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

[alsa-devel] [PATCH 00/14] soundwire: intel: implement new ASoC interfaces
by Pierre-Louis Bossart 08 Nov '19

08 Nov '19

this patchset applies on top of the series "[PATCH 0/4] soundwire: update ASoC interfaces" and previously submitted cleanups "[PATCH v3 0/5] soundwire: intel/cadence: better initialization" The changes are essentially a removal of the platform devices and the implementation of the new interfaces required to scan the ACPI tables, probe the links and start them. The missing prepare, trigger and setup ASoC callbacks are also implemented. The hw_params and free callbacks use the new interfaces as well. While there are quite a few lines of code changed, this is mostly about interface changes. The next series will contain more functional changes and deal with race conditions on probe, enumeration and suspend/resume issues. Bard Liao (1): soundwire: add device driver to sdw_md_driver Pierre-Louis Bossart (10): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and provide new interface soundwire: intel: free all resources on hw_free() soundwire: intel_init: add implementation of sdw_intel_enable_irq() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus_type.c | 60 +++--- drivers/soundwire/intel.c | 280 ++++++++++++++++++++++----- drivers/soundwire/intel.h | 8 +- drivers/soundwire/intel_init.c | 300 ++++++++++++++++++++++------- drivers/soundwire/master.c | 64 ++++++ drivers/soundwire/slave.c | 9 +- include/linux/soundwire/sdw.h | 39 +++- include/linux/soundwire/sdw_type.h | 34 +++- 11 files changed, 642 insertions(+), 160 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

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[alsa-devel] [PATCH] ASoC: fsi: switch to yaml base Documentation
by Kuninori Morimoto 07 Nov '19

07 Nov '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document for FSI. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,fsi.txt | 31 --------- .../devicetree/bindings/sound/renesas,fsi.yaml | 76 ++++++++++++++++++++++ 2 files changed, 76 insertions(+), 31 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/renesas,fsi.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,fsi.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt deleted file mode 100644 index 0cf0f81..0000000 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.txt +++ /dev/null @@ -1,31 +0,0 @@ -Renesas FSI - -Required properties: -- compatible : "renesas,fsi2-<soctype>", - "renesas,sh_fsi2" or "renesas,sh_fsi" as - fallback. - Examples with soctypes are: - - "renesas,fsi2-r8a7740" (R-Mobile A1) - - "renesas,fsi2-sh73a0" (SH-Mobile AG5) -- reg : Should contain the register physical address and length -- interrupts : Should contain FSI interrupt - -- fsia,spdif-connection : FSI is connected by S/PDIF -- fsia,stream-mode-support : FSI supports 16bit stream mode. -- fsia,use-internal-clock : FSI uses internal clock when master mode. - -- fsib,spdif-connection : same as fsia -- fsib,stream-mode-support : same as fsia -- fsib,use-internal-clock : same as fsia - -Example: - -sh_fsi2: sh_fsi2@ec230000 { - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupts = <0 146 0x4>; - - fsia,spdif-connection; - fsia,stream-mode-support; - fsia,use-internal-clock; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml new file mode 100644 index 0000000..140a37f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,fsi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas FSI Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for FSI2 SoC + - items: + - enum: + - renesas,fsi2-sh73a0 + - renesas,fsi2-r8a7740 + - enum: + - renesas,sh_fsi2 + # for Generic + - items: + - enum: + - renesas,sh_fsi + - renesas,sh_fsi2 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + fsia,spdif-connection: + $ref: /schemas/types.yaml#/definitions/flag + description: FSI is connected by S/PDIF + + fsia,stream-mode-support: + $ref: /schemas/types.yaml#/definitions/flag + description: FSI supports 16bit stream mode + + fsia,use-internal-clock: + $ref: /schemas/types.yaml#/definitions/flag + description: FSI uses internal clock when master mode + + fsib,spdif-connection: + $ref: /schemas/types.yaml#/definitions/flag + description: same as fsia + + fsib,stream-mode-support: + $ref: /schemas/types.yaml#/definitions/flag + description: same as fsia + + fsib,use-internal-clock: + $ref: /schemas/types.yaml#/definitions/flag + description: same as fsia + +required: + - compatible + - reg + - interrupts + +examples: + - | + sh_fsi2: sound@ec230000 { + compatible = "renesas,fsi2-r8a7740", "renesas,sh_fsi2"; + reg = <0xec230000 0x400>; + interrupts = <0 146 0x4>; + + fsia,spdif-connection; + fsia,stream-mode-support; + fsia,use-internal-clock; + }; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: wm_adsp: Expose mixer control API
by David Rhodes 07 Nov '19

