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

[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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[alsa-devel] [PATCH v4 0/2] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 05 Nov '19

05 Nov '19

Thanks for reviewing the v3 patchset. Here is new patchset addressing all the comments from v3 This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is tested on on DragonBoard DB845c based of SDM845 SoC and Lenovo YOGA C630 Laptop based on SDM850. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since v3: - Updated bindings as suggested by Rob. - Fixed sdw_disable/unprepare order. - removed setting stream_name as suggested by Pierre - removed v1.5.0 controller support for now, will be added after testing. Srinivas Kandagatla (2): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller .../bindings/soundwire/qcom,sdw.txt | 179 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 904 ++++++++++++++++++ 4 files changed, 1096 insertions(+) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

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[alsa-devel] [PATCH] dt-bindings: sound: adau7118: Fix example warning
by Maxime Ripard 04 Nov '19

04 Nov '19

The ADAU7118 has an example where the codec has an i2c address of 14, and the unit address set to 14 as well. However, while the address is expressed in decimal, the unit-address is supposed to be in hexadecimal, which ends up with two different addresses that trigger a DTC warning. Fix this by setting the unit address to (0x)e. Cc: Nuno Sá <nuno.sa(a)analog.com> Fixes: 969d49b2cdc8 ("dt-bindings: asoc: Add ADAU7118 documentation") Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Documentation/devicetree/bindings/sound/adi,adau7118.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index c3f10afbdd6f..65f6844a0c6d 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -65,7 +65,7 @@ examples: /* example with i2c support */ #address-cells = <1>; #size-cells = <0>; - adau7118_codec: audio-codec@14 { + adau7118_codec: audio-codec@e { compatible = "adi,adau7118"; reg = <14>; #sound-dai-cells = <0>; -- 2.23.0

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[alsa-devel] [GIT PULL] soundwire fixes for v5.4-rc6
by Vinod Koul 02 Nov '19

02 Nov '19

Hi Greg, Please pull to receive couple of fixes for soundwire. All these changes have been sitting in linux-next as well The following changes since commit 54ecb8f7028c5eb3d740bb82b0f1d90f2df63c5c: Linux 5.4-rc1 (2019-09-30 10:35:40 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.4-rc6 for you to fetch changes up to 7b47ad3322d114736e08cc244a84fe7af9b861f6: soundwire: slave: fix scanf format (2019-10-24 16:55:45 +0530) ---------------------------------------------------------------- soundwire fixes for v5.4-rc6 - Kconfig fixes to ensure soundwire is built only for ACPI and DT platform - fix for intel PDI offsets and numbers - slave scanf format fix ---------------------------------------------------------------- Bard Liao (1): soundwire: intel: fix intel_register_dai PDI offsets and numbers Michal Suchanek (2): soundwire: depend on ACPI soundwire: depend on ACPI || OF Pierre-Louis Bossart (1): soundwire: slave: fix scanf format drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 4 ++-- drivers/soundwire/slave.c | 3 ++- 3 files changed, 5 insertions(+), 3 deletions(-) Thanks -- ~Vinod

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[alsa-devel] Question about FE & BE DPCM
by Daniel Baluta 01 Nov '19

01 Nov '19

Hi Morimoto-san, I'm really confused on what exactly is Front End and Back End. Is FE a DAI or a DAI link or part of a DAI link? In simple-card.c there is simple_dai_link_of_dpcm function: struct snd_soc_dai_link_component *codecs = dai_link->codecs; » /* BE is dummy */ » » codecs->of_node»» = NULL; » » codecs->dai_name» = "snd-soc-dummy-dai"; » » codecs->name» » = "snd-soc-dummy"; » » /* FE settings */ » » dai_link->dynamic» » = 1; » » dai_link->dpcm_merged_format» = 1; So, from here it looks like BE/FE are components being part of the same link. Another thing is: asoc_simple_canonicalize_platform function where it does say: » * DPCM BE can be no platform. » * Alloced memory will be waste, but not leak. » */ » if (!dai_link->platforms->of_node) » » dai_link->num_platforms = 0; But I do want to create a BE with a platform. So, should I try to revert the following patch: commit 794fcee8da3c0c8a01b08ecad1c181cb0a622868 Author: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed Jul 10 17:01:12 2019 +0900 ASoC: simple-card-utils: care no Platform for DPCM commit 34614739988ad ("ASoC: soc-core: support dai_link with platforms_num != 1") supports multi Platform, and commit 9f3eb91753451 ("ASoC: simple-card-utils: consider CPU-Platform possibility") removed no Platform from simple-card. Multi Platform is now checking both Platform name/of_node are NULL case. But in normal case, DPCM be doesn't have Platform. Basically, what I want to do is: after the DAI link is created to call: snd_soc_fixup_dai_links_platform_name function to set a proper name for BE platform. The problem is that above mentioned patch sets dai_link->num_platforms = 0; so even the name is set the platform component is not probed. thanks, Daniel.

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