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[alsa-devel] [PATCH] ASoC: rt5514: Eliminate the noise in the ASRC case
by Oder Chiou 09 Aug '17

09 Aug '17

In the ASRC case, there is the noise if the clock is asynchronous with the LRCK. The default settings will eliminate the noise. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 21cbb56..0eabe56 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,6 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, }; static const struct reg_default rt5514_reg[] = { @@ -89,10 +91,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_DELAY_BUF_CTRL3, 0x00000000}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x00000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x00000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: remove 16ch support for now
by Kuninori Morimoto 09 Aug '17

09 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car Gen3 sound device can use TDM 16ch, but its usage is very limited. It is not yet supported. Let's remove unexpected/unsupported channel settings for now. This patch makes xx_ADINR register settings sanity without runtime check. R-Car Gen3 datasheet 40.4.7 Note on Usage of the SSIU Module When an SSIU module is used in "TDM-16ch mode", only the transfer with the external memory is available. Reported-by: Nguyen Viet Dung <dung.nguyen.aj(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4117b56..cd8db39 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -726,7 +726,6 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, case 2: case 6: case 8: - case 16: /* TDM Extend Mode */ rsnd_rdai_channels_set(rdai, slots); rsnd_rdai_ssi_lane_set(rdai, 1); @@ -740,7 +739,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, } static unsigned int rsnd_soc_hw_channels_list[] = { - 2, 6, 8, 16, + 2, 6, 8, }; static unsigned int rsnd_soc_hw_rate_list[] = { @@ -1025,7 +1024,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, drv->playback.rates = RSND_RATES; drv->playback.formats = RSND_FMTS; drv->playback.channels_min = 2; - drv->playback.channels_max = 16; + drv->playback.channels_max = 8; drv->playback.stream_name = rdai->playback.name; snprintf(rdai->capture.name, RSND_DAI_NAME_SIZE, @@ -1033,7 +1032,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, drv->capture.rates = RSND_RATES; drv->capture.formats = RSND_FMTS; drv->capture.channels_min = 2; - drv->capture.channels_max = 16; + drv->capture.channels_max = 8; drv->capture.stream_name = rdai->capture.name; rdai->playback.rdai = rdai; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: avoid duplicate free_irq()
by Kuninori Morimoto 09 Aug '17

09 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on Parent SSI") fixed Parent SSI duplicate free_irq(). But on Renesas Sound, not only Parent SSI but also Multi SSI have same issue. This patch avoid duplicate free_irq() if it was not pure SSI. Fixes: 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on Parent SSI") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 25c351f..0309de5 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -846,10 +846,10 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io); + struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io); - /* Do nothing for SSI parent mod */ - if (ssi_parent_mod == mod) + /* Do nothing if non SSI (= SSI parent, multi SSI) mod */ + if (pure_ssi_mod != mod) return 0; /* PIO will request IRQ again */ -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 09 Aug '17

09 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can reorder/disable them in the dts. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codec in the dts. Jeffy Chen (3): ASoC: rockchip: Parse dai links from dts arm64: dts: rockchip: Add rockchip,codec-names property dt-bindings: ASoC: rockchip: Add rockchip,codec-names property .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 + arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 1 + sound/soc/rockchip/rk3399_gru_sound.c | 129 ++++++++++++++------- 3 files changed, 89 insertions(+), 43 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 0/3] Add support for playing a given number of samples
by Daniel Baluta 09 Aug '17

09 Aug '17

We introduce (patch 1) the option of playing only a given number of samples in order to reduce our simple test running time. Until now we can only limit them at 1 second (-d parameter). This patch series also refactors the playback function (patch 2) and fixes a bug with small raw files (patch 3). Previous discussions about this started here: https://www.spinics.net/lists/alsa-devel/msg64463.html Daniel Baluta (2): aplay: Refactor playback code aplay: Fix playback for small raw files Ion-Horia Petrisor (1): aplay: Add samples argument for playing/recording a given number of samples aplay/aplay.c | 185 +++++++++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 143 insertions(+), 42 deletions(-) -- 2.7.4

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[alsa-devel] XRUN support in alsabat
by Zhang, Keqiao 08 Aug '17

08 Aug '17

Hi Takashi and all, Sorry to bother you. I'm trying to get alsabat(alsa-utils:BAT) to support XRUN for driver testing and there have been no effective solutions on how to inject XRUN for playback or capture. So could you give me some suggestions on this? Thanks in advance~ Thanks Keqiao

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[alsa-devel] Boss Katana amps
by Keith A. Milner 08 Aug '17

