrences and new ones were added by copy/paste inertia. While we are at it, make checkpatch happy with a sizeof(*msg) Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 5 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index af93244b4868..00a37a09dc9b 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -166,11 +166,11 @@ int sst_create_ipc_msg(struct ipc_post **arg, bool large) { struct ipc_post *msg; - msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); + msg = kzalloc(sizeof(*msg), GFP_KERNEL); if (!msg) return -ENOMEM; if (large) { - msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); + msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); if (!msg->mailbox_data) { kfree(msg); return -ENOMEM; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 51f0d45d6f8f..9de64f447e7b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -403,7 +403,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) const char *i2c_name; int i; - drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c4b94e2617c5..c74c4f17316f 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -603,7 +603,7 @@ static int geminilake_audio_probe(struct platform_device *pdev) { struct glk_card_private *ctx; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3bc78e3a91ee..58eb0fe69978 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -935,7 +935,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index b6f287fa9505..15c502d6774d 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -140,7 +140,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s: entry\n", __func__); - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: common: add SOF information for APL RVP" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: common: add SOF information for APL RVP 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 97bb91ae2f3325fd06ea2c28fb8b5b4e023b4b5d Mon Sep 17 00:00:00 2001 From: Bard liao <bard.liao(a)intel.com> Date: Thu, 1 Nov 2018 16:34:53 -0500 Subject: [PATCH] ASoC: Intel: common: add SOF information for APL RVP Add firmware/topology information for APL RVP Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Bard liao <bard.liao(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index f39386e540d3..2756fa4ab552 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -19,6 +19,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { .id = "INT343A", .drv_name = "bxt_alc298s_i2s", .fw_filename = "intel/dsp_fw_bxtn.bin", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-rt298.tplg", + .asoc_plat_name = "0000:00:0e.0", }, { .id = "DLGS7219", -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2 0/7] ASoC: Intel: cleanups
by Pierre-Louis Bossart 06 Nov '18

06 Nov '18

This patchset contain 4 different cleanups: a) change of pdata definition to avoid using SST-specific structures and allow for machine driver reuse w/ SOF. No new features, just different data structures. b) removal of remaining GFP_ATOMIC usages c) Cleanup of machine driver tables for APL (missing SOF information and quirk for APL RVP/LeafHill CRB boards). These boards are used outside of Intel so it's worth sharing the fixes for the general public. d) typo v2: Added Andy Shevchenko's Reviewed-by tag removed global variable w/ .driver_data in DMI quirk Bard liao (1): ASoC: Intel: common: add SOF information for APL RVP Pierre-Louis Bossart (6): ASoC: acpi: define common interface for machine driver configuration ASoC: Intel: use standard interface for Hdaudio machine driver ASoC: Intel: use standard interface for Atom machine drivers ASoC: Intel: boards: fix Skylake typo ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL ASoC: Intel: common: add quirk for APL RVP boards include/sound/soc-acpi.h | 14 ++++++++ sound/soc/intel/atom/sst/sst_acpi.c | 4 +++ sound/soc/intel/atom/sst/sst_pvt.c | 4 +-- sound/soc/intel/boards/bytcr_rt5640.c | 6 +--- sound/soc/intel/boards/bytcr_rt5651.c | 6 +--- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +--- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++--- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +-- sound/soc/intel/boards/skl_hda_dsp_generic.c | 22 ++++++------ .../soc/intel/boards/skl_nau88l25_max98357a.c | 4 +-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 +-- .../intel/common/soc-acpi-intel-bxt-match.c | 36 +++++++++++++++++++ sound/soc/intel/skylake/skl.c | 10 ++---- 15 files changed, 85 insertions(+), 47 deletions(-) -- 2.17.1

2 8

[alsa-devel] [PATCH 0/3] audio-graph-card: remove endpoint bidirectional check
by Kuninori Morimoto 06 Nov '18

06 Nov '18

Hi Mark These are cleanup for audio-graph-card. It removes unneeded error check code Kuninori Morimoto (3): ASoC: audio-graph-card: remove endpoint bidirectional check ASoC: audio-graph-scu-card: remove endpoint bidirectional check ASoC: audio-graph-scu-card: remove error check which never happen sound/soc/generic/audio-graph-card.c | 9 --------- sound/soc/generic/audio-graph-scu-card.c | 19 +------------------ 2 files changed, 1 insertion(+), 27 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH 00/10] ASoC: rsnd: add TDM Split mode
by Kuninori Morimoto 06 Nov '18

06 Nov '18

Hi Mark Cc: Jiada These patches are for TDM Split mode support for Renesas rsnd driver. 1) - 6) : tidyup / prepare for TDM Split mode suport 7), 9) : DT binding patch for TDM Split mode 8), 10) : TDM Split mode support Kuninori Morimoto (10): 1) ASoC: rsnd: remove endpoint bidirectional check 2) ASoC: rsnd: rename rsnd_runtime_is_ssi_xxx() 3) ASoC: rsnd: use defined io_playback/io_capture 4) ASoC: rsnd: move HDMI information from ssi.c to core.c 5) ASoC: rsnd: handle DPCM converted rate/chan under core 6) ASoC: rsnd: tidyup Document for DT binding 7) ASoC: rsnd: add SSIU BUSIF support for Document 8) ASoC: rsnd: add SSIU BUSIF support 9) ASoC: rsnd: add TDM Split mode support for Document 10) ASoC: rsnd: add TDM Split mode support .../devicetree/bindings/sound/renesas,rsnd.txt | 388 ++++++++++++--------- sound/soc/sh/rcar/core.c | 112 +++++- sound/soc/sh/rcar/ctu.c | 45 --- sound/soc/sh/rcar/dma.c | 37 +- sound/soc/sh/rcar/rsnd.h | 30 +- sound/soc/sh/rcar/src.c | 40 +-- sound/soc/sh/rcar/ssi.c | 115 ++---- sound/soc/sh/rcar/ssiu.c | 212 +++++++++-- 8 files changed, 604 insertions(+), 375 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 19

[alsa-devel] Applied "ASoC: rsnd: add TDM Split mode support for Document" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: rsnd: add TDM Split mode support for Document 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 cf704dc83fd6600758e98ac71769eaeb00dfc770 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 6 Nov 2018 05:21:22 +0000 Subject: [PATCH] ASoC: rsnd: add TDM Split mode support for Document This patch adds TDM Split mode support. rsnd driver is assuming audio-graph-scu-card is used for Sound Card. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/renesas,rsnd.txt | 91 +++++++++++++++++++ 1 file changed, 91 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 6ed1a3658ffd..453e8ec2ac78 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -188,6 +188,25 @@ In DMA mode, you can select not to use SSIU by using "no-busif" via SSI. SSIU handles BUSIF which will be used for TDM Split mode. This driver is assuming that audio-graph card will be used. +TDM Split mode merges 4 sounds. You can see 4 sound interface on system, +and these sounds will be merged SSIU/SSI. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 xxxx.wav & + aplay -D plughw:0,2 xxxx.wav & + aplay -D plughw:0,3 xxxx.wav + + 2ch 8ch + [MEM] -> [SSIU 30] -+-> [SSIU 3] --> [Codec] + 2ch | + [MEM] -> [SSIU 31] -+ + 2ch | + [MEM] -> [SSIU 32] -+ + 2ch | + [MEM] -> [SSIU 33] -+ + +see "Example: simple sound card for TDM Split" + ============================================= * SSI (Serial Sound Interface) ============================================= @@ -648,6 +667,78 @@ rsnd_tdm: sound { }; }; +============================================= +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 ============================================= -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH - dmix v3 0/1] pcm: dmix: Align slave_hw_ptr to slave period boundary
by Timo Wischer 06 Nov '18

06 Nov '18

On 10/30/18 14:21, Takashi Iwai wrote: > Yes, and is this correct? Suppose you start a stream at the position > one sample before the next period (psize * N + (psize-1)). Then > slave_hw_ptr is psize * N. At the next moment, the dpcm->hw.ptr > reaches to psize * (N+1), and snd_pcm_dmix_sync_ptr() gets called. > Then slave_hw_ptr will be updated to psize * (N+1) although only one > sample has been processed? I have created a spreadsheet to simulate the behavior of dmix (see attachment). I used this sheet to test different corner cases with different implementations. Column B-D describes 3 corner cases with the original implementation. Column E-H describes the corner cases with patch v2 applied and column I-L describes the corner cases with patch v3. With patch v3 I was not able to find a corner case which would fail. All our internal audio test are also passing fine. The corner cases are described in the form N*period+period-1 (N+1)period+1 (N+2)period+1 The first line describes the value of dmix->spcm->hw.ptr when snd_pcm_dmix_start() will be called. The second line describes the hw_ptr when poll() returns for the first time and the third line describes the hw_ptr when poll() returns for the second time. In column B is the issue case described when the patch is not applied. snd_pcm_avail() returns 2 frames only (B27) but the buffer only contains 3 frames (B18). Therefore up to 5 frames are available (a buffer size of 8 frames is choosen). The issue case of patch v2 is shown in column H. snd_pcm_avail() returns 7 frames (H46) but the buffer contains still 3 frames (H37). Therefore there are only 5 frames free to overwrite. Do you see any corner cases which I missed or any other drawbacks? Best regards and thanks for your time Timo

3 5

[alsa-devel] [PATCH] ASoC: pxa-ssp: make suspend call resilient against multiple invocations
by Daniel Mack 06 Nov '18

06 Nov '18

The suspend callback is called twice when the machine is suspended, once through the ASoC layer and once through the platform device layer. Make sure not to confuse the clock layer with unbalanced calls, and bail out early if we are already suspended. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/pxa-ssp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index adcf8ba9d287..fd93706702a6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -51,6 +51,7 @@ struct ssp_priv { uint32_t cr1; uint32_t to; uint32_t psp; + bool suspended; #endif }; @@ -142,6 +143,9 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; + if (priv->suspended) + return 0; + if (!cpu_dai->active) clk_prepare_enable(ssp->clk); @@ -152,6 +156,8 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) pxa_ssp_disable(ssp); clk_disable_unprepare(ssp->clk); + priv->suspended = true; + return 0; } @@ -161,6 +167,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; + if (!priv->suspended) + return 0; + clk_prepare_enable(ssp->clk); __raw_writel(sssr, ssp->mmio_base + SSSR); @@ -174,6 +183,8 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) else clk_disable_unprepare(ssp->clk); + priv->suspended = false; + return 0; } -- 2.17.2

2 1

[alsa-devel] [PATCH] fix the kernel panic due to wrong use the dev memory API
by He, Bo 06 Nov '18

06 Nov '18

skl->dais is allocated with devm_kcalloc, can't free with the krealloc. Memory allocated with devm API is automatically freed on driver detach, Like all other devres resources. Refer to drivers/base/devres.c devm_kmalloc for more details. Signed-off-by: he, bo <bo.he(a)intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e391..928d314 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1438,7 +1438,8 @@ int skl_platform_register(struct device *dev) } if (!skl->use_tplg_pcm) { - dais = krealloc(skl->dais, sizeof(skl_fe_dai) + + devm_kfree(dev, skl->dais); + dais = devm_kcalloc(dev, skl->dais, sizeof(skl_fe_dai) + sizeof(skl_platform_dai), GFP_KERNEL); if (!dais) { ret = -ENOMEM; @@ -1472,7 +1473,5 @@ int skl_platform_unregister(struct device *dev) } } - kfree(skl->dais); - return 0; } -- 2.7.4

3 4

[alsa-devel] [PATCH] ALSA: hda - Fix incorrect clearance of thinkpad_acpi hooks
by Takashi Iwai 06 Nov '18

06 Nov '18

Since the commit c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls") we allow enabling the mic mute LED with multiple ADCs. The commit changed the function return value to be zero or a negative error, while this change was overlooked in the thinkpad_acpi helper code where it still expects a positive return value for success. This eventually leads to a NULL dereference on a system that has only a mic mute LED. This patch corrects the return value check in the corresponding code as well. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201621 Fixes: c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/thinkpad_helper.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 97f49b751e6e..568575b72f2f 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -58,8 +58,8 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = false; } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0 && - snd_hda_gen_add_micmute_led(codec, - update_tpacpi_micmute) > 0) + !snd_hda_gen_add_micmute_led(codec, + update_tpacpi_micmute)) removefunc = false; } -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: skl: always select SND_SOC_HDAC_HDA
by Arnd Bergmann 06 Nov '18

06 Nov '18

The skylake sound support is written to work both with or without CONFIG_SND_SOC_HDAC_HDA, and uses an #ifdef to decide whether it should link against that. However, this fails with SND_SOC_ALL_CODECS=m or SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH=m when the Skylake support itself is built-in, with this link error: sound/soc/intel/skylake/skl.o: In function `skl_probe': skl.c:(.text+0x56c): undefined reference to `snd_soc_hdac_hda_get_ops' Using an explicit 'select' here simplifies the logic and prevents it from happening, at the cost of always including the compile time dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/skl.c | 2 -- 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0caa1f4eb94d..c21ce7624af1 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_INTEL_SKYLAKE depends on PCI && ACPI select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER + select SND_SOC_HDAC_HDA select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST select SND_SOC_ACPI_INTEL_MATCH diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 29225623b4b4..1069ee265ce5 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -870,9 +870,7 @@ static int skl_create(struct pci_dev *pci, hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); -#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA) ext_ops = snd_soc_hdac_hda_get_ops(); -#endif snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops); bus->use_posbuf = 1; skl->pci = pci; -- 2.18.0

5 10

[alsa-devel] plughw behaviour not consistent with hw
by Stefano Antonelli 06 Nov '18

06 Nov '18

Hello list, I'm working on an application that requires a period time of 10ms. The hardware at the other end of the bus is a ulaw codec running at 8kHz. The application runs at 16kHz S16_LE and I was hoping to use the plughw device to do sample rate conversion and ulaw encode/decode. This way the application doesn't need to change to accommodate this ulaw codec. However, it's not working and I don't know how to go about debugging. I'm hoping I can get some pointers here. The problem is best illustrated by this output, first using plughw: alsa_read: 10 alsa_read: 20 alsa_read: 0 alsa_read: 20 alsa_read: 0 alsa_read: 20 alsa_read: 0 alsa_read: 20 alsa_read: 0 alsa_read: 20 It would appear that plughw gives the correct period time on the first call, but then subsequently buffers 2 periods before returning. Using hw with matching sample rate, sample format, and number of samples: alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_r ead: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 This code snippet generated the above output: for (i = 0; i < 10; ++i) { gettimeofday(&time1, NULL); err = snd_pcm_readi (capture_handle, buf, NSMP); if (err != NSMP) { printf ("read from audio interface failed " " (%s)\n", snd_strerror (err)); exit (1); } gettimeofday(&time2, NULL); printf("alsa_read: %d\n", getTimeDiff(time1, time2)); } I have tried using the patch from here (without the environment variable switch): http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/01170 3.html to print out the configuration of plughw's plugin chain, but it generates no output. I've verified the md5sum of the library after compilation and it matches the one I compiled. And I ran ldconfig after replacing libasound.so. So I don't know what's going on there. I definitely call snd_pcm_hw_params() when setting up the capture device. This is using alsa-lib-1.0.25. Does anyone know why plughw might be acting the way it is and how I can get it to deliver 10ms periods? Equally appreciated would be any tips on how to debug this problem. Thanks, Stef

2 1

[alsa-devel] [PATCH][RESEND] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new
by Kuninori Morimoto 06 Nov '18

06 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. These information are needed for driver not for user. Let's remove these from snd_kcontrol_new. There is no detail explanation for these register on datasheet, but, judging from its behavior, it seems these register needs clock to read. Otherwise, I2C returns error. Therefor, amixer command will be failed under non working timing. Without this patch, user often fail amixer command. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark 2weeks passed. repost this patch sound/soc/codecs/pcm3168a.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 5f4327f..115f2165 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), - SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), - SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), - SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), - SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +172,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), - SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), - SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), - SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.7.4

1 0

[alsa-devel] aplay: fix description about options for process duration
by GitHub pull_request - opened 06 Nov '18

06 Nov '18

alsa-project/alsa-utils pull request #1 was opened from takaswie: Hi, At present, aplay has two options for duration to process PCM frames. However, the recent one ('--samples') option is not described yet in its manual. Furthermore, '-s' option was used for '--sleep-min' option once and description for this option was left as is. This can puzzle uses. This patchset improves description about options for process duration. Request URL : https://github.com/alsa-project/alsa-utils/pull/1 Patch URL : https://github.com/alsa-project/alsa-utils/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH v2 0/3] Huawei laptops WMI & sound fixes
by Ayman Bagabas 05 Nov '18

05 Nov '18

This patch set fixes some of the issues with Huawei laptops. [PATCH v2 1/3] The first patch adds support for missing hotkeys on some models. Some hotkeys, like brightness keys, work out of the box on these models. [PATCH v2 2/3] This one enables the front speakers on the Huawei Matebook X Pro (MBXP). This solves bug 200501 https://bugzilla.kernel.org/show_bug.cgi?id=200501 It simply uses the pins configurations generated by hdajackretast using the settings posted on the bug page https://imgur.com/a/N1xsCVZ [PATCH v2 3/3] This enables the micmute LED on Huawei laptops. It calls an WMI method, using PATCH #1, to turn the micmute LED on/off. Ayman Bagabas (3): x86: add support for Huawei WMI hotkeys. ALSA: hda: fix front speakers on Huawei MBXP. ALSA: hda: add support for Huawei WMI MicMute LED drivers/platform/x86/Kconfig | 13 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei_wmi.c | 224 ++++++++++++++++++++++++++++++ include/linux/huawei_wmi.h | 7 + sound/pci/hda/huawei_wmi_helper.c | 48 +++++++ sound/pci/hda/patch_realtek.c | 28 ++++ 6 files changed, 321 insertions(+) create mode 100644 drivers/platform/x86/huawei_wmi.c create mode 100644 include/linux/huawei_wmi.h create mode 100644 sound/pci/hda/huawei_wmi_helper.c -- 2.17.2

4 13

[alsa-devel] linux-4.20-rc1/sound/soc/stm/stm32_sai_sub.c:343: Faulty error check
by David Binderman 05 Nov '18

05 Nov '18

Hello there, linux-4.20-rc1/sound/soc/stm/stm32_sai_sub.c:343]: (style) Checking if unsigned variable 'div' is less than zero. Source code is div = stm32_sai_get_clk_div(sai, parent_rate, rate); if (div < 0) return div; Suggest store return value from function in int local variable, test it and only then assign it into an unsigned variable. Regards David Binderman

2 1

[alsa-devel] [PATCH] vga_switcheroo: Fix missing gpu_bound call at audio client registration
by Takashi Iwai 05 Nov '18

05 Nov '18

The commit 37a3a98ef601 ("ALSA: hda - Enable runtime PM only for discrete GPU") added a new ops gpu_bound to be called when GPU gets bound. The patch overlooked, however, that vga_switcheroo_enable() is called only once at GPU is bound. When an audio client is registered after that point, it would miss the gpu_bound call. This leads to the unexpected lack of runtime PM in HD-audio side. For addressing that regression, just call gpu_bound callback manually at vga_switcheroo_register_audio_client() when the GPU was already bound. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201615 Fixes: 37a3a98ef601 ("ALSA: hda - Enable runtime PM only for discrete GPU") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/vga/vga_switcheroo.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index cf2a18571d48..a132c37d7334 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -380,6 +380,9 @@ int vga_switcheroo_register_audio_client(struct pci_dev *pdev, mutex_unlock(&vgasr_mutex); return -EINVAL; } + /* notify if GPU has been already bound */ + if (ops->gpu_bound) + ops->gpu_bound(pdev, id); } mutex_unlock(&vgasr_mutex); -- 2.19.1

2 6

[alsa-devel] [PATCH 0/5] ASoC: omap-mcbsp: Cleanup and split core/sidetone support
by Peter Ujfalusi 05 Nov '18

05 Nov '18

Hi, While preparing for merging the davinci and omap directories (I will send the series for that shortly) I have taken some time to do cleanup on the McBSP driver. The mcbsp.c/h files were the result when we moved code from arch/arm/plat-omap to sound/soc/omap/ a long time ago and it contained code for McBSP core and the OMAP3 sidetone functionality. With this series I tried to split the core and sidetone code to separate files. There were two reasons for this: 1. to have the OMAP specific DAI driver files prefixed with omap-* under the new sound/soc/ti/ directory 2. easier to find functions as all sidetone code is in separate file. Regards, Peter --- Peter Ujfalusi (5): ASoC: omap-mcbsp: Clean up dma_data addr initialization code ASoC: omap-mcbsp: Clean up the interrupt handlers ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1104 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 855 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1281 insertions(+), 1330 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 5

[alsa-devel] [PATCH] ASoC: sunxi: rename SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP
by Vasily Khoruzhick 05 Nov '18

05 Nov '18

SND_SUN50I_CODEC_ANALOG selects SND_SUNXI_ADDA_PR_REGMAP which is leftover of renaming SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP. Replace it with SND_SUN8I_ADDA_PR_REGMAP to fix possible link errors for some configurations: sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe': sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init' Fixes: 42371f327df0 ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls") Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/sunxi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 66aad0d3f9c7..8134c3c94229 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -31,7 +31,7 @@ config SND_SUN8I_CODEC_ANALOG config SND_SUN50I_CODEC_ANALOG tristate "Allwinner sun50i Codec Analog Controls Support" depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select SND_SUNXI_ADDA_PR_REGMAP + select SND_SUN8I_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in Allwinner A64 SoC. -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: stm32: sai: fix less than zero comparison on unsigned int
by Colin King 05 Nov '18

05 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> Currently, the comparison of div < 0 is always false because div is an unsigned int. Fix this by making div an int. Detected by CoverityScan, CID#1475309 ("Unsigned compared against 0") Fixes: 8307b2afd386 "(ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai_sub.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 211589b0b2ef..d4825700b63f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -336,8 +336,7 @@ static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, { struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); struct stm32_sai_sub_data *sai = mclk->sai_data; - unsigned int div; - int ret; + int div, ret; div = stm32_sai_get_clk_div(sai, parent_rate, rate); if (div < 0) -- 2.19.1

2 1

[alsa-devel] [PATCH] ASoC: smd845: constify snd_soc_ops structure
by Julia Lawall 05 Nov '18

05 Nov '18

The snd_soc_ops structure can be const as it is only stored in the ops field of a snd_soc_dai_link structure and this field is const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/qcom/sdm845.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 9effbecc571f..8d0cdff64377 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -171,7 +171,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) } } -static struct snd_soc_ops sdm845_be_ops = { +static const struct snd_soc_ops sdm845_be_ops = { .hw_params = sdm845_snd_hw_params, .startup = sdm845_snd_startup, .shutdown = sdm845_snd_shutdown,

2 1

[alsa-devel] [PATCH] [v2] ASoC: wm97xx: fix uninitialized regmap pointer problem
by Arnd Bergmann 05 Nov '18

05 Nov '18

gcc notices that without either the ac97 bus or the pdata, we never initialize the regmap pointer, which leads to an uninitialized variable access: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized] Since that configuration is invalid, it's better to return an error here. I tried to avoid adding complexity to the conditions, and turned the #ifdef into a regular if(IS_ENABLED()) check for readability. This in turn requires moving some header file declarations out of an #ifdef. The same code is used in three drivers, all of which I'm changing the same way. Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: fix a missing check that Charles pointed out --- include/sound/soc.h | 2 +- sound/soc/codecs/wm9705.c | 10 ++++------ sound/soc/codecs/wm9712.c | 10 ++++------ sound/soc/codecs/wm9713.c | 10 ++++------ 4 files changed, 13 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 70c10a8f3e90..3e0ac310a3df 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -553,12 +553,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -#ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_component(struct snd_ac97 *ac97); +#ifdef CONFIG_SND_SOC_AC97_BUS int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ccdf088461b7..54c306707c02 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -325,8 +325,7 @@ static int wm9705_soc_probe(struct snd_soc_component *component) if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; regmap = wm9705->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (IS_ERR(wm9705->ac97)) { @@ -339,7 +338,8 @@ static int wm9705_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9705->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_set_drvdata(component, wm9705->ac97); @@ -350,14 +350,12 @@ static int wm9705_soc_probe(struct snd_soc_component *component) static void wm9705_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); - if (!wm9705->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9705->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9705->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9705 = { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e873baa9e778..01949eaba4fd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -642,8 +642,7 @@ static int wm9712_soc_probe(struct snd_soc_component *component) if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { int ret; wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, @@ -660,7 +659,8 @@ static int wm9712_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9712->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -673,14 +673,12 @@ static int wm9712_soc_probe(struct snd_soc_component *component) static void wm9712_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); - if (!wm9712->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9712->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9712->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9712 = { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 643863bb32e0..5a2fdf4f69bf 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1214,8 +1214,7 @@ static int wm9713_soc_probe(struct snd_soc_component *component) if (wm9713->mfd_pdata) { wm9713->ac97 = wm9713->mfd_pdata->ac97; regmap = wm9713->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) @@ -1225,7 +1224,8 @@ static int wm9713_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9713->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -1238,14 +1238,12 @@ static int wm9713_soc_probe(struct snd_soc_component *component) static void wm9713_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); - if (!wm9713->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9713->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9713->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9713 = { -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: Intel: mrfld: fix uninitialized variable access
by Arnd Bergmann 05 Nov '18

05 Nov '18

Randconfig testing revealed a very old bug, with gcc-8: sound/soc/intel/atom/sst/sst_loader.c: In function 'sst_load_fw': sound/soc/intel/atom/sst/sst_loader.c:357:5: error: 'fw' may be used uninitialized in this function [-Werror=maybe-uninitialized] if (fw == NULL) { ^ sound/soc/intel/atom/sst/sst_loader.c:354:25: note: 'fw' was declared here const struct firmware *fw; We must check the return code of request_firmware() before we look at the pointer result that may be uninitialized when the function fails. Fixes: 9012c9544eea ("ASoC: Intel: mrfld - Add DSP load and management") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/atom/sst/sst_loader.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 27413ebae956..b8c456753f01 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst) const struct firmware *fw; retval = request_firmware(&fw, sst->firmware_name, sst->dev); - if (fw == NULL) { - dev_err(sst->dev, "fw is returning as null\n"); - return -EINVAL; - } if (retval) { dev_err(sst->dev, "request fw failed %d\n", retval); return retval; } + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } mutex_lock(&sst->sst_lock); retval = sst_cache_and_parse_fw(sst, fw); mutex_unlock(&sst->sst_lock); -- 2.18.0

3 2

[alsa-devel] Applied "ASoC: nau8825: remove unnecessary unlikely()" to the asoc tree
by Mark Brown 05 Nov '18

05 Nov '18

The patch ASoC: nau8825: remove unnecessary unlikely() 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 0b6277e6343e192aaa7d452ab933281eb0d420dc Mon Sep 17 00:00:00 2001 From: Yangtao Li <tiny.windzz(a)gmail.com> Date: Sun, 4 Nov 2018 07:55:29 -0500 Subject: [PATCH] ASoC: nau8825: remove unnecessary unlikely() WARN_ON() already contains an unlikely(), so it's not necessary to use unlikely. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8825.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index b9fed99d8b5e..7bbcbf5f05c8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -424,10 +424,8 @@ static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) { u32 gain, sidetone; - if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { - WARN_ON(1); + if (WARN_ON(sig_org == 0 || sig_cros == 0)) return 0; - } sig_org = nau8825_intlog10_dec3(sig_org); sig_cros = nau8825_intlog10_dec3(sig_cros); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: AMD: constify regulator_desc structure
by Julia Lawall 05 Nov '18

05 Nov '18

The regulator_desc structure can be const as it is only passed as the second argument of devm_regulator_register and the corresponding parameter is declared as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3f813ea5210a..a5daad973ce5 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -403,7 +403,7 @@ static struct regulator_config acp_da7219_cfg = { static struct regulator_ops acp_da7219_ops = { }; -static struct regulator_desc acp_da7219_desc = { +static const struct regulator_desc acp_da7219_desc = { .name = "reg-fixed-1.8V", .type = REGULATOR_VOLTAGE, .owner = THIS_MODULE,

2 1

[alsa-devel] [PATCH] ASoC: omap-mcbsp: On OMAP1 and OMAP2420 only 16bit format is supported
by Peter Ujfalusi 05 Nov '18

05 Nov '18

For samples sizes of 20/24/32 bit the DRR2/DXR2 registers must be used to receive or transmit the most significant part of the 32bit sample. We can not trick the system by using the DRR2/DXR2 register offset and use 32bit element size since these (and other McBSP) registers are 16bit ones. We would need support for port_window in cyclic mode from the sDMA driver, but it is not straight forward as we would need to have the sDMA frame to cover the 32bit (DRR2+DRR1/DXR2+DXR1), but the frames must cover the ALSA period to be able to receive periodic interrupts. Since the 32bit samples are not working (DMA timeout), just remove it in case we have McBSP with reg_size of 2. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have noticed that on Nokia n810 32bit samples just don't work at all for audio Regards, Peter sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d0ebb6b9bfac..caa24c463b3e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -862,6 +862,11 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; + if (mcbsp->pdata->reg_size == 2) { + omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 4

[alsa-devel] [PATCH] ASoC: pcm3168a: cleanup hw_constraint for LEFT_J/I2S
by Kuninori Morimoto 05 Nov '18

05 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> hw constraint for LEFT_J/I2S are same, but had duplicated code. This code cleanup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e1c3a18..115f2165 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -497,6 +497,10 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, unsigned int fmt; unsigned int sample_min; unsigned int channel_max; + unsigned int channel_maxs[] = { + 6, /* rx */ + 8 /* tx */ + }; if (tx) fmt = pcm3168a->dac_fmt; @@ -521,18 +525,9 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, channel_max = 2; break; case PCM3168A_FMT_LEFT_J: - sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; - break; case PCM3168A_FMT_I2S: sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; + channel_max = channel_maxs[tx]; break; default: sample_min = 24; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/3] ASoC: rsnd: prepare patches for BUSIF support
by Kuninori Morimoto 05 Nov '18

05 Nov '18

Hi Mark These are prepare/cleanup for for Renesas rsnd driver BUSIF support. I will post BUSIF support patch soon, but it needs Jiada Wang <jiada_wang(a)mentor.com> 's check. If it was OK, I will focus to TDM split mode. Kuninori Morimoto (3): ASoC: rsnd: move .get_status under rsnd_mod_ops ASoC: rsnd: add .get_id/.get_id_sub ASoC: rsnd: rsnd_mod_name() handles both name and ID sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/cmd.c | 11 ++-- sound/soc/sh/rcar/core.c | 72 +++++++++++++++++------ sound/soc/sh/rcar/ctu.c | 23 +++++++- sound/soc/sh/rcar/dma.c | 47 +++++++-------- sound/soc/sh/rcar/dvc.c | 3 +- sound/soc/sh/rcar/gen.c | 12 ++-- sound/soc/sh/rcar/mix.c | 3 +- sound/soc/sh/rcar/rsnd.h | 21 +++---- sound/soc/sh/rcar/src.c | 29 +++++----- sound/soc/sh/rcar/ssi.c | 147 +++++++++++++++++++++++------------------------ sound/soc/sh/rcar/ssiu.c | 37 ++++++------ 12 files changed, 232 insertions(+), 175 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 6

[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0
by Hans de Goede 05 Nov '18

05 Nov '18

Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the mclk instead of pmc_plt_clk_3. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Swanky that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++++++++++++++++++-- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index db6976f4ddaa..9d9f6e41d81c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -19,6 +19,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -35,6 +36,8 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "HiFi" +#define QUIRK_PMC_PLT_CLK_0 0x01 + struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; @@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = { .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Swanky model Chromebook (Toshiba Chromebook 2) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + {} +}; + static int snd_cht_mc_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; int ret_val = 0; struct cht_mc_private *drv; + const char *mclk_name; + int quirks = 0; + + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (quirks & QUIRK_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.19.0

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6afe: Fix wrong MI2S SD line mask
by Rohit kumar 05 Nov '18