07 Nov '19

From: Li Xu <li.xu(a)cirrus.com> Expose mixer control API for reading and writing. The exposed API can be used by codec driver for interacting with mixer control in kernel space. This allows codec driver to implement more involved interactions with DSP firmware, such as Fast Use Case Switching. Signed-off-by: Li Xu <li.xu(a)cirrus.com> Signed-off-by: David Rhodes <david.rhodes(a)cirrus.com> --- sound/soc/codecs/wm_adsp.c | 70 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 4 +++ 2 files changed, 73 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index aa74fd1..77b8aea 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -635,6 +635,9 @@ struct wm_coeff_ctl_ops { struct wm_coeff_ctl { const char *name; const char *fw_name; + /* Subname is needed to match with firmware */ + const char *subname; + unsigned int subname_len; struct wm_adsp_alg_region alg_region; struct wm_coeff_ctl_ops ops; struct wm_adsp *dsp; @@ -1520,6 +1523,7 @@ static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) { kfree(ctl->cache); kfree(ctl->name); + kfree(ctl->subname); kfree(ctl); } @@ -1597,6 +1601,15 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ret = -ENOMEM; goto err_ctl; } + if (subname) { + ctl->subname_len = subname_len; + ctl->subname = kmemdup(subname, + strlen(subname) + 1, GFP_KERNEL); + if (!ctl->subname) { + ret = -ENOMEM; + goto err_ctl_name; + } + } ctl->enabled = 1; ctl->set = 0; ctl->ops.xget = wm_coeff_get; @@ -1610,7 +1623,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { ret = -ENOMEM; - goto err_ctl_name; + goto err_ctl_subname; } list_add(&ctl->list, &dsp->ctl_list); @@ -1633,6 +1646,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, err_ctl_cache: kfree(ctl->cache); +err_ctl_subname: + kfree(ctl->subname); err_ctl_name: kfree(ctl->name); err_ctl: @@ -2194,6 +2209,59 @@ static int wm_adsp_load(struct wm_adsp *dsp) return ret; } +/* + * Find wm_coeff_ctl with input name as its subname + * If not found, return NULL + */ +static struct wm_coeff_ctl *wm_adsp_get_ctl(struct wm_adsp *dsp, + const char *name) +{ + struct wm_coeff_ctl *pos, *rslt = NULL; + + list_for_each_entry(pos, &dsp->ctl_list, list) { + if (!pos->subname) + continue; + if (strncmp(pos->subname, name, pos->subname_len) == 0) { + rslt = pos; + break; + } + } + + return rslt; +} + +int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, const void *buf, + size_t len) +{ + struct wm_coeff_ctl *ctl; + + ctl = wm_adsp_get_ctl(dsp, name); + if (!ctl) + return -EINVAL; + + if (len > ctl->len) + return -EINVAL; + + return wm_coeff_write_control(ctl, buf, len); +} +EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); + +int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, void *buf, + size_t len) +{ + struct wm_coeff_ctl *ctl; + + ctl = wm_adsp_get_ctl(dsp, name); + if (!ctl) + return -EINVAL; + + if (len > ctl->len) + return -EINVAL; + + return wm_coeff_read_control(ctl, buf, len); +} +EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); + static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region) { diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 5275a49..2a8cd09 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -223,5 +223,9 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); int wm_adsp_compr_copy(struct snd_compr_stream *stream, char __user *buf, size_t count); +int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, const void *buf, + size_t len); +int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, void *buf, + size_t len); #endif -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: SOF: Intel: Soundwire integration
by Pierre-Louis Bossart 06 Nov '19

06 Nov '19

This patchset applies on top of the series "[PATCH 0/4] soundwire: update ASoC interfaces". The SOF/Intel code makes use of the interfaces defined for initialization. Build support for SoundWire is not provided for now, all Soundwire-related code will be handled with a dummy fallback. We will enable SoundWire interfaces in the Kconfigs when the functionality is enabled in the soundwire tree. In short, if the interfaces are agreed on, there is no risk with the integration of these patches on the ASoC side. Pierre-Louis Bossart (5): ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend include/sound/sof/dai-intel.h | 18 +-- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 13 ++ sound/soc/sof/intel/hda.c | 230 ++++++++++++++++++++++++++++++- sound/soc/sof/intel/hda.h | 44 ++++++ 5 files changed, 295 insertions(+), 12 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v2 00/11] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 06 Nov '19