08 Aug '17

Unfortunately the new Boss Katana amps do not work fully with ALSA. The MIDI interfaces are exposed, but the audio interfaces are not. I suspect this is for similar reasons that the Boss GT-001 and GT-100 don't work. I've been meaning to try to find some time to look at this, but failing badly so far. If anyone else wants to chip in with ideas, the lsusb output is below: Bus 001 Device 004: ID 0582:01d8 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x01d8 bcdDevice 0.00 iManufacturer 1 BOSS iProduct 2 KATANA iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 188 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 44 ac 00 ** UNRECOGNIZED: 06 24 f1 04 16 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x0d EP 13 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 44 ac 00 ** UNRECOGNIZED: 06 24 f1 04 16 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8e EP 14 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 03 03 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered Cheers, Keith

1 0

[alsa-devel] [PATCH v4 00/17] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 08 Aug '17

08 Aug '17

The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Summary of changes since V3: - Manually stripped register definitions we're not using right now from registers.h to reduce the size and included some explanation in the commit message for why we do it the way we do - Fixed review comments on MFD code - Fixed Kconfig dependency error on pinctrl (missing dependency in MFD) - minor bugfixes to routing in CS47L85 and CS47L90 codec drivers Richard Fitzgerald (17): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add DT bindings for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add DT bindings for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 Documentation/devicetree/bindings/mfd/madera.txt | 102 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 99 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 67 + MAINTAINERS | 23 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 301 ++ drivers/mfd/Kconfig | 48 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1610 +++++++ drivers/mfd/cs47l85-tables.c | 3010 +++++++++++++ drivers/mfd/cs47l90-tables.c | 2675 ++++++++++++ drivers/mfd/madera-core.c | 617 +++ drivers/mfd/madera-i2c.c | 135 + drivers/mfd/madera-spi.c | 135 + drivers/mfd/madera.h | 54 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1074 +++++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 27 + include/linux/irqchip/irq-madera-pdata.h | 23 + include/linux/irqchip/irq-madera.h | 96 + include/linux/mfd/madera/core.h | 179 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 3987 ++++++++++++++++++ include/sound/madera-pdata.h | 61 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1747 ++++++++ sound/soc/codecs/cs47l85.c | 2722 ++++++++++++ sound/soc/codecs/cs47l90.c | 2661 ++++++++++++ sound/soc/codecs/madera.c | 4438 ++++++++++++++++++++ sound/soc/codecs/madera.h | 470 +++ 46 files changed, 27066 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: fsl: Convert to using %pOF instead of full_name
by Rob Herring 08 Aug '17

08 Aug '17

Now that we have a custom printf format specifier, convert users of full_name to use %pOF instead. This is preparation to remove storing of the full path string for each node. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org --- v3: Split ASoC changes to separate patch sound/soc/fsl/fsl_dma.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 16 ++++++++-------- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ccadefceeff2..ed8ea002902d 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -897,8 +897,8 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) ret = of_address_to_resource(ssi_np, 0, &res); if (ret) { - dev_err(&pdev->dev, "could not determine resources for %s\n", - ssi_np->full_name); + dev_err(&pdev->dev, "could not determine resources for %pOF\n", + ssi_np); of_node_put(ssi_np); return ret; } diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da341d61..392d5eef356d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -268,13 +268,13 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, ret = of_property_read_u32(child, "fsl,audmux-port", &port); if (ret) { - dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n", - child->full_name); + dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%pOF\"\n", + child); continue; } if (!of_property_read_bool(child, "fsl,port-config")) { - dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n", - child->full_name); + dev_warn(&pdev->dev, "child node \"%pOF\" does not have property fsl,port-config\n", + child); continue; } @@ -292,15 +292,15 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, } if (ret != -EOVERFLOW) { - dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", - i, child->full_name); + dev_err(&pdev->dev, "Failed to read u32 at index %d of child %pOF\n", + i, child); continue; } if (audmux_type == IMX31_AUDMUX) { if (i % 2) { - dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n", - child->full_name); + dev_err(&pdev->dev, "One pdcr value is missing in child node %pOF\n", + child); continue; } imx_audmux_v2_configure_port(port, ptcr, pdcr); -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: samsung: odroid: Drop requirement of clocks in the sound node
by Sylwester Nawrocki 08 Aug '17