05 Nov '18

SD line mask for MI2S starts from BIT 0 instead of BIT 1. Fix all bit mask for MI2S SD lines. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/qdsp6/q6afe.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 000775b..829b5e987 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -49,14 +49,14 @@ #define AFE_PORT_I2S_SD1 0x2 #define AFE_PORT_I2S_SD2 0x3 #define AFE_PORT_I2S_SD3 0x4 -#define AFE_PORT_I2S_SD0_MASK BIT(0x1) -#define AFE_PORT_I2S_SD1_MASK BIT(0x2) -#define AFE_PORT_I2S_SD2_MASK BIT(0x3) -#define AFE_PORT_I2S_SD3_MASK BIT(0x4) -#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1) -#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3) -#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1) -#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1) +#define AFE_PORT_I2S_SD0_MASK BIT(0x0) +#define AFE_PORT_I2S_SD1_MASK BIT(0x1) +#define AFE_PORT_I2S_SD2_MASK BIT(0x2) +#define AFE_PORT_I2S_SD3_MASK BIT(0x3) +#define AFE_PORT_I2S_SD0_1_MASK GENMASK(1, 0) +#define AFE_PORT_I2S_SD2_3_MASK GENMASK(3, 2) +#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(2, 0) +#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(3, 0) #define AFE_PORT_I2S_QUAD01 0x5 #define AFE_PORT_I2S_QUAD23 0x6 #define AFE_PORT_I2S_6CHS 0x7 -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[alsa-devel] [PATCH] ALSA: usb-audio: Define registers for CM6206
by Linus Walleij 05 Nov '18

05 Nov '18

The register map for CM6206 is known and we can define what the values written to the different registers actually mean. I tested to print the hex values before/after this change, there is no functional change. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/usb/quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 124 insertions(+), 5 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8a945ece9869..3d0f09108c98 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -19,6 +19,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/midi.h> +#include <linux/bits.h> #include <sound/control.h> #include <sound/core.h> @@ -668,15 +669,133 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) } /* - * C-Media CM6206 is based on CM106 with two additional - * registers that are not documented in the data sheet. - * Values here are chosen based on sniffing USB traffic - * under Windows. + * CM6206 registers from the CM6206 datasheet rev 2.1 */ +#define CM6206_REG0_DMA_MASTER BIT(15) +#define CM6206_REG0_SPDIFO_RATE_48K (2 << 12) +#define CM6206_REG0_SPDIFO_RATE_96K (7 << 12) +/* Bit 4 thru 11 is the S/PDIF category code */ +#define CM6206_REG0_SPDIFO_CAT_CODE_GENERAL (0 << 4) +#define CM6206_REG0_SPDIFO_EMPHASIS_CD BIT(3) +#define CM6206_REG0_SPDIFO_COPYRIGHT_NA BIT(2) +#define CM6206_REG0_SPDIFO_NON_AUDIO BIT(1) +#define CM6206_REG0_SPDIFO_PRO_FORMAT BIT(0) + +#define CM6206_REG1_TEST_SEL_CLK BIT(14) +#define CM6206_REG1_PLLBIN_EN BIT(13) +#define CM6206_REG1_SOFT_MUTE_EN BIT(12) +#define CM6206_REG1_GPIO4_OUT BIT(11) +#define CM6206_REG1_GPIO4_OE BIT(10) +#define CM6206_REG1_GPIO3_OUT BIT(9) +#define CM6206_REG1_GPIO3_OE BIT(8) +#define CM6206_REG1_GPIO2_OUT BIT(7) +#define CM6206_REG1_GPIO2_OE BIT(6) +#define CM6206_REG1_GPIO1_OUT BIT(5) +#define CM6206_REG1_GPIO1_OE BIT(4) +#define CM6206_REG1_SPDIFO_INVALID BIT(3) +#define CM6206_REG1_SPDIF_LOOP_EN BIT(2) +#define CM6206_REG1_SPDIFO_DIS BIT(1) +#define CM6206_REG1_SPDIFI_MIX BIT(0) + +#define CM6206_REG2_DRIVER_ON BIT(15) +#define CM6206_REG2_HEADP_SEL_SIDE_CHANNELS (0 << 13) +#define CM6206_REG2_HEADP_SEL_SURROUND_CHANNELS (1 << 13) +#define CM6206_REG2_HEADP_SEL_CENTER_SUBW (2 << 13) +#define CM6206_REG2_HEADP_SEL_FRONT_CHANNELS (3 << 13) +#define CM6206_REG2_MUTE_HEADPHONE_RIGHT BIT(12) +#define CM6206_REG2_MUTE_HEADPHONE_LEFT BIT(11) +#define CM6206_REG2_MUTE_REAR_SURROUND_RIGHT BIT(10) +#define CM6206_REG2_MUTE_REAR_SURROUND_LEFT BIT(9) +#define CM6206_REG2_MUTE_SIDE_SURROUND_RIGHT BIT(8) +#define CM6206_REG2_MUTE_SIDE_SURROUND_LEFT BIT(7) +#define CM6206_REG2_MUTE_SUBWOOFER BIT(6) +#define CM6206_REG2_MUTE_CENTER BIT(5) +#define CM6206_REG2_MUTE_RIGHT_FRONT BIT(3) +#define CM6206_REG2_MUTE_LEFT_FRONT BIT(3) +#define CM6206_REG2_EN_BTL BIT(2) +#define CM6206_REG2_MCUCLKSEL_1_5_MHZ (0) +#define CM6206_REG2_MCUCLKSEL_3_MHZ (1) +#define CM6206_REG2_MCUCLKSEL_6_MHZ (2) +#define CM6206_REG2_MCUCLKSEL_12_MHZ (3) + +/* Bit 11..13 sets the sensitivity to FLY tuner volume control VP/VD signal */ +#define CM6206_REG3_FLYSPEED_DEFAULT (2 << 11) +#define CM6206_REG3_VRAP25EN BIT(10) +#define CM6206_REG3_MSEL1 BIT(9) +#define CM6206_REG3_SPDIFI_RATE_44_1K BIT(0 << 7) +#define CM6206_REG3_SPDIFI_RATE_48K BIT(2 << 7) +#define CM6206_REG3_SPDIFI_RATE_32K BIT(3 << 7) +#define CM6206_REG3_PINSEL BIT(6) +#define CM6206_REG3_FOE BIT(5) +#define CM6206_REG3_ROE BIT(4) +#define CM6206_REG3_CBOE BIT(3) +#define CM6206_REG3_LOSE BIT(2) +#define CM6206_REG3_HPOE BIT(1) +#define CM6206_REG3_SPDIFI_CANREC BIT(0) + +#define CM6206_REG5_DA_RSTN BIT(13) +#define CM6206_REG5_AD_RSTN BIT(12) +#define CM6206_REG5_SPDIFO_AD2SPDO BIT(12) +#define CM6206_REG5_SPDIFO_SEL_FRONT (0 << 9) +#define CM6206_REG5_SPDIFO_SEL_SIDE_SUR (1 << 9) +#define CM6206_REG5_SPDIFO_SEL_CEN_LFE (2 << 9) +#define CM6206_REG5_SPDIFO_SEL_REAR_SUR (3 << 9) +#define CM6206_REG5_CODECM BIT(8) +#define CM6206_REG5_EN_HPF BIT(7) +#define CM6206_REG5_T_SEL_DSDA4 BIT(6) +#define CM6206_REG5_T_SEL_DSDA3 BIT(5) +#define CM6206_REG5_T_SEL_DSDA2 BIT(4) +#define CM6206_REG5_T_SEL_DSDA1 BIT(3) +#define CM6206_REG5_T_SEL_DSDAD_NORMAL 0 +#define CM6206_REG5_T_SEL_DSDAD_FRONT 4 +#define CM6206_REG5_T_SEL_DSDAD_S_SURROUND 5 +#define CM6206_REG5_T_SEL_DSDAD_CEN_LFE 6 +#define CM6206_REG5_T_SEL_DSDAD_R_SURROUND 7 + static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { int err = 0, reg; - int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + int val[] = { + /* + * Values here are chosen based on sniffing USB traffic + * under Windows. + * + * REG0: DAC is master, sample rate 48kHz, no copyright + */ + CM6206_REG0_SPDIFO_RATE_48K | + CM6206_REG0_SPDIFO_COPYRIGHT_NA, + /* + * REG1: PLL binary search enable, soft mute enable. + */ + CM6206_REG1_PLLBIN_EN | + CM6206_REG1_SOFT_MUTE_EN | + /* + * REG2: enable output drivers, + * select front channels to the headphone output, + * then mute the headphone channels, run the MCU + * at 1.5 MHz. + */ + CM6206_REG2_DRIVER_ON | + CM6206_REG2_HEADP_SEL_FRONT_CHANNELS | + CM6206_REG2_MUTE_HEADPHONE_RIGHT | + CM6206_REG2_MUTE_HEADPHONE_LEFT, + /* + * REG3: default flyspeed, set 2.5V mic bias + * enable all line out ports and enable SPDIF + */ + CM6206_REG3_FLYSPEED_DEFAULT | + CM6206_REG3_VRAP25EN | + CM6206_REG3_FOE | + CM6206_REG3_ROE | + CM6206_REG3_CBOE | + CM6206_REG3_LOSE | + CM6206_REG3_HPOE | + CM6206_REG3_SPDIFI_CANREC, + /* REG4 is just a bunch of GPIO lines */ + 0x0000, + /* REG5: de-assert AD/DA reset signals */ + CM6206_REG5_DA_RSTN | + CM6206_REG5_AD_RSTN }; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); -- 2.19.1

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8996: fix small typo
by Mark Brown 05 Nov '18

05 Nov '18

On Fri, Nov 02, 2018 at 09:30:06AM +0800, sh liu wrote: > From e9b923690675ca8fa883fd25dcead5b457856735 Mon Sep 17 00:00:00 2001 > From: liush <liush(a)allwinnertech.com> > Date: Fri, 2 Nov 2018 08:57:00 +0800 > Subject: [PATCH] ASoC: wm8996: fix small typo > > atleast -> at least > > Change-Id: Icc970b438166daef13518b7d1a62b13eb8752f5f > Signed-off-by: liush <liush(a)allwinnertech.com> This patch is corrupted and can't be applied: > *codec, int fll_id, int source, > if (i2c->irq) > timeout *= 10; > else > - /* ensure timeout of atleast 1 jiffies */ > + /* ensure timeout of at least 1 jiffies */ > timeout = timeout/2 ? : 1; > It looks like tabs have been replaced by spaces in whatever you're using to send it.

1 0

[alsa-devel] [PATCH] SoC: nau8825: remove unnecessary unlikely()
by Yangtao Li 04 Nov '18

04 Nov '18

WARN_ON() already contains an unlikely(), so it's not necessary to use unlikely. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> --- sound/soc/codecs/nau8825.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index b9fed99d8b5e..7bbcbf5f05c8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -424,10 +424,8 @@ static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) { u32 gain, sidetone; - if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { - WARN_ON(1); + if (WARN_ON(sig_org == 0 || sig_cros == 0)) return 0; - } sig_org = nau8825_intlog10_dec3(sig_org); sig_cros = nau8825_intlog10_dec3(sig_cros); -- 2.17.0

1 0

[alsa-devel] Trying to get Roland TD-30 soundcard to work on Linux
by Cupertino Miranda 02 Nov '18

02 Nov '18

Hi, I have the Rolands TD-30 edrum module that includes a USB sound card in it. The card lists but no capture or recording devices work, although they are listed. I am new to audio device drivers and for that reason need some guidance to contribute to such driver. A full log can be found at: www.alsa-project.org/db/?f=e1a28cbd7e9f1201d3b0cbe44d8ec1908272a1d4 Looking forward to ear from you. Best regards, Cupertino Miranda

1 0

[alsa-devel] [PATCH 1/2] ALSA: x86: Fix runtime PM for hdmi-lpe-audio
by Ville Syrjala 02 Nov '18

02 Nov '18

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Commit 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") broke runtime PM with lpe audio. We can no longer runtime suspend the GPU since the sysfs power/control for the lpe-audio device no longer exists and the device is considered always active. We can fix this by not marking the device as active. Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Fixes: 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/x86/intel_hdmi_audio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 83d76c345940..bbed4acaafd1 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1877,7 +1877,6 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); - pm_runtime_set_active(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { -- 2.18.1

3 11

[alsa-devel] Pull request testing (please ignore)
by GitHub pull_request-opened 02 Nov '18

02 Nov '18

alsa-project/alsa-lib pull request #1 was opened from perexg: Comment body Issue URL : https://github.com/alsa-project/alsa-lib/pull/1 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH] ASoC: wm97xx: fix uninitialized regmap pointer problem
by Arnd Bergmann 02 Nov '18

02 Nov '18

gcc notices that without either the ac97 bus or the pdata, we never initialize the regmap pointer, which leads to an uninitialized variable access: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized] Since that configuration is invalid, it's better to return an error here. I tried to avoid adding complexity to the conditions, and turned the #ifdef into a regular if(IS_ENABLED()) check for readability. This in turn requires moving some header file declarations out of an #ifdef. The same code is used in three drivers, all of which I'm changing the same way. Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- include/sound/soc.h | 2 +- sound/soc/codecs/wm9705.c | 10 ++++------ sound/soc/codecs/wm9712.c | 10 ++++------ sound/soc/codecs/wm9713.c | 10 ++++------ 4 files changed, 13 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 70c10a8f3e90..3e0ac310a3df 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -553,12 +553,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -#ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_component(struct snd_ac97 *ac97); +#ifdef CONFIG_SND_SOC_AC97_BUS int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ccdf088461b7..54c306707c02 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -325,8 +325,7 @@ static int wm9705_soc_probe(struct snd_soc_component *component) if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; regmap = wm9705->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (IS_ERR(wm9705->ac97)) { @@ -339,7 +338,8 @@ static int wm9705_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9705->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_set_drvdata(component, wm9705->ac97); @@ -350,14 +350,12 @@ static int wm9705_soc_probe(struct snd_soc_component *component) static void wm9705_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); - if (!wm9705->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9705->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9705->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9705 = { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e873baa9e778..01949eaba4fd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -642,8 +642,7 @@ static int wm9712_soc_probe(struct snd_soc_component *component) if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { int ret; wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, @@ -660,7 +659,8 @@ static int wm9712_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9712->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -673,14 +673,12 @@ static int wm9712_soc_probe(struct snd_soc_component *component) static void wm9712_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); - if (!wm9712->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9712->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9712->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9712 = { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 643863bb32e0..b2bb465c8e5a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1214,8 +1214,7 @@ static int wm9713_soc_probe(struct snd_soc_component *component) if (wm9713->mfd_pdata) { wm9713->ac97 = wm9713->mfd_pdata->ac97; regmap = wm9713->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) @@ -1225,7 +1224,8 @@ static int wm9713_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9713->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -1238,14 +1238,12 @@ static int wm9713_soc_probe(struct snd_soc_component *component) static void wm9713_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); - if (!wm9713->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9713->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9713 = { -- 2.18.0

2 2

[alsa-devel] [PATCH v2 0/1] Introduce dmic mode switch delay parameter
by Jenny TC 02 Nov '18

02 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message Jenny TC (1): ASoC: dmic: Introduce mode switch delay sound/soc/codecs/dmic.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) -- 1.9.1

3 3

[alsa-devel] [PATCH alsa-utils] topology/topology.c: drop unneeded <dlfcn.h> include
by Thomas Petazzoni 02 Nov '18

02 Nov '18

This include is not used/needed and prevents building on systems that don't provide <dlfcn.h>. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- topology/topology.c | 1 - 1 file changed, 1 deletion(-) diff --git a/topology/topology.c b/topology/topology.c index 097c255..b72e341 100644 --- a/topology/topology.c +++ b/topology/topology.c @@ -27,7 +27,6 @@ #include <errno.h> #include <string.h> #include <sys/stat.h> -#include <dlfcn.h> #include <getopt.h> #include <assert.h> -- 2.14.4

2 1

[alsa-devel] [PATCH alsa-lib] utils/alsa.m4: conditionally enable libdl in AM_PATH_ALSA m4 macro
by Thomas Petazzoni 02 Nov '18

02 Nov '18

From: Thomas De Schampheleire <thomas.de.schampheleire(a)gmail.com> The AM_PATH_ALSA macro in utils/alsa.m4 unconditionally uses -ldl. This breaks compilation of alsa-utils (and probably other packages using this macro) for targets that do not support dynamic loading. This patch updates the macro to check if dlopen is available, and use that result to conditionally add -ldl to the list of libraries. Signed-off-by: Thomas De Schampheleire <thomas.de.schampheleire(a)gmail.com> Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- utils/alsa.m4 | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/utils/alsa.m4 b/utils/alsa.m4 index e12310df..a5c5a292 100644 --- a/utils/alsa.m4 +++ b/utils/alsa.m4 @@ -44,6 +44,8 @@ if test "$alsa_inc_prefix" != "" ; then fi AC_MSG_RESULT($ALSA_CFLAGS) +AC_CHECK_LIB(c, dlopen, LIBDL="", [AC_CHECK_LIB(dl, dlopen, LIBDL="-ldl")]) + dnl add any special lib dirs AC_MSG_CHECKING(for ALSA LDFLAGS) if test "$alsa_prefix" != "" ; then @@ -52,7 +54,7 @@ if test "$alsa_prefix" != "" ; then fi dnl add the alsa library -ALSA_LIBS="$ALSA_LIBS -lasound -lm -ldl -lpthread" +ALSA_LIBS="$ALSA_LIBS -lasound -lm $LIBDL -lpthread" LIBS="$ALSA_LIBS $LIBS" AC_MSG_RESULT($ALSA_LIBS) -- 2.14.4

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8996: fix small typo
by Charles Keepax 02 Nov '18

02 Nov '18

On Fri, Nov 02, 2018 at 09:30:06AM +0800, sh liu wrote: > >From e9b923690675ca8fa883fd25dcead5b457856735 Mon Sep 17 00:00:00 2001 > From: liush <liush(a)allwinnertech.com> > Date: Fri, 2 Nov 2018 08:57:00 +0800 > Subject: [PATCH] ASoC: wm8996: fix small typo > > atleast -> at least > > Change-Id: Icc970b438166daef13518b7d1a62b13eb8752f5f > Signed-off-by: liush <liush(a)allwinnertech.com> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Thanks, Charles

1 0

[alsa-devel] Bug report - Dell-WD15-Dock not working caused by ucm profile in alsa-lib 1.1.7
by René Santing 02 Nov '18

02 Nov '18

Hello Alsa developers, I would like to file a bug report about the Dell-WD15-Dock. This dock stopped working for me on archlinux/kde. The device was not picked up at all by pulse audio after some updates I installed last week. By trial and error I found out it was caused by the ucm profile that was added in alsa-lib 1.1.7. Running the following restored functionality. sudo mv /usr/share/alsa/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf /usr/share/alsa/ucm/Dell-WD15-Dock/Dell-WD15-Dock.bak pulseaudio -k I uploaded 3 alsa-info.sh profiles. 1.1.6 http://www.alsa-project.org/db/?f=a9a4c53d1e417c3d2ef304fe0778581c35b7a95e (working) 1.1.7 http://www.alsa-project.org/db/?f=bb474a687692e25f5e54c475f16f7ce4ef210f7a (not working) 1.1.7 http://www.alsa-project.org/db/?f=586dd059aaa08e839eb9df5f7b49990a5f4ad1a5 (working but moved the wd15 ucm profile) https://alsa-project.org/main/index.php/Bug_Tracking pointed me to this mailing list to submit this report. If this is the wrong place or there is more information need could you let me know so I can send it to the proper channels? Best regards, René Santing

2 9

[alsa-devel] [PATCH alsa-lib] Don't use fork() on noMMU platforms
by Thomas Petazzoni 02 Nov '18

02 Nov '18

fork() is not available on noMMU platforms, but in the specific case of pcm_direct.c, vfork() can be used instead. This commit adds an autoconf check for fork(), and falls back to vfork() is fork() is not available. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- configure.ac | 2 ++ src/pcm/pcm_direct.c | 10 +++++++++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 4c9d860f..ab659490 100644 --- a/configure.ac +++ b/configure.ac @@ -51,6 +51,8 @@ dnl Checks for library functions. AC_PROG_GCC_TRADITIONAL AC_CHECK_FUNCS([uselocale]) +AC_CHECK_FUNC([fork]) + SAVE_LIBRARY_VERSION AC_SUBST(LIBTOOL_VERSION_INFO) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 2b07eff9..4dc3ea26 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -431,13 +431,21 @@ int snd_pcm_direct_server_create(snd_pcm_direct_t *dmix) close(dmix->server_fd); return ret; } - + +#ifdef HAVE_FORK ret = fork(); +#else + ret = vfork(); +#endif if (ret < 0) { close(dmix->server_fd); return ret; } else if (ret == 0) { +#ifdef HAVE_FORK ret = fork(); +#else + ret = vfork(); +#endif if (ret == 0) server_job(dmix); _exit(EXIT_SUCCESS); -- 2.14.4

2 1

[alsa-devel] [PATCH 1/2] CHROMIUM: ASoC: rt5663: Add documentation for power supply support
by Cheng-Yi Chiang 02 Nov '18

02 Nov '18

rt5663 codec driver will support setting CPVDD and AVDD power supply from device tree. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Documentation/devicetree/bindings/sound/rt5663.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 23386446c63d6..d4058dfde0392 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -36,6 +36,9 @@ Optional properties: "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of impedance in the impedance sensing function. +- avdd-supply: Power supply for AVDD, providing 1.8V. +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + Pins on the device (for linking into audio routes) for RT5663: * IN1P @@ -51,4 +54,6 @@ rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; }; -- 2.19.1.568.g152ad8e336-goog

1 2

[alsa-devel] [PATCH] ASoC: wm8996: fix small typo
by lshua312 02 Nov '18

02 Nov '18

From e9b923690675ca8fa883fd25dcead5b457856735 Mon Sep 17 00:00:00 2001 From: liush <liush(a)allwinnertech.com> Date: Fri, 2 Nov 2018 08:57:00 +0800 Subject: [PATCH] ASoC: wm8996: fix small typo atleast -> at least Change-Id: Icc970b438166daef13518b7d1a62b13eb8752f5f Signed-off-by: liush <liush(a)allwinnertech.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8affa49..d039835 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2109,7 +2109,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (i2c->irq) timeout *= 10; else - /* ensure timeout of atleast 1 jiffies */ + /* ensure timeout of at least 1 jiffies */ timeout = timeout/2 ? : 1; for (retry = 0; retry < 10; retry++) { -- 1.9.1

1 0

[alsa-devel] [PATCH 0/7] ASoC: Intel: cleanups
by Pierre-Louis Bossart 01 Nov '18

01 Nov '18

This patchset contain 4 different cleanups: a) change of pdata definition to avoid using SST-specific structures and allow for machine driver reuse w/ SOF. No new features, just different data structures. b) removal of remaining GFP_ATOMIC usages c) Cleanup of machine driver tables for APL (missing SOF information and quirk for APL RVP/LeafHill CRB boards). These boards are used outside of Intel so it's worth sharing the fixes for the general public. d) typo Bard liao (1): ASoC: Intel: common: add SOF information for APL RVP Pierre-Louis Bossart (6): ASoC: acpi: define common interface for machine driver configuration ASoC: Intel: use standard interface for Hdaudio machine driver ASoC: Intel: use standard interface for Atom machine drivers ASoC: Intel: boards: fix Skylake typo ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL ASoC: Intel: common: add quirk for APL RVP boards include/sound/soc-acpi.h | 14 +++++++ sound/soc/intel/atom/sst/sst_acpi.c | 4 ++ sound/soc/intel/atom/sst/sst_pvt.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +-- sound/soc/intel/boards/bytcr_rt5651.c | 6 +-- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +-- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 22 +++++------ .../soc/intel/boards/skl_nau88l25_max98357a.c | 4 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 +- .../intel/common/soc-acpi-intel-bxt-match.c | 38 +++++++++++++++++++ sound/soc/intel/skylake/skl.c | 10 ++--- 15 files changed, 87 insertions(+), 47 deletions(-) -- 2.17.1

3 11

[alsa-devel] [USB-Audio - PX USB, playback] snd-usb-audio: probe of 2-3:1.0 failed with error -22 after kernel upgrade
by Mutz Heuer 01 Nov '18

01 Nov '18

USB-Audio Connection on Bowers & Wilkens PX Bluetooth Headphones started working when upgrading kernel from 4.13 to 4.15 series. Then it suddenly stopped working again on newer kernels. The bug report has been generated on the latest working mainline kernel 4.15.7. All newer kernels have some usb quirks in place, which should fix the playback sample rates to 48 kHz. The quirk seems not to be working correctly - it seems to hinder the audio system to recognize the USB-device as a sound device. The Headphones work on kernels before 4,15.7 when manually configuring pulseaudio to a default samplerate of 48 kHz. PX Headphones are on the newest firmware level. See https://patchwork.kernel.org/patch/10265345/. Dropping that patch and recompiling the kernel leads to a situation where alsa and pulseaudio seem to to be working again. Adding default-sample-rate = 48000 to /etc/pulse/daemon.conf leads to working headphones. Maybe the patch has to be adjusted to newer firmware? A quirk seems to be really necessary to avoid the specific patch to daemon.conf. More details can be found in https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1773697 What would be the best way to proceed?

2 4

[alsa-devel] [PATCH 3/3] ALSA:hda: add support for Huawei WMI MicMute LED
by Ayman Bagabas 01 Nov '18

01 Nov '18

Some of Huawei laptops come with a LED in the mic mute key. This patch enables and disable this LED when the internal microphone status is changed. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- include/linux/huawei_wmi.h | 7 ++++ sound/pci/hda/huawei_wmi_helper.c | 66 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 12 ++++++ 3 files changed, 85 insertions(+) create mode 100644 include/linux/huawei_wmi.h create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/include/linux/huawei_wmi.h b/include/linux/huawei_wmi.h new file mode 100644 index 000000000000..69b656c5029b --- /dev/null +++ b/include/linux/huawei_wmi.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __HUAWEI_WMI_H__ +#define __HUAWEI_WMI_H__ + +int huawei_wmi_micmute_led_set(bool on); + +#endif diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..57256f51fd88 --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,66 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI Mic Mute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_LAPTOP) +#include <linux/huawei_wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (!ucontrol || !huawei_wmi_micmute_led_set_func) + return; + if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { + /* TODO: How do I verify if it's a mono or stereo here? */ + bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; + huawei_wmi_micmute_led_set_func(!val); + } +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct hda_gen_spec *spec = codec->spec; + bool removefunc = false; + + codec_info(codec, "In alc_fixup_huawei_wmi\n"); + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = true; + if (huawei_wmi_micmute_led_set_func(false) >= 0) { + if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch) + codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); + else { + spec->cap_sync_hook = update_huawei_wmi_micmute_led; + removefunc = false; + } + } + codec_info(codec, "In alc_fixup_huawei_wmi IF\n"); + + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void hda_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f7a39c7883c..0dab00c0df50 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5371,6 +5371,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for dell wmi mic mute led */ #include "dell_wmi_helper.c" +/* for huawei wmi mic mute led */ +#include "huawei_wmi_helper.c" + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -6348,6 +6352,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi, + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6363,6 +6371,8 @@ static const struct hda_fixup alc269_fixups[] = { {0x21, 0x04211020}, { }, }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC295_FIXUP_HP_X360] = { .type = HDA_FIXUP_FUNC, @@ -6610,6 +6620,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ @@ -6775,6 +6786,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, .name = "alc274-dell-aio"}, {.id = ALC255_FIXUP_DUMMY_LINEOUT_VERB, .name = "alc255-dummy-lineout"}, {.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"}, + {.id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, .name = "alc256-huawei-micmute"}, {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, {} }; -- 2.17.2

2 2

[alsa-devel] [PATCH 2/3] ALSA:hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 01 Nov '18

01 Nov '18

This patch enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ac7ba9b342d..4f7a39c7883c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -6347,6 +6348,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { }, + }, + }, [ALC295_FIXUP_HP_X360] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_hp_top_speakers, @@ -6592,6 +6609,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

3 2

[alsa-devel] Bug report - snd_hda_intel - snd_pcm_delay() returned a value that is exceptionally large -
by Sergey Popol 31 Oct '18

31 Oct '18

Hello, After a few hours my ubuntu workstation becomes unresponsive (requiring a Sysreq reisub to reboot): ================================================================= $ cat /var/log/syslog | grep 20:25 Oct 31 20:25:01 soulwax CRON[1587]: (root) CMD (command -v debian-sa1 > /dev/null && debian-sa1 1 1) Oct 31 20:25:28 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -417964 bytes (-2369 ms). Oct 31 20:25:28 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Oct 31 20:25:29 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event3 - Logitech USB Optical Mouse: SYN_DROPPED event - some input events have been lost. Oct 31 20:25:31 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: snd_pcm_dump(): Oct 31 20:25:32 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Soft volume PCM Oct 31 20:25:34 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Control: PCM Playback Volume Oct 31 20:25:34 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: min_dB: -51 Oct 31 20:25:35 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: max_dB: 0 Oct 31 20:25:36 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event4 - Logitech USB Keyboard: SYN_DROPPED event - some input events have been lost. Oct 31 20:25:36 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: resolution: 256 Oct 31 20:25:37 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Its setup is: Oct 31 20:25:37 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: stream : PLAYBACK Oct 31 20:25:37 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "21" Oct 31 20:25:37 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event2 - Power Button: device removed Oct 31 20:25:38 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: access : MMAP_INTERLEAVED Oct 31 20:25:38 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "24" Oct 31 20:25:39 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event5 - Video Bus: device removed Oct 31 20:25:39 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "25" Oct 31 20:25:39 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event1 - Power Button: device removed Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "26" Oct 31 20:25:40 soulwax kernel: [35165.869886] rfkill: input handler enabled Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event0 - Sleep Button: device removed Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "27" Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event3 - Logitech USB Optical Mouse: device removed Oct 31 20:25:41 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "28" Oct 31 20:25:41 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event4 - Logitech USB Keyboard: device removed Oct 31 20:25:41 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: format : S16_LE Oct 31 20:25:42 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "29" Oct 31 20:25:42 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event6 - Logitech USB Keyboard: device removed Oct 31 20:25:43 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "30" Oct 31 20:25:54 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event7 - ITE8713 CIR transceiver: device removed ================================================================= http://www.alsa-project.org/db/?f=d5e5889f64b41a208f4dad2c1ac4b49a9b9ec558 Hope that can help. Sergei

1 0

[alsa-devel] [PATCH 3/3] ALSA:hda: add support for Huawei WMI MicMute LED
by ayman.bagabas＠gmail.com 31 Oct '18

31 Oct '18

1 0

[alsa-devel] [PATCH] ALSA:hda: fix front speakers on Huawei MBXP.
by ayman.bagabas＠gmail.com 31 Oct '18

31 Oct '18

1 0

[alsa-devel] [PATCH] ASoC: rsnd: fixup clock start checker
by Kuninori Morimoto 31 Oct '18

31 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4d230d12710646 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") fixuped clock start timing. But it exchanged clock start checker from ssi->usrcnt to ssi->rate. Current rsnd_ssi_master_clk_start() is called from .prepare, but some player (for example GStreamer) might calls it many times. In such case, the checker might returns error even though it was not error. It should check ssi->usrcnt instead of ssi->rate. This patch fixup it. Without this patch, GStreamer can't switch 48kHz / 44.1kHz. Reported-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 95acc8b..dc85fff 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -300,7 +300,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->rate) { + if (ssi->usrcnt > 1) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; -- 2.7.4

2 1

[alsa-devel] Bug in hdmi codec driver in DPCM
by Rohit Kumar 31 Oct '18

31 Oct '18

Hi, We are facing crash in hdmi codec driver when we use DPCM based dai-links. Crash is happening in hdmi_codec_eld_chmap() while accessing chmap_info struct during hdmi_codec_startup(). However, chmap_info struct is allocated in hdmi_codec_pcm_new() which is only called in non-DPCM based dai-links. Below is the crash stack: [61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018 ..... [61.666696]CM = 0, WnR = 1 [61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000 [61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000 [61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP [61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21 .. [61.740269] PC is at hdmi_codec_startup+0x124/0x164 [61.745308] LR is at hdmi_codec_startup+0xe4/0x164 .. We would probably need to move the implementation on pcm_new() to some other function to make it compatible with DPCM. Can you please provide suggestions in solving the issue. Thanks, Rohit

2 2

[alsa-devel] [PATCH] ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider()
by Wei Yongjun 30 Oct '18

30 Oct '18

sizeof() when applied to a pointer typed expression gives the size of the pointer, not that of the pointed data. Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ea05cc9..211589b 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -390,7 +390,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) char *mclk_name, *p, *s = (char *)pname; int ret, i = 0; - mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); + mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL); if (!mclk) return -ENOMEM;

2 1

[alsa-devel] [PATCH v2] ASoC: AMD: Fix race condition between register access
by Agrawal, Akshu 30 Oct '18