06 Nov '19

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. Pin Controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. If anyone is interested to try, here are working set of patches on top of rc3. https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=audio/v5… alsa ucm files: https://git.linaro.org/people/srinivas.kandagatla/alsa-lib.git/log/?h=DB845c Thanks, srini Changes since v1: - Code cleanup and convert to proper dpcm widgets where possible. - converted to mfd driver. - added pinctrl driver to this series - added dts changes in this series - bindings converted to yaml. Srinivas Kandagatla (10): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: pinctrl: qcom-wcd934x: Add bindings for gpio ASoC: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga Yeleswarapu Nagaradhesh (1): pinctrl: qcom-wcd934x: Add support to wcd934x pinctrl driver. .../pinctrl/qcom,wcd934x-pinctrl.yaml | 51 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 169 + drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 330 ++ drivers/pinctrl/qcom/Kconfig | 7 + drivers/pinctrl/qcom/Makefile | 1 + drivers/pinctrl/qcom/pinctrl-wcd934x-gpio.c | 365 ++ include/linux/mfd/wcd934x/registers.h | 529 ++ include/linux/mfd/wcd934x/wcd934x.h | 24 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5218 +++++++++++++++++ sound/soc/qcom/sdm845.c | 71 +- 15 files changed, 6789 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,wcd934x-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/mfd/wcd934x.c create mode 100644 drivers/pinctrl/qcom/pinctrl-wcd934x-gpio.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

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[alsa-devel] [PATCH 1/7] ASoC: amd: Create multiple I2S platform device endpoints.
by Ravulapati Vishnu vardhan rao 06 Nov '19

06 Nov '19

Creates Platform Device endpoints for multiple I2S instances: SP and BT endpoints device. Pass PCI resources like MMIO, irq to the platform devices. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 5 +++ sound/soc/amd/raven/pci-acp3x.c | 83 +++++++++++++++++++++++++++-------------- 2 files changed, 60 insertions(+), 28 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 4f2cadd..2f15fe1 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -7,10 +7,15 @@ #include "chip_offset_byte.h" +#define ACP3x_DEVS 3 #define ACP3x_PHY_BASE_ADDRESS 0x1240000 #define ACP3x_I2S_MODE 0 #define ACP3x_REG_START 0x1240000 #define ACP3x_REG_END 0x1250200 +#define ACP3x_I2STDM_REG_START 0x1242400 +#define ACP3x_I2STDM_REG_END 0x1242410 +#define ACP3x_BT_TDM_REG_START 0x1242800 +#define ACP3x_BT_TDM_REG_END 0x1242810 #define I2S_MODE 0x04 #define BT_TX_THRESHOLD 26 #define BT_RX_THRESHOLD 25 diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..7f435b3 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -16,16 +16,16 @@ struct acp3x_dev_data { void __iomem *acp3x_base; bool acp3x_audio_mode; struct resource *res; - struct platform_device *pdev; + struct platform_device *pdev[ACP3x_DEVS]; }; static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr, val; + u32 addr, val, i; struct acp3x_dev_data *adata; - struct platform_device_info pdevinfo; + struct platform_device_info pdevinfo[ACP3x_DEVS]; unsigned int irqflags; if (pci_enable_device(pci)) { @@ -68,7 +68,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, switch (val) { case I2S_MODE: adata->res = devm_kzalloc(&pci->dev, - sizeof(struct resource) * 2, + sizeof(struct resource) * 4, GFP_KERNEL); if (!adata->res) { ret = -ENOMEM; @@ -80,39 +80,61 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->res[0].start = addr; adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); - adata->res[1].name = "acp3x_i2s_irq"; - adata->res[1].flags = IORESOURCE_IRQ; - adata->res[1].start = pci->irq; - adata->res[1].end = pci->irq; + adata->res[1].name = "acp3x_i2s_sp"; + adata->res[1].flags = IORESOURCE_MEM; + adata->res[1].start = addr + ACP3x_I2STDM_REG_START; + adata->res[1].end = addr + ACP3x_I2STDM_REG_END; + + adata->res[2].name = "acp3x_i2s_bt"; + adata->res[2].flags = IORESOURCE_MEM; + adata->res[2].start = addr + ACP3x_BT_TDM_REG_START; + adata->res[2].end = addr + ACP3x_BT_TDM_REG_END; + + adata->res[3].name = "acp3x_i2s_irq"; + adata->res[3].flags = IORESOURCE_IRQ; + adata->res[3].start = pci->irq; + adata->res[3].end = adata->res[3].start; adata->acp3x_audio_mode = ACP3x_I2S_MODE; memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo.name = "acp3x_rv_i2s"; - pdevinfo.id = 0; - pdevinfo.parent = &pci->dev; - pdevinfo.num_res = 2; - pdevinfo.res = adata->res; - pdevinfo.data = &irqflags; - pdevinfo.size_data = sizeof(irqflags); - - adata->pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(adata->pdev)) { - dev_err(&pci->dev, "cannot register %s device\n", - pdevinfo.name); - ret = PTR_ERR(adata->pdev); - goto unmap_mmio; + pdevinfo[0].name = "acp3x_rv_i2s_dma"; + pdevinfo[0].id = 0; + pdevinfo[0].parent = &pci->dev; + pdevinfo[0].num_res = 4; + pdevinfo[0].res = &adata->res[0]; + pdevinfo[0].data = &irqflags; + pdevinfo[0].size_data = sizeof(irqflags); + + pdevinfo[1].name = "acp3x_i2s_playcap"; + pdevinfo[1].id = 0; + pdevinfo[1].parent = &pci->dev; + pdevinfo[1].num_res = 1; + pdevinfo[1].res = &adata->res[1]; + + pdevinfo[2].name = "acp3x_i2s_playcap"; + pdevinfo[2].id = 1; + pdevinfo[2].parent = &pci->dev; + pdevinfo[2].num_res = 1; + pdevinfo[2].res = &adata->res[2]; + for (i = 0; i < ACP3x_DEVS ; i++) { + adata->pdev[i] = + platform_device_register_full(&pdevinfo[i]); + if (adata->pdev[i] == NULL) { + dev_err(&pci->dev, "cannot register %s device\n", + pdevinfo[i].name); + ret = -ENODEV; + goto unmap_mmio; + } } break; - default: - dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); - ret = -ENODEV; - goto unmap_mmio; } return 0; unmap_mmio: - pci_disable_msi(pci); + for (i = 0 ; i < ACP3x_DEVS ; i++) + platform_device_unregister(adata->pdev[i]); + kfree(adata->res); iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); @@ -124,9 +146,13 @@ static int snd_acp3x_probe(struct pci_dev *pci, static void snd_acp3x_remove(struct pci_dev *pci) { + int i; struct acp3x_dev_data *adata = pci_get_drvdata(pci); - platform_device_unregister(adata->pdev); + if (adata->acp3x_audio_mode == ACP3x_I2S_MODE) { + for (i = 0 ; i < ACP3x_DEVS ; i++) + platform_device_unregister(adata->pdev[i]); + } iounmap(adata->acp3x_base); pci_disable_msi(pci); @@ -151,6 +177,7 @@ static struct pci_driver acp3x_driver = { module_pci_driver(acp3x_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_DESCRIPTION("AMD ACP3x PCI driver"); MODULE_LICENSE("GPL v2"); -- 2.7.4