08 Aug '17

As suggested in reviews the requirement of clocks in the 'sound' node is dropped and instead a leaf clock is used to configure frequency of the audio root clock PLL. This can work now after the clock tree definitions have been updated to allow clock rate setting propagation on the path from the I2S controller up to the EPLL. This patch also lowers the CODEC master clock frequency so as to not exceed the maximum allowed 60 MHz at maximum audio sampling rates. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../devicetree/bindings/sound/samsung,odroid.txt | 6 --- sound/soc/samsung/odroid.c | 44 +++++++++++++++------- 2 files changed, 30 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt index c30934d..625b1b1 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt @@ -7,9 +7,6 @@ Required properties: - model - the user-visible name of this sound complex - clocks - should contain entries matching clock names in the clock-names property - - clock-names - should contain following entries: - - "epll" - indicating the EPLL output clock - - "i2s_rclk" - indicating the RCLK (root) clock of the I2S0 controller - samsung,audio-widgets - this property specifies off-codec audio elements like headphones or speakers, for details see widgets.txt - samsung,audio-routing - a list of the connections between audio @@ -46,9 +43,6 @@ sound { "IN1", "Mic Jack", "Mic Jack", "MICBIAS"; - clocks = <&clock CLK_FOUT_EPLL>, <&i2s0 CLK_I2S_RCLK_SRC>; - clock-names = "epll", "sclk_i2s"; - cpu { sound-dai = <&i2s0 0>; }; diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 0834319..44b6de5 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -19,8 +19,8 @@ struct odroid_priv { struct snd_soc_card card; struct snd_soc_dai_link dai_link; - struct clk *pll; - struct clk *rclk; + struct clk *clk_i2s_bus; + struct clk *sclk_i2s; }; static int odroid_card_startup(struct snd_pcm_substream *substream) @@ -58,13 +58,18 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - ret = clk_set_rate(priv->pll, pll_freq + 1); + ret = clk_set_rate(priv->clk_i2s_bus, pll_freq / 2 + 1); if (ret < 0) return ret; - rclk_freq = params_rate(params) * 256 * 4; + /* + * We add 1 to the rclk_freq value in order to avoid too low clock + * frequency values due to the EPLL output frequency not being exact + * multiple of the audio sampling rate. + */ + rclk_freq = params_rate(params) * 256 + 1; - ret = clk_set_rate(priv->rclk, rclk_freq); + ret = clk_set_rate(priv->sclk_i2s, rclk_freq); if (ret < 0) return ret; @@ -118,14 +123,6 @@ static int odroid_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); - priv->pll = devm_clk_get(dev, "epll"); - if (IS_ERR(priv->pll)) - return PTR_ERR(priv->pll); - - priv->rclk = devm_clk_get(dev, "i2s_rclk"); - if (IS_ERR(priv->rclk)) - return PTR_ERR(priv->rclk); - ret = snd_soc_of_parse_card_name(card, "model"); if (ret < 0) return ret; @@ -171,14 +168,31 @@ static int odroid_audio_probe(struct platform_device *pdev) link->name = "Primary"; link->stream_name = link->name; + + priv->sclk_i2s = of_clk_get_by_name(link->cpu_of_node, "i2s_opclk1"); + if (IS_ERR(priv->sclk_i2s)) { + ret = PTR_ERR(priv->sclk_i2s); + goto err_put_i2s_n; + } + + priv->clk_i2s_bus = of_clk_get_by_name(link->cpu_of_node, "iis"); + if (IS_ERR(priv->clk_i2s_bus)) { + ret = PTR_ERR(priv->clk_i2s_bus); + goto err_put_sclk; + } + ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { dev_err(dev, "snd_soc_register_card() failed: %d\n", ret); - goto err_put_i2s_n; + goto err_put_clk_i2s; } return 0; +err_put_clk_i2s: + clk_put(priv->clk_i2s_bus); +err_put_sclk: + clk_put(priv->sclk_i2s); err_put_i2s_n: of_node_put(link->cpu_of_node); err_put_codec_n: @@ -192,6 +206,8 @@ static int odroid_audio_remove(struct platform_device *pdev) of_node_put(priv->dai_link.cpu_of_node); odroid_put_codec_of_nodes(&priv->dai_link); + clk_put(priv->sclk_i2s); + clk_put(priv->clk_i2s_bus); return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH 0/3] ASoC: rsnd: control SSICR::EN correctly
by Kuninori Morimoto 08 Aug '17