30 Oct '18

During simultaneous running of playback and capture, we got hit by incorrect value write on common register. This was due to race condition between 2 streams. Fixing this by locking the common register access. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Added 2 helper functions, removed locking in ch enable/disable as they are separate registers. sound/soc/amd/acp-pcm-dma.c | 63 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 48 insertions(+), 15 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 0ac4b5b..5be9c2d 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -121,6 +121,9 @@ .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +/* Lock to protect access to registers */ +static DEFINE_SPINLOCK(lock); + static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -131,6 +134,33 @@ static void acp_reg_write(u32 val, void __iomem *acp_mmio, u32 reg) writel(val, acp_mmio + (reg * 4)); } +static void acp_reg_write_srbm_targ(void __iomem *acp_mmio, + u32 addr, u32 data) +{ + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + acp_reg_write(addr, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); + acp_reg_write(data, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + spin_unlock_irqrestore(&lock, flags); +} + +static void acp_reg_read_mod_write(void __iomem *acp_mmio, u32 addr, + u32 mask, bool set) +{ + u32 val; + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, addr); + if (set) + val |= mask; + else + val &= ~mask; + acp_reg_write(val, acp_mmio, addr); + spin_unlock_irqrestore(&lock, flags); +} + /* * Configure a given dma channel parameters - enable/disable, * number of descriptors, priority @@ -172,15 +202,12 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, sram_offset = (descr_idx * sizeof(acp_dma_dscr_transfer_t)); /* program the source base address. */ - acp_reg_write(sram_offset, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->src, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset, descr_info->src); /* program the destination base address. */ - acp_reg_write(sram_offset + 4, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->dest, acp_mmio, mmACP_SRBM_Targ_Idx_Data); - + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 4, descr_info->dest); /* program the number of bytes to be transferred for this descriptor. */ - acp_reg_write(sram_offset + 8, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->xfer_val, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 8, + descr_info->xfer_val); } static void pre_config_reset(void __iomem *acp_mmio, u16 ch_num) @@ -309,8 +336,12 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, u32 low; u32 high; u32 offset; + unsigned long flags; offset = ACP_DAGB_GRP_SRBM_SRAM_BASE_OFFSET + (pte_offset * 8); + + spin_lock_irqsave(&lock, flags); + for (page_idx = 0; page_idx < (num_of_pages); page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), @@ -333,6 +364,8 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Move to next physically contiguos page */ pg++; } + + spin_unlock_irqrestore(&lock, flags); } static void config_acp_dma(void __iomem *acp_mmio, @@ -870,29 +903,29 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } } if (adata->asic_type == CHIP_STONEY) { - val = acp_reg_read(adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + val = ACP_I2S_SP_16BIT_RESOLUTION_EN; } } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + val = ACP_I2S_MIC_16BIT_RESOLUTION_EN; } } - acp_reg_write(val, adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); + acp_reg_read_mod_write(adata->acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN, + val, + true); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: AMD: Fix race condition between register access
by Agrawal, Akshu 30 Oct '18

30 Oct '18

During simultaneous running of playback and capture, we got hit by incorrect value write on common register. This was due to race condition between 2 streams. Fixing this by locking the common register access. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 0ac4b5b..993a7db 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -121,6 +121,9 @@ .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +/* Lock to protect access to registers */ +static DEFINE_SPINLOCK(lock); + static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -168,9 +171,12 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, acp_dma_dscr_transfer_t *descr_info) { u32 sram_offset; + unsigned long flags; sram_offset = (descr_idx * sizeof(acp_dma_dscr_transfer_t)); + spin_lock_irqsave(&lock, flags); + /* program the source base address. */ acp_reg_write(sram_offset, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); acp_reg_write(descr_info->src, acp_mmio, mmACP_SRBM_Targ_Idx_Data); @@ -181,6 +187,8 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, /* program the number of bytes to be transferred for this descriptor. */ acp_reg_write(sram_offset + 8, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); acp_reg_write(descr_info->xfer_val, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + + spin_unlock_irqrestore(&lock, flags); } static void pre_config_reset(void __iomem *acp_mmio, u16 ch_num) @@ -309,8 +317,12 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, u32 low; u32 high; u32 offset; + unsigned long flags; offset = ACP_DAGB_GRP_SRBM_SRAM_BASE_OFFSET + (pte_offset * 8); + + spin_lock_irqsave(&lock, flags); + for (page_idx = 0; page_idx < (num_of_pages); page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), @@ -333,6 +345,8 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Move to next physically contiguos page */ pg++; } + + spin_unlock_irqrestore(&lock, flags); } static void config_acp_dma(void __iomem *acp_mmio, @@ -367,6 +381,7 @@ static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, u16 cap_channel) { u32 val, ch_reg, imr_reg, res_reg; + unsigned long flags; switch (cap_channel) { case CAP_CHANNEL1: @@ -381,6 +396,8 @@ static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, imr_reg = mmACP_I2SMICSP_IMR0; break; } + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { @@ -393,12 +410,15 @@ static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; acp_reg_write(val, acp_mmio, imr_reg); acp_reg_write(0x1, acp_mmio, ch_reg); + + spin_unlock_irqrestore(&lock, flags); } static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, u16 cap_channel) { u32 val, ch_reg, imr_reg; + unsigned long flags; switch (cap_channel) { case CAP_CHANNEL1: @@ -411,11 +431,15 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, ch_reg = mmACP_I2SMICSP_RER0; break; } + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, imr_reg); val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; acp_reg_write(val, acp_mmio, imr_reg); acp_reg_write(0x0, acp_mmio, ch_reg); + + spin_unlock_irqrestore(&lock, flags); } /* Start a given DMA channel transfer */ @@ -847,6 +871,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, int status; uint64_t size; u32 val = 0; + unsigned long flags; struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; @@ -870,6 +895,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } } if (adata->asic_type == CHIP_STONEY) { + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -893,6 +920,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + + spin_unlock_irqrestore(&lock, flags); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: fix oops w/ for_each_rtd_codec_dai_rollback() macro
by Pierre-Louis Bossart 30 Oct '18

30 Oct '18

A kernel oops happens on an error case (usual missing BE mixer configuration required by Intel SST driver). Git bisect points to this macro and an operator precedence issue. for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);) The initial code replaced by this macro was while (--i >= 0) { codec_dai = rtd->codec_dais[i]; Fix the C operator precedence difference by reverting to pre-decrement Fixes: 0b7990e3897 ('ASoC: add for_each_rtd_codec_dai() macro') Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f1dab1f4b194..70c10a8f3e90 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1192,7 +1192,7 @@ struct snd_soc_pcm_runtime { ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ (i)++) #define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ - for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);) + for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);) /* mixer control */ -- 2.17.1

2 1

[alsa-devel] [PATCH - dmix v2 1/1] pcm: dmix: Align slave_hw_ptr to slave period boundary
by twischer＠de.adit-jv.com 30 Oct '18

30 Oct '18

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> These changes are required due to the kernel commit 07b7acb51d283d8469696c906b91f1882696a4d4 ("ASoC: rsnd: update pointer more accurate") Issue is that snd_pcm_wait() goes back to waiting because the hw_ptr is not period aligned. Therefore snd_pcm_wait() will block for a longer time as required. With these rcar driver changes the exact position of the dma is returned. During snd_pcm_start they read hw_ptr as reference, and this hw_ptr is now not period aligned, and is a little ahead over the period while it is read. Therefore when the avail is calculated during snd_pcm_wait(), it is missing the avail_min by a few frames. Consider the below example: Considering the application is trying to write 0x120 frames and the period_size = 0x60, avail_min = 0x120 and buffersize = 0x360 : rsnd_pointer=0x12c -> dma pointer at the end of second write during snd_pcm_dmix_start(). Since another 0x120 buffer is available, application writes 0x120 and goes to snd_pcm_wait(). It is woken up after 3 snd_pcm_period_elapsed() to see rsnd_pointer=0x248. So hw_ptr = new_slave_hw_ptr - reference_slave_hw_ptr = 0x248 - 0x12c = 0x11c. It needs 4 more frames to be able to write. And so it goes back to waiting. But since 3 snd_pcm_period_elapsed(), 3 periods should be available and it should have been able to write. If rsnd_pointer during the start was 0x120 which is 3 periods then 0x248 - 0x120 = 128 it could go on with write. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> --- > On 10/29/18 16:54, Takashi Iwai wrote: > The problem is that aligning the start essentially imposes an > artificial long latency, and changes the behavior out of sudden. Now, we are only align the salve_hw_ptr which also solves our delay issue. But this change should not increase the latency because the write position is given by slave_appl_ptr. Do you see any other drawbacks? Best regards Timo diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index a6a8f3a..eaa0b0f 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -560,6 +560,8 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; + dmix->slave_hw_ptr = ((dmix->slave_hw_ptr / dmix->slave_period_size) + * dmix->slave_period_size); if (pcm->buffer_size > pcm->period_size * 2) return; /* If we have too litte periods, better to align the start position -- 2.7.4

3 3

[alsa-devel] [PATCH] ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size
by Takashi Sakamoto 30 Oct '18

30 Oct '18

In a former commit, PCM constraint based on LCM of SYT_INTERVAL was obsoleted with PCM rule. However, the new PCM rule brings -EINVAL in some cases that max/min values of size of buffer/period is not multiples of one of values of SYT_INTERVAL. For example, pulseaudio always fail to configure PCM substream. This commit changes strategy for the PCM rule. Although the buggy rules had a single dependency (rate from period, period from rate, rate from buffer, buffer from rate), a revised rule has double dependencies (period from period/rate, buffer from buffer/rate). A step of value is calculated with table of SYT_INTERVAL and list of available rates. This prevents interval template which brings -EINVAL to a call of snd_interval_refine(). Fixes: 5950229582bc('ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 57 ++++++----------------------------- 1 file changed, 9 insertions(+), 48 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index fcd965f1d69e..9be76c808fcc 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -146,53 +146,22 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, struct snd_interval *s = hw_param_interval(params, rule->var); const struct snd_interval *r = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval t = { - .min = s->min, .max = s->max, .integer = 1, - }; + struct snd_interval t = {0}; + unsigned int step = 0; int i; for (i = 0; i < CIP_SFC_COUNT; ++i) { - unsigned int rate = amdtp_rate_table[i]; - unsigned int step = amdtp_syt_intervals[i]; - - if (!snd_interval_test(r, rate)) - continue; - - t.min = roundup(t.min, step); - t.max = rounddown(t.max, step); + if (snd_interval_test(r, amdtp_rate_table[i])) + step = max(step, amdtp_syt_intervals[i]); } - if (snd_interval_checkempty(&t)) - return -EINVAL; + t.min = roundup(s->min, step); + t.max = rounddown(s->max, step); + t.integer = 1; return snd_interval_refine(s, &t); } -static int apply_constraint_to_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval *r = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]); - struct snd_interval t = { - .min = UINT_MAX, .max = 0, .integer = 1, - }; - int i; - - for (i = 0; i < CIP_SFC_COUNT; ++i) { - unsigned int step = amdtp_syt_intervals[i]; - unsigned int rate = amdtp_rate_table[i]; - - if (s->min % step || s->max % step) - continue; - - t.min = min(t.min, rate); - t.max = max(t.max, rate); - } - - return snd_interval_refine(r, &t); -} - /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. @@ -250,24 +219,16 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, */ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - apply_constraint_to_rate, NULL, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); - if (err < 0) - goto end; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - apply_constraint_to_rate, NULL, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); - if (err < 0) - goto end; end: return err; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: ca0106: Disable IZD on SB0570 DAC to fix audio pops
by Alex Stanoev 29 Oct '18

29 Oct '18

The Creative Audigy SE (SB0570) card currently exhibits an audible pop whenever playback is stopped or resumed, or during silent periods of an audio stream. Initialise the IZD bit to the 0 to eliminate these pops. The Infinite Zero Detection (IZD) feature on the DAC causes the output to be shunted to Vcap after 2048 samples of silence. This discharges the AC coupling capacitor through the output and causes the aforementioned pop/click noise. The behaviour of the IZD bit is described on page 15 of the WM8768GEDS datasheet: "With IZD=1, applying MUTE for 1024 consecutive input samples will cause all outputs to be connected directly to VCAP. This also happens if 2048 consecutive zero input samples are applied to all 6 channels, and IZD=0. It will be removed as soon as any channel receives a non-zero input". I believe the second sentence might be referring to IZD=1 instead of IZD=0 given the observed behaviour of the card. This change should make the DAC initialisation consistent with Creative's Windows driver, as this popping persists when initialising the card in Linux and soft rebooting into Windows, but is not present on a cold boot to Windows. Signed-off-by: Alex Stanoev <alex(a)astanoev.com> --- sound/pci/ca0106/ca0106.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index 04402c14cb23..9847b669cf3c 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -582,7 +582,7 @@ #define SPI_PL_BIT_R_R (2<<7) /* right channel = right */ #define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */ #define SPI_IZD_REG 2 -#define SPI_IZD_BIT (1<<4) /* infinite zero detect */ +#define SPI_IZD_BIT (0<<4) /* infinite zero detect */ #define SPI_FMT_REG 3 #define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */ -- 2.19.1

2 1

[alsa-devel] [PATCH] pcm: dmix: Align slave_hw_ptr to slave period boundary
by twischer＠de.adit-jv.com 29 Oct '18

29 Oct '18

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> These changes are required due to the kernel commit 07b7acb51d283d8469696c906b91f1882696a4d4 ("ASoC: rsnd: update pointer more accurate") Issue is that snd_pcm_wait() goes back to waiting because the hw_ptr is not period aligned. Therefore snd_pcm_wait() will block for a longer time as required. With these rcar driver changes the exact position of the dma is returned. During snd_pcm_start they read hw_ptr as reference, and this hw_ptr is now not period aligned, and is a little ahead over the period while it is read. Therefore when the avail is calculated during snd_pcm_wait(), it is missing the avail_min by a few frames. Consider the below example: Considering the application is trying to write 0x120 frames and the period_size = 0x60, avail_min = 0x120 and buffersize = 0x360 : rsnd_pointer=0x12c -> dma pointer at the end of second write during snd_pcm_dmix_start(). Since another 0x120 buffer is available, application writes 0x120 and goes to snd_pcm_wait(). It is woken up after 3 snd_pcm_period_elapsed() to see rsnd_pointer=0x248. So hw_ptr = new_slave_hw_ptr - reference_slave_hw_ptr = 0x248 - 0x12c = 0x11c. It needs 4 more frames to be able to write. And so it goes back to waiting. But since 3 snd_pcm_period_elapsed(), 3 periods should be available and it should have been able to write. If rsnd_pointer during the start was 0x120 which is 3 periods then 0x248 - 0x120 = 128 it could go on with write. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- We are not completely sure if this is the best approach but we had also no better idea to solve this problem (with similar CPU usage). Therefore if anyone has a better solution please feel free to describe this one. src/pcm/pcm_dmix.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index a6a8f3a..d3e9319 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -560,11 +560,10 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; - if (pcm->buffer_size > pcm->period_size * 2) - return; - /* If we have too litte periods, better to align the start position - * to the period boundary so that the interrupt can be handled properly - * at the right time. + dmix->slave_hw_ptr = ((dmix->slave_hw_ptr / dmix->slave_period_size) + * dmix->slave_period_size); + /* Better to align the start position to the period boundary so that + * the interrupt can be handled properly at the right time. */ dmix->slave_appl_ptr = ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) / dmix->slave_period_size) * dmix->slave_period_size; -- 2.7.4

2 1

[alsa-devel] alsaloop problems; ALSA streaming tutorial (resend)
by frederik＠ofb.net 29 Oct '18

29 Oct '18

Dear Mark (cc'ing ALSA-user, ALSA-devel) Thank you for your tutorial http://www.pogo.org.uk/~mark/trx/streaming-desktop-audio.html I found it helpful and clearly-written (although missing a "}" brace and I think it should be "-P" instead of "-p"...). I got as far as trying the 'alsaloop' example. It seems very finicky, sometimes I am able to play a test.wav cleanly through the loopback device, and sometimes there are a lot of crackles and I get errors like playback hw:0: change avail_min=2968 playback hw:0: change avail_min=2972 playback hw:0: change avail_min=2976 underrun for playback hw:0 last write before 9.8370ms, queued 10.0000ms/0.0000ms -> missing -0.1630ms expected 10.0000ms, processing 0.1910ms, max missing 0.0443ms last wake 0.0350ms, last check 0.0350ms, avail_min 32.0000ms max buf 40.0000ms, pfilled 0.0000ms, cfilled 10.0000ms job started before 0.0110ms For 'aplay' invocations where there are errors, it generally never recovers; the whole (10 second) wav file is played with crackles and errors being printed. I tried to figure out if this is due to some different rate settings being chosen at random with each 'aplay' invocation, but all the alsaloop debugging output looks the same between working and non-working runs of aplay (except for the above underrun errors). I also tried various settings for -A and -S in 'alsaloop', but nothing would alleviate this problem. The only thing that fixes it is to change the latency '-t' setting to a higher value than the default 10ms. However this seems like a bad solution, given that it is able (50% of the time) to play a whole sound file at the lower latency with no errors; I would rather have it recover from the underruns when they happen, than increase the latency for every invocation. Enabling realtime via the Arch Wiki info (a bit tricky as I had to edit some of the files by hand) didn't change this behavior. I also encountered a problem when trying to make alsaloop output to a dmix device. I wanted to have it output to dmix so that I could use my audio normally while experimenting with alsaloop. I got a "Poll FD initialization failed." error: playback plug:hw0mix/capture hubcap sync type: CAPTRATESHIFT New pitch for playback plug:hw0mix/capture hubcap: 1.00000000 (min/max samples = 0/0) underrun for playback plug:hw0mix last write before 19.7440ms, queued -1.0000ms/-1.0000ms -> missing -1.0000ms expected 10.0000ms, processing 0.0290ms, max missing 0.0000ms last wake 1539883911239.1819ms, last check 19.6460ms, avail_min 2.5000ms max buf 250.0000ms, pfilled 0.0000ms, cfilled 0.0000ms job started before 0.0310ms Poll FD initialization failed. Is this error understood? I would have thought that enough others would also have been wanting to use alsaloop in such a simple configuration, to warrant a better error message. I'm somewhat confused by the alsaloop manual page, my version says it is from "5 Aug 2010" but it has glaring errors, it says "-B" is a synonym for "--buffer" but alsaloop tells me "alsaloop: invalid option -- 'B'". Also it says that I can use "-c" to set both channels and rate. Are these typos really unfixed for eight years? Here are the commands I used: $ alsaloop -C hubcap -P hw:0 -v -v -U |& tee alsaloloop.out ... $ aplay -v test.wav |& tee aplay.out ... $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.14.71-1-lts. The earlier version of this email had output files attached, but they caused the message to be held for "moderator approval". Here are some HTTP versions of the same files: alsaloop.out: http://ix.io/1pUj aplay.out: http://ix.io/1pUk asound-remote (the config): http://ix.io/1pUl I think I'll investigate network audio with JACK next but I wanted to share my experiences in case there is something I'm doing wrong here, or in case there are bugs that can be fixed in ALSA. Thank you, Frederick

2 2

Re: [alsa-devel] [linux-sunxi] [BUG] On Allwinner A20 Board Cubietruck distorted SPDIF opical output
by Markus Mitsch 28 Oct '18

28 Oct '18

hi. i posted my answer wrong so here is my finding: i made the output work by disabling the sigma delta modulation. after compiling the kernel i can output through spdif with any rate. greetings markus 2018-05-26 18:38 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: > 2018-05-26 16:57 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >> 2018-05-23 17:12 GMT+02:00 Code Kipper <codekipper(a)gmail.com>: >>> Hi All, >>> I can see that the problem was introduced somewhere here >>> >>> DOESN'T WORK >>> ffc3eb6f3a83 Merge tag 'sunxi-clk-for-4.15' of >>> https://git.kernel.org/pub/scm/linux/kernel/git/sunxi/linux into >>> clk-next >>> ae74ac082886 Merge tag 'clk-v4.15-exynos-pm' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> faa865f18cb7 Merge tag 'clk-v4.15-samsung' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> f09a6b86fdae Merge tag 'clk-renesas-for-v4.15-tag1' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/geert/renesas-drivers >>> into clk-next >>> 4328a2186e51 clk: sunxi-ng: sun4i: Export video PLLs >>> 553c7d5ba2fe clk: sunxi-ng: Add A83T display clocks >>> 7679eb20353d clk: samsung: Add a separate driver for Exynos4412 ISP clocks >>> 8ca8ac102484 clk: samsung: Add dt bindings for Exynos4412 ISP clock controller >>> 75920aac275a clk: samsung: Instantiate Exynos4412 ISP clocks only when available >>> 24ea78a09f89 clk: sunxi-ng: sun8i: a23: Use sigma-delta modulation for audio PLL >>> ee6501d69217 clk: sunxi-ng: sun6i: Use sigma-delta modulation for audio PLL >>> 042f7f8f9715 clk: sunxi-ng: sun5i: Use sigma-delta modulation for audio PLL >>> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >>> a5e3e2b2ef85 clk: sunxi-ng: sun8i: h3: Use sigma-delta modulation for audio PLL >>> 392ba5fafcdf clk: sunxi-ng: nm: Add support for sigma-delta modulation >>> 05d2eaac96d4 clk: sunxi-ng: Add sigma-delta modulation support >>> 4cdbc40d64d4 clk: sunxi-ng: nm: Check if requested rate is supported >>> by fractional clock >>> d51fe3ba9773 clk: sunxi-ng: sun5i: Fix bit offset of audio PLL post-divider >>> 4e8975cbb516 clk: samsung: exynos5433: mark PM functions as __maybe_unused >>> a4f21e9ceb5c clk: samsung: Remove obsolete clkdev alias support >>> 45d882daf88a clk: samsung: Add explicit MPLL, EPLL clkdev aliases in >>> S3C2443 driver >>> cd05417f728b clk: samsung: Rework clkdev alias handling in S3C2443 driver >>> efea8d377184 clk: samsung: Rework clkdev alias handling in Exynos5440 driver >>> 36ba48240b19 clk: samsung: Drop useless alias in Exynos5420 clk driver >>> 29964890f31c clk: samsung: Remove clkdev alias support in Exynos5250 clk driver >>> 6de08891c896 clk: samsung: Remove double assignment of CLK_ARM_CLK in >>> Exynos4 driver >>> 58f4a5ff3a62 clk: samsung: Remove clkdev alias support in Exynos4 clk driver >>> c9194fb623b0 clk: samsung: Remove support for obsolete Exynos4212 CPU clock >>> d5cd103b06f9 clk: samsung: Remove support for Exynos4212 SoCs in >>> Exynos CLKOUT driver >>> 7c4f63ec94a1 clk: sunxi-ng: a83t: Fix invalid csi-mclk mux offset >>> WORKS >>> 0a4e632b6f9d Merge branch 'clk-fixes' into clk-next >>> >>> BR, >>> CK >>> >>> >>> On 22 May 2018 at 21:27, Code Kipper <codekipper(a)gmail.com> wrote: >>>> Hi Markus, >>>> I've been able to reproduce this on my A31 board with linux-next(plays >>>> at 44100 but not at 96K)so it is an issue. I also tested the same code >>>> on my A64 and it worked fine, The blocks are very similar so it maybe >>>> a clocking issue, I'll look into this. >>>> BR, >>>> CK >>>> >>>> On 21 May 2018 at 20:17, Markus Mitsch <markusmitsch86(a)googlemail.com> wrote: >>>>> 2018-05-21 15:17 GMT+02:00 Maxime Ripard <maxime.ripard(a)bootlin.com>: >>>>>> On Mon, May 21, 2018 at 02:43:55PM +0200, Markus Mitsch wrote: >>>>>>> 2018-05-21 11:43 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >>>>>>> >>> Hello, >>>>>>> >>> >>>>>>> >>> i have a new problem. if i want to output sound from optical i get >>>>>>> >>> distorted sound. At 44.1 khz its like an old vinly, but at 48 khz and >>>>>>> >>> higher it is very bad. thanks in advance for your time. >>>>>>> > >>>>>>> >> Hi Markus, >>>>>>> >> I haven't tested it for a while but when I did it worked as high as >>>>>>> >> 96K with no issues. Does it work ok with the original cubietruck >>>>>>> >> software?, >>>>>>> >> CK >>>>>>> > >>>>>>> > Hello, >>>>>>> > i am using arch linux with mainline kernel 4.16.8. parhaps i can try >>>>>>> > today with official image. >>>>>>> > mfg >>>>>>> >>>>>>> Hi. >>>>>>> So, with debian image from 2014 with 3.4.79 kernel the sound is clear >>>>>>> up to 96 khz. >>>>>> >>>>>> You said that it is a new problem, when was it introduced? Could you >>>>>> bisect the issue? >>>>>> >>>>>> Maxime >>>>> >>>>> Hello. >>>>> I dont know since when this problem appears. When saying "new" i >>>>> refered to my older post. I tried to look into the dts file and the >>>>> code >>>>> but i am afraid this is a bit too high for me. I can say that the >>>>> analog output works, even at high sampling rates. >>>>> And I have read something about a clock source could be the problem. >>>>> I'm sorry I can't tell you any more. >>>>> >>>>> mfg >>>>> Markus >> >> Hello, >> i tried undoing part of the commit >> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >> by setting the value for the divider from 1 to 4 again on both places. >> with no success. >> >> markus > > i now had a little success. the output works!! > i disabled the sigma-delta support by making > ccu_sdm_helper_enable in drivers/clk/sunxi-ng/ccu_sdm.c return imediately. > > markus

2 2

[alsa-devel] [PATCH] ALSA: dice: fix to wait for releases of all ALSA character devices
by Takashi Sakamoto 27 Oct '18

27 Oct '18

In a development period for Linux kernel v4.20, drivers in ALSA firewire stack were changed to wait for releases of all ALSA character devices at .remove callback of bus driver. However, ALSA dice driver is partly out of this change. This bug can bring fault to user process which holds the last of character device in unplugging. This commit fixes the driver to wait in the callback. Fixes: 61ccc6f6b27c ('ALSA: firewire: block .remove callback of bus driver till all of ALSA character devices are released') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 0f6dbcffe711..ed50b222d36e 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -240,8 +240,8 @@ static void dice_remove(struct fw_unit *unit) cancel_delayed_work_sync(&dice->dwork); if (dice->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(dice->card); + // Block till all of ALSA character devices are released. + snd_card_free(dice->card); } mutex_destroy(&dice->mutex); -- 2.19.1

2 1

[alsa-devel] QA issue with media-sound/alsa-utils-1.1.7 Gentoo package
by Francesco Turco 27 Oct '18

27 Oct '18

The Portage package manager returns the following QA warning when installing the media-sound/alsa-utils-1.1.7 Gentoo package: * QA Notice: Package triggers severe warnings which indicate that it * may exhibit random runtime failures. * ../alsamixer/volume_mapping.c:116:15: warning: incompatible implicit declaration of built-in function ‘exp10’ * ../alsamixer/volume_mapping.c:151:14: warning: incompatible implicit declaration of built-in function ‘exp10’ * Please do not file a Gentoo bug and instead report the above QA * issues directly to the upstream developers of this software. * Homepage: https://alsa-project.org/ -- https://fturco.gitlab.io/

1 0

[alsa-devel] [PATCH 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 25 Oct '18

25 Oct '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to mormal, off to normal etc. Jenny TC (2): ASoC: dmic: Enable ACPI device entry ASoC: dmic: Introduce mode switch delay sound/soc/codecs/dmic.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) -- 1.9.1

6 11

[alsa-devel] [PATCH] alsa-utils: fix identation indentation on speaker-test.c
by Eduardo Abinader 25 Oct '18

25 Oct '18

Signed-off-by: Eduardo Abinader <eduardoabinader(a)gmail.com> --- speaker-test/speaker-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 773af0a..4aac30c 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -1207,7 +1207,7 @@ int main(int argc, char *argv[]) { gettimeofday(&tv1, NULL); for(chn = 0; chn < channels; chn++) { - int channel = get_speaker_channel(chn); + int channel = get_speaker_channel(chn); printf(" %d - %s\n", channel, get_channel_name(channel)); err = write_loop(handle, channel, ((rate*3)/period_size), frames); -- 2.17.1

2 1

[alsa-devel] [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 25 Oct '18

25 Oct '18

Hi all, This new version addresses the feedback from v1 and fixes a bug on capture mode. Code-wise, the changes are very trivial, no major diffs. Below follows the highlights: * Standard headers like linux/if_ether.h are now checked by configure script. * The 'static' modifier was added to array variables in aaf_hw_constraint(). * aaf_close() callback was simplified. * aaf_rx_pdu() was modified so it doesn't return an error in case the number of bytes read by recv() is different from aaf->pdu_size. This is an expected situation and it means the received AVTPDU doesn't belong to the AAF stream the plugin is interested in so it should simply drop that AVTPDU. This fixes a bug when running multiple streams (with different AVPTU sizes) concurrently. Additionally, this new version improves the instructions in doc/aaf.txt in order to make it easier to test the plugin. Anyone should be able to test it just by copying and pasting the instructions. To recap what has been discussed about the plugin so far, check the comments in RFC and PATCH series archive [1-3]. All versions of this series can be also found in my alsa-plugins tree in github [4]. Regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140667.ht… [4] https://github.com/aguedes/alsa-plugins Andre Guedes (7): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support aaf: Implement dump() ioplug callback aaf: Add support for direct read/write transfers Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1200 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 11 + doc/aaf.txt | 157 ++++++++ 5 files changed, 1380 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

3 11

[alsa-devel] KASAN: use-after-free Read in snd_seq_timer_interrupt
by syzbot 24 Oct '18

24 Oct '18

Hello, syzbot found the following crash on: HEAD commit: 44786880df19 Merge branch 'parisc-4.20-1' of git://git.ker.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=178c8bad400000 kernel config: https://syzkaller.appspot.com/x/.config?x=5071a2e3cb82c943 dashboard link: https://syzkaller.appspot.com/bug?extid=d5136d4d3240cbe45a2a compiler: gcc (GCC) 8.0.1 20180413 (experimental) 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+d5136d4d3240cbe45a2a(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in snd_seq_timer_interrupt+0x364/0x3e0 sound/core/seq/seq_timer.c:145 Read of size 8 at addr ffff8801cf71c118 by task syz-executor0/24085 CPU: 1 PID: 24085 Comm: syz-executor0 Not tainted 4.19.0+ #300 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x244/0x39d lib/dump_stack.c:113 print_address_description.cold.7+0x9/0x1ff mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report.cold.8+0x242/0x309 mm/kasan/report.c:412 __asan_report_load8_noabort+0x14/0x20 mm/kasan/report.c:433 snd_seq_timer_interrupt+0x364/0x3e0 sound/core/seq/seq_timer.c:145 snd_timer_interrupt+0x9ff/0x1660 sound/core/timer.c:855 snd_timer_s_function+0xcd/0x130 sound/core/timer.c:1094 call_timer_fn+0x272/0x920 kernel/time/timer.c:1326 expire_timers kernel/time/timer.c:1363 [inline] __run_timers+0x7e5/0xc70 kernel/time/timer.c:1682 run_timer_softirq+0x52/0xb0 kernel/time/timer.c:1695 __do_softirq+0x308/0xb7e kernel/softirq.c:292 invoke_softirq kernel/softirq.c:373 [inline] irq_exit+0x17f/0x1c0 kernel/softirq.c:413 exiting_irq arch/x86/include/asm/apic.h:536 [inline] smp_apic_timer_interrupt+0x1cb/0x760 arch/x86/kernel/apic/apic.c:1061 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:801 </IRQ> RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:761 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0xaf/0xd0 kernel/locking/spinlock.c:184 Code: 30 f6 31 89 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 80 3c 02 00 75 28 48 83 3d 28 cd 7c 01 00 74 15 48 89 df 57 9d <0f> 1f 44 00 00 eb ad e8 7c 52 4b f9 eb bb 0f 0b 0f 0b e8 8a 54 0e RSP: 0018:ffff880181eaf508 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: dffffc0000000000 RBX: 0000000000000282 RCX: 0000000000000000 RDX: 1ffffffff1263ec6 RSI: 0000000000000001 RDI: 0000000000000282 RBP: ffff880181eaf518 R08: ffff88018378a140 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffff8801c0b94ac0 R13: ffff8801c0b94300 R14: ffff8801daf2cc40 R15: 0000000000000001 try_to_wake_up+0x10a/0x12e0 kernel/sched/core.c:2056 wake_up_process kernel/sched/core.c:2124 [inline] wake_up_q+0xa4/0x100 kernel/sched/core.c:440 futex_wake+0x61f/0x760 kernel/futex.c:1556 do_futex+0x2e4/0x26d0 kernel/futex.c:3533 __do_sys_futex kernel/futex.c:3589 [inline] __se_sys_futex kernel/futex.c:3557 [inline] __x64_sys_futex+0x472/0x6a0 kernel/futex.c:3557 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457569 Code: fd b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:0000000000a3fbf8 EFLAGS: 00000246 ORIG_RAX: 00000000000000ca RAX: ffffffffffffffda RBX: 000000000072bfa0 RCX: 0000000000457569 RDX: 0000000000000000 RSI: 0000000000000081 RDI: 000000000072bfa8 RBP: 00000000000003b7 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000a3fcb0 R11: 0000000000000246 R12: 000000000072bfac R13: 000000000072bfac R14: 0000000000000003 R15: 0000000000000001 Allocated by task 24073: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] kasan_kmalloc+0xc7/0xe0 mm/kasan/kasan.c:553 kmem_cache_alloc_trace+0x152/0x750 mm/slab.c:3620 kmalloc include/linux/slab.h:513 [inline] kzalloc include/linux/slab.h:707 [inline] queue_new sound/core/seq/seq_queue.c:113 [inline] snd_seq_queue_alloc+0xab/0x5b0 sound/core/seq/seq_queue.c:185 snd_seq_ioctl_create_queue+0xb0/0x320 sound/core/seq/seq_clientmgr.c:1522 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:501 [inline] do_vfs_ioctl+0x1de/0x1720 fs/ioctl.c:685 ksys_ioctl+0xa9/0xd0 fs/ioctl.c:702 __do_sys_ioctl fs/ioctl.c:709 [inline] __se_sys_ioctl fs/ioctl.c:707 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:707 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe Freed by task 24072: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] __kasan_slab_free+0x102/0x150 mm/kasan/kasan.c:521 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:528 __cache_free mm/slab.c:3498 [inline] kfree+0xcf/0x230 mm/slab.c:3813 queue_delete+0x90/0xb0 sound/core/seq/seq_queue.c:156 snd_seq_queue_client_leave+0x36/0x170 sound/core/seq/seq_queue.c:584 seq_free_client1.part.9+0xf4/0x270 sound/core/seq/seq_clientmgr.c:263 seq_free_client1 sound/core/seq/seq_clientmgr.c:256 [inline] seq_free_client+0x82/0x100 sound/core/seq/seq_clientmgr.c:285 snd_seq_release+0x51/0xe0 sound/core/seq/seq_clientmgr.c:366 __fput+0x385/0xa30 fs/file_table.c:278 ____fput+0x15/0x20 fs/file_table.c:309 task_work_run+0x1e8/0x2a0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:193 [inline] exit_to_usermode_loop+0x318/0x380 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_64+0x6be/0x820 arch/x86/entry/common.c:293 entry_SYSCALL_64_after_hwframe+0x49/0xbe The buggy address belongs to the object at ffff8801cf71c0c0 which belongs to the cache kmalloc-512 of size 512 The buggy address is located 88 bytes inside of 512-byte region [ffff8801cf71c0c0, ffff8801cf71c2c0) The buggy address belongs to the page: page:ffffea00073dc700 count:1 mapcount:0 mapping:ffff8801da800940 index:0x0 flags: 0x2fffc0000000100(slab) raw: 02fffc0000000100 ffffea00073a4fc8 ffffea000761f4c8 ffff8801da800940 raw: 0000000000000000 ffff8801cf71c0c0 0000000100000006 0000000000000000 kobject: 'loop5' (00000000172d3417): kobject_uevent_env page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8801cf71c000: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff8801cf71c080: fc fc fc fc fc fc fc fc fb fb fb fb fb fb fb fb > ffff8801cf71c100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8801cf71c180: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8801cf71c200: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- 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#bug-status-tracking for how to communicate with syzbot.