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[alsa-devel] [PATCH V2 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm
by Shengjiu Wang 06 Nov '19

06 Nov '19

In order to support the two asrc modules in imx8qm, we need to add compatible string "fsl,imx8qm-asrc0" and "fsl,imx8qm-asrc1" Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..cd2bd3daa7e1 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,8 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc0" or "fsl,imx8qm-asrc1". - reg : Offset and length of the register set for the device. -- 2.21.0

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[alsa-devel] [PATCH] ASoC: bdw-rt5677: enable runtime channel merge
by Brent Lu 05 Nov '19

05 Nov '19

In the DAI link "Capture PCM", the FE DAI "Capture Pin" supports 4-channel capture but the BE DAI supports only 2-channel capture. To fix the channel mismatch, we need to enable the runtime channel merge for this DAI link. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d335..8d07038 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -273,6 +273,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { }, .dpcm_capture = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, SND_SOC_DAILINK_REG(fe, dummy, platform), }, -- 2.7.4

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[alsa-devel] tegra30 tdm support
by Ben Dooks 05 Nov '19

05 Nov '19

New series for TDM audio support on tegra30 v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch v4: - fix comment style issues - change tdm-a to data-offset 1

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[alsa-devel] [PATCH v2 resend 00/18] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 05 Nov '19

05 Nov '19

Hi Mark These are resend of soc-core cleanup step4. These are based on mark/for-5.5 branch. I already got report from Intel that these had no issue on Intel CI machine. Kuninori Morimoto (18): ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: tidyup soc_init_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component_unlocked() ASoC: soc-core: remove snd_soc_component_add/del() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 15 +- sound/soc/soc-core.c | 536 +++++++++++++++++++++-------------------------- sound/soc/soc-topology.c | 17 +- 3 files changed, 263 insertions(+), 305 deletions(-) -- 2.7.4

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