08 Aug '17

Hi Mark These are bugfix patches for Renesas sound SSI driver. SSI1 can share clock pin with SSI0, and then, I had assumed that SSI0 clock output needs SSICR::EN bit. But, it wasn't. SSI0 clock output for SSI1 needs other settings which already sets, but EN was not needed. Because of this assumption, and SSICR::EN operation bug, it gets underrun error in some sound route. This patch solves this issue. Kuninori Morimoto (3): ASoC: rsnd: control SSICR::EN correctly ASoC: rsnd: move rsnd_ssi_config_init() execute condition into it. ASoC: rsnd: rsnd_ssi_can_output_clk() macro sound/soc/sh/rcar/ssi.c | 53 +++++++++++++++++++++++++++++++++++-------------- 1 file changed, 38 insertions(+), 15 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH 00/11] ALSA: vsnd: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 08 Aug '17

08 Aug '17

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Thank you, Oleksandr [1] https://xenproject.org/ Oleksandr Andrushchenko (11): ALSA: vsnd: Implement driver's probe/remove ALSA: vsnd: Implement Xen bus state handling ALSA: vsnd: Read sound driver configuration from Xen store ALSA: vsnd: Implement Xen event channel handling ALSA: vsnd: Implement handling of shared buffers ALSA: vsnd: Introduce ALSA virtual sound driver ALSA: vsnd: Initialize virtul sound card ALSA: vsnd: Add timer for period interrupt emulation ALSA: vsnd: Implement ALSA PCM operations ALSA: vsnd: Implement communication with backend ALSA: vsnd: Introduce Kconfig option to enable Xen PV sound sound/drivers/Kconfig | 12 + sound/drivers/Makefile | 2 + sound/drivers/xen-front.c | 2029 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 2043 insertions(+) -- 2.7.4

3 20

[alsa-devel] [PATCH] ASoC: compress: Set reasonable compress id string
by Donglin Peng 08 Aug '17

08 Aug '17

For dynamic compress rtd, the codec_dai may not have any actual meaning(like snd-soc-dummy-dai), so compress id can just use the value of dai_link->stream_name. But for others, its codec_dai has actual meaning, so compress id can include codec_dai name. Signed-off-by: pengdonglin <pengdonglin137(a)163.com> --- sound/soc/soc-compress.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36bf43e8..589d6412c9f0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) @@ -797,19 +794,18 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) else if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STRE AM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); - } else + } else { + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); + } /* Add copy callback for not memory mapped DSPs */ if (platform->driver->compr_ops && platform->driver->compr_ops->c opy) compr->ops->copy = soc_compr_copy; mutex_init(&compr->lock); - - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, - rtd->codec_dai->name, num); - ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: compress: Remove duplicated code
by Donglin Peng 07 Aug '17

07 Aug '17

The value of the variable new_name is assigned repeatedly. Signed-off-by: Peng Donglin <pengdonglin137(a)163.com> --- sound/soc/soc-compress.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36b..2c16f42 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) -- 1.9.1

4 5

[alsa-devel] [PATCH] SPI: fix building SPI_PXA on MMP
by Arnd Bergmann 07 Aug '17

07 Aug '17

When the audio driver selects CONFIG_PXA_SSP on ARCH_MMP as a loadable module, and the PXA SPI driver is built-in, we get a link error in the SPI driver: drivers/spi/spi-pxa2xx.o: In function `pxa2xx_spi_remove': spi-pxa2xx.c:(.text+0x5f0): undefined reference to `pxa_ssp_free' drivers/spi/spi-pxa2xx.o: In function `pxa2xx_spi_probe': spi-pxa2xx.c:(.text+0xeac): undefined reference to `pxa_ssp_request' spi-pxa2xx.c:(.text+0x1468): undefined reference to `pxa_ssp_free' spi-pxa2xx.c:(.text+0x15bc): undefined reference to `pxa_ssp_free' The problem is that the PXA SPI driver only uses 'select SSP' specifically when building it for PXA, but we can also build it for PCI, which is meant for Intel x86 SoCs that use the same SPI block. When the sound driver forces the SSP to be a loadable module, the IS_ENABLED() check in include/linux/pxa2xx_ssp.h triggers but the spi driver can't reference the exported symbols. I had a different approach before, making the PCI case depend on X86, which fixed the problem by avoiding the MMP case. This goes a different route, making the driver select PXA_SSP also on MMP, which has an SSP that none of the boards in mainline Linux use for SPI. There is no harm in always enabling the build on MMP (PCI or not PCI), so I do that too, to document that this hardware is actually available on MMP. Link: https://patchwork.kernel.org/patch/8879921/ Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/spi/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/spi/Kconfig b/drivers/spi/Kconfig index 9b31351fe429..f9f9b0940746 100644 --- a/drivers/spi/Kconfig +++ b/drivers/spi/Kconfig @@ -518,8 +518,8 @@ config SPI_PPC4xx config SPI_PXA2XX tristate "PXA2xx SSP SPI master" - depends on (ARCH_PXA || PCI || ACPI) - select PXA_SSP if ARCH_PXA + depends on (ARCH_PXA || ARCH_MMP || PCI || ACPI) + select PXA_SSP if ARCH_PXA || ARCH_MMP help This enables using a PXA2xx or Sodaville SSP port as a SPI master controller. The driver can be configured to use any SSP port and -- 2.9.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: soc-core: Remove unneeded dentry member from snd_soc_codec
by Fabio Estevam 07 Aug '17