1 0

[alsa-devel] [PATCH v4 00/14] ASoC: Add support to WCD9335 Audio Codec
by srinivas.kandagatla＠linaro.org 24 Oct '18

24 Oct '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thankyou for reviewing v3 patchset, here is v4 addressing comments from v3. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v3 (https://lkml.org/lkml/2018/9/4/318/) - Updated mbhc dt bindings according to Robs Comments. - Added a cleanup patch to common up fixups. - Addressed various comments on mfd driver by Lee Jones. - Added acks on bindings and irq support. - fixed module loading Srinivas Kandagatla (14): ASoC: dt-bindings: update wcd9335 bindings. mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: dt-bindings: Add WCD9335 MBHC specific properties ASoC: wcd9335: add mbhc support ASoC: apq8096: add slim support ASoC: apq8096: add headset JACK support ASoC: qcom: common: move be_hw_fixup to common .../devicetree/bindings/sound/qcom,wcd9335.txt | 30 +- drivers/mfd/Kconfig | 13 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 394 ++ include/linux/mfd/wcd9335/registers.h | 640 +++ include/linux/mfd/wcd9335/wcd9335.h | 81 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 577 +++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5241 ++++++++++++++++++++ sound/soc/qcom/apq8096.c | 128 +- sound/soc/qcom/common.c | 17 + sound/soc/qcom/sdm845.c | 22 +- 14 files changed, 7168 insertions(+), 35 deletions(-) create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.9.3

5 21

[alsa-devel] [asoc:for-4.20 519/528] sound/soc/stm/stm32_sai_sub.c:393:26-32: ERROR: application of sizeof to pointer
by kbuild test robot 24 Oct '18

24 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 6be0f96d799f487f05eb412d362d5a1747d665c2 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/528] ASoC: stm32: sai: set sai as mclk clock provider coccinelle warnings: (new ones prefixed by >>) >> sound/soc/stm/stm32_sai_sub.c:393:26-32: ERROR: application of sizeof to pointer -- >> sound/soc/stm/stm32_sai_sub.c:343:5-8: WARNING: Unsigned expression compared with zero: div < 0 Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] Audio miniconference schedule
by Mark Brown 24 Oct '18

24 Oct '18

As previously announced there's an audio miniconference happening on the 21st in Edinburgh: https://blog.sirena.org.uk/2018/09/19/2018-linux-audio-miniconfernce (yes there is a typo in the URL). A number of people have proposed topics so I just went through them and pulled them into a proposed agenda: - Virtualization (Liam) - SoundWire (Liam) - Byte controls (Vinod) - GObject introspection for language bindings (Sakamoto-san) - Integrating the ALSA control core (Sakamoto-san) - AGL (Stephane/Ronan) - Y2038 (broonie) - AVB (Liam) - DSPs (Liam) - Sound Open Firmware What do people think? Any other topics? There's still time to sign up if you're interested in attending, you can sign up right up to the day of the event. It also seems good to mention that there's another audio event going on around ELC: https://blogs.gnome.org/uraeus/2018/09/24/getting-the-team-together-to-revo… which is a hackathon for a new userspace stack called PipeWire - if we have someone attending who knows about that I think it'd be good to talk about it too.

8 24

[alsa-devel] [PATCH - Intervals v2 1/1] interval: Interpret (x x+1] correctly and return x+1
by twischer＠de.adit-jv.com 24 Oct '18

24 Oct '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this change an interval of (x x+1] will be interpreted as an empty interval but the right value would be x+1. This leads to a failing snd_pcm_hw_params() call which returns -EINVAL. An example issue log is given in the following: snd_pcm_hw_params failed with err -22 (Invalid argument) ACCESS: MMAP_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 16 CHANNELS: 1 RATE: 16000 PERIOD_TIME: (15999 16000] PERIOD_SIZE: (255 256] PERIOD_BYTES: (510 512] PERIODS: [2 3) BUFFER_TIME: 32000 BUFFER_SIZE: 512 BUFFER_BYTES: 1024 In case of (x x+1) we have to interpret it anyway as a single value of x to compensate rounding issues. For example the period size will result in an interval of (352 353) when the period time is 16ms and the sample rate 22050 Hz (16ms * 22,05 kHz = 352,8 frames). But 352 has to be chosen to allow a buffer size of 705 (32ms * 22,05 kHz = 705,6 frames) which has to be >= 2x period size to avoid Xruns. The buffer size will not end up with an interval of (705 706) simular to the period size because snd_pcm_rate_hw_refine_cchange() calls snd_interval_floor() for the buffer size. Therefore this value will be interpreted as an integer interval instead of a real interval further on. This issue seems to exist since the change of 9bb985c38 ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- > On 10/18/18 12:57, Takashi Iwai wrote: > This change looks risky. The snd_interval_value() might be called > even if the interval isn't reduced to a single value. Rather check > openmin instead. If I would do "if (i->openmin)" on an interval of (x x+1) x+1 would be returned. But this would result in the second issue which I have tried to describe in the commit message: x has to be returned otherwise it is not guaranteed that buffer_size >= 2x period_size. And this would result in continuously Xruns. Therefore I would like to prefer "if (i->openmin && !i->openmax)" diff --git a/src/pcm/interval_inline.h b/src/pcm/interval_inline.h index a68e292..d9a30b2 100644 --- a/src/pcm/interval_inline.h +++ b/src/pcm/interval_inline.h @@ -51,12 +51,14 @@ INTERVAL_INLINE int snd_interval_single(const snd_interval_t *i) { assert(!snd_interval_empty(i)); return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } INTERVAL_INLINE int snd_interval_value(const snd_interval_t *i) { assert(snd_interval_single(i)); + if (i->openmin && !i->openmax) + return i->max; return i->min; } -- 2.7.4

5 15

Re: [alsa-devel] GFP_ATOMIC in Intel code
by Takashi Iwai 24 Oct '18

24 Oct '18

On Tue, 23 Oct 2018 10:49:27 +0200, Vinod wrote: > > On 22-10-18, 12:22, Pierre-Louis Bossart wrote: > > Looking at the latest for-next branch, by accident I saw a number of > > GFP_ATOMIC references. > > > > With my extended break I don't remember much but didn't we agree that none > > of those were needed? If yes I'll send a patch to fix all this. > > > I thought that someone had fixed up all these instances, Looking back at the git history, "someone" was mostly you :) A few of them below seem to be new codes that copied the bad past examples, some are overlooked one, and some are mailbox codes that aren't 100% sure to safe to replace (although I guess it is). > looks like few > missed, so yes please do fix them up and Welcome back :-) I hope you enjoyed the time-off, Pierre. thanks, Takashi > > > Thanks > > > > -Pierre > > > > > > /data/pbossart/ktest/broonie-next/sound/soc/intel$ git grep ATOMIC > > atom/sst/sst_pvt.c: msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); > > atom/sst/sst_pvt.c: msg->mailbox_data = > > kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); > > boards/cht_bsw_rt5672.c: drv = devm_kzalloc(&pdev->dev, sizeof(*drv), > > GFP_ATOMIC); > > boards/glk_rt5682_max98357a.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > boards/kbl_da7219_max98927.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > boards/skl_hda_dsp_generic.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > -- > ~Vinod >

2 1

[alsa-devel] [PATCH - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 23 Oct '18

23 Oct '18

Hi all, The AAF plugin RFC series v2 has been around on alsa-devel ML for more than two weeks so I'm moving forward and sending it as PATCH series now. The first 5 patches from this series are pretty much the same from the previous version. The news are the top 2 patches which add more functionality to the plugin. Patch 6/7 implements the dump() callback which has been useful for debugging purposes and Patch 7/7 adds support for direct read/write transfers. For general context information about the AAF plugin, I'm quoting below the cover letter from the RFC series v1: "This patchset introduces the AAF plugin to the alsa-plugins project which enables TSN audio applications on Linux. For those not familiarized with TSN, it stands for Time-Sensitive Networking, and it's a set of IEEE technologies (enhancements to IEEE 802.1Q and Ethernet protocols) that provide precise time synchronization, bounded latency and application interoperability to bridged networks. Those technologies enable time-sensitive applications such as audio, video and control loops to run on top of bridged networks, co-existing with regular applications. TSN technologies are a super set of Audio Video Bridging (AVB) technologies also developed by IEEE. AVTP is the protocol defined to transport data in a TSN system, and AAF is one of the formats defined by AVTP to encapsulate audio data. AAF is specified in Chapter 7 from the AVTP spec [1]. This work is part of the effort to enable TSN technologies on upstream Linux ecosystem. All building-blocks required to enable TSN audio applications have been already developed and pushed to upstream projects. Time synchronization features are provided by linuxptp project [2], bounded latency features are provided by Linux Traffic Control subsystem since kernel version 4.15 [3], and AVTP packetization is provided by libavtp [4]. What is currently missing in the ecosystem to enable TSN audio applications is a piece of software that plumbs it all together and interfaces with Linux Audio system properly. That's the point of the AAF plugin introduced here. The AAF plugin is a PCM plugin that uses AVTP protocol to transmit / receive audio data through a TSN capable network. When operating in playback mode, the plugin reads PCM samples from the audio buffer, encapsulates into AVTP packets and transmits to the network, mimicking a typical AVTP Talker. When operating in capture mode, the plugin receives AVTP packets from the network, retrieves the PCM samples, and present them to alsa-lib layer at the presentation time, mimicking a typical AVTP Listener." For further information about what has been previously discussed, please refer to RFC series v1 and v2 archives in [5] and [6]. Finally, all versions of this series can be also found in my alsa-plugins tree in github [7]. Best regards, Andre [1] 1722-2016 - IEEE Standard for a Transport Protocol for Time-Sensitive Applications in Bridged Local Area Networks [2] http://linuxptp.sourceforge.net [3] https://patchwork.ozlabs.org/cover/826678/ [4] https://github.com/AVnu/libavtp [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.html [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… [7] https://github.com/aguedes/alsa-plugins Andre Guedes (7): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support aaf: Implement dump() ioplug callback aaf: Add support for direct read/write transfers Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1197 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 142 +++++++ 5 files changed, 1360 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

5 20

[alsa-devel] [GIT PULL] ASoC fixes for the next release
by Mark Brown 23 Oct '18

23 Oct '18

The following changes since commit 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399: ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks (2018-10-21 16:46:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.0-2 for you to fetch changes up to 6be0f96d799f487f05eb412d362d5a1747d665c2: ASoC: stm32: sai: fix master clock naming (2018-10-23 10:51:45 +0100) ---------------------------------------------------------------- ASoC: Additional fixes for the next release A couple of fixes for build bot issues in the STM32 SAI driver. ---------------------------------------------------------------- Olivier Moysan (2): ASoC: stm32: add clock dependency for sai ASoC: stm32: sai: fix master clock naming sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 5 ++--- 2 files changed, 3 insertions(+), 3 deletions(-)

2 1

[alsa-devel] Segfault in pulseaudio since Alsa 1.1.7
by Hyperion 23 Oct '18

23 Oct '18

Hi, Â Since the upgrade to Alsa 1.1.7, I get a segfault at pulseaudio startup. This have been discussed and documented here : [1]https://www.linuxquestions.org/questions/slackware-14/current-fri-oc t-19-puseaudio-segfauts-4175640773/ Â It seems that I found the origin of the bug : reverting these 2 commits fixes the problem : [2]http://git.alsa-project.org/?p=alsa-...792dc809c908cc [3]http://git.alsa-project.org/?p=alsa-...5dc6e1594291e7 Then no more segfault in pulseaudio when accessing the USB soundcard. Â Please could you fix it ? Â Regards jp References Visible links 1. https://www.linuxquestions.org/questions/slackware-14/current-fri-oct-19-pu… 2. http://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff;h=fc3f5d51062f… 3. http://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff;h=6b996865126d… Hidden links: 4. https://www.linuxquestions.org/questions/slackware-14/current-fri-oct-19-pu…

2 6

[alsa-devel] [alsa-plugins-1.17] Regression in JACK plugin
by tet 23 Oct '18

23 Oct '18

Hi, after updating to alsa-plugins-1.17 ALSA throws: "ALSA lib pcm_plug.c:1273:(_snd_pcm_plug_open) Unknown field playback_ports My ~/.asourdrc is working in 1.16 and it does not look like the syntax changed for the plugin when looking at the readme. pcm.jack { type plug slave.pcm { type jack capture_ports { 0 system:capture_1 1 system:capture_1 } playback_ports { 0 system:playback_1 1 system:playback_2 } } hint.description "JACK Audio Kit" } pcm.!default { @func getenv vars [ PCM ] default jack } Any ideas?

2 6

[alsa-devel] [RFC PATCH 0/4] Initial prototype of DAPM sample rate domains
by Charles Keepax 23 Oct '18

23 Oct '18

Here is a first pass at adding dapm sample rate domain support. Things are still pretty rough in quite a few places and I expect some major refactoring before they are ready to merge, but things are far enough along for people to look at the approach I am taking to the problem. And should provide enough for discussion at the mini-conference, although hopefully I will get some more done next week as well. The basic high level summary is two new concepts are added to ASoC, one being a sample rate domain and the other being a domain group. The domain groups represent widgets that are limited by hardware restrictions to always run at the same sample rate as each other. Sample rate domains represent a slot on a device for a sample rate. Both of these are currently limited to be within the scope of a single component. The core will track which groups are connected together, then as a group is powered up it will be allocated a domain based on the other groups it is connected to. Domain groups are implemented as a new type of DAPM widget and the domains themselves are a new thing which devices add through their component drivers. As we have previously discussed my intention would be that we also add bridges between domains although I haven't started work on that yet. The current code is limited to within a single component and I think that the bridges will allow better support for propagating rates between components as well as obviously for SRCs. Currently the SRCs on wm5110 are just being handled as points where the domain groups don't connect to another peer. This allows them to function but doesn't accurately convey things like the limitations which input rates support which output rates on the ISRC. Thanks, Charles Charles Keepax (4): ASoC: dapm: Add support for hw_free on CODEC to CODEC links ASoC: dapm: Add support for a rate domain widget ASoC: domain: Add sample rate domain support ASoC: arizona: Add rate domain support include/sound/soc-dapm.h | 14 ++ include/sound/soc-domain.h | 98 +++++++++++ include/sound/soc.h | 8 + sound/soc/Makefile | 2 +- sound/soc/codecs/arizona.c | 255 ++++++++++++++++++++++------ sound/soc/codecs/arizona.h | 76 ++++++++- sound/soc/codecs/wm5110.c | 414 ++++++++++++++++++++++++++++++++++----------- sound/soc/soc-core.c | 8 + sound/soc/soc-dapm.c | 57 ++++++- sound/soc/soc-domain.c | 412 ++++++++++++++++++++++++++++++++++++++++++++ 10 files changed, 1183 insertions(+), 161 deletions(-) create mode 100644 include/sound/soc-domain.h create mode 100644 sound/soc/soc-domain.c -- 2.11.0

3 8

[alsa-devel] [asoc:for-4.20 333/424] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 23 Oct '18

23 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 1c8bc7b3de5e76cb89aacdc7be1475a028af505f commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/424] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

4 7

[alsa-devel] [PATCH 0/2] ASoC: stm32: sai: fix kbuild errors
by Olivier Moysan 23 Oct '18

23 Oct '18

These patches fix following compilation errors: - >> sound/soc/stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type - >> sound/soc/stm/stm32_sai_sub.c:412:3: warning: 'strncat' specified bound 6 equals source length [-Wstringop-overflow=] Olivier Moysan (2): ASoC: stm32: add clock dependency for sai ASoC: stm32: sai: fix master clock naming sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 5 ++--- 2 files changed, 3 insertions(+), 3 deletions(-) -- 2.7.4

2 4

[alsa-devel] [asoc:for-4.20 519/526] sound/soc/stm/stm32_sai_sub.c:343 stm32_sai_mclk_set_rate() warn: unsigned 'div' is never less than zero.
by kbuild test robot 23 Oct '18

23 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/526] ASoC: stm32: sai: set sai as mclk clock provider smatch warnings: sound/soc/stm/stm32_sai_sub.c:343 stm32_sai_mclk_set_rate() warn: unsigned 'div' is never less than zero. vim +/div +343 sound/soc/stm/stm32_sai_sub.c 333 334 static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, 335 unsigned long parent_rate) 336 { 337 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 338 struct stm32_sai_sub_data *sai = mclk->sai_data; 339 unsigned int div; 340 int ret; 341 342 div = stm32_sai_get_clk_div(sai, parent_rate, rate); > 343 if (div < 0) 344 return div; 345 346 ret = stm32_sai_set_clk_div(sai, div); 347 if (ret) 348 return ret; 349 350 mclk->freq = rate; 351 352 return 0; 353 } 354 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [GIT PULL] ASoC updates for v5.0
by Mark Brown 23 Oct '18

23 Oct '18

The following changes since commit 35a7f35ad1b150ddf59a41dcac7b2fa32982be0e: Linux 4.19-rc8 (2018-10-15 07:20:24 +0200) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.0 for you to fetch changes up to 65dfb6d6dd2850f3f99f08536d2b14190350c854: Merge branch 'asoc-4.20' into asoc-next (2018-10-21 16:59:23 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.0/v4.20 As ever there's a lot of small and driver specific changes going on here, but we do also have some relatively large changes in the core thanks to the hard work of Charles and Morimoto-san: - More component transitions from Morimoto-san, I think we're about finished with this. Thanks for all the hard work! - Morimoto-san also added a bunch of for_each_foo macros - A bunch of cleanups and fixes for DAPM from Charles. - MCLK support for several different devices, including CS42L51, STM32 SAI, and MAX98373. - Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and TI PCM3060. ---------------------------------------------------------------- Akshu Agrawal (5): ASoC: AMD: Set constraints for DMIC and MAX98357a codec ASoC: AMD: Change MCLK to 48Mhz ASoC: AMD: Fix simultaneous playback and capture on different channel ASoC: AMD: Fix capture unstable in beginning for some runs ASoC: AMD: Add SND_JACK_LINEOUT jack type Alexandru Ardelean (1): ASoC: dt-bindings: add dt bindings for adau1977 audio codec Andreas Dannenberg (2): ASoC: codecs: tas5720: add TAS5722 specific volume control ASoC: codecs: tas5720: add TAS5722 TDM slot width setting support Andreas Färber (3): ASoC: dt-bindings: add max98088 audio codec ASoC: max98088: add OF support ASoC: max98088: Add master clock handling Andrew F. Davis (4): ASoC: tas6424: Print full register name in error message ASoC: tas6424: Add channel fault reporting ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support Arnd Bergmann (4): ASoC: pxa: avoid AC97_BUS build warning ASoC: atmel: add SND_SOC_I2C_AND_SPI dependency ASoC: wm9712: fix unused variable warning ASoC: max98988: add I2C dependency Biju Das (1): ASoC: rsnd: Add r8a7744 support Charles Keepax (11): ASoC: dapm: Remove clock framework ifdefs ASoC: dapm: Don't fail creating new DAPM control on NULL pinctrl ASoC: dpcm: Properly initialise hw->rate_max ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control ASoC: dapm: Move connection of CODEC to CODEC DAIs ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime ASoC: dapm: Avoid uninitialised variable warning ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai ASoC: dapm: Add support for hw_free on CODEC to CODEC links Christoph Hellwig (1): ASoC: intel: don't pass GFP_DMA32 to dma_alloc_coherent Codrin Ciubotariu (2): ASoC: Add driver for PROTO Audio CODEC (with a WM8731) ASoC: mikroe-proto: dt-bindings: add DT bindings for PROTO board Colin Ian King (2): ASoC: hdac_hdmi: remove redundant check for !port condition ASoC: sgtl5000: avoid division by zero if lo_vag is zero Dan Carpenter (4): ASoC: dapm: Fix a couple uninitialized ret variables ASoC: qdsp6: q6asm-dai: clean up a return ASoC: qdsp6: q6asm-dai: checking NULL vs IS_ERR() ASoC: Intel: kbl_da7219_max98927: minor white space clean up Daniel Mack (6): ASoC: pxa-ssp: enable and disable extclk if given ASoC: add fault detect recovery property to DT bindings ASoC: Add device tree documentation file for wm8782 stereo DAC ASoC: wm8782: add support for regulators ASoC: sta32x: set ->component pointer in private struct ASoC: sta32x: Add support for XTI clock Danny Smith (1): ASoC: adau17x1: Implemented safeload support David Lin (1): ASoC: nau8822: new codec driver Dmytro Prokopchuk (1): ASoC: rsnd: fixup SSI clock during suspend/resume modes Dragos Tarcatu (3): ASoC: rsnd: add support for 16/24 bit slot widths ASoC: rsnd: add support for the DSP_A/DSP_B formats ASoC: rsnd: add support for 8 bit S8 format Fabrizio Castro (1): ASoC: rsnd: Add r8a774a1 support Grant Grundler (1): ASoC: max98373: usleep_range() needs include/delay.h Gustavo A. R. Silva (2): ASoC: es8328: Fix fall-through annotations ASoC: hisilicon: fix fall-through annotations Hans de Goede (3): ASoC: rt5670: Add quirk for Thinkpad 8 tablet ASoC: Intel: common: Add quirk for Thinkpad 8 tablet ASoC: Intel: cht-bsw-rt5672: Add key-mappings for the headset buttons Hiroyuki Yokoyama (1): ASoC: rsnd: Add device tree binding for r8a77990 Jerome Brunet (7): ASoC: meson: imply clock and reset controllers ASoC: meson: axg-fifo: report interrupt request failure ASoC: meson: axg-tdm: restrict formats depending on slot width ASoC: dmic: add Kconfig prompt for the generic dmic codec. ASoC: dmic: add DT module alias ASoC: meson: add axg pdm input DT binding documentation ASoC: meson: add axg pdm input Jiada Wang (8): ASoC: rsnd: add warning message to rsnd_kctrl_accept_runtime() ASoC: rsnd: remove is_play parameter from hw_rule function ASoC: rsnd: ssi: Fix issue in dma data address assignment ASoC: rsnd: ssi: Check runtime channel number rather than hw_params ASoc: rsnd: dma: Calculate dma address with consider of BUSIF ASoc: rsnd: dma: Calculate PDMACHCRE with consider of BUSIF ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0 ASoC: rsnd: ssiu: Support to init different BUSIF instance Jon Hunter (1): ASoC: core: Don't schedule DAPM work if already in target state Katsuhiro Suzuki (1): ASoC: rockchip: add missing INTERLEAVED PCM attribute Kirill Marinushkin (3): ASoC: pcm3060: Add codec driver ASoC: pcm3060: Improve stylistics of file comments ASoC: pcm3060: Improve legibility of if-statements Kuninori Morimoto (55): ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec ASoC: simple-card: support snd_soc_dai_link_component style for codec ASoC: simple-scu-card: use simple_dai_props ASoC: simple-scu-card: support snd_soc_dai_link_component style for codec ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec ASoC: audio-graph-scu-card: use simple_dai_props ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for codec ASoC: simple-card-util: remove dai_link compatible code for codec ASoC: soc-core: use snd_soc_dai_link_component for platform ASoC: simple-card-util: support snd_soc_dai_link_component style for platform ASoC: simple-card: support snd_soc_dai_link_component style for platform ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform ASoC: simple-card-util: remove dai_link compatible code for platform ASoC: rsnd: merge .nolock_start and .prepare ASoC: add for_each_link_codecs() macro ASoC: add for_each_rtd_codec_dai() macro ASoC: rsnd: gen: use tab instead of white-space ASoC: rsnd: don't use %p for dev_dbg() ASoC: atmel_ssc_dai: use devm_snd_soc_register_component() ASoC: bcm: use devm_snd_soc_register_component() ASoC: hdac_hda: use devm_snd_soc_register_component() ASoC: rt5668: use devm_snd_soc_register_component() ASoC: tscs454: use devm_snd_soc_register_component() ASoC: nuc900: use devm_snd_soc_register_component() ASoC: omap: use devm_snd_soc_register_component() ASoC: sh: use devm_snd_soc_register_component() ASoC: txx9: use devm_snd_soc_register_component() ASoC: rt5668: remove empty rt5668_i2c_remove() ASoC: soc-core: avoid nested code on soc_remove_dai() ASoC: soc-core: remove unused num_dai_links ASoC: soc-core: remove dai->driver NULL check ASoC: soc-core: manage platform name under snd_soc_init_platform() ASoC: soc-core: add snd_soc_is_matching_component() ASoC: convert for_each_rtd_codec_dai() for missing part ASoC: rename for_each_rtd_codec_dai_reverse to rollback ASoC: add for_each_card_prelinks() macro ASoC: add for_each_card_links() macro ASoC: add for_each_card_rtds() macro ASoC: add for_each_card_components() macro ASoC: add for_each_comp_order() macro ASoC: add for_each_dpcm_fe() macro ASoC: add for_each_dpcm_be() macro ASoC: tidyup for_each_card_prelinks() dai_link ASoC: add for_each_component() macro ASoC: add for_each_component_dais() macro ASoC: rsnd: use 32bit TDM width as default ASoC: audio-graph-card: enable mclk-fs on codec node ASoC: pcm3168a: add HW constraint for non RIGHT_J ASoC: pcm3168a: add hw constraint for channel ASoC: pcm3168a: add I2S/Left_J TDM support ASoC: pcm3168a: add hw constraint for capture channel ASoC: rsnd: enable TDM settings for SSI parent ASoC: rsnd: tidyup SSICR::SWSP for TDM Linus Walleij (1): ASoC: rt5677-spi: Drop unused GPIO include Lucas Tanure (1): ASoC: wm2000: Remove wm2000_read helper function Mac Chiang (2): ASoC: Intel: Boards: Add KBL Dialog Maxim I2S machine driver ASoC: Intel: common: Add Kabylake Dialog+Maxim machine driver entry Marcel Ziswiler (2): ASoC: tegra_sgtl5000: fix device_node refcounting ASoC: soc-core: fix trivial checkpatch issues Marco Felsch (2): ASoC: dt-bindings: max98088: add external clock binding ASoC: max98988: make it selectable Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Mark Brown (10): Merge tag 'v4.19-rc1' into asoc-4.19 Merge tag 'hda-codec-h-move' into asoc-4.20 Merge branch 'asoc-4.19' into asoc-4.20 tas dependency Merge branch 'asoc-4.19' into asoc-4.20 for rcar dep Merge branch 'asoc-4.19' into asoc-4.20 Cirrus conflict ALSA: hda: Fix implicit definition of pci_iomap() on SH Merge branch 'asoc-4.19' into asoc-4.20 for rt5682 deps. Merge remote-tracking branch 'asoc/for-4.19' into asoc-4.20 Merge branch 'asoc-4.19' into asoc-linus Merge branch 'asoc-4.20' into asoc-next Matt Flax (4): ASoC: cs4265: SOC_SINGLE register value error fix ASoC: cs4265: Add native 32bit I2S transport ASoC: cs4265: Add a S/PDIF enable switch ASoC: cs4265: Add a MIC pre. route Matthias Kaehlcke (1): ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_dma_ops Oder Chiou (2): ASoC: rt5514-spi: Get the period_bytes in the copy work to make sure the value correctly ASoC: rt5663: Remove the boost volume in the beginning of playback Olivier Moysan (5): ASoC: dt-bindings: add mclk provider support to stm32 sai ASoC: dt-bindings: add mclk support to cs42l51 ASoC: stm32: sai: set sai as mclk clock provider ASoC: cs42l51: add mclk support ASoC: cs42l51: fix mclk support Peter Rosin (1): ASoC: atmel: tse850: switch to SPDX license identifier Peter Ujfalusi (1): ASoC: davinci-mcasp: Add support for FIFO usage caused delay reporting Pierre-Louis Bossart (3): ALSA: hda: move hda_codec.h to include/sound ASoC: Intel: common: add table for HDA-based platforms ALSA: hda: move hda_codec.h to include/sound Rakesh Ughreja (5): ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers Randy Dunlap (1): ASoC: fix soc-core.c kernel-doc warning Richard Fitzgerald (2): ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate Rob Herring (1): ASoC: Convert to using %pOFn instead of device_node.name Robert Jarzmik (1): ASoC: pxa: switch to new ac97 bus support Robert Rosengren (1): ASoC: adau17x1: Unused exported functions changed to internal Rohit kumar (2): ASoC: Fix UBSAN warning at snd_soc_get/put_volsw_sx() ASoC: q6afe: dt-bindings: Update input range for qcom,sd-lines Ryan Lee (4): ASoC: max98373: Sort Digital Volume in reverse order ASoC: max98373: Sort BDE Limiter Thresh Volume in reverse order ASoC: max98373: Sort max98373_bde_gain_tlv in reverse order ASoC: max98373: Sort DHT Rot Pnt Volume in reverse order Shuming Fan (6): ASoC: rt5682: Update calibration function ASoC: rt5682: Improve HP performance ASoC: rt5682: Remove HP volume control ASoC: rt5682: Update calibration function ASoC: rt5682: Fix the boost volume at the begining of playback ASoC: rt5682: Minor code modification Srinivas Kandagatla (1): ASoC: core: add support to card rebind Takashi Iwai (6): ASoC: intel: skylake: Add missing break in skl_tplg_get_token() ASoC: adau1761: Use the standard fall-through annotation ASoC: pcm186x: Use the standard fall-through annotation ASoC: rt274: Add fall-through annotations ASoC: intel: skylake: Add fall-through annotation ASoC: topology: Use the standard fall-through annotations Tzung-Bi Shih (1): ASoC: ts3a227e: fix a typo in documentation Vasily Khoruzhick (6): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks Yong Zhi (1): ASoC: Intel: hdac_hdmi: Limit sampling rates at dai creation YueHaibing (2): ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c ASoC: qcom: qdsp6: remove duplicated include zhong jiang (5): ASoC: q6core: Use kmemdup to replace kzalloc + memcpy ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy ASoC: remove unneeded static set .owner field in platform_driver ASoC: remove redundant include ASoC: qcom: qdsp6: remove duplicated include from q6adm.c .../devicetree/bindings/sound/adi,adau1977.txt | 54 + .../devicetree/bindings/sound/amlogic,axg-pdm.txt | 24 + .../devicetree/bindings/sound/cs42l51.txt | 17 + .../devicetree/bindings/sound/maxim,max98088.txt | 23 + .../bindings/sound/mikroe,mikroe-proto.txt | 23 + .../devicetree/bindings/sound/nau8822.txt | 16 + .../devicetree/bindings/sound/pcm3060.txt | 17 + .../devicetree/bindings/sound/qcom,q6afe.txt | 18 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 5 +- .../devicetree/bindings/sound/st,sta32x.txt | 9 + .../devicetree/bindings/sound/st,stm32-sai.txt | 7 + .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../devicetree/bindings/sound/ts3a227e.txt | 2 +- Documentation/devicetree/bindings/sound/wm8782.txt | 17 + .../devicetree/bindings/trivial-devices.txt | 1 - .../devicetree/bindings/vendor-prefixes.txt | 1 + MAINTAINERS | 7 + {sound/pci/hda => include/sound}/hda_codec.h | 0 include/sound/simple_card_utils.h | 27 +- include/sound/soc-acpi-intel-match.h | 6 + include/sound/soc-dapm.h | 9 - include/sound/soc-dpcm.h | 10 + include/sound/soc.h | 45 +- sound/arm/Kconfig | 1 - sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/hda_bind.c | 14 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_ca0132.c | 3 +- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 2 +- sound/soc/amd/acp-da7219-max98357a.c | 77 +- sound/soc/amd/acp-pcm-dma.c | 30 +- sound/soc/amd/acp.h | 3 +- sound/soc/atmel/Kconfig | 12 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/mikroe-proto.c | 165 +++ sound/soc/atmel/tse850-pcm5142.c | 78 +- sound/soc/bcm/cygnus-ssp.c | 13 +- sound/soc/codecs/Kconfig | 36 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/adau1761.c | 3 +- sound/soc/codecs/adau17x1.c | 86 +- sound/soc/codecs/adau17x1.h | 4 - sound/soc/codecs/cs4265.c | 12 +- sound/soc/codecs/cs42l51.c | 21 + sound/soc/codecs/dmic.c | 1 + sound/soc/codecs/es8328.c | 4 +- sound/soc/codecs/hdac_hda.c | 483 +++++++++ sound/soc/codecs/hdac_hda.h | 24 + sound/soc/codecs/hdac_hdmi.c | 11 +- sound/soc/codecs/max98088.c | 36 + sound/soc/codecs/max98373.c | 47 +- sound/soc/codecs/nau8822.c | 1136 ++++++++++++++++++++ sound/soc/codecs/nau8822.h | 204 ++++ sound/soc/codecs/pcm186x.c | 3 +- sound/soc/codecs/pcm3060-i2c.c | 60 ++ sound/soc/codecs/pcm3060-spi.c | 59 + sound/soc/codecs/pcm3060.c | 295 +++++ sound/soc/codecs/pcm3060.h | 88 ++ sound/soc/codecs/pcm3168a.c | 82 ++ sound/soc/codecs/rt274.c | 2 + sound/soc/codecs/rt5514-spi.c | 14 +- sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5663.c | 7 +- sound/soc/codecs/rt5668.c | 10 +- sound/soc/codecs/rt5670.c | 12 + sound/soc/codecs/rt5677-spi.c | 1 - sound/soc/codecs/rt5682.c | 86 +- sound/soc/codecs/rt5682.h | 14 + sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sta32x.c | 30 + sound/soc/codecs/tas5720.c | 103 +- sound/soc/codecs/tas6424.c | 58 +- sound/soc/codecs/tas6424.h | 10 + sound/soc/codecs/tlv320aic31xx.c | 85 ++ sound/soc/codecs/tlv320aic31xx.h | 23 + sound/soc/codecs/tscs454.c | 2 +- sound/soc/codecs/wm2000.c | 54 +- sound/soc/codecs/wm8782.c | 63 ++ sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm9712.c | 3 +- sound/soc/codecs/wm_adsp.c | 26 +- sound/soc/davinci/davinci-mcasp.c | 37 + sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_utils.c | 4 +- sound/soc/fsl/pcm030-audio-fabric.c | 5 +- sound/soc/generic/audio-graph-card.c | 21 +- sound/soc/generic/audio-graph-scu-card.c | 55 +- sound/soc/generic/simple-card-utils.c | 53 +- sound/soc/generic/simple-card.c | 30 +- sound/soc/generic/simple-scu-card.c | 54 +- sound/soc/hisilicon/hi6210-i2s.c | 4 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +- sound/soc/intel/boards/Kconfig | 22 + sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 9 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 983 +++++++++++++++++ sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 127 +++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 183 ++++ sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 7 + sound/soc/intel/common/soc-acpi-intel-hda-match.c | 40 + sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 13 + sound/soc/intel/common/sst-firmware.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 71 +- sound/soc/intel/skylake/skl-topology.c | 4 +- sound/soc/intel/skylake/skl.c | 96 +- sound/soc/intel/skylake/skl.h | 12 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 13 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 14 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 14 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 12 +- sound/soc/meson/Kconfig | 13 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 16 +- sound/soc/meson/axg-fifo.c | 2 + sound/soc/meson/axg-pdm.c | 654 +++++++++++ sound/soc/meson/axg-tdm-interface.c | 50 +- sound/soc/nuc900/nuc900-ac97.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 4 +- sound/soc/pxa/Kconfig | 13 +- sound/soc/pxa/pxa-ssp.c | 6 + sound/soc/pxa/pxa2xx-ac97.c | 48 +- sound/soc/qcom/apq8096.c | 7 +- sound/soc/qcom/qdsp6/q6adm.c | 17 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 8 +- sound/soc/qcom/qdsp6/q6asm.c | 1 - sound/soc/qcom/qdsp6/q6core.c | 9 +- sound/soc/qcom/sdm845.c | 7 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 1 - sound/soc/rockchip/rockchip_pcm.c | 3 +- sound/soc/samsung/tm2_wm5110.c | 13 +- sound/soc/sh/hac.c | 3 +- sound/soc/sh/rcar/adg.c | 4 +- sound/soc/sh/rcar/core.c | 124 ++- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/dma.c | 109 +- sound/soc/sh/rcar/gen.c | 33 +- sound/soc/sh/rcar/rsnd.h | 63 +- sound/soc/sh/rcar/src.c | 2 +- sound/soc/sh/rcar/ssi.c | 112 +- sound/soc/sh/rcar/ssiu.c | 92 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 582 +++++----- sound/soc/soc-dapm.c | 437 ++++---- sound/soc/soc-ops.c | 4 +- sound/soc/soc-pcm.c | 253 +++-- sound/soc/soc-topology.c | 15 +- sound/soc/soc-utils.c | 4 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai.h | 3 + sound/soc/stm/stm32_sai_sub.c | 282 ++++- sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 82 +- sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +- sound/soc/sunxi/sun8i-codec.c | 22 +- sound/soc/tegra/tegra_sgtl5000.c | 17 +- sound/soc/txx9/txx9aclc-ac97.c | 3 +- 193 files changed, 8358 insertions(+), 1542 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt create mode 100644 Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt create mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm3060.txt create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8782.txt rename {sound/pci/hda => include/sound}/hda_codec.h (100%) create mode 100644 sound/soc/atmel/mikroe-proto.c create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/codecs/nau8822.c create mode 100644 sound/soc/codecs/nau8822.h create mode 100644 sound/soc/codecs/pcm3060-i2c.c create mode 100644 sound/soc/codecs/pcm3060-spi.c create mode 100644 sound/soc/codecs/pcm3060.c create mode 100644 sound/soc/codecs/pcm3060.h create mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c create mode 100644 sound/soc/meson/axg-pdm.c create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Actually fix microphone issue
by Connor McAdams 22 Oct '18

22 Oct '18

This patch fixes the microphone issue for all cards. The previous fix worked on the ZxR, but not on the AE-5 or Z. This patch has been tested to work for all cards. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 73 ++++++++++++++++++++++---------------------- 1 file changed, 37 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 1a13cea..f0ef52e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -6983,37 +6983,44 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - unsigned int tmp; + unsigned int tmp, i; - switch (spec->quirk) { - case QUIRK_SBZ: - case QUIRK_AE5: - tmp = 0x00000003; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); - tmp = 0x00000001; - dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); - tmp = 0x00000004; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000005; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - break; - case QUIRK_R3D: - case QUIRK_R3DI: - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); - tmp = 0x00000001; - dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); - tmp = 0x00000004; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000005; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - break; + /* + * Gotta run these twice, or else mic works inconsistently. Not clear + * why this is, but multiple tests have confirmed it. + */ + for (i = 0; i < 2; i++) { + switch (spec->quirk) { + case QUIRK_SBZ: + case QUIRK_AE5: + tmp = 0x00000003; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + break; + case QUIRK_R3D: + case QUIRK_R3DI: + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + break; + } + msleep(100); } } @@ -7246,8 +7253,6 @@ static void r3d_setup_defaults(struct hda_codec *codec) int num_fx; int idx, i; - msleep(100); - if (spec->dsp_state != DSP_DOWNLOADED) return; @@ -7292,8 +7297,6 @@ static void sbz_setup_defaults(struct hda_codec *codec) int num_fx; int idx, i; - msleep(100); - if (spec->dsp_state != DSP_DOWNLOADED) return; @@ -7351,8 +7354,6 @@ static void ae5_setup_defaults(struct hda_codec *codec) int num_fx; int idx, i; - msleep(100); - if (spec->dsp_state != DSP_DOWNLOADED) return; -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks" to the asoc tree
by Mark Brown 21 Oct '18

21 Oct '18

The patch ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks 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 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Sun, 21 Oct 2018 08:39:11 -0700 Subject: [PATCH] ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks startup() and shutdown() hooks are called for both substreams, so stopping either substream when another is running breaks the latter. E.g. playback breaks if capture is stopped when playback is running. Move code from startup() and shutdown() to resume() and suspend() hooks respectively to fix this issue Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 61 +++++++++++++++---------------------- 1 file changed, 25 insertions(+), 36 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index c63d226e2436..d5ec1a20499d 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); -- 2.19.0

1 0

[alsa-devel] [PATCH RESEND v2] ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
by Vasily Khoruzhick 21 Oct '18

21 Oct '18

startup() and shutdown() hooks are called for both substreams, so stopping either substream when another is running breaks the latter. E.g. playback breaks if capture is stopped when playback is running. Move code from startup() and shutdown() to resume() and suspend() hooks respectively to fix this issue Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- v2: - rename from "ASoC: sun4i-i2s: don't try to start up or shut down DAI if it's active" - move code from startup/shutdown hooks into pm_runtime sound/soc/sunxi/sun4i-i2s.c | 61 +++++++++++++++---------------------- 1 file changed, 25 insertions(+), 36 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..daa6c08cffbc 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); -- 2.19.0

1 0

[alsa-devel] [PATCH v2] ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
by Vasily Khoruzhick 21 Oct '18

21 Oct '18

startup() and shutdown() hooks are called for both substreams, so stopping either substream when another is running breaks the latter. E.g. playback breaks if capture is stopped when playback is running. Move code from startup() and shutdown() to resume() and suspend() hooks respectively to fix this issue Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v2: - rename from "ASoC: sun4i-i2s: don't try to start up or shut down DAI if it's active" - move code from startup/shutdown hooks into pm_runtime sound/soc/sunxi/sun4i-i2s.c | 61 +++++++++++++++---------------------- 1 file changed, 25 insertions(+), 36 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..daa6c08cffbc 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); -- 2.19.0

3 9

[alsa-devel] Applied "ASoC: wm2000: Remove wm2000_read helper function" to the asoc tree
by Mark Brown 21 Oct '18

21 Oct '18

The patch ASoC: wm2000: Remove wm2000_read helper function 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 c5d09485def41cab9e75ba23abbf87080183183c Mon Sep 17 00:00:00 2001 From: Lucas Tanure <tanureal(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 17:44:22 +0100 Subject: [PATCH] ASoC: wm2000: Remove wm2000_read helper function The return type "unsigned int" was used by the wm2000_read() function despite of the aspect that it will eventually return a negative error code. The resulting function doesn't add much to the code, so replace wm2000_read with regmap_read. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm2000.c | 54 ++++++++++++++++++++++----------------- 1 file changed, 30 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c5ae07234a00..bba330e30162 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg, return regmap_write(wm2000->regmap, reg, value); } -static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r) -{ - struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); - unsigned int val; - int ret; - - ret = regmap_read(wm2000->regmap, r, &val); - if (ret < 0) - return -1; - - return val; -} - static void wm2000_reset(struct wm2000_priv *wm2000) { struct i2c_client *i2c = wm2000->i2c; @@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000) static int wm2000_poll_bit(struct i2c_client *i2c, unsigned int reg, u8 mask) { + struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); int timeout = 4000; - int val; + unsigned int val; - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); while (!(val & mask) && --timeout) { msleep(1); - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); } if (timeout == 0) @@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); unsigned long rate; + unsigned int val; int ret; if (WARN_ON(wm2000->anc_mode != ANC_OFF)) @@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) WM2000_MODE_THERMAL_ENABLE); } - ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); + ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret); + regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies); + return ret; + } if (wm2000->speech_clarity) - ret |= WM2000_SPEECH_CLARITY; + val |= WM2000_SPEECH_CLARITY; else - ret &= ~WM2000_SPEECH_CLARITY; - wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); + val &= ~WM2000_SPEECH_CLARITY; + wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02); @@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, const char *filename; const struct firmware *fw = NULL; int ret, i; - int reg; + unsigned int reg; u16 id; wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL); @@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, } /* Verify that this is a WM2000 */ - reg = wm2000_read(i2c, WM2000_REG_ID1); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret); + return ret; + } id = reg << 8; - reg = wm2000_read(i2c, WM2000_REG_ID2); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret); + return ret; + } id |= reg & 0xff; if (id != 0x2000) { @@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, goto err_supplies; } - reg = wm2000_read(i2c, WM2000_REG_REVISON); + ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret); + return ret; + } dev_info(&i2c->dev, "revision %c\n", reg + 'A'); wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK"); -- 2.19.0

1 0

Re: [alsa-devel] [PATCH] ASoC: wm2000: Remove wm2000_read helper function
by Charles Keepax 21 Oct '18

21 Oct '18

On Fri, Oct 19, 2018 at 05:44:22PM +0100, Lucas Tanure wrote: > The return type "unsigned int" was used by the wm2000_read() > function despite of the aspect that it will eventually return > a negative error code. > The resulting function doesn't add much to the code, so replace > wm2000_read with regmap_read. > > Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Thanks, Charles

1 0

Re: [alsa-devel] Tascam FW-1884 driver
by Takashi Sakamoto 20 Oct '18

20 Oct '18

Hi, (C.C.ed to alsa-devel, ffado-devel and Scott) On Sat, 20 Oct 2018, Cédric Mialaret wrote: > I stumbled up the source code of a linux driver for the Tascam FW-1884 audio interface: > > https://elixir.bootlin.com/linux/v4.19-rc8/source/sound/firewire/tascam/tas… > > a. Does it work ? It, ALSA firewire-tascam driver, works for systems in which Linux kernel is used. It's already included in Linux kernel source since kernel v4.4. > b. Does it implement all functionalities of the FW-1884 ? As of kernel v4.19, it supports below functionalities: - data transmission for PCM frames via ALSA PCM interface - data transmission for MIDI messages via ALSA rawmidi interface Recently I got Scott's help and work to support handling control messages from faders/knobs/buttons on control surface of FW-1884/FW-1082. You can see recent patches in my remote repository: - https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspac… - https://github.com/takaswie/libhinawa/tree/topic/tascam-userspace-take3 You can see our discussion in list: - http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140593.ht… > c. Is this the original code from Tascam/Frontier Design? Apparently, no. It comes from my work for reverse engineering, by sniffing packet communications over IEEE 1394 bus. > d. Do you by any chance have the source code of the Windows driver ? I have no such chance, as well as the others. Regards Takashi Sakamoto

1 0

[alsa-devel] [asoc:for-4.20 333/524] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 20 Oct '18

20 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 318e741ee13b5a72f3051d9bb6852b1f4d02d0bb commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/524] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-4.20 519/524] sound/soc//stm/stm32_sai_sub.c:379:2: error: unknown field 'recalc_rate' specified in initializer
by kbuild test robot 19 Oct '18

19 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 318e741ee13b5a72f3051d9bb6852b1f4d02d0bb commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/524] ASoC: stm32: sai: set sai as mclk clock provider config: openrisc-allyesconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8307b2afd386ccce369821daa2196068c47fe8cd # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc//stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/err.h:5:0, from include/linux/clk.h:15, from sound/soc//stm/stm32_sai_sub.c:19: sound/soc//stm/stm32_sai_sub.c: In function 'stm32_sai_mclk_round_rate': include/linux/kernel.h:997:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:335:18: note: in definition of macro '__compiletime_assert' int __cond = !(condition); \ ^~~~~~~~~ include/linux/compiler.h:358:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:45:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:997:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:997:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:265:27: note: in expansion of macro 'container_of' #define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) ^~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:313:37: note: in expansion of macro 'to_mclk_data' struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); ^~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c: At top level: sound/soc//stm/stm32_sai_sub.c:376:21: error: variable 'mclk_ops' has initializer but incomplete type static const struct clk_ops mclk_ops = { ^~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:377:2: error: unknown field 'enable' specified in initializer .enable = stm32_sai_mclk_enable, ^ sound/soc//stm/stm32_sai_sub.c:377:12: warning: excess elements in struct initializer .enable = stm32_sai_mclk_enable, ^~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:377:12: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:378:2: error: unknown field 'disable' specified in initializer .disable = stm32_sai_mclk_disable, ^ sound/soc//stm/stm32_sai_sub.c:378:13: warning: excess elements in struct initializer .disable = stm32_sai_mclk_disable, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:378:13: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:379:2: error: unknown field 'recalc_rate' specified in initializer .recalc_rate = stm32_sai_mclk_recalc_rate, ^ sound/soc//stm/stm32_sai_sub.c:379:17: warning: excess elements in struct initializer .recalc_rate = stm32_sai_mclk_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:379:17: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:380:2: error: unknown field 'round_rate' specified in initializer .round_rate = stm32_sai_mclk_round_rate, ^ sound/soc//stm/stm32_sai_sub.c:380:16: warning: excess elements in struct initializer .round_rate = stm32_sai_mclk_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:380:16: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:381:2: error: unknown field 'set_rate' specified in initializer .set_rate = stm32_sai_mclk_set_rate, ^ sound/soc//stm/stm32_sai_sub.c:381:14: warning: excess elements in struct initializer .set_rate = stm32_sai_mclk_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:381:14: note: (near initialization for 'mclk_ops') sound/soc//stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc//stm/stm32_sai_sub.c:389:22: error: implicit declaration of function '__clk_get_name' [-Werror=implicit-function-declaration] const char *pname = __clk_get_name(sai->sai_ck); ^~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:389:22: warning: initialization makes pointer from integer without a cast [-Wint-conversion] >> sound/soc//stm/stm32_sai_sub.c:414:18: error: implicit declaration of function 'CLK_HW_INIT' [-Werror=implicit-function-declaration] mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); ^~~~~~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:419:8: error: implicit declaration of function 'devm_clk_hw_register' [-Werror=implicit-function-declaration] ret = devm_clk_hw_register(&sai->pdev->dev, hw); ^~~~~~~~~~~~~~~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:427:9: error: implicit declaration of function 'devm_of_clk_add_hw_provider' [-Werror=implicit-function-declaration] return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:427:42: error: 'of_clk_hw_simple_get' undeclared (first use in this function) return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:427:42: note: each undeclared identifier is reported only once for each function it appears in sound/soc//stm/stm32_sai_sub.c: At top level: sound/soc//stm/stm32_sai_sub.c:376:29: error: storage size of 'mclk_ops' isn't known static const struct clk_ops mclk_ops = { ^~~~~~~~ sound/soc//stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': sound/soc//stm/stm32_sai_sub.c:428:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/recalc_rate +379 sound/soc//stm/stm32_sai_sub.c 309 310 static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, 311 unsigned long *prate) 312 { > 313 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 314 struct stm32_sai_sub_data *sai = mclk->sai_data; 315 int div; 316 317 div = stm32_sai_get_clk_div(sai, *prate, rate); 318 if (div < 0) 319 return div; 320 321 mclk->freq = *prate / div; 322 323 return mclk->freq; 324 } 325 326 static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw, 327 unsigned long parent_rate) 328 { 329 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 330 331 return mclk->freq; 332 } 333 334 static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, 335 unsigned long parent_rate) 336 { 337 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 338 struct stm32_sai_sub_data *sai = mclk->sai_data; 339 unsigned int div; 340 int ret; 341 342 div = stm32_sai_get_clk_div(sai, parent_rate, rate); 343 if (div < 0) 344 return div; 345 346 ret = stm32_sai_set_clk_div(sai, div); 347 if (ret) 348 return ret; 349 350 mclk->freq = rate; 351 352 return 0; 353 } 354 355 static int stm32_sai_mclk_enable(struct clk_hw *hw) 356 { 357 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 358 struct stm32_sai_sub_data *sai = mclk->sai_data; 359 360 dev_dbg(&sai->pdev->dev, "Enable master clock\n"); 361 362 return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, 363 SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); 364 } 365 366 static void stm32_sai_mclk_disable(struct clk_hw *hw) 367 { 368 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 369 struct stm32_sai_sub_data *sai = mclk->sai_data; 370 371 dev_dbg(&sai->pdev->dev, "Disable master clock\n"); 372 373 regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); 374 } 375 376 static const struct clk_ops mclk_ops = { > 377 .enable = stm32_sai_mclk_enable, > 378 .disable = stm32_sai_mclk_disable, > 379 .recalc_rate = stm32_sai_mclk_recalc_rate, > 380 .round_rate = stm32_sai_mclk_round_rate, > 381 .set_rate = stm32_sai_mclk_set_rate, 382 }; 383 384 static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) 385 { 386 struct clk_hw *hw; 387 struct stm32_sai_mclk_data *mclk; 388 struct device *dev = &sai->pdev->dev; > 389 const char *pname = __clk_get_name(sai->sai_ck); 390 char *mclk_name, *p, *s = (char *)pname; 391 int ret, i = 0; 392 393 mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); 394 if (!mclk) 395 return -ENOMEM; 396 397 mclk_name = devm_kcalloc(dev, sizeof(char), 398 SAI_MCLK_NAME_LEN, GFP_KERNEL); 399 if (!mclk_name) 400 return -ENOMEM; 401 402 /* 403 * Forge mclk clock name from parent clock name and suffix. 404 * String after "_" char is stripped in parent name. 405 */ 406 p = mclk_name; 407 while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { 408 *p++ = *s++; 409 i++; 410 } 411 STM_SAI_IS_SUB_A(sai) ? 412 strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); 413 > 414 mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); 415 mclk->sai_data = sai; 416 hw = &mclk->hw; 417 418 dev_dbg(dev, "Register master clock %s\n", mclk_name); > 419 ret = devm_clk_hw_register(&sai->pdev->dev, hw); 420 if (ret) { 421 dev_err(dev, "mclk register returned %d\n", ret); 422 return ret; 423 } 424 sai->sai_mclk = hw->clk; 425 426 /* register mclk provider */ > 427 return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); 428 } 429 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 19 Oct '18

19 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 30 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 30 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 35 +- sound/soc/sunxi/sunxi-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sunxi-adda-pr-regmap.h | 7 + 14 files changed, 798 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.h -- 2.19.0

2 18

[alsa-devel] [PATCH 1/1] ASoC: cs42l51: fix mclk support
by Olivier Moysan 19 Oct '18

19 Oct '18

Fixes: 5e8d63a726f8 ("ASoC: cs42l51: add mclk support") The MCLK clock is made optional for cs42l51 codec. However, ASoC DAPM clock supply widget, expects the clock to be defined unconditionally. Register MCLK DAPM conditionally in codec driver, depending on clock presence in DT. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/codecs/cs42l51.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index eb40bff54cec..fd2bd74024c1 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -21,6 +21,7 @@ * - master mode *NOT* supported */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/slab.h> #include <sound/core.h> @@ -41,6 +42,7 @@ enum master_slave_mode { struct cs42l51_private { unsigned int mclk; + struct clk *mclk_handle; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; }; @@ -492,9 +494,13 @@ static int cs42l51_component_probe(struct snd_soc_component *component) { int ret, reg; struct snd_soc_dapm_context *dapm; + struct cs42l51_private *cs42l51; + cs42l51 = snd_soc_component_get_drvdata(component); dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); + + if (cs42l51->mclk_handle) + snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); /* * DAC configuration @@ -548,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, cs42l51); + cs42l51->mclk_handle = devm_clk_get(dev, "MCLK"); + if (IS_ERR(cs42l51->mclk_handle)) { + if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT) + return PTR_ERR(cs42l51->mclk_handle); + cs42l51->mclk_handle = NULL; + } + /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { -- 2.7.4

2 2

[alsa-devel] [asoc:for-4.20 519/523] sound/soc/stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type
by kbuild test robot 19 Oct '18

19 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 7ece50c9f790270e82b77589d34e54547c895a66 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/523] ASoC: stm32: sai: set sai as mclk clock provider config: powerpc-allyesconfig (attached as .config) compiler: powerpc64-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8307b2afd386ccce369821daa2196068c47fe8cd # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=powerpc All error/warnings (new ones prefixed by >>): >> sound/soc/stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/err.h:5:0, from include/linux/clk.h:15, from sound/soc/stm/stm32_sai_sub.c:19: sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_mclk_round_rate': include/linux/kernel.h:997:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:335:18: note: in definition of macro '__compiletime_assert' int __cond = !(condition); \ ^~~~~~~~~ include/linux/compiler.h:358:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:45:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:997:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:997:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ >> sound/soc/stm/stm32_sai_sub.c:265:27: note: in expansion of macro 'container_of' #define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) ^~~~~~~~~~~~ >> sound/soc/stm/stm32_sai_sub.c:313:37: note: in expansion of macro 'to_mclk_data' struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); ^~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c: At top level: >> sound/soc/stm/stm32_sai_sub.c:376:21: error: variable 'mclk_ops' has initializer but incomplete type static const struct clk_ops mclk_ops = { ^~~~~~~ >> sound/soc/stm/stm32_sai_sub.c:377:3: error: 'const struct clk_ops' has no member named 'enable' .enable = stm32_sai_mclk_enable, ^~~~~~ >> sound/soc/stm/stm32_sai_sub.c:377:12: warning: excess elements in struct initializer .enable = stm32_sai_mclk_enable, ^~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:377:12: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:378:3: error: 'const struct clk_ops' has no member named 'disable' .disable = stm32_sai_mclk_disable, ^~~~~~~ sound/soc/stm/stm32_sai_sub.c:378:13: warning: excess elements in struct initializer .disable = stm32_sai_mclk_disable, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:378:13: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:379:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = stm32_sai_mclk_recalc_rate, ^~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:379:17: warning: excess elements in struct initializer .recalc_rate = stm32_sai_mclk_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:379:17: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:380:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = stm32_sai_mclk_round_rate, ^~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:380:16: warning: excess elements in struct initializer .round_rate = stm32_sai_mclk_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:380:16: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:381:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = stm32_sai_mclk_set_rate, ^~~~~~~~ sound/soc/stm/stm32_sai_sub.c:381:14: warning: excess elements in struct initializer .set_rate = stm32_sai_mclk_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:381:14: note: (near initialization for 'mclk_ops') sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc/stm/stm32_sai_sub.c:389:22: error: implicit declaration of function '__clk_get_name'; did you mean 'clk_get_rate'? [-Werror=implicit-function-declaration] const char *pname = __clk_get_name(sai->sai_ck); ^~~~~~~~~~~~~~ clk_get_rate >> sound/soc/stm/stm32_sai_sub.c:389:22: warning: initialization makes pointer from integer without a cast [-Wint-conversion] >> sound/soc/stm/stm32_sai_sub.c:414:18: error: implicit declaration of function 'CLK_HW_INIT'; did you mean 'KLIST_INIT'? [-Werror=implicit-function-declaration] mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); ^~~~~~~~~~~ KLIST_INIT >> sound/soc/stm/stm32_sai_sub.c:419:8: error: implicit declaration of function 'devm_clk_hw_register'; did you mean 'device_unregister'? [-Werror=implicit-function-declaration] ret = devm_clk_hw_register(&sai->pdev->dev, hw); ^~~~~~~~~~~~~~~~~~~~ device_unregister >> sound/soc/stm/stm32_sai_sub.c:427:9: error: implicit declaration of function 'devm_of_clk_add_hw_provider'; did you mean 'of_clk_get_from_provider'? [-Werror=implicit-function-declaration] return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ of_clk_get_from_provider >> sound/soc/stm/stm32_sai_sub.c:427:42: error: 'of_clk_hw_simple_get' undeclared (first use in this function); did you mean 'ida_simple_get'? return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~ ida_simple_get sound/soc/stm/stm32_sai_sub.c:427:42: note: each undeclared identifier is reported only once for each function it appears in sound/soc/stm/stm32_sai_sub.c: At top level: >> sound/soc/stm/stm32_sai_sub.c:376:29: error: storage size of 'mclk_ops' isn't known static const struct clk_ops mclk_ops = { ^~~~~~~~ sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc/stm/stm32_sai_sub.c:428:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/hw +260 sound/soc/stm/stm32_sai_sub.c 258 259 struct stm32_sai_mclk_data { > 260 struct clk_hw hw; 261 unsigned long freq; 262 struct stm32_sai_sub_data *sai_data; 263 }; 264 > 265 #define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) 266 #define STM32_SAI_MAX_CLKS 1 267 268 static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, 269 unsigned long input_rate, 270 unsigned long output_rate) 271 { 272 int version = sai->pdata->conf->version; 273 int div; 274 275 div = DIV_ROUND_CLOSEST(input_rate, output_rate); 276 if (div > SAI_XCR1_MCKDIV_MAX(version)) { 277 dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); 278 return -EINVAL; 279 } 280 dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div); 281 282 if (input_rate % div) 283 dev_dbg(&sai->pdev->dev, 284 "Rate not accurate. requested (%ld), actual (%ld)\n", 285 output_rate, input_rate / div); 286 287 return div; 288 } 289 290 static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, 291 unsigned int div) 292 { 293 int version = sai->pdata->conf->version; 294 int ret, cr1, mask; 295 296 if (div > SAI_XCR1_MCKDIV_MAX(version)) { 297 dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); 298 return -EINVAL; 299 } 300 301 mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); 302 cr1 = SAI_XCR1_MCKDIV_SET(div); 303 ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); 304 if (ret < 0) 305 dev_err(&sai->pdev->dev, "Failed to update CR1 register\n"); 306 307 return ret; 308 } 309 310 static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, 311 unsigned long *prate) 312 { > 313 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 314 struct stm32_sai_sub_data *sai = mclk->sai_data; 315 int div; 316 317 div = stm32_sai_get_clk_div(sai, *prate, rate); 318 if (div < 0) 319 return div; 320 321 mclk->freq = *prate / div; 322 323 return mclk->freq; 324 } 325 326 static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw, 327 unsigned long parent_rate) 328 { 329 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 330 331 return mclk->freq; 332 } 333 334 static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, 335 unsigned long parent_rate) 336 { 337 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 338 struct stm32_sai_sub_data *sai = mclk->sai_data; 339 unsigned int div; 340 int ret; 341 342 div = stm32_sai_get_clk_div(sai, parent_rate, rate); 343 if (div < 0) 344 return div; 345 346 ret = stm32_sai_set_clk_div(sai, div); 347 if (ret) 348 return ret; 349 350 mclk->freq = rate; 351 352 return 0; 353 } 354 355 static int stm32_sai_mclk_enable(struct clk_hw *hw) 356 { 357 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 358 struct stm32_sai_sub_data *sai = mclk->sai_data; 359 360 dev_dbg(&sai->pdev->dev, "Enable master clock\n"); 361 362 return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, 363 SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); 364 } 365 366 static void stm32_sai_mclk_disable(struct clk_hw *hw) 367 { 368 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 369 struct stm32_sai_sub_data *sai = mclk->sai_data; 370 371 dev_dbg(&sai->pdev->dev, "Disable master clock\n"); 372 373 regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); 374 } 375 > 376 static const struct clk_ops mclk_ops = { > 377 .enable = stm32_sai_mclk_enable, > 378 .disable = stm32_sai_mclk_disable, > 379 .recalc_rate = stm32_sai_mclk_recalc_rate, > 380 .round_rate = stm32_sai_mclk_round_rate, > 381 .set_rate = stm32_sai_mclk_set_rate, 382 }; 383 384 static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) 385 { 386 struct clk_hw *hw; 387 struct stm32_sai_mclk_data *mclk; 388 struct device *dev = &sai->pdev->dev; > 389 const char *pname = __clk_get_name(sai->sai_ck); 390 char *mclk_name, *p, *s = (char *)pname; 391 int ret, i = 0; 392 393 mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); 394 if (!mclk) 395 return -ENOMEM; 396 397 mclk_name = devm_kcalloc(dev, sizeof(char), 398 SAI_MCLK_NAME_LEN, GFP_KERNEL); 399 if (!mclk_name) 400 return -ENOMEM; 401 402 /* 403 * Forge mclk clock name from parent clock name and suffix. 404 * String after "_" char is stripped in parent name. 405 */ 406 p = mclk_name; 407 while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { 408 *p++ = *s++; 409 i++; 410 } 411 STM_SAI_IS_SUB_A(sai) ? 412 strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); 413 > 414 mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); 415 mclk->sai_data = sai; 416 hw = &mclk->hw; 417 418 dev_dbg(dev, "Register master clock %s\n", mclk_name); > 419 ret = devm_clk_hw_register(&sai->pdev->dev, hw); 420 if (ret) { 421 dev_err(dev, "mclk register returned %d\n", ret); 422 return ret; 423 } 424 sai->sai_mclk = hw->clk; 425 426 /* register mclk provider */ > 427 return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); > 428 } 429 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-4.20 519/523] sound/soc/stm/stm32_sai_sub.c:412:3: warning: 'strncat' specified bound 6 equals source length
by kbuild test robot 19 Oct '18