07 Aug '17

There is no need to have the *debugfs_reg dentry member as part of the snd_soc_codec structure as its only usage is inside soc_init_codec_debugfs(). Use a local dentry variable instead. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- include/sound/soc.h | 4 ---- sound/soc/soc-core.c | 9 +++++---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9d6e1db..42e2e50 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -897,10 +897,6 @@ struct snd_soc_codec { /* component */ struct snd_soc_component component; - -#ifdef CONFIG_DEBUG_FS - struct dentry *debugfs_reg; -#endif }; /* codec driver */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 13c875e..e93a72c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -339,11 +339,12 @@ static void soc_cleanup_component_debugfs(struct snd_soc_component *component) static void soc_init_codec_debugfs(struct snd_soc_component *component) { struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + struct dentry *debugfs_reg; - codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->component.debugfs_root, - codec, &codec_reg_fops); - if (!codec->debugfs_reg) + debugfs_reg = debugfs_create_file("codec_reg", 0644, + codec->component.debugfs_root, + codec, &codec_reg_fops); + if (!debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); } -- 2.7.4

2 2

[alsa-devel] [PATCH v3 1/2] ASoC: soc-core: Remove unneeded dentry member from snd_soc_codec
by Fabio Estevam 07 Aug '17

07 Aug '17

There is no need to have the *debugfs_reg dentry member as part of the snd_soc_codec structure as its only usage is inside soc_init_codec_debugfs(). Use a local dentry variable instead. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Changes since v2: - None include/sound/soc.h | 4 ---- sound/soc/soc-core.c | 9 +++++---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9d6e1db..42e2e50 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -897,10 +897,6 @@ struct snd_soc_codec { /* component */ struct snd_soc_component component; - -#ifdef CONFIG_DEBUG_FS - struct dentry *debugfs_reg; -#endif }; /* codec driver */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 13c875e..e93a72c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -339,11 +339,12 @@ static void soc_cleanup_component_debugfs(struct snd_soc_component *component) static void soc_init_codec_debugfs(struct snd_soc_component *component) { struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + struct dentry *debugfs_reg; - codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->component.debugfs_root, - codec, &codec_reg_fops); - if (!codec->debugfs_reg) + debugfs_reg = debugfs_create_file("codec_reg", 0644, + codec->component.debugfs_root, + codec, &codec_reg_fops); + if (!debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); } -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: pxa: don't select SSP for COMPILE_TESTing
by Arnd Bergmann 07 Aug '17

07 Aug '17

When the audio driver selects CONFIG_PXA_SSP to be a loadable module on a platform other than PXA, and the PXA SPI driver is built-in, we get a link error in the SPI driver: drivers/spi/spi-pxa2xx.o: In function `pxa2xx_spi_remove': spi-pxa2xx.c:(.text+0x5f0): undefined reference to `pxa_ssp_free' drivers/spi/spi-pxa2xx.o: In function `pxa2xx_spi_probe': spi-pxa2xx.c:(.text+0xeac): undefined reference to `pxa_ssp_request' spi-pxa2xx.c:(.text+0x1468): undefined reference to `pxa_ssp_free' spi-pxa2xx.c:(.text+0x15bc): undefined reference to `pxa_ssp_free' Fixes: 73d7ee2e831f ("ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/pxa/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484ab3c2ad67..1d392582559d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -33,7 +33,7 @@ config SND_PXA2XX_SOC_I2S config SND_PXA_SOC_SSP tristate - select PXA_SSP + select PXA_SSP if ARCH_PXA config SND_MMP_SOC_SSPA tristate @@ -141,7 +141,6 @@ config SND_PXA910_SOC config SND_SOC_TTC_DKB tristate "SoC Audio support for TTC DKB" depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y - select PXA_SSP select SND_PXA_SOC_SSP select SND_MMP_SOC select MFD_88PM860X -- 2.9.0