19 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 7ece50c9f790270e82b77589d34e54547c895a66 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/523] ASoC: stm32: sai: set sai as mclk clock provider config: microblaze-allyesconfig (attached as .config) compiler: microblaze-linux-gcc (GCC) 8.1.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8307b2afd386ccce369821daa2196068c47fe8cd # save the attached .config to linux build tree GCC_VERSION=8.1.0 make.cross ARCH=microblaze All warnings (new ones prefixed by >>): sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc/stm/stm32_sai_sub.c:412:3: warning: 'strncat' specified bound 6 equals source length [-Wstringop-overflow=] strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:412:29: warning: 'strncat' specified bound 6 equals source length [-Wstringop-overflow=] strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); ^~~~~~~~~~~~~~~~~~~~~~~ vim +/strncat +412 sound/soc/stm/stm32_sai_sub.c 383 384 static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) 385 { 386 struct clk_hw *hw; 387 struct stm32_sai_mclk_data *mclk; 388 struct device *dev = &sai->pdev->dev; 389 const char *pname = __clk_get_name(sai->sai_ck); 390 char *mclk_name, *p, *s = (char *)pname; 391 int ret, i = 0; 392 393 mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); 394 if (!mclk) 395 return -ENOMEM; 396 397 mclk_name = devm_kcalloc(dev, sizeof(char), 398 SAI_MCLK_NAME_LEN, GFP_KERNEL); 399 if (!mclk_name) 400 return -ENOMEM; 401 402 /* 403 * Forge mclk clock name from parent clock name and suffix. 404 * String after "_" char is stripped in parent name. 405 */ 406 p = mclk_name; 407 while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { 408 *p++ = *s++; 409 i++; 410 } 411 STM_SAI_IS_SUB_A(sai) ? > 412 strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); 413 414 mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); 415 mclk->sai_data = sai; 416 hw = &mclk->hw; 417 418 dev_dbg(dev, "Register master clock %s\n", mclk_name); 419 ret = devm_clk_hw_register(&sai->pdev->dev, hw); 420 if (ret) { 421 dev_err(dev, "mclk register returned %d\n", ret); 422 return ret; 423 } 424 sai->sai_mclk = hw->clk; 425 426 /* register mclk provider */ 427 return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); 428 } 429 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Applied "ASoC: tegra_sgtl5000: fix device_node refcounting" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: tegra_sgtl5000: fix device_node refcounting 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 a85227da2dcc291b762c8482a505bc7d0d2d4b07 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Date: Tue, 16 Oct 2018 12:47:29 +0200 Subject: [PATCH] ASoC: tegra_sgtl5000: fix device_node refcounting Similar to the following: commit 4321723648b0 ("ASoC: tegra_alc5632: fix device_node refcounting") commit 7c5dfd549617 ("ASoC: tegra: fix device_node refcounting") Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/tegra/tegra_sgtl5000.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: nau8822: new codec driver
by David Lin 19 Oct '18

19 Oct '18

Add driver for NAU88C22. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8822.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8822.c | 1136 +++++++++++++++++ sound/soc/codecs/nau8822.h | 204 +++ 5 files changed, 1363 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 sound/soc/codecs/nau8822.c create mode 100644 sound/soc/codecs/nau8822.h diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt new file mode 100644 index 000000000000..a471d162d4e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8822.txt @@ -0,0 +1,16 @@ +NAU8822 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8822" + + - reg : the I2C address of the device. + +Example: + +codec: nau8822@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..57365dbc1581 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C + select SND_SOC_NAU8822 if I2C select SND_SOC_NAU8824 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC @@ -1299,6 +1300,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8822 + tristate "Nuvoton Technology Corporation NAU88C22 CODEC" + depends on I2C + config SND_SOC_NAU8824 tristate "Nuvoton Technology Corporation NAU88L24 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..181a7d222fd7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -106,6 +106,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o @@ -366,6 +367,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c new file mode 100644 index 000000000000..622ce947f134 --- /dev/null +++ b/sound/soc/codecs/nau8822.c @@ -0,0 +1,1136 @@ +/* + * nau8822.c -- NAU8822 ALSA Soc Audio Codec driver + * + * Copyright 2017 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> +#include "nau8822.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + +static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8822_reg_defaults[] = { + { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 }, + { NAU8822_REG_AUDIO_INTERFACE, 0x0050 }, + { NAU8822_REG_COMPANDING_CONTROL, 0x0000 }, + { NAU8822_REG_CLOCKING, 0x0140 }, + { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 }, + { NAU8822_REG_GPIO_CONTROL, 0x0000 }, + { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 }, + { NAU8822_REG_DAC_CONTROL, 0x0000 }, + { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 }, + { NAU8822_REG_ADC_CONTROL, 0x0100 }, + { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_EQ1, 0x012c }, + { NAU8822_REG_EQ2, 0x002c }, + { NAU8822_REG_EQ3, 0x002c }, + { NAU8822_REG_EQ4, 0x002c }, + { NAU8822_REG_EQ5, 0x002c }, + { NAU8822_REG_DAC_LIMITER_1, 0x0032 }, + { NAU8822_REG_DAC_LIMITER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_1, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_3, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_4, 0x0000 }, + { NAU8822_REG_ALC_CONTROL_1, 0x0038 }, + { NAU8822_REG_ALC_CONTROL_2, 0x000b }, + { NAU8822_REG_ALC_CONTROL_3, 0x0032 }, + { NAU8822_REG_NOISE_GATE, 0x0010 }, + { NAU8822_REG_PLL_N, 0x0008 }, + { NAU8822_REG_PLL_K1, 0x000c }, + { NAU8822_REG_PLL_K2, 0x0093 }, + { NAU8822_REG_PLL_K3, 0x00e9 }, + { NAU8822_REG_3D_CONTROL, 0x0000 }, + { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_CONTROL, 0x0033 }, + { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_OUTPUT_CONTROL, 0x0002 }, + { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_LHP_VOLUME, 0x0039 }, + { NAU8822_REG_RHP_VOLUME, 0x0039 }, + { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_AUX2_MIXER, 0x0001 }, + { NAU8822_REG_AUX1_MIXER, 0x0001 }, + { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 }, + { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_MISC, 0x0020 }, + { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_DEVICE_REVISION, 0x007f }, + { NAU8822_REG_DEVICE_ID, 0x001a }, + { NAU8822_REG_DAC_DITHER, 0x0114 }, + { NAU8822_REG_ALC_ENHANCE_1, 0x0000 }, + { NAU8822_REG_ALC_ENHANCE_2, 0x0000 }, + { NAU8822_REG_192KHZ_SAMPLING, 0x0008 }, + { NAU8822_REG_MISC_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_TIEOFF, 0x0000 }, + { NAU8822_REG_POWER_REDUCTION, 0x0000 }, + { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 }, + { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 }, + { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 }, + { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 }, +}; + +static bool nau8822_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET: + case NAU8822_REG_DEVICE_REVISION: + case NAU8822_REG_DEVICE_ID: + case NAU8822_REG_AGC_PEAK2PEAK: + case NAU8822_REG_AGC_PEAK_DETECT: + case NAU8822_REG_AUTOMUTE_CONTROL: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8822_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg; + u16 reg_val, *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + reg_val = snd_soc_component_read32(component, reg + i); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8822_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = snd_soc_component_write(component, reg + i, value); + if (ret) { + dev_err(component->dev, + "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + kfree(data); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8822_companding[] = { + "Off", "NC", "u-law", "A-law"}; + +static const struct soc_enum nau8822_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const struct soc_enum nau8822_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const char * const nau8822_eqmode[] = {"Capture", "Playback"}; + +static const struct soc_enum nau8822_eqmode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT, + ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode); + +static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"}; +static const char * const nau8822_alc3[] = {"Normal", "Limiter"}; + +static const struct soc_enum nau8822_alc_enable_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT, + ARRAY_SIZE(nau8822_alc1), nau8822_alc1); + +static const struct soc_enum nau8822_alc_mode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT, + ARRAY_SIZE(nau8822_alc3), nau8822_alc3); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); + +static const struct snd_kcontrol_new nau8822_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8822_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8822_companding_dac_enum), + + SOC_ENUM("EQ Function", nau8822_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8822_eq_get, nau8822_eq_put), + + SOC_DOUBLE("DAC Inversion Switch", + NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("PCM Volume", + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", + NAU8822_REG_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", + NAU8822_REG_ADC_CONTROL, 4, 7, 0), + + SOC_DOUBLE("ADC Inversion Switch", + NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("ADC Volume", + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("DAC Limiter Switch", + NAU8822_REG_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Limiter Decay", + NAU8822_REG_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Limiter Attack", + NAU8822_REG_DAC_LIMITER_1, 0, 15, 0), + SOC_SINGLE("DAC Limiter Threshold", + NAU8822_REG_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE_TLV("DAC Limiter Volume", + NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), + + SOC_ENUM("ALC Mode", nau8822_alc_mode_enum), + SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum), + SOC_SINGLE("ALC Min Gain", + NAU8822_REG_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Max Gain", + NAU8822_REG_ALC_CONTROL_1, 3, 7, 0), + SOC_SINGLE("ALC Hold", + NAU8822_REG_ALC_CONTROL_2, 4, 10, 0), + SOC_SINGLE("ALC Target", + NAU8822_REG_ALC_CONTROL_2, 0, 15, 0), + SOC_SINGLE("ALC Decay", + NAU8822_REG_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Attack", + NAU8822_REG_ALC_CONTROL_3, 0, 10, 0), + SOC_SINGLE("ALC Noise Gate Switch", + NAU8822_REG_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", + NAU8822_REG_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("PGA ZC Switch", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + SOC_DOUBLE_R_TLV("PGA Volume", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv), + + SOC_DOUBLE_R("Headphone ZC Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Headphone Playback Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Headphone Volume", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("Speaker ZC Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Speaker Playback Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Speaker Volume", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("AUXOUT Playback Switch", + NAU8822_REG_AUX2_MIXER, + NAU8822_REG_AUX1_MIXER, 6, 1, 1), + + SOC_DOUBLE_R_TLV("PGA Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv), + + SOC_SINGLE("DAC 128x Oversampling Switch", + NAU8822_REG_DAC_CONTROL, 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", + NAU8822_REG_ADC_CONTROL, 5, 1, 0), +}; + +/* LMAIN and RMAIN Mixer */ +static const struct snd_kcontrol_new nau8822_left_out_mixer[] = { + SOC_DAPM_SINGLE("LINMIX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("LAUX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_right_out_mixer[] = { + SOC_DAPM_SINGLE("RINMIX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("RAUX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0), +}; + +/* AUX1 and AUX2 Mixer */ +static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = { + SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = { + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("AUX1MIX Output Switch", + NAU8822_REG_AUX2_MIXER, 3, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8822_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new nau8822_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0), +}; + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8822_loopback = + SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL, + NAU8822_ADDAP_SFT, 1, 0); + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + unsigned int value; + + value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING); + + return (value & NAU8822_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 1, 0), + + SOC_MIXER_ARRAY("Left Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer), + SOC_MIXER_ARRAY("AUX1 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer), + SOC_MIXER_ARRAY("AUX2 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 2, 0, nau8822_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 3, 0, nau8822_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", + NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("AUX1 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", + NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", + NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8822_loopback), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), + SND_SOC_DAPM_OUTPUT("AUXOUT1"), + SND_SOC_DAPM_OUTPUT("AUXOUT2"), +}; + +static const struct snd_soc_dapm_route nau8822_dapm_routes[] = { + {"Right DAC", NULL, "PLL", check_mclk_select_pll}, + {"Left DAC", NULL, "PLL", check_mclk_select_pll}, + + /* LMAIN and RMAIN Mixer */ + {"Right Output Mixer", "LDAC Switch", "Left DAC"}, + {"Right Output Mixer", "RDAC Switch", "Right DAC"}, + {"Right Output Mixer", "RAUX Switch", "RAUX"}, + {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "LDAC Switch", "Left DAC"}, + {"Left Output Mixer", "RDAC Switch", "Right DAC"}, + {"Left Output Mixer", "LAUX Switch", "LAUX"}, + {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + + /* AUX1 and AUX2 Mixer */ + {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"}, + {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"}, + {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + + {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + {"AUX1 Out", NULL, "AUX1 Output Mixer"}, + {"AUX2 Out", NULL, "AUX2 Output Mixer"}, + {"AUXOUT1", NULL, "AUX1 Out"}, + {"AUXOUT2", NULL, "AUX2 Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "PLL", check_mclk_select_pll}, + {"Left ADC", NULL, "PLL", check_mclk_select_pll}, + + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + /* Enable Microphone Power */ + {"Right Capture PGA", NULL, "Mic Bias"}, + {"Left Capture PGA", NULL, "Mic Bias"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, + + /* Digital Loopback */ + {"Digital Loopback", "Switch", "Left ADC"}, + {"Digital Loopback", "Switch", "Right ADC"}, + {"Left DAC", NULL, "Digital Loopback"}, + {"Right DAC", NULL, "Digital Loopback"}, +}; + +static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + nau8822->div_id = clk_id; + nau8822->sysclk = freq; + dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq, + clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs, + struct nau8822_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8822_mclk_scaler); + + for (i = 0; i < scal_sel; i++) { + f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + + if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll = &nau8822->pll; + int i, sclk, imclk; + + switch (nau8822->div_id) { + case NAU8822_CLK_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) { + sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(component->dev, "master clock prescaler %x for fs %d\n", + div, rate); + + /* master clock from MCLK and disable PLL */ + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_MCLK); + break; + + case NAU8822_CLK_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(component->dev, + "master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_PLL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll_param = &nau8822->pll; + int ret, fs; + + fs = freq_out / 256; + + ret = nau8822_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); + return ret; + } + + dev_info(component->dev, + "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, + pll_param->mclk_scaler, pll_param->pre_factor); + + snd_soc_component_update_bits(component, + NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, + (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + snd_soc_component_write(component, + NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) & + NAU8822_PLLK1_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) & + NAU8822_PLLK2_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + + return 0; +} + +static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + u16 ctrl1_val = 0, ctrl2_val = 0; + + dev_dbg(component->dev, "%s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl2_val &= ~1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= 0x18; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, + NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK, + ctrl1_val); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val); + + return 0; +} + +static int nau8822_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + int val_len = 0, val_rate = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8822_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8822_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8822_WLEN_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8822_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8822_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8822_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8822_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8822_SMPLR_32K; + break; + case 44100: + case 48000: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len); + snd_soc_component_update_bits(component, + NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8822->div_id == NAU8822_CLK_MCLK) + nau8822_config_clkdiv(dai, 0, params_rate(params)); + + return 0; +} + +static int nau8822_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + dev_dbg(component->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int nau8822_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN); + + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K); + mdelay(100); + } + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_1, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_2, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(component->dev, "%s: %d\n", __func__, level); + + return 0; +} + +#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8822_dai_ops = { + .hw_params = nau8822_hw_params, + .digital_mute = nau8822_mute, + .set_fmt = nau8822_set_dai_fmt, + .set_sysclk = nau8822_set_dai_sysclk, + .set_pll = nau8822_set_pll, +}; + +static struct snd_soc_dai_driver nau8822_dai = { + .name = "nau8822-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .ops = &nau8822_dai_ops, + .symmetric_rates = 1, +}; + +static int nau8822_suspend(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + + regcache_mark_dirty(nau8822->regmap); + + return 0; +} + +static int nau8822_resume(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + regcache_sync(nau8822->regmap); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, +}; + +static int nau8822_probe(struct snd_soc_component *component) +{ + int i; + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + snd_soc_component_update_bits(component, + update_reg[i], 0x100, 0x100); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_nau8822 = { + .probe = nau8822_probe, + .suspend = nau8822_suspend, + .resume = nau8822_resume, + .set_bias_level = nau8822_set_bias_level, + .controls = nau8822_snd_controls, + .num_controls = ARRAY_SIZE(nau8822_snd_controls), + .dapm_widgets = nau8822_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8822_dapm_widgets), + .dapm_routes = nau8822_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8822_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config nau8822_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8822_REG_MAX_REGISTER, + .volatile_reg = nau8822_volatile, + + .readable_reg = nau8822_readable_reg, + .writeable_reg = nau8822_writeable_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8822_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults), +}; + +static int nau8822_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8822 *nau8822 = dev_get_platdata(dev); + int ret; + + if (!nau8822) { + nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL); + if (nau8822 == NULL) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8822); + + nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config); + if (IS_ERR(nau8822->regmap)) { + ret = PTR_ERR(nau8822->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + nau8822->dev = dev; + + /* Reset the codec */ + ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822, + &nau8822_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct i2c_device_id nau8822_i2c_id[] = { + { "nau8822", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8822_of_match[] = { + { .compatible = "nuvoton,nau8822", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8822_of_match); +#endif + +static struct i2c_driver nau8822_i2c_driver = { + .driver = { + .name = "nau8822", + .of_match_table = of_match_ptr(nau8822_of_match), + }, + .probe = nau8822_i2c_probe, + .id_table = nau8822_i2c_id, +}; +module_i2c_driver(nau8822_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8822 codec driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h new file mode 100644 index 000000000000..aa79c969cd44 --- /dev/null +++ b/sound/soc/codecs/nau8822.h @@ -0,0 +1,204 @@ +/* + * nau8822.h -- NAU8822 Soc Audio Codec driver + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8822_H__ +#define __NAU8822_H__ + +#define NAU8822_REG_RESET 0x00 +#define NAU8822_REG_POWER_MANAGEMENT_1 0x01 +#define NAU8822_REG_POWER_MANAGEMENT_2 0x02 +#define NAU8822_REG_POWER_MANAGEMENT_3 0x03 +#define NAU8822_REG_AUDIO_INTERFACE 0x04 +#define NAU8822_REG_COMPANDING_CONTROL 0x05 +#define NAU8822_REG_CLOCKING 0x06 +#define NAU8822_REG_ADDITIONAL_CONTROL 0x07 +#define NAU8822_REG_GPIO_CONTROL 0x08 +#define NAU8822_REG_JACK_DETECT_CONTROL_1 0x09 +#define NAU8822_REG_DAC_CONTROL 0x0A +#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define NAU8822_REG_JACK_DETECT_CONTROL_2 0x0D +#define NAU8822_REG_ADC_CONTROL 0x0E +#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define NAU8822_REG_EQ1 0x12 +#define NAU8822_REG_EQ2 0x13 +#define NAU8822_REG_EQ3 0x14 +#define NAU8822_REG_EQ4 0x15 +#define NAU8822_REG_EQ5 0x16 +#define NAU8822_REG_DAC_LIMITER_1 0x18 +#define NAU8822_REG_DAC_LIMITER_2 0x19 +#define NAU8822_REG_NOTCH_FILTER_1 0x1B +#define NAU8822_REG_NOTCH_FILTER_2 0x1C +#define NAU8822_REG_NOTCH_FILTER_3 0x1D +#define NAU8822_REG_NOTCH_FILTER_4 0x1E +#define NAU8822_REG_ALC_CONTROL_1 0x20 +#define NAU8822_REG_ALC_CONTROL_2 0x21 +#define NAU8822_REG_ALC_CONTROL_3 0x22 +#define NAU8822_REG_NOISE_GATE 0x23 +#define NAU8822_REG_PLL_N 0x24 +#define NAU8822_REG_PLL_K1 0x25 +#define NAU8822_REG_PLL_K2 0x26 +#define NAU8822_REG_PLL_K3 0x27 +#define NAU8822_REG_3D_CONTROL 0x29 +#define NAU8822_REG_RIGHT_SPEAKER_CONTROL 0x2B +#define NAU8822_REG_INPUT_CONTROL 0x2C +#define NAU8822_REG_LEFT_INP_PGA_CONTROL 0x2D +#define NAU8822_REG_RIGHT_INP_PGA_CONTROL 0x2E +#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL 0x2F +#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL 0x30 +#define NAU8822_REG_OUTPUT_CONTROL 0x31 +#define NAU8822_REG_LEFT_MIXER_CONTROL 0x32 +#define NAU8822_REG_RIGHT_MIXER_CONTROL 0x33 +#define NAU8822_REG_LHP_VOLUME 0x34 +#define NAU8822_REG_RHP_VOLUME 0x35 +#define NAU8822_REG_LSPKOUT_VOLUME 0x36 +#define NAU8822_REG_RSPKOUT_VOLUME 0x37 +#define NAU8822_REG_AUX2_MIXER 0x38 +#define NAU8822_REG_AUX1_MIXER 0x39 +#define NAU8822_REG_POWER_MANAGEMENT_4 0x3A +#define NAU8822_REG_LEFT_TIME_SLOT 0x3B +#define NAU8822_REG_MISC 0x3C +#define NAU8822_REG_RIGHT_TIME_SLOT 0x3D +#define NAU8822_REG_DEVICE_REVISION 0x3E +#define NAU8822_REG_DEVICE_ID 0x3F +#define NAU8822_REG_DAC_DITHER 0x41 +#define NAU8822_REG_ALC_ENHANCE_1 0x46 +#define NAU8822_REG_ALC_ENHANCE_2 0x47 +#define NAU8822_REG_192KHZ_SAMPLING 0x48 +#define NAU8822_REG_MISC_CONTROL 0x49 +#define NAU8822_REG_INPUT_TIEOFF 0x4A +#define NAU8822_REG_POWER_REDUCTION 0x4B +#define NAU8822_REG_AGC_PEAK2PEAK 0x4C +#define NAU8822_REG_AGC_PEAK_DETECT 0x4D +#define NAU8822_REG_AUTOMUTE_CONTROL 0x4E +#define NAU8822_REG_OUTPUT_TIEOFF 0x4F +#define NAU8822_REG_MAX_REGISTER NAU8822_REG_OUTPUT_TIEOFF + +/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */ +#define NAU8822_REFIMP_MASK 0x3 +#define NAU8822_REFIMP_80K 0x1 +#define NAU8822_REFIMP_300K 0x2 +#define NAU8822_REFIMP_3K 0x3 +#define NAU8822_IOBUF_EN (0x1 << 2) +#define NAU8822_ABIAS_EN (0x1 << 3) + +/* NAU8822_REG_AUDIO_INTERFACE (0x4) */ +#define NAU8822_AIFMT_MASK (0x3 << 3) +#define NAU8822_WLEN_MASK (0x3 << 5) +#define NAU8822_WLEN_20 (0x1 << 5) +#define NAU8822_WLEN_24 (0x2 << 5) +#define NAU8822_WLEN_32 (0x3 << 5) +#define NAU8822_LRP_MASK (0x1 << 7) +#define NAU8822_BCLKP_MASK (0x1 << 8) + +/* NAU8822_REG_COMPANDING_CONTROL (0x5) */ +#define NAU8822_ADDAP_SFT 0 +#define NAU8822_ADCCM_SFT 1 +#define NAU8822_DACCM_SFT 3 + +/* NAU8822_REG_CLOCKING (0x6) */ +#define NAU8822_CLKIOEN_MASK 0x1 +#define NAU8822_MCLKSEL_SFT 5 +#define NAU8822_MCLKSEL_MASK (0x7 << 5) +#define NAU8822_BCLKSEL_SFT 2 +#define NAU8822_BCLKSEL_MASK (0x7 << 2) +#define NAU8822_CLKM_MASK (0x1 << 8) +#define NAU8822_CLKM_MCLK (0x0 << 8) +#define NAU8822_CLKM_PLL (0x1 << 8) + +/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */ +#define NAU8822_SMPLR_SFT 1 +#define NAU8822_SMPLR_MASK (0x7 << 1) +#define NAU8822_SMPLR_48K (0x0 << 1) +#define NAU8822_SMPLR_32K (0x1 << 1) +#define NAU8822_SMPLR_24K (0x2 << 1) +#define NAU8822_SMPLR_16K (0x3 << 1) +#define NAU8822_SMPLR_12K (0x4 << 1) +#define NAU8822_SMPLR_8K (0x5 << 1) + +/* NAU8822_REG_EQ1 (0x12) */ +#define NAU8822_EQ1GC_SFT 0 +#define NAU8822_EQ1CF_SFT 5 +#define NAU8822_EQM_SFT 8 + +/* NAU8822_REG_EQ2 (0x13) */ +#define NAU8822_EQ2GC_SFT 0 +#define NAU8822_EQ2CF_SFT 5 +#define NAU8822_EQ2BW_SFT 8 + +/* NAU8822_REG_EQ3 (0x14) */ +#define NAU8822_EQ3GC_SFT 0 +#define NAU8822_EQ3CF_SFT 5 +#define NAU8822_EQ3BW_SFT 8 + +/* NAU8822_REG_EQ4 (0x15) */ +#define NAU8822_EQ4GC_SFT 0 +#define NAU8822_EQ4CF_SFT 5 +#define NAU8822_EQ4BW_SFT 8 + +/* NAU8822_REG_EQ5 (0x16) */ +#define NAU8822_EQ5GC_SFT 0 +#define NAU8822_EQ5CF_SFT 5 + +/* NAU8822_REG_ALC_CONTROL_1 (0x20) */ +#define NAU8822_ALCMINGAIN_SFT 0 +#define NAU8822_ALCMXGAIN_SFT 3 +#define NAU8822_ALCEN_SFT 7 + +/* NAU8822_REG_ALC_CONTROL_2 (0x21) */ +#define NAU8822_ALCSL_SFT 0 +#define NAU8822_ALCHT_SFT 4 + +/* NAU8822_REG_ALC_CONTROL_3 (0x22) */ +#define NAU8822_ALCATK_SFT 0 +#define NAU8822_ALCDCY_SFT 4 +#define NAU8822_ALCM_SFT 8 + +/* NAU8822_REG_PLL_N (0x24) */ +#define NAU8822_PLLMCLK_DIV2 (0x1 << 4) +#define NAU8822_PLLN_MASK 0xF + +#define NAU8822_PLLK1_SFT 18 +#define NAU8822_PLLK1_MASK 0x3F + +/* NAU8822_REG_PLL_K2 (0x26) */ +#define NAU8822_PLLK2_SFT 9 +#define NAU8822_PLLK2_MASK 0x1FF + +/* NAU8822_REG_PLL_K3 (0x27) */ +#define NAU8822_PLLK3_MASK 0x1FF + +/* System Clock Source */ +enum { + NAU8822_CLK_MCLK, + NAU8822_CLK_PLL, +}; + +struct nau8822_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +/* Codec Private Data */ +struct nau8822 { + struct device *dev; + struct regmap *regmap; + int mclk_idx; + struct nau8822_pll pll; + int sysclk; + int div_id; +}; + +#endif /* __NAU8822_H__ */ -- 2.18.0

3 5

[alsa-devel] Applied "ASoC: sta32x: Add support for XTI clock" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: sta32x: Add support for XTI clock 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 fce9ec954a8af7e04cbf5b9daa8bec9c1df5cfe6 Mon Sep 17 00:00:00 2001 From: Daniel Mack <daniel(a)zonque.org> Date: Wed, 17 Oct 2018 13:37:03 +0200 Subject: [PATCH] ASoC: sta32x: Add support for XTI clock The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/st,sta32x.txt | 6 ++++ sound/soc/codecs/sta32x.c | 28 +++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1097,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: pcm3168a: add hw constraint for capture channel
by Kuninori Morimoto 19 Oct '18

19 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> LEFT_J / I2S only can use TDM. commit 594680ea4a394 ("ASoC: pcm3168a: add hw constraint for channel") commit 3809688980205 ("ASoC: pcm3168a: add HW constraint for non RIGHT_J") added channel constraint for it, but, it was only for playback. This patch adds constraint for capture. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 63aa025..52cc950 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -529,11 +529,17 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, break; case PCM3168A_FMT_LEFT_J: sample_min = 24; - channel_max = 8; + if (tx) + channel_max = 8; + else + channel_max = 6; break; case PCM3168A_FMT_I2S: sample_min = 24; - channel_max = 8; + if (tx) + channel_max = 8; + else + channel_max = 6; break; default: sample_min = 24; @@ -559,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { }; static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = { + .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt_adc, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params -- 2.7.4

2 2

[alsa-devel] Applied "ASoC: rsnd: enable TDM settings for SSI parent" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: rsnd: enable TDM settings for SSI parent 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 6817d7593f3e3c8a9c11e7a07cb5646c70371f0a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 17 Oct 2018 01:55:37 +0000 Subject: [PATCH] ASoC: rsnd: enable TDM settings for SSI parent Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..b42a0e0feab7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -399,6 +399,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->sys_delay) cr_own |= DEL; + /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: rsnd: tidyup SSICR::SWSP for TDM
by Kuninori Morimoto 19 Oct '18

19 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car datasheet is indicating that WS output settings of SSICR::SWSP is inverted on TDM mode from non TDM mode settings. But, it is meaning that TDM should use 0 here. Without this patch, sound input/output 1ch will be 2ch, 2ch will be 3ch ..., be jumbled on I2S + TDM settings. This patch fixup it. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index b42a0e0..fcb4df2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -392,7 +392,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->bit_clk_inv) cr_own |= SCKP; - if (rdai->frm_clk_inv ^ is_tdm) + if (rdai->frm_clk_inv && !is_tdm) cr_own |= SWSP; if (rdai->data_alignment) cr_own |= SDTA; -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98927: minor white space clean up
by Dan Carpenter 19 Oct '18

19 Oct '18

I just added a couple missing tabs. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3ab96ee7bd3c..3fa1c3ca6d37 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -180,14 +180,14 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); - return ret; + return ret; } } if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); - return ret; + return ret; } } }

2 2

[alsa-devel] [PATCH 0/4] ASoC: stm32: sai: add mclk clock provider
by Olivier Moysan 19 Oct '18

19 Oct '18

This patch set allows to use STM32 SAI interface as a master clock provider. The Cirrus cs42l51 codec patch gives the example of the source code required on codec side, to use SAI mclk provider. These patches allow to cover the following use cases: - SAI is master or slave and is master clock provider - SAI is master or slave and is not master clock provider (another sai provides the mclk clock) Implementation overview: - mclk configuration SAI is registered in clock framework: - either as a mclk provider (optional in DT). SAI generates mclk when requested by DAPM - or as a mclk consumer (optional in DT). cs42l51 codec is configured as a mclk consumer (optional in DT) - mclk power A DAPM is registered in codec to handle mclk enable/disable (DAPM route is set in soundcard node) - mclk rate: mclk rate is computed at runtime from stream rate and mclk-fs ratio provided in DT mclk rate is propagated through sysclk callback to the SAI which is mclk consumer. The mclk rate is set exclusively to avoid concurrent rate requests on SAI master. Regards Olivier Olivier Moysan (4): ASoC: dt-bindings: add mclk provider support to stm32 sai ASoC: dt-bindings: add mclk support to cs42l51 ASoC: stm32: sai: set sai as mclk clock provider ASoC: cs42l51: add mclk support .../devicetree/bindings/sound/cs42l51.txt | 17 ++ .../devicetree/bindings/sound/st,stm32-sai.txt | 7 + .../devicetree/bindings/trivial-devices.txt | 1 - sound/soc/codecs/cs42l51.c | 8 + sound/soc/stm/stm32_sai.h | 3 + sound/soc/stm/stm32_sai_sub.c | 275 ++++++++++++++++++--- 6 files changed, 274 insertions(+), 37 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt -- 2.7.4

2 12

[alsa-devel] Applied "ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer 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 2a2aefa41ce48ace8e1e963cb10c3f5ff43aa994 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 13:25:15 +0100 Subject: [PATCH] ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer The wm_adsp_buffer struct is the control header of a circular buffer used to transfer data from the firmware over the control interface to an ALSA compressed stream. The original names of the fields pointing to the data buffer were based on ADSP2V2 memory layout where they correspond to {XM, XM, YM}. But this circular buffer could be used on other types of DSP core that have different memory region types. Also the names and description of the size fields were not very clear. The field names and descriptions have been changed to be generic and not imply any particular memory types. This patch updates the wm_adsp driver to the new field names. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..7ae10c632614 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate 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 e3a360b8cdede74d25807fc405e5d8bfb025692f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 13:25:16 +0100 Subject: [PATCH] ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate Increase the address value width in the debug log from 4 digits to 8 digits to allow for DSP cores with larger memory address ranges. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7ae10c632614..a53dc174bbf0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) region->cumulative_size = offset; adsp_dbg(buf->dsp, - "region=%d type=%d base=%04x off=%04x size=%04x\n", + "region=%d type=%d base=%08x off=%08x size=%08x\n", i, region->mem_type, region->base_addr, region->offset, region->cumulative_size); } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: tegra_sgtl5000: fix device_node refcounting" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: tegra_sgtl5000: fix device_node refcounting 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 a85227da2dcc291b762c8482a505bc7d0d2d4b07 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Date: Tue, 16 Oct 2018 12:47:29 +0200 Subject: [PATCH] ASoC: tegra_sgtl5000: fix device_node refcounting Similar to the following: commit 4321723648b0 ("ASoC: tegra_alc5632: fix device_node refcounting") commit 7c5dfd549617 ("ASoC: tegra: fix device_node refcounting") Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/tegra/tegra_sgtl5000.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: sta32x: Add support for XTI clock" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: sta32x: Add support for XTI clock 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 fce9ec954a8af7e04cbf5b9daa8bec9c1df5cfe6 Mon Sep 17 00:00:00 2001 From: Daniel Mack <daniel(a)zonque.org> Date: Wed, 17 Oct 2018 13:37:03 +0200 Subject: [PATCH] ASoC: sta32x: Add support for XTI clock The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/st,sta32x.txt | 6 ++++ sound/soc/codecs/sta32x.c | 28 +++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1097,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: rsnd: enable TDM settings for SSI parent" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: rsnd: enable TDM settings for SSI parent 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 6817d7593f3e3c8a9c11e7a07cb5646c70371f0a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 17 Oct 2018 01:55:37 +0000 Subject: [PATCH] ASoC: rsnd: enable TDM settings for SSI parent Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..b42a0e0feab7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -399,6 +399,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->sys_delay) cr_own |= DEL; + /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
by Richard Fitzgerald 19 Oct '18

19 Oct '18

The wm_adsp_buffer struct is the control header of a circular buffer used to transfer data from the firmware over the control interface to an ALSA compressed stream. The original names of the fields pointing to the data buffer were based on ADSP2V2 memory layout where they correspond to {XM, XM, YM}. But this circular buffer could be used on other types of DSP core that have different memory region types. Also the names and description of the size fields were not very clear. The field names and descriptions have been changed to be generic and not imply any particular memory types. This patch updates the wm_adsp driver to the new field names. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..7ae10c632614 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; -- 2.11.0

1 1

[alsa-devel] AGL / Questions about the upcoming features of Alsa
by Thierry Bultel 19 Oct '18

19 Oct '18

Hi, My name is Thierry Bultel from Iot.bzh, and I am part of the responsible team for the audio framework (aka, 4A) in Automotive Grade Linux. For the ones interested in the audio architecture in AGL (and the ones that will read my list of questions below), these are useful links to follow: New AVIRT driver in replacement of SND_ALOOP: https://iot.bzh/download/public/2018/amm-dresden/ALSA_Virtual_Dynamic_Sound… 4A framework kickstart: https://iot.bzh/download/public/2018/amm-dresden/4a-kickstart-Loic-IoTbzh-D… In AGL, we have used the Alsa framework intensively. I wish I could meet the Alsa team next week in Edimburg but unfortunately I will not be available at that date. Other colleagues of mine will be there, however. Here are some points that I would like to be addressed at this meeting: - ioplug PCMs: There are known issues about potential (effective !) deadlocks when using ioplug PCMs. We have worked around this by setting LIBASOUND_THREAD_SAFE=0 Is there any pending work to avoid doing this and fix the bug ? - capture dispatching According to the experts, what would be the most effective way to implement a capture dispatcher ? The need is to have several applications listening to the same microphone. Is JACK the most suitable solution (at this time) ? (more precisely, this could behave as a 1to1 switch or 1toN, by duplicating the audio stream). A kernel solution based on AVIRT with routing could easily be done, too. - Documentation: Is there any pending rework of the documentation for having it a little bit more detailed ? For instance, having a page explaining the roles of period_size and buffer_size, would be great, since it impossible to perform sound playback correctly in an application, without understanding those notions. - Could something like a dynamic setting of the parameters of the rate converter plugin, become a reality one day ? - What is you overall mind about the AVIRT driver that we (with Fiberdyne) are developing ? - Is there a release features roadmap for Alsa (kernel & library) ? - What is your feeling about PipeWire, which could eventually replace PulseAudio (and seems to also invade the JACK land) ? Thanks ! Thierry Bultel

1 0

[alsa-devel] Audio miniconference schedule
by Mark Brown 18 Oct '18

18 Oct '18

As previously announced there's an audio miniconference happening on the 21st in Edinburgh, meeting around 9am and getting started once enough people have arrived (Cirrus people, please let me know if we need to start later, some people have said they'll be arriving late so we won't wait for everyone): https://blog.sirena.org.uk/2018/09/19/2018-linux-audio-miniconfernce (yes there is a typo in the URL). A number of people have proposed topics so I just went through them and pulled them into a proposed agenda, revised again after further feedback: - Introductions - Community governence issues (Takashi) - Virtualization (Liam) - AVB (Liam) - Testing (Takashi) - PCM improvements (Takashi) - Granularity - Timestamping - SoundWire (Liam) - Integrating the ALSA control core (Sakamoto-san) - Obsolete 'dimen' member in container for information to an element - Remove limitation on a container for value array to an element - User-defined element set left by finished process - Expand restriction to the number of user-defined control element set per sound card - Another approach to produce language bindings by GObject Introspection (Sakamoto-san) - ASoC rate domains/core refactoring (Charles) - ASoC/ALSA core merging (Takashi) - DSPs (Liam) - Sound Open Firmware (Liam) - Byte controls (Vinod/Patrick) - Calibration loading (Patrick) - Exposing graph at runtime (Patrick) - Batching commands (Patrick) - AGL (Stephane/Ronan) I hope I've not missed anything people suggested, the schedule is now looking pretty full but I'm hoping some of these topics can be relatively brief. I've dropped Y2038 from the expliit schedule, since Arnd Bergmann (who's been driving a lot of the general Y2038 work) will be in Edinburgh and may be free to join I'm suggesting that if he's free we cover that either first thing or just after lunch and otherwise Takashi and I will try to coordinate with him during the Maintainer's Summit on Monday (unless others have a great enthusiasm for the subject). What do people think? Any other topics? There's still time to sign up if you're interested in attending, you can sign up right up to the day of the event though registering early would help a lot with logistics.

1 0

[alsa-devel] [PATCH] conf/ucm/Dell-WD15-Dock: Fix incorrect device names
by Takashi Iwai 18 Oct '18

18 Oct '18

The device name string for Dell WD15 (and its variants) dock is set as "WD15Dock", while the actual device name to be used is "Dock". Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1112292 Fixes: 8ebb40c96970 ("conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/conf/ucm/Dell-WD15-Dock/HiFi.conf | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf index b07f34400086..1b6bd21237f1 100644 --- a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf +++ b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf @@ -3,7 +3,7 @@ SectionDevice."Headphone" { Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,0" + PlaybackPCM "hw:Dock,0" } } @@ -12,7 +12,7 @@ SectionDevice."LineOut" { Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,1" + PlaybackPCM "hw:Dock,1" } } @@ -21,6 +21,6 @@ SectionDevice."Mic" { Value { CaptureChannels "2" - CapturePCM "hw:WD15Dock,0" + CapturePCM "hw:Dock,0" } } -- 2.19.0

1 0

[alsa-devel] [PATCH - Intervals 1/1] interval: Interpret (x x+1] correctly and return x+1
by twischer＠de.adit-jv.com 18 Oct '18

18 Oct '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this change an interval of (x x+1] will be interpreted as an empty interval but the right value would be x+1. This leads to a failing snd_pcm_hw_params() call which returns -EINVAL. An example issue log is given in the following: snd_pcm_hw_params failed with err -22 (Invalid argument) ACCESS: MMAP_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 16 CHANNELS: 1 RATE: 16000 PERIOD_TIME: (15999 16000] PERIOD_SIZE: (255 256] PERIOD_BYTES: (510 512] PERIODS: [2 3) BUFFER_TIME: 32000 BUFFER_SIZE: 512 BUFFER_BYTES: 1024 In case of (x x+1) we have to interpret it anyway as a single value of x to compensate rounding issues. For example the period size will result in an interval of (352 353) when the period time is 16ms and the sample rate 22050 Hz (16ms * 22,05 kHz = 352,8 frames). But 352 has to be chosen to allow a buffer size of 705 (32ms * 22,05 kHz = 705,6 frames) which has to be >= 2x period size to avoid Xruns. The buffer size will not end up with an interval of (705 706) similar to the period size because snd_pcm_rate_hw_refine_cchange() calls snd_interval_floor() for the buffer size. Therefore this value will be interpreted as an integer interval instead of a real interval further on. This issue seems to exist since the change of 9bb985c38 ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/interval_inline.h b/src/pcm/interval_inline.h index a68e292..5e59d72 100644 --- a/src/pcm/interval_inline.h +++ b/src/pcm/interval_inline.h @@ -51,12 +51,14 @@ INTERVAL_INLINE int snd_interval_single(const snd_interval_t *i) { assert(!snd_interval_empty(i)); return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } INTERVAL_INLINE int snd_interval_value(const snd_interval_t *i) { assert(snd_interval_single(i)); + if (!i->openmax) + return i->max; return i->min; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: doc: Brush up the old writing-an-alsa-driver
by Takashi Iwai 18 Oct '18

18 Oct '18

Slightly brushing up and throw the old dust away from my ancient writing-an-alsa-driver document. The contents aren't changed so much but the obsoleted parts are dropped. Also, remove the date and the version number. It's useless. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a patch included in my devres patchset. Since this is an independent fix for documentation, let's apply this at first. .../kernel-api/writing-an-alsa-driver.rst | 307 +++++++++--------- 1 file changed, 149 insertions(+), 158 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index a0b268466cb1..b60a26807420 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3,8 +3,6 @@ Writing an ALSA Driver ====================== :Author: Takashi Iwai <tiwai(a)suse.de> -:Date: Oct 15, 2007 -:Edition: 0.3.7 Preface ======= @@ -21,11 +19,6 @@ explain the general topic of linux kernel coding and doesn't cover low-level driver implementation details. It only describes the standard way to write a PCI sound driver on ALSA. -If you are already familiar with the older ALSA ver.0.5.x API, you can -check the drivers such as ``sound/pci/es1938.c`` or -``sound/pci/maestro3.c`` which have also almost the same code-base in -the ALSA 0.5.x tree, so you can compare the differences. - This document is still a draft version. Any feedback and corrections, please!! @@ -35,24 +28,7 @@ File Tree Structure General ------- -The ALSA drivers are provided in two ways. - -One is the trees provided as a tarball or via cvs from the ALSA's ftp -site, and another is the 2.6 (or later) Linux kernel tree. To -synchronize both, the ALSA driver tree is split into two different -trees: alsa-kernel and alsa-driver. The former contains purely the -source code for the Linux 2.6 (or later) tree. This tree is designed -only for compilation on 2.6 or later environment. The latter, -alsa-driver, contains many subtle files for compiling ALSA drivers -outside of the Linux kernel tree, wrapper functions for older 2.2 and -2.4 kernels, to adapt the latest kernel API, and additional drivers -which are still in development or in tests. The drivers in alsa-driver -tree will be moved to alsa-kernel (and eventually to the 2.6 kernel -tree) when they are finished and confirmed to work fine. - -The file tree structure of ALSA driver is depicted below. Both -alsa-kernel and alsa-driver have almost the same file structure, except -for “core” directory. It's named as “acore” in alsa-driver tree. +The file tree structure of ALSA driver is depicted below. :: @@ -61,14 +37,11 @@ for “core” directory. It's named as “acore” in alsa-driver tree. /oss /seq /oss - /instr - /ioctl32 /include /drivers /mpu401 /opl3 /i2c - /l3 /synth /emux /pci @@ -80,6 +53,7 @@ for “core” directory. It's named as “acore” in alsa-driver tree. /sparc /usb /pcmcia /(cards) + /soc /oss @@ -99,13 +73,6 @@ directory. The rawmidi OSS emulation is included in the ALSA rawmidi code since it's quite small. The sequencer code is stored in ``core/seq/oss`` directory (see `below <#core-seq-oss>`__). -core/ioctl32 -~~~~~~~~~~~~ - -This directory contains the 32bit-ioctl wrappers for 64bit architectures -such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures, -these are not compiled. - core/seq ~~~~~~~~ @@ -119,11 +86,6 @@ core/seq/oss This contains the OSS sequencer emulation codes. -core/seq/instr -~~~~~~~~~~~~~~ - -This directory contains the modules for the sequencer instrument layer. - include directory ----------------- @@ -161,11 +123,6 @@ Although there is a standard i2c layer on Linux, ALSA has its own i2c code for some cards, because the soundcard needs only a simple operation and the standard i2c API is too complicated for such a purpose. -i2c/l3 -~~~~~~ - -This is a sub-directory for ARM L3 i2c. - synth directory --------------- @@ -209,11 +166,19 @@ The PCMCIA, especially PCCard drivers will go here. CardBus drivers will be in the pci directory, because their API is identical to that of standard PCI cards. +soc directory +------------- + +This directory contains the codes for ASoC (ALSA System on Chip) +layer including ASoC core, codec and machine drivers. + oss directory ------------- -The OSS/Lite source files are stored here in Linux 2.6 (or later) tree. -In the ALSA driver tarball, this directory is empty, of course :) +Here contains OSS/Lite codes. +All codes have been deprecated except for dmasound on m68k as of +writing this. + Basic Flow for PCI Drivers ========================== @@ -352,10 +317,8 @@ to details explained in the following section. /* (3) */ err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; /* (4) */ strcpy(card->driver, "My Chip"); @@ -368,22 +331,23 @@ to details explained in the following section. /* (6) */ err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; /* (7) */ pci_set_drvdata(pci, card); dev++; return 0; + + error: + snd_card_free(card); + return err; } /* destructor -- see the "Destructor" sub-section */ static void snd_mychip_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } @@ -445,14 +409,26 @@ In this part, the PCI resources are allocated. struct mychip *chip; .... err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; The details will be explained in the section `PCI Resource Management`_. +When something goes wrong, the probe function needs to deal with the +error. In this example, we have a single error handling path placed +at the end of the function. + +:: + + error: + snd_card_free(card); + return err; + +Since each component can be properly freed, the single +:c:func:`snd_card_free()` call should suffice in most cases. + + 4) Set the driver ID and name strings. ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -486,10 +462,8 @@ too. :: err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; Will be explained in the section `Management of Cards and Components`_, too. @@ -513,14 +487,13 @@ The destructor, remove callback, simply releases the card instance. Then the ALSA middle layer will release all the attached components automatically. -It would be typically like the following: +It would be typically just :c:func:`calling snd_card_free()`: :: static void snd_mychip_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } @@ -546,7 +519,7 @@ in the source file. If the code is split into several files, the files without module options don't need them. In addition to these headers, you'll need ``<linux/interrupt.h>`` for -interrupt handling, and ``<asm/io.h>`` for I/O access. If you use the +interrupt handling, and ``<linux/io.h>`` for I/O access. If you use the :c:func:`mdelay()` or :c:func:`udelay()` functions, you'll need to include ``<linux/delay.h>`` too. @@ -720,6 +693,13 @@ function, which will call the real destructor. where :c:func:`snd_mychip_free()` is the real destructor. +The demerit of this method is the obviously more amount of codes. +The merit is, however, you can trigger the own callback at registering +and disconnecting the card via setting in snd_device_ops. +About the registering and disconnecting the card, see the subsections +below. + + Registration and Release ------------------------ @@ -905,10 +885,8 @@ Resource Allocation ------------------- The allocation of I/O ports and irqs is done via standard kernel -functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And -these resources must be released in the destructor function (see below). -Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI -like in ALSA 0.5.x. +functions. These resources must be released in the destructor +function (see below). Now assume that the PCI device has an I/O port with 8 bytes and an interrupt. Then :c:type:`struct mychip <mychip>` will have the @@ -1064,7 +1042,8 @@ and the allocation would be like below: :: - if ((err = pci_request_regions(pci, "My Chip")) < 0) { + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { kfree(chip); return err; } @@ -1086,6 +1065,21 @@ and the corresponding destructor would be: .... } +Of course, a modern way with :c:func:`pci_iomap()` will make things a +bit easier, too. + +:: + + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + return err; + } + chip->iobase_virt = pci_iomap(pci, 0, 0); + +which is paired with :c:func:`pci_iounmap()` at destructor. + + PCI Entries ----------- @@ -1154,13 +1148,6 @@ And at last, the module entries: Note that these module entries are tagged with ``__init`` and ``__exit`` prefixes. -Oh, one thing was forgotten. If you have no exported symbols, you need -to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). - -:: - - EXPORT_NO_SYMBOLS; - That's all! PCM Interface @@ -2113,6 +2100,16 @@ non-contiguous buffers. The mmap calls this callback to get the page address. Some examples will be explained in the later section `Buffer and Memory Management`_, too. +mmap calllback +~~~~~~~~~~~~~~ + +This is another optional callback for controlling mmap behavior. +Once when defined, PCM core calls this callback when a page is +memory-mapped instead of dealing via the standard helper. +If you need special handling (due to some architecture or +device-specific issues), implement everything here as you like. + + PCM Interrupt Handler --------------------- @@ -2370,6 +2367,27 @@ to define the inverse rule: hw_rule_format_by_channels, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1); +One typical usage of the hw constraints is to align the buffer size +with the period size. As default, ALSA PCM core doesn't enforce the +buffer size to be aligned with the period size. For example, it'd be +possible to have a combination like 256 period bytes with 999 buffer +bytes. + +Many device chips, however, require the buffer to be a multiple of +periods. In such a case, call +:c:func:`snd_pcm_hw_constraint_integer()` for +``SNDRV_PCM_HW_PARAM_PERIODS``. + +:: + + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + +This assures that the number of periods is integer, hence the buffer +size is aligned with the period size. + +The hw constraint is a very much powerful mechanism to define the +preferred PCM configuration, and there are relevant helpers. I won't give more details here, rather I would like to say, “Luke, use the source.” @@ -3712,7 +3730,14 @@ example, for an intermediate buffer. Since the allocated pages are not contiguous, you need to set the ``page`` callback to obtain the physical address at every offset. -The implementation of ``page`` callback would be like this: +The easiest way to achieve it would be to use +:c:func:`snd_pcm_lib_alloc_vmalloc_buffer()` for allocating the buffer +via :c:func:`vmalloc()`, and set :c:func:`snd_pcm_sgbuf_ops_page()` to +the ``page`` callback. At release, you need to call +:c:func:`snd_pcm_lib_free_vmalloc_buffer()`. + +If you want to implementation the ``page`` manually, it would be like +this: :: @@ -3848,7 +3873,9 @@ Power Management If the chip is supposed to work with suspend/resume functions, you need to add power-management code to the driver. The additional code for -power-management should be ifdef-ed with ``CONFIG_PM``. +power-management should be ifdef-ed with ``CONFIG_PM``, or annotated +with __maybe_unused attribute; otherwise the compiler will complain +you. If the driver *fully* supports suspend/resume that is, the device can be properly resumed to its state when suspend was called, you can set the @@ -3879,18 +3906,16 @@ the case of PCI drivers, the callbacks look like below: :: - #ifdef CONFIG_PM - static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) + static int __maybe_unused snd_my_suspend(struct device *dev) { .... /* do things for suspend */ return 0; } - static int snd_my_resume(struct pci_dev *pci) + static int __maybe_unused snd_my_resume(struct device *dev) { .... /* do things for suspend */ return 0; } - #endif The scheme of the real suspend job is as follows. @@ -3909,18 +3934,14 @@ The scheme of the real suspend job is as follows. 6. Stop the hardware if necessary. -7. Disable the PCI device by calling - :c:func:`pci_disable_device()`. Then, call - :c:func:`pci_save_state()` at last. - A typical code would be like: :: - static int mychip_suspend(struct pci_dev *pci, pm_message_t state) + static int __maybe_unused mychip_suspend(struct device *dev) { /* (1) */ - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct mychip *chip = card->private_data; /* (2) */ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -3932,9 +3953,6 @@ A typical code would be like: snd_mychip_save_registers(chip); /* (6) */ snd_mychip_stop_hardware(chip); - /* (7) */ - pci_disable_device(pci); - pci_save_state(pci); return 0; } @@ -3943,44 +3961,35 @@ The scheme of the real resume job is as follows. 1. Retrieve the card and the chip data. -2. Set up PCI. First, call :c:func:`pci_restore_state()`. Then - enable the pci device again by calling - :c:func:`pci_enable_device()`. Call - :c:func:`pci_set_master()` if necessary, too. +2. Re-initialize the chip. -3. Re-initialize the chip. +3. Restore the saved registers if necessary. -4. Restore the saved registers if necessary. +4. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. -5. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. +5. Restart the hardware (if any). -6. Restart the hardware (if any). - -7. Call :c:func:`snd_power_change_state()` with +6. Call :c:func:`snd_power_change_state()` with ``SNDRV_CTL_POWER_D0`` to notify the processes. A typical code would be like: :: - static int mychip_resume(struct pci_dev *pci) + static int __maybe_unused mychip_resume(struct pci_dev *pci) { /* (1) */ - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct mychip *chip = card->private_data; /* (2) */ - pci_restore_state(pci); - pci_enable_device(pci); - pci_set_master(pci); - /* (3) */ snd_mychip_reinit_chip(chip); - /* (4) */ + /* (3) */ snd_mychip_restore_registers(chip); - /* (5) */ + /* (4) */ snd_ac97_resume(chip->ac97); - /* (6) */ + /* (5) */ snd_mychip_restart_chip(chip); - /* (7) */ + /* (6) */ snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -4046,15 +4055,14 @@ And next, set suspend/resume callbacks to the pci_driver. :: + static SIMPLE_DEV_PM_OPS(snd_my_pm_ops, mychip_suspend, mychip_resume); + static struct pci_driver driver = { .name = KBUILD_MODNAME, .id_table = snd_my_ids, .probe = snd_my_probe, .remove = snd_my_remove, - #ifdef CONFIG_PM - .suspend = snd_my_suspend, - .resume = snd_my_resume, - #endif + .driver.pm = &snd_my_pm_ops, }; Module Parameters @@ -4078,7 +4086,7 @@ variables, instead. ``enable`` option is not always necessary in this case, but it would be better to have a dummy option for compatibility. The module parameters must be declared with the standard -``module_param()()``, ``module_param_array()()`` and +``module_param()``, ``module_param_array()`` and :c:func:`MODULE_PARM_DESC()` macros. The typical coding would be like below: @@ -4094,15 +4102,14 @@ The typical coding would be like below: module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); -Also, don't forget to define the module description, classes, license -and devices. Especially, the recent modprobe requires to define the +Also, don't forget to define the module description and the license. +Especially, the recent modprobe requires to define the module license as GPL, etc., otherwise the system is shown as “tainted”. :: - MODULE_DESCRIPTION("My Chip"); + MODULE_DESCRIPTION("Sound driver for My Chip"); MODULE_LICENSE("GPL"); - MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}"); How To Put Your Driver Into ALSA Tree @@ -4117,21 +4124,17 @@ a question now: how to put my own driver into the ALSA driver tree? Here Suppose that you create a new PCI driver for the card “xyz”. The card module name would be snd-xyz. The new driver is usually put into the -alsa-driver tree, ``alsa-driver/pci`` directory in the case of PCI -cards. Then the driver is evaluated, audited and tested by developers -and users. After a certain time, the driver will go to the alsa-kernel -tree (to the corresponding directory, such as ``alsa-kernel/pci``) and -eventually will be integrated into the Linux 2.6 tree (the directory -would be ``linux/sound/pci``). +alsa-driver tree, ``sound/pci`` directory in the case of PCI +cards. In the following sections, the driver code is supposed to be put into -alsa-driver tree. The two cases are covered: a driver consisting of a +Linux kernel tree. The two cases are covered: a driver consisting of a single source file and one consisting of several source files. Driver with A Single Source File -------------------------------- -1. Modify alsa-driver/pci/Makefile +1. Modify sound/pci/Makefile Suppose you have a file xyz.c. Add the following two lines @@ -4160,52 +4163,43 @@ Driver with A Single Source File For the details of Kconfig script, refer to the kbuild documentation. -3. Run cvscompile script to re-generate the configure script and build - the whole stuff again. - Drivers with Several Source Files --------------------------------- Suppose that the driver snd-xyz have several source files. They are -located in the new subdirectory, pci/xyz. +located in the new subdirectory, sound/pci/xyz. -1. Add a new directory (``xyz``) in ``alsa-driver/pci/Makefile`` as - below +1. Add a new directory (``sound/pci/xyz``) in ``sound/pci/Makefile`` + as below :: - obj-$(CONFIG_SND) += xyz/ + obj-$(CONFIG_SND) += sound/pci/xyz/ -2. Under the directory ``xyz``, create a Makefile +2. Under the directory ``sound/pci/xyz``, create a Makefile :: - ifndef SND_TOPDIR - SND_TOPDIR=../.. - endif - - include $(SND_TOPDIR)/toplevel.config - include $(SND_TOPDIR)/Makefile.conf - snd-xyz-objs := xyz.o abc.o def.o - obj-$(CONFIG_SND_XYZ) += snd-xyz.o - include $(SND_TOPDIR)/Rules.make - 3. Create the Kconfig entry This procedure is as same as in the last section. -4. Run cvscompile script to re-generate the configure script and build - the whole stuff again. Useful Functions ================ :c:func:`snd_printk()` and friends ---------------------------------------- +---------------------------------- + +.. note:: This subsection describes a few helper functions for +decorating a bit more on the standard :c:func:`printk()` & co. +However, in general, the use of such helpers is no longer recommended. +If possible, try to stick with the standard functions like +:c:func:`dev_err()` or :c:func:`pr_err()`. ALSA provides a verbose version of the :c:func:`printk()` function. If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function @@ -4221,13 +4215,10 @@ just like :c:func:`snd_printk()`. If the ALSA is compiled without the debugging flag, it's ignored. :c:func:`snd_printdd()` is compiled in only when -``CONFIG_SND_DEBUG_VERBOSE`` is set. Please note that -``CONFIG_SND_DEBUG_VERBOSE`` is not set as default even if you configure -the alsa-driver with ``--with-debug=full`` option. You need to give -explicitly ``--with-debug=detect`` option instead. +``CONFIG_SND_DEBUG_VERBOSE`` is set. :c:func:`snd_BUG()` ------------------------- +------------------- It shows the ``BUG?`` message and stack trace as well as :c:func:`snd_BUG_ON()` at the point. It's useful to show that a @@ -4236,7 +4227,7 @@ fatal error happens there. When no debug flag is set, this macro is ignored. :c:func:`snd_BUG_ON()` ----------------------------- +---------------------- :c:func:`snd_BUG_ON()` macro is similar with :c:func:`WARN_ON()` macro. For example, snd_BUG_ON(!pointer); or -- 2.19.0

3 3

[alsa-devel] [PATCH v2] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new
by Kuninori Morimoto 18 Oct '18

18 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. There is no detail explanation for these register on datasheet, but, judging from its behavior, it seems these register needs clock to read. Otherwise, I2C returns error. Therefor, amixer command will be failed under non working timing. These information are needed for driver, not for user. So, let's remove these from snd_kcontrol_new. Without this patch, user often fail amixer command. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - removed directly, instead of using #if 0 - tidyup git-log sound/soc/codecs/pcm3168a.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 52cc950..e1c3a18 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), - SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), - SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), - SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), - SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +172,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), - SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), - SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), - SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new
by Kuninori Morimoto 18 Oct '18

18 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. There is no detail explanation of these register on datasheet, but, judging from it behavior, it seems these register needs clock to reading. Therefor, amixer command will be failed under non working timing. Let's ignore these from snd_kcontrol_new so far, but keep it under #if 0. Without this patch, amixer command offten failed and can't sound settings. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 52cc950..29eaf9d 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,13 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), +#if 0 + /* read-only status register needs clocks */ SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), +#endif SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +179,12 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), +#if 0 + /* read-only status register needs clocks */ SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), +#endif SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.7.4

2 2

[alsa-devel] [PATCH v2 0/9] ARM: dts: tegra: last minute fixes
by Marcel Ziswiler 18 Oct '18

18 Oct '18

This series comes with some last minutes fixes and further clean-up. Changes in v2: - Dropped "[PATCH v1 3/8] ARM: tegra: apalis/colibri_t30: fix hdmi regulator" as suggested by Russell et. al. - Added 2 new patches improving/fixing audio on Apalis TK1. Changes in v1: - Remove simple-panel compatible as suggested by Rob. Marcel Ziswiler (9): ARM: tegra: fix simple-panel compatibles ARM: tegra: apalis-tk1/colibri_t20/t30: eval/iris: fix regulator gpio enable ARM: tegra: colibri_t20: reorder pmic properties ARM: tegra: apalis-tk1: further regulator clean-up ARM: tegra: apalis_t30/tk1: annotate power I2C being on-module ARM: tegra: colibri_t30: further regulator clean-up ARM: tegra: apalis_t30: further regulator clean-up ASoC: tegra_sgtl5000: fix device_node refcounting ASoC: tegra_sgtl5000: fix platform name vs. of_node assignement arch/arm/boot/dts/tegra114-dalmore.dts | 3 +- arch/arm/boot/dts/tegra124-apalis-eval.dts | 4 +- arch/arm/boot/dts/tegra124-apalis-v1.2-eval.dts | 4 +- arch/arm/boot/dts/tegra124-apalis-v1.2.dtsi | 17 +++--- arch/arm/boot/dts/tegra124-apalis.dtsi | 17 +++--- arch/arm/boot/dts/tegra124-venice2.dts | 2 +- arch/arm/boot/dts/tegra20-colibri-eval-v3.dts | 4 +- arch/arm/boot/dts/tegra20-colibri-iris.dts | 4 +- arch/arm/boot/dts/tegra20-colibri.dtsi | 4 +- arch/arm/boot/dts/tegra20-harmony.dts | 2 +- arch/arm/boot/dts/tegra20-medcom-wide.dts | 2 +- arch/arm/boot/dts/tegra20-paz00.dts | 2 +- arch/arm/boot/dts/tegra20-seaboard.dts | 2 +- arch/arm/boot/dts/tegra20-ventana.dts | 2 +- arch/arm/boot/dts/tegra30-apalis-eval.dts | 2 +- arch/arm/boot/dts/tegra30-apalis-v1.1-eval.dts | 4 +- arch/arm/boot/dts/tegra30-apalis-v1.1.dtsi | 72 +++++++++++-------------- arch/arm/boot/dts/tegra30-apalis.dtsi | 70 ++++++++++-------------- arch/arm/boot/dts/tegra30-cardhu.dtsi | 2 +- arch/arm/boot/dts/tegra30-colibri-eval-v3.dts | 4 +- arch/arm/boot/dts/tegra30-colibri.dtsi | 45 ++++++++-------- sound/soc/tegra/tegra_sgtl5000.c | 22 +++++++- 22 files changed, 143 insertions(+), 147 deletions(-) -- 2.14.4