2 6

[alsa-devel] [PATCH] ASoC: compress: Remove duplicated code
by Peng Donglin 07 Aug '17

07 Aug '17

The value of the variable new_name is assigned repeatedly. Signed-off-by: Peng Donglin <pengdonglin137(a)163.com> --- sound/soc/soc-compress.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36b..2c16f42 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) -- 1.9.1

1 0

[alsa-devel] [PATCH RESEND 00/11] ALSA: vsnd: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 07 Aug '17

07 Aug '17

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Thank you, Oleksandr Resending because of rebase onto [2] [1] https://xenproject.org/ [2] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/log/?h=for-… Oleksandr Andrushchenko (11): ALSA: vsnd: Implement driver's probe/remove ALSA: vsnd: Implement Xen bus state handling ALSA: vsnd: Read sound driver configuration from Xen store ALSA: vsnd: Implement Xen event channel handling ALSA: vsnd: Implement handling of shared buffers ALSA: vsnd: Introduce ALSA virtual sound driver ALSA: vsnd: Initialize virtul sound card ALSA: vsnd: Add timer for period interrupt emulation ALSA: vsnd: Implement ALSA PCM operations ALSA: vsnd: Implement communication with backend ALSA: vsnd: Introduce Kconfig option to enable Xen PV sound sound/drivers/Kconfig | 12 + sound/drivers/Makefile | 2 + sound/drivers/xen-front.c | 2029 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 2043 insertions(+) -- 2.7.4

2 15

[alsa-devel] [PATCH] ASoC: mediatek: Fix an error checking code
by Christophe JAILLET 07 Aug '17

07 Aug '17

Check the value returned by 'devm_clk_get()' instead of the clock identifier which can never be an ERR code. Fixes: d6f3710a56e1 ("ASoC: mediatek: add structure define and clock control for 2701") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c b/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c index b815ecc6bbf6..affa7fb25dd9 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c @@ -75,7 +75,7 @@ int mt2701_init_clock(struct mtk_base_afe *afe) for (i = 0; i < MT2701_CLOCK_NUM; i++) { afe_priv->clocks[i] = devm_clk_get(afe->dev, aud_clks[i]); - if (IS_ERR(aud_clks[i])) { + if (IS_ERR(afe_priv->clocks[i])) { dev_warn(afe->dev, "%s devm_clk_get %s fail\n", __func__, aud_clks[i]); return PTR_ERR(aud_clks[i]); -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: rt5514: reset dma_offset at hw_params
by Oder Chiou 07 Aug '17

07 Aug '17

dma_offset needs reset every time otherwise for consecutive PCM open the last dma_offset value will cause incorrect pointer position be reported to user space. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 6f513cc..640193d 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -151,7 +151,6 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data) u8 buf[8]; rt5514_dsp->get_size = 0; - rt5514_dsp->dma_offset = 0; /** * The address area x1800XXXX is the register address, and it cannot @@ -205,6 +204,7 @@ static int rt5514_spi_hw_params(struct snd_pcm_substream *substream, ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); rt5514_dsp->substream = substream; + rt5514_dsp->dma_offset = 0; mutex_unlock(&rt5514_dsp->dma_lock); return ret; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rockchip: add bindings for i2s
by Sugar Zhang 07 Aug '17

07 Aug '17

This patch add bindings for rk3036/rk3228/rk3328/rk3366 i2s. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 206aba1..b208a75 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -7,8 +7,12 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 + - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 + - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 + - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 - reg: physical base address of the controller and length of memory mapped -- 2.7.4

3 3

[alsa-devel] [PATCH v2 1/2] ASoC: soc-core: Remove unneeded dentry member from snd_soc_codec
by Fabio Estevam 07 Aug '17

07 Aug '17

There is no need to have the *debugfs_reg dentry member as part of the snd_soc_codec structure as its only usage is inside soc_init_codec_debugfs(). Use a local dentry variable instead. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Changes since v1: - None include/sound/soc.h | 4 ---- sound/soc/soc-core.c | 9 +++++---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9d6e1db..42e2e50 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -897,10 +897,6 @@ struct snd_soc_codec { /* component */ struct snd_soc_component component; - -#ifdef CONFIG_DEBUG_FS - struct dentry *debugfs_reg; -#endif }; /* codec driver */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 13c875e..e93a72c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -339,11 +339,12 @@ static void soc_cleanup_component_debugfs(struct snd_soc_component *component) static void soc_init_codec_debugfs(struct snd_soc_component *component) { struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + struct dentry *debugfs_reg; - codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->component.debugfs_root, - codec, &codec_reg_fops); - if (!codec->debugfs_reg) + debugfs_reg = debugfs_create_file("codec_reg", 0644, + codec->component.debugfs_root, + codec, &codec_reg_fops); + if (!debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); } -- 2.7.4

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Re: [alsa-devel] [PATCH v4 15/17] ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35
by Richard Fitzgerald 07 Aug '17

07 Aug '17

On Sun, 2017-07-30 at 07:01 +0800, kbuild test robot wrote: > Hi Richard, > > [auto build test ERROR on linus/master] > [also build test ERROR on v4.13-rc2 next-20170728] > [cannot apply to ljones-mfd/for-mfd-next asoc/for-next] > [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] > > url: https://github.com/0day-ci/linux/commits/Richard-Fitzgerald/Add-support-for… > config: x86_64-allmodconfig (attached as .config) > compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 > reproduce: > # save the attached .config to linux build tree > make ARCH=x86_64 > > Note: the linux-review/Richard-Fitzgerald/Add-support-for-Cirrus-Logic-CS47L35-L85-L90-L91-codecs/20170730-062737 HEAD 15487144eec6910ecfbbbbd97bb5a2c7efdb259f builds fine. > It only hurts bisectibility. > > All errors (new ones prefixed by >>): > > >> sound/soc//codecs/madera.c:146:2: error: 'MADERA_IRQ_DSP1_BUS_ERROR' undeclared here (not in a function) > MADERA_IRQ_DSP1_BUS_ERROR, > ^~~~~~~~~~~~~~~~~~~~~~~~~ > >> sound/soc//codecs/madera.c:147:2: error: 'MADERA_IRQ_DSP2_BUS_ERROR' undeclared here (not in a function) > MADERA_IRQ_DSP2_BUS_ERROR, > ^~~~~~~~~~~~~~~~~~~~~~~~~ > >> sound/soc//codecs/madera.c:148:2: error: 'MADERA_IRQ_DSP3_BUS_ERROR' undeclared here (not in a function) > MADERA_IRQ_DSP3_BUS_ERROR, > ^~~~~~~~~~~~~~~~~~~~~~~~~ > >> sound/soc//codecs/madera.c:149:2: error: 'MADERA_IRQ_DSP4_BUS_ERROR' undeclared here (not in a function) > MADERA_IRQ_DSP4_BUS_ERROR, > ^~~~~~~~~~~~~~~~~~~~~~~~~ > >> sound/soc//codecs/madera.c:150:2: error: 'MADERA_IRQ_DSP5_BUS_ERROR' undeclared here (not in a function) > MADERA_IRQ_DSP5_BUS_ERROR, > ^~~~~~~~~~~~~~~~~~~~~~~~~ > >> sound/soc//codecs/madera.c:151:2: error: 'MADERA_IRQ_DSP6_BUS_ERROR' undeclared here (not in a function) > MADERA_IRQ_DSP6_BUS_ERROR, > ^~~~~~~~~~~~~~~~~~~~~~~~~ > >> sound/soc//codecs/madera.c:152:2: error: 'MADERA_IRQ_DSP7_BUS_ERROR' undeclared here (not in a function) > MADERA_IRQ_DSP7_BUS_ERROR, > ^~~~~~~~~~~~~~~~~~~~~~~~~ > Oops. The rename of these somehow got merged into the top patch instead of going into #14 where it should be. I'll wait a while and see what comments I get on the earlier patches, maybe they will all be accepted for merge and then I'll only have to reissue the top 4 patches to fix this. > vim +/MADERA_IRQ_DSP1_BUS_ERROR +146 sound/soc//codecs/madera.c > > da1efc4cc Richard Fitzgerald 2017-07-28 130 > da1efc4cc Richard Fitzgerald 2017-07-28 131 #define madera_fll_err(_fll, fmt, ...) \ > da1efc4cc Richard Fitzgerald 2017-07-28 132 dev_err(_fll->madera->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) > da1efc4cc Richard Fitzgerald 2017-07-28 133 #define madera_fll_warn(_fll, fmt, ...) \ > da1efc4cc Richard Fitzgerald 2017-07-28 134 dev_warn(_fll->madera->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) > da1efc4cc Richard Fitzgerald 2017-07-28 135 #define madera_fll_dbg(_fll, fmt, ...) \ > da1efc4cc Richard Fitzgerald 2017-07-28 136 dev_dbg(_fll->madera->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) > da1efc4cc Richard Fitzgerald 2017-07-28 137 > da1efc4cc Richard Fitzgerald 2017-07-28 138 #define madera_aif_err(_dai, fmt, ...) \ > da1efc4cc Richard Fitzgerald 2017-07-28 139 dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) > da1efc4cc Richard Fitzgerald 2017-07-28 140 #define madera_aif_warn(_dai, fmt, ...) \ > da1efc4cc Richard Fitzgerald 2017-07-28 141 dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) > da1efc4cc Richard Fitzgerald 2017-07-28 142 #define madera_aif_dbg(_dai, fmt, ...) \ > da1efc4cc Richard Fitzgerald 2017-07-28 143 dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) > da1efc4cc Richard Fitzgerald 2017-07-28 144 > da1efc4cc Richard Fitzgerald 2017-07-28 145 static const int madera_dsp_bus_error_irqs[MADERA_MAX_ADSP] = { > da1efc4cc Richard Fitzgerald 2017-07-28 @146 MADERA_IRQ_DSP1_BUS_ERROR, > da1efc4cc Richard Fitzgerald 2017-07-28 @147 MADERA_IRQ_DSP2_BUS_ERROR, > da1efc4cc Richard Fitzgerald 2017-07-28 @148 MADERA_IRQ_DSP3_BUS_ERROR, > da1efc4cc Richard Fitzgerald 2017-07-28 @149 MADERA_IRQ_DSP4_BUS_ERROR, > da1efc4cc Richard Fitzgerald 2017-07-28 @150 MADERA_IRQ_DSP5_BUS_ERROR, > da1efc4cc Richard Fitzgerald 2017-07-28 @151 MADERA_IRQ_DSP6_BUS_ERROR, > da1efc4cc Richard Fitzgerald 2017-07-28 @152 MADERA_IRQ_DSP7_BUS_ERROR, > da1efc4cc Richard Fitzgerald 2017-07-28 153 }; > da1efc4cc Richard Fitzgerald 2017-07-28 154 > > :::::: The code at line 146 was first introduced by commit > :::::: da1efc4cc715281f9599274bb5ff408b19cd3335 ASoC: madera: Add common support for Cirrus Logic Madera codecs > > :::::: TO: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > :::::: CC: 0day robot <fengguang.wu(a)intel.com> > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v3 00/12] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 07 Aug '17