4 11

Re: [alsa-devel] [PATCH 2/3] MIPS: Ci20: Enable SND_JZ4740_SOC driver
by Mathieu Malaterre 17 Oct '18

17 Oct '18

On Wed, Jun 6, 2018 at 9:38 PM Mathieu Malaterre <malat(a)debian.org> wrote: > > Update the Ci20's defconfig to enable the JZ4780's SND driver. > > Signed-off-by: Mathieu Malaterre <malat(a)debian.org> > --- > arch/mips/configs/ci20_defconfig | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/arch/mips/configs/ci20_defconfig b/arch/mips/configs/ci20_defconfig > index e1c14f6af824..0c08c7675b42 100644 > --- a/arch/mips/configs/ci20_defconfig > +++ b/arch/mips/configs/ci20_defconfig > @@ -104,6 +104,10 @@ CONFIG_REGULATOR=y > CONFIG_REGULATOR_DEBUG=y > CONFIG_REGULATOR_FIXED_VOLTAGE=y > # CONFIG_VGA_CONSOLE is not set > +CONFIG_SOUND=y > +CONFIG_SND=y > +CONFIG_SND_SOC=y > +CONFIG_SND_JZ4740_SOC=y technically I am also missing: CONFIG_SND_JZ4740_SOC_I2S=y > # CONFIG_HID is not set > # CONFIG_USB_SUPPORT is not set > CONFIG_MMC=y > -- > 2.11.0 >

1 0

Re: [alsa-devel] [PATCH 1/3] MIPS: jz4780: Allow access to jz4740-i2s
by Paul Burton 17 Oct '18

17 Oct '18

Hi Matthieu, On Wed, Jun 06, 2018 at 09:38:08PM +0200, Mathieu Malaterre wrote: > diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig > index 1a354a6b6e87..35d82d96e781 100644 > --- a/sound/soc/jz4740/Kconfig > +++ b/sound/soc/jz4740/Kconfig > @@ -1,20 +1,20 @@ > config SND_JZ4740_SOC > - tristate "SoC Audio for Ingenic JZ4740 SoC" > - depends on MACH_JZ4740 || COMPILE_TEST > + tristate "SoC Audio for Ingenic JZ4740/JZ4780 SoC" > + depends on MACH_JZ4740 || MACH_JZ4780 || COMPILE_TEST Perhaps this could be MACH_INGENIC, or even just MIPS? Thanks, Paul

2 4

[alsa-devel] [PATCH] ASoC: sta32x: Add support for XTI clock
by Daniel Mack 17 Oct '18

17 Oct '18

The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../devicetree/bindings/sound/st,sta32x.txt | 6 +++++ sound/soc/codecs/sta32x.c | 25 +++++++++++++++++++ 2 files changed, 31 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index a878c8d92e68..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -882,6 +884,15 @@ static int sta32x_probe(struct snd_soc_component *component) sta32x->component = component; + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -984,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1100,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- 2.17.2

2 1

[alsa-devel] Applied "ASoC: sun4i-i2s: Add compatibility with A64 codec I2S" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sun4i-i2s: Add compatibility with A64 codec I2S 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 66ecce3325383c8304063e7d5a30f4374ef5a33e Mon Sep 17 00:00:00 2001 From: Marcus Cooper <codekipper(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:05 -0700 Subject: [PATCH] ASoC: sun4i-i2s: Add compatibility with A64 codec I2S The I2S block used for the audio codec in the A64 differs from other 3 I2S modules in A64 and isn't compatible with H3. But it is very similar to what is found in A10(sun4i). However, its TX FIFO is located at a different address. Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/sun4i-i2s.txt | 2 ++ sound/soc/sunxi/sun4i-i2s.c | 21 +++++++++++++++++++ 2 files changed, 23 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt index b9d50d6cdef3..61e71c1729e0 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt @@ -10,6 +10,7 @@ Required properties: - "allwinner,sun6i-a31-i2s" - "allwinner,sun8i-a83t-i2s" - "allwinner,sun8i-h3-i2s" + - "allwinner,sun50i-a64-codec-i2s" - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the I2S interrupt. @@ -26,6 +27,7 @@ Required properties for the following compatibles: - "allwinner,sun6i-a31-i2s" - "allwinner,sun8i-a83t-i2s" - "allwinner,sun8i-h3-i2s" + - "allwinner,sun50i-a64-codec-i2s" - resets: phandle to the reset line for this codec Example: diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..c63d226e2436 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -961,6 +961,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { .field_rxchansel = REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2), }; +static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .has_slave_select_bit = true, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5), + .field_fmt_bclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6), + .field_fmt_lrclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7), + .field_fmt_mode = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1), + .field_txchanmap = REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31), + .field_rxchanmap = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31), + .field_txchansel = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2), + .field_rxchansel = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2), +}; + static int sun4i_i2s_init_regmap_fields(struct device *dev, struct sun4i_i2s *i2s) { @@ -1169,6 +1186,10 @@ static const struct of_device_id sun4i_i2s_match[] = { .compatible = "allwinner,sun8i-h3-i2s", .data = &sun8i_h3_i2s_quirks, }, + { + .compatible = "allwinner,sun50i-a64-codec-i2s", + .data = &sun50i_a64_codec_i2s_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun4i_i2s_match); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v5 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 17 Oct '18

17 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags - drop the change to sun8i_codec_clk_div Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 29 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 29 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 22 +- 14 files changed, 785 insertions(+), 81 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h -- 2.19.0

3 11

[alsa-devel] Applied "ASoC: sun8i-codec-analog: split regmap code into separate driver" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sun8i-codec-analog: split regmap code into separate driver 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 55b407f6468c481edc4a26b21c9f567ae57f01a4 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:07 -0700 Subject: [PATCH] ASoC: sun8i-codec-analog: split regmap code into separate driver It will be reused by sun50i-codec-analog later. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/Kconfig | 7 +- sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 +++++++++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 ++ sound/soc/sunxi/sun8i-codec-analog.c | 79 +------------------ 5 files changed, 119 insertions(+), 77 deletions(-) create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 22408bc2d6ec..83b770cdfdaa 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -23,7 +23,7 @@ config SND_SUN8I_CODEC config SND_SUN8I_CODEC_ANALOG tristate "Allwinner sun8i Codec Analog Controls Support" depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select REGMAP + select SND_SUN8I_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. @@ -45,4 +45,9 @@ config SND_SUN4I_SPDIF help Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. + +config SND_SUN8I_ADDA_PR_REGMAP + tristate + select REGMAP + endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 4a9ef67386ca..74b99d55cfca 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -4,3 +4,4 @@ obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o +obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c new file mode 100644 index 000000000000..e68ce9d2884d --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c @@ -0,0 +1,102 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver provides regmap to access to analog part of audio codec + * found on Allwinner A23, A31s, A33, H3 and A64 Socs + * + * Copyright 2016 Chen-Yu Tsai <wens(a)csie.org> + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul(a)gmail.com> + */ + +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "sun8i-adda-pr-regmap.h" + +/* Analog control register access bits */ +#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ +#define ADDA_PR_RESET BIT(28) +#define ADDA_PR_WRITE BIT(24) +#define ADDA_PR_ADDR_SHIFT 16 +#define ADDA_PR_ADDR_MASK GENMASK(4, 0) +#define ADDA_PR_DATA_IN_SHIFT 8 +#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) +#define ADDA_PR_DATA_OUT_SHIFT 0 +#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) + +/* regmap access bits */ +static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Read back value */ + *val = readl(base) & ADDA_PR_DATA_OUT_MASK; + + return 0; +} + +static int adda_reg_write(void *context, unsigned int reg, unsigned int val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Set data to write */ + tmp = readl(base); + tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); + tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; + writel(tmp, base); + + /* Set write bit to signal a write */ + writel(readl(base) | ADDA_PR_WRITE, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + return 0; +} + +static const struct regmap_config adda_pr_regmap_cfg = { + .name = "adda-pr", + .reg_bits = 5, + .reg_stride = 1, + .val_bits = 8, + .reg_read = adda_reg_read, + .reg_write = adda_reg_write, + .fast_io = true, + .max_register = 31, +}; + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base) +{ + return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg); +} +EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init); + +MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver"); +MODULE_AUTHOR("Vasily Khoruzhick <anarsoul(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sunxi-adda-pr"); diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h new file mode 100644 index 000000000000..a5ae95dfebc1 --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul(a)gmail.com> + */ + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base); diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 485e79f292c4..916a46bbc1c8 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -27,6 +27,8 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include "sun8i-adda-pr-regmap.h" + /* Codec analog control register offsets and bit fields */ #define SUN8I_ADDA_HP_VOLC 0x00 #define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7 @@ -120,81 +122,6 @@ #define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6 #define SUN8I_ADDA_ADC_AP_EN_ADCG 0 -/* Analog control register access bits */ -#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ -#define ADDA_PR_RESET BIT(28) -#define ADDA_PR_WRITE BIT(24) -#define ADDA_PR_ADDR_SHIFT 16 -#define ADDA_PR_ADDR_MASK GENMASK(4, 0) -#define ADDA_PR_DATA_IN_SHIFT 8 -#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) -#define ADDA_PR_DATA_OUT_SHIFT 0 -#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) - -/* regmap access bits */ -static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Read back value */ - *val = readl(base) & ADDA_PR_DATA_OUT_MASK; - - return 0; -} - -static int adda_reg_write(void *context, unsigned int reg, unsigned int val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Set data to write */ - tmp = readl(base); - tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); - tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; - writel(tmp, base); - - /* Set write bit to signal a write */ - writel(readl(base) | ADDA_PR_WRITE, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - return 0; -} - -static const struct regmap_config adda_pr_regmap_cfg = { - .name = "adda-pr", - .reg_bits = 5, - .reg_stride = 1, - .val_bits = 8, - .reg_read = adda_reg_read, - .reg_write = adda_reg_write, - .fast_io = true, - .max_register = 24, -}; - /* mixer controls */ static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = { SOC_DAPM_DOUBLE_R("DAC Playback Switch", @@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev) return PTR_ERR(base); } - regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg); + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); if (IS_ERR(regmap)) { dev_err(&pdev->dev, "Failed to create regmap\n"); return PTR_ERR(regmap); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path 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 af2c06c41970eb0110a03696fd7f8f07854719db Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:08 -0700 Subject: [PATCH] ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls The internal codec on Allwinner A64 is split into 2 parts. The analog path controls are routed through an embedded custom register bus accessed through the PRCM block just as on A23/A33/H3. Add a binding for this hardware. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/sun50i-codec-analog.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt diff --git a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt new file mode 100644 index 000000000000..4f8ad0e04d20 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt @@ -0,0 +1,12 @@ +* Allwinner A64 Codec Analog Controls + +Required properties: +- compatible: must be one of the following compatibles: + - "allwinner,sun50i-a64-codec-analog" +- reg: must contain the registers location and length + +Example: + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun50i-a64-codec-analog"; + reg = <0x01f015c0 0x4>; + }; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path 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 42371f327df0b8e9d479b929c5cd301846dd0f70 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:09 -0700 Subject: [PATCH] ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls The internal codec on A64 is split into 2 parts. The analog path controls are routed through an embedded custom register bus accessed through the PRCM block. Add an ASoC component driver for it. This should be tied to the codec audio card as an auxiliary device. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++++++++++ 3 files changed, 453 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 83b770cdfdaa..8a055ca1819a 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -28,6 +28,14 @@ config SND_SUN8I_CODEC_ANALOG Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. +config SND_SUN50I_CODEC_ANALOG + tristate "Allwinner sun50i Codec Analog Controls Support" + depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST + select SND_SUNXI_ADDA_PR_REGMAP + help + Say Y or M if you want to add support for the analog controls for + the codec embedded in Allwinner A64 SoC. + config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 74b99d55cfca..a86be340a076 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -3,5 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o +obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c new file mode 100644 index 000000000000..8f5f999df631 --- /dev/null +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -0,0 +1,444 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver supports the analog controls for the internal codec + * found in Allwinner's A64 SoC. + * + * Copyright (C) 2016 Chen-Yu Tsai <wens(a)csie.org> + * Copyright (C) 2017 Marcus Cooper <codekipper(a)gmail.com> + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul(a)gmail.com> + * + * Based on sun8i-codec-analog.c + * + */ + +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "sun8i-adda-pr-regmap.h" + +/* Codec analog control register offsets and bit fields */ +#define SUN50I_ADDA_HP_CTRL 0x00 +#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE 7 +#define SUN50I_ADDA_HP_CTRL_HPPA_EN 6 +#define SUN50I_ADDA_HP_CTRL_HPVOL 0 + +#define SUN50I_ADDA_OL_MIX_CTRL 0x01 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN 3 +#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL 2 +#define SUN50I_ADDA_OL_MIX_CTRL_DACL 1 +#define SUN50I_ADDA_OL_MIX_CTRL_DACR 0 + +#define SUN50I_ADDA_OR_MIX_CTRL 0x02 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP 3 +#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR 2 +#define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 +#define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 + +#define SUN50I_ADDA_LINEOUT_CTRL0 0x05 +#define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 +#define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 +#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL 5 +#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL 4 + +#define SUN50I_ADDA_LINEOUT_CTRL1 0x06 +#define SUN50I_ADDA_LINEOUT_CTRL1_VOL 0 + +#define SUN50I_ADDA_MIC1_CTRL 0x07 +#define SUN50I_ADDA_MIC1_CTRL_MIC1G 4 +#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN 3 +#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST 0 + +#define SUN50I_ADDA_MIC2_CTRL 0x08 +#define SUN50I_ADDA_MIC2_CTRL_MIC2G 4 +#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN 3 +#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST 0 + +#define SUN50I_ADDA_LINEIN_CTRL 0x09 +#define SUN50I_ADDA_LINEIN_CTRL_LINEING 0 + +#define SUN50I_ADDA_MIX_DAC_CTRL 0x0a +#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN 7 +#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN 6 +#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN 5 +#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN 4 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE 3 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE 2 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS 1 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS 0 + +#define SUN50I_ADDA_L_ADCMIX_SRC 0x0b +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN 3 +#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL 2 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL 1 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR 0 + +#define SUN50I_ADDA_R_ADCMIX_SRC 0x0c +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP 3 +#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR 2 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR 1 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL 0 + +#define SUN50I_ADDA_ADC_CTRL 0x0d +#define SUN50I_ADDA_ADC_CTRL_ADCREN 7 +#define SUN50I_ADDA_ADC_CTRL_ADCLEN 6 +#define SUN50I_ADDA_ADC_CTRL_ADCG 0 + +#define SUN50I_ADDA_HS_MBIAS_CTRL 0x0e +#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7 + +#define SUN50I_ADDA_JACK_MIC_CTRL 0x1d +#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5 + +/* mixer controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("DAC Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0), + SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0), +}; + +/* ADC mixer controls */ +static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Mixer Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0), + SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0), +}; + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale, + -450, 150, 0); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0), +); + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1); + +static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); + + +/* volume / mute controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { + SOC_SINGLE_TLV("Headphone Playback Volume", + SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0, + sun50i_codec_hp_vol_scale), + + SOC_DOUBLE("Headphone Playback Switch", + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE, + SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic2 Playback Volume", + SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* ADC */ + SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0, + sun50i_codec_out_mixer_pregain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL, + SUN50I_ADDA_LINEIN_CTRL_LINEING, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + SOC_SINGLE_TLV("Line Out Playback Volume", + SUN50I_ADDA_LINEOUT_CTRL1, + SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0, + sun50i_codec_lineout_vol_scale), + + SOC_DOUBLE("Line Out Playback Switch", + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LEN, + SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + +}; + +static const char * const sun50i_codec_hp_src_enum_text[] = { + "DAC", "Mixer", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum, + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPIS, + SUN50I_ADDA_MIX_DAC_CTRL_RHPIS, + sun50i_codec_hp_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_hp_src[] = { + SOC_DAPM_ENUM("Headphone Source Playback Route", + sun50i_codec_hp_src_enum), +}; + +static const char * const sun50i_codec_lineout_src_enum_text[] = { + "Stereo", "Mono Differential", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum, + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL, + SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL, + sun50i_codec_lineout_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { + SOC_DAPM_ENUM("Line Out Source Playback Route", + sun50i_codec_lineout_src_enum), +}; + +static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { + /* DAC */ + SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0), + /* ADC */ + SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0), + /* + * Due to this component and the codec belonging to separate DAPM + * contexts, we need to manually link the above widgets to their + * stream widgets at the card level. + */ + + SND_SOC_DAPM_MUX("Headphone Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src), + SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HP"), + + SND_SOC_DAPM_MUX("Line Out Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + + /* Microphone inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL, + SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0), + + /* Microphone input */ + SND_SOC_DAPM_INPUT("MIC2"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL, + SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0), + + /* Line input */ + SND_SOC_DAPM_INPUT("LINEIN"), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), + SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), +}; + +static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { + /* Left Mixer Routes */ + { "Left Mixer", "DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" }, + { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Right Mixer Routes */ + { "Right Mixer", "DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" }, + { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Left ADC Mixer Routes */ + { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" }, + { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" }, + { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* Right ADC Mixer Routes */ + { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" }, + { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" }, + { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* ADC Routes */ + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + /* Headphone Routes */ + { "Headphone Source Playback Route", "DAC", "Left DAC" }, + { "Headphone Source Playback Route", "DAC", "Right DAC" }, + { "Headphone Source Playback Route", "Mixer", "Left Mixer" }, + { "Headphone Source Playback Route", "Mixer", "Right Mixer" }, + { "Headphone Amp", NULL, "Headphone Source Playback Route" }, + { "HP", NULL, "Headphone Amp" }, + + /* Microphone Routes */ + { "Mic1 Amplifier", NULL, "MIC1"}, + + /* Microphone Routes */ + { "Mic2 Amplifier", NULL, "MIC2"}, + { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* Line-in Routes */ + { "Left Mixer", "Line In Playback Switch", "LINEIN" }, + { "Right Mixer", "Line In Playback Switch", "LINEIN" }, + { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" }, + + /* Line-out Routes */ + { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, + { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", + "Right Mixer" }, + { "LINEOUT", NULL, "Line Out Source Playback Route" }, +}; + +static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { + .controls = sun50i_a64_codec_controls, + .num_controls = ARRAY_SIZE(sun50i_a64_codec_controls), + .dapm_widgets = sun50i_a64_codec_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun50i_a64_codec_widgets), + .dapm_routes = sun50i_a64_codec_routes, + .num_dapm_routes = ARRAY_SIZE(sun50i_a64_codec_routes), +}; + +static const struct of_device_id sun50i_codec_analog_of_match[] = { + { + .compatible = "allwinner,sun50i-a64-codec-analog", + }, + {} +}; +MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match); + +static int sun50i_codec_analog_probe(struct platform_device *pdev) +{ + struct resource *res; + struct regmap *regmap; + void __iomem *base; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); + if (IS_ERR(regmap)) { + dev_err(&pdev->dev, "Failed to create regmap\n"); + return PTR_ERR(regmap); + } + + return devm_snd_soc_register_component(&pdev->dev, + &sun50i_codec_analog_cmpnt_drv, + NULL, 0); +} + +static struct platform_driver sun50i_codec_analog_driver = { + .driver = { + .name = "sun50i-codec-analog", + .of_match_table = sun50i_codec_analog_of_match, + }, + .probe = sun50i_codec_analog_probe, +}; +module_platform_driver(sun50i_codec_analog_driver); + +MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64"); +MODULE_AUTHOR("Vasily Khoruzhick <anarsoul(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-codec-analog"); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 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 7e95aac96b554193760aaeb64e263da58127bf27 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:10 -0700 Subject: [PATCH] ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 Allwinner A64 uses the same digital codec part as in A33, so we need to build this driver on ARM64 as well. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 8a055ca1819a..66aad0d3f9c7 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -12,7 +12,7 @@ config SND_SUN4I_CODEC config SND_SUN8I_CODEC tristate "Allwinner SUN8I audio codec" depends on OF - depends on MACH_SUN8I || COMPILE_TEST + depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST select REGMAP_MMIO help This option enables the digital part of the internal audio codec for -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v4 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 17 Oct '18

17 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 29 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 29 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 26 +- 14 files changed, 787 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h -- 2.19.0

2 14

[alsa-devel] [PATCH] ASoC: rsnd: enable TDM settings for SSI parent
by Kuninori Morimoto 17 Oct '18

17 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f..b42a0e0 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -400,6 +400,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own |= DEL; /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. */ @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: wm8904: fix spelling mistake "Inveting" -> "Inverting"
by Colin King 16 Oct '18

16 Oct '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in sound control text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 2a3e5fbd04e4..ef83dae6e5ae 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -870,7 +870,7 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, lin_text); static const struct snd_kcontrol_new lin_inv_mux = - SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum); static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" @@ -886,7 +886,7 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, rin_text); static const struct snd_kcontrol_new rin_inv_mux = - SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum); static const char *aif_text[] = { "Left", "Right" -- 2.17.1

2 1

[alsa-devel] [ANNOUNCE] 2018 Linux Audio miniconference
by Mark Brown 16 Oct '18

16 Oct '18

Hi, As in previous years we're trying to organize an audio miniconference so we can get together and talk through issues, especially design decisons, face to face. This year's event will be held on Sunday October 21st in Edinburgh, the day before ELC Europe starts there. We're still finalizing the venue (we expect to be able to confirm next week) but it'll be outside the ELC venue so there will be no need to purchase an ELC ticket to attend. As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/spreadsheets/d/11p2pSNxS7sJysg1JwErWi0T2Dq7XKRtAGng… Thanks, Mark

6 10

[alsa-devel] [PATCH] ALSA: hda: Add 2 more models to the power_save blacklist
by Hans de Goede 16 Oct '18

16 Oct '18

Power-saving is causing plops on audio start/stop on Dell Precision T3600 laptops and Intel DZ77BH boards, add these to the power_save blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Note both these models were reported through: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 as usual alsa-info.sh output for both is available as attachment there. --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index aa4c672dbaf7..454ef6feee3a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2257,8 +2257,12 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x8086, 0x2040, "Intel DZ77BH-55K", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */ SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ -- 2.19.0

2 1

[alsa-devel] [PATCH v3 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 15 Oct '18

15 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 30 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 30 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 26 +- 14 files changed, 789 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h -- 2.19.0

1 9

[alsa-devel] [PATCH] ASoC: sun4i-i2s: don't try to start up or shut down DAI if it's active
by Vasily Khoruzhick 15 Oct '18

15 Oct '18

Otherwise we may end up with shutting down I2S if shutdown() was called for capture substream, but playback is still running. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/sunxi/sun4i-i2s.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a9ea329dc046..787b67c4f845 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -649,6 +649,9 @@ static int sun4i_i2s_startup(struct snd_pcm_substream *substream, { struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); + if (dai->active) + return 0; + /* Enable the whole hardware block */ regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); @@ -667,6 +670,9 @@ static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, { struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); + if (dai->active) + return; + clk_disable_unprepare(i2s->mod_clk); /* Disable our output lines */ -- 2.19.0

3 5

[alsa-devel] [PATCH] ASoC: AMD: Add SND_JACK_LINEOUT jack type
by Akshu Agrawal 15 Oct '18

15 Oct '18

Some 3 pole connectors report impedance greater than threshold of 1000Ohm. Thus, da7219 reports them as LINEOUT. Adding the SND_JACK_LINEOUT type so that we don't fail to detect any 3 pole jack type. Also, changing SND_JACK_HEADPHONE | SND_JACK_MICROPHONE -> SND_JACK_HEADSET Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 0d97d00698a0..eced4820b0bc 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks"); ret = snd_soc_card_jack_new(card, "Headset Jack", - SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &cz_jack, NULL, 0); -- 2.16.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Boards: Add KBL Dialog Maxim I2S machine driver
by mac.chiang＠intel.com 15 Oct '18

15 Oct '18

From: Mac Chiang <mac.chiang(a)intel.com> This patch adds Kabylake I2S machine driver with: DA7219 audio codec(SSP1) and MAXIM98927(SSP0) speaker amplifier. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 13 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_da7219_max98927.c | 983 +++++++++++++++++++++++++++ 3 files changed, 998 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 88e4b42..73ca135 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -280,6 +280,19 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH create an alsa sound card for DA7219 + MAX98357A I2S audio codec. Say Y if you have such a device. +config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH + tristate "KBL with DA7219 and MAX98927 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_DA7219 + select SND_SOC_MAX98927 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for DA7219 + MAX98927 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6e88373..5381e27 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -17,6 +17,7 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o +snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c new file mode 100644 index 0000000..3ab96ee --- /dev/null +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -0,0 +1,983 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs + * + * Modified from: + * Intel Kabylake I2S Machine driver supporting MAX98927 and + * RT5663 codecs + */ + +#include <linux/input.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 "../../codecs/da7219.h" +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "../../codecs/da7219-aad.h" + +#define KBL_DIALOG_CODEC_DAI "da7219-hifi" +#define MAX98927_CODEC_DAI "max98927-aif1" +#define MAXIM_DEV0_NAME "i2c-MX98927:00" +#define MAXIM_DEV1_NAME "i2c-MX98927:01" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_card *kabylake_audio_card; +static struct snd_soc_jack kabylake_hdmi[3]; + +struct kbl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct kbl_codec_private { + struct snd_soc_jack kabylake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + KBL_DPCM_AUDIO_PB = 0, + KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_ECHO_REF_CP, + KBL_DPCM_AUDIO_REF_CP, + KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_HDMI1_PB, + KBL_DPCM_AUDIO_HDMI2_PB, + KBL_DPCM_AUDIO_HDMI3_PB, + KBL_DPCM_AUDIO_HS_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(card->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + +static const struct snd_kcontrol_new kabylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget kabylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("DP", NULL), + SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route kabylake_map[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, + + /* other jacks */ + { "DMic", NULL, "SoC DMIC" }, + + { "HDMI", NULL, "hif5 Output" }, + { "DP", NULL, "hif6 Output" }, + + /* CODEC BE connections */ + { "Left HiFi Playback", NULL, "ssp0 Tx" }, + { "Right HiFi Playback", NULL, "ssp0 Tx" }, + { "ssp0 Tx", NULL, "spk_out" }, + + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + + /* AEC capture path */ + { "echo_ref_out", NULL, "ssp0 Rx" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, + + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, +}; + +static const struct snd_soc_dapm_route kabylake_ssp1_map[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, + + /* CODEC BE connections */ + { "Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec1_out" }, + + { "hs_in", NULL, "ssp1 Rx" }, + { "ssp1 Rx", NULL, "Capture" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + int ret = 0, j; + + for (j = 0; j < runtime->num_codecs; j++) { + struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + + if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static struct snd_soc_ops kabylake_ssp0_ops = { + .hw_params = kabylake_ssp0_hw_params, +}; + +static int kabylake_ssp_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); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_soc_dpcm *dpcm = container_of( + params, struct snd_soc_dpcm, hw_params); + struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; + struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + + /* + * The ADSP will convert the FE rate to 48k, stereo, 24 bit + */ + if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + } + + /* + * The speaker on the SSP0 supports S16_LE and not S24_LE. + * thus changing the mask here + */ + if (!strcmp(be_dai_link->name, "SSP0-Codec")) + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + struct snd_soc_card *card = rtd->card; + int ret; + + + ret = snd_soc_dapm_add_routes(&card->dapm, + kabylake_ssp1_map, + ARRAY_SIZE(kabylake_ssp1_map)); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + da7219_aad_jack_det(component, &ctx->kabylake_headset); + + ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); + if (ret) + dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); + + return ret; +} + +static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct kbl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = device; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); +} + +static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); +} + +static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); +} + +static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_component *component = rtd->cpu_dai->component; + + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + + return 0; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int kbl_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 48Khz + * stereo + * 16 bit audio + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + + return 0; +} + +static const struct snd_soc_ops kabylake_da7219_fe_ops = { + .startup = kbl_fe_startup, +}; + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static int kabylake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +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, +}; +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + +static int kabylake_refcap_startup(struct snd_pcm_substream *substream) +{ + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +} + + +static struct snd_soc_ops skylaye_refcap_ops = { + .startup = kabylake_refcap_startup, +}; + +static struct snd_soc_codec_conf max98927_codec_conf[] = { + + { + .dev_name = MAXIM_DEV0_NAME, + .name_prefix = "Right", + }, + + { + .dev_name = MAXIM_DEV1_NAME, + .name_prefix = "Left", + }, +}; + +static struct snd_soc_dai_link_component ssp0_codec_components[] = { + { /* Left */ + .name = MAXIM_DEV0_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + + { /* For Right */ + .name = MAXIM_DEV1_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HS_PB] = { + .name = "Kbl Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .ops = &kabylake_da7219_fe_ops, + + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 1, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-DLGS7219:00", + .codec_dai_name = KBL_DIALOG_CODEC_DAI, + .init = kabylake_da7219_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_max98927_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + .name = "dmic01", + .id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 2, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 3, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 4, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int kabylake_card_late_probe(struct snd_soc_card *card) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); + struct kbl_hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &kabylake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &kabylake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); + + return 0; +} + +/* kabylake audio machine driver for SPT + DA7219 */ +static struct snd_soc_card kbl_audio_card_da7219_m98927 = { + .name = "kblda7219m98927", + .owner = THIS_MODULE, + .dai_link = kabylake_dais, + .num_links = ARRAY_SIZE(kabylake_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +/* kabylake audio machine driver for Maxim98927 */ +static struct snd_soc_card kbl_audio_card_max98927 = { + .name = "kblmax98927", + .owner = THIS_MODULE, + .dai_link = kabylake_max98927_dais, + .num_links = ARRAY_SIZE(kabylake_max98927_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static int kabylake_audio_probe(struct platform_device *pdev) +{ + struct kbl_codec_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + kabylake_audio_card = + (struct snd_soc_card *)pdev->id_entry->driver_data; + + kabylake_audio_card->dev = &pdev->dev; + snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); +} + +static const struct platform_device_id kbl_board_ids[] = { + { + .name = "kbl_da7219_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_da7219_m98927, + }, + { + .name = "kbl_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_max98927, + }, + { } +}; + +static struct platform_driver kabylake_audio = { + .probe = kabylake_audio_probe, + .driver = { + .name = "kbl_da7219_max98927", + .pm = &snd_soc_pm_ops, + }, + .id_table = kbl_board_ids, +}; + +module_platform_driver(kabylake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:kbl_da7219_max98927"); +MODULE_ALIAS("platform:kbl_max98927"); -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: Intel: common: Add Kabylake Dialog+Maxim machine driver entry
by mac.chiang＠intel.com 15 Oct '18

15 Oct '18

From: Mac Chiang <mac.chiang(a)intel.com> This patch adds da7219_max98927 machine driver entry into machine table Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index 0ee173c..a317b77 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { .codecs = {"MX98357A"} }; +static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = { + .num_codecs = 1, + .codecs = {"MX98927"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { { .id = "INT343A", @@ -83,6 +88,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .quirk_data = &kbl_7219_98357_codecs, .pdata = &skl_dmic_data, }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98927_codecs, + .pdata = &skl_dmic_data + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); -- 2.7.4

2 1

[alsa-devel] [alsa-utils][PATCH 0/9] alsactl: friendly to pluggable devices
by Takashi Sakamoto 14 Oct '18

14 Oct '18

Hi, A bug was reported that monitor mode of alsactl consumes much CPU time after disconnection of any sound card[1]. This patchset improves the mode to handle connection/disconnection of sound card. Observed control nodes are maintained by list structure instead of array. Linux specific epoll(7) is used to dispatch events. Linux specific inotify(7) is used to detect connection of new sound card. Linux specific signalfd(2) is used to catch Unix signals for termination. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140580.… Takashi Sakamoto (9): alsactl: split event loop code to a function alsactl: add an iterator of registered instances of sound card alsactl: use epoll(7) instead of poll(2) alsactl: use link list to maintain source of events alsactl: use a list of source for event dispatcher instead of an array of source alsactl: obsolete array for maintenance of handlers alsactl: handle disconnection of sound card alsactl: handle detection of new sound card alsactl: use signalfd to catch UNIX signal alsactl/monitor.c | 422 ++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 375 insertions(+), 47 deletions(-) -- 2.19.1

2 10