07 Aug '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 I don't own a A33 device which uses the i2s block for the audio codec so if someone could test against that it would be much appreciated. I'm also wondering if there is a preferred way of setting the lrclk size in the dts?..currently it is set to the sample width but for example the pcm5102a wants it to be 32 bits whatever the sample rate. Thanks in advance, CK --- v3 changes compared to v2 are: - initial changes to prepare driver for newer SoCs has been broken down into smaller patches - reduce use of regmap fields to where just needed. - clkdiv expansion will be delivered later. - defines for H3 variant segregated. - fixed regmap config issue with SUN8I_I2S_FIFO_TX_REG. v2 changes compared to v1 are: - massive refactoring to remove duplicate code making use of regmap_fields. - extending the clock divisors. - removed code that should be delivered when we support 20/24bits --- Marcus Cooper (12): ASoC: sun4i-i2s: Extend quirks scope ASoC: sun4i-i2s: Add clkdiv offsets to quirks ASoC: sun4i-i2s: Add regmap config to quirks ASoC: sun4i-i2s: Add TX FIFO offset to quirks ASoC: sun4i-i2s: Add regmap fields for channels ASoC: sun4i-i2s: Add changes for wss and sr ASoC: sun4i-i2s: bclk and lrclk polarity tidyup ASoC: sun4i-i2s: Add mclk enable regmap field ASoC: sun4i-i2s: Add regmap field to set format ASoC: sun4i-i2s: Check for slave select bit ASoC: sun4i-i2s: Update global enable with bitmask ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 460 ++++++++++++++++++--- 2 files changed, 398 insertions(+), 64 deletions(-) -- 2.13.3

6 32

[alsa-devel] [PATCH v2] ASoC: Intel: Reset hw_ptr on resume trigger
by Cheng-Yi Chiang 07 Aug '17

07 Aug '17

From: "U. Artie Eoff" <ullysses.a.eoff(a)intel.com> Reset the hw_ptr before queuing the restore_stream_context work to eradicate a nasty white audio noise on resume. Tested-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 4765ad474544..e0db7070cd42 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -187,8 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - if (pdata->restore_stream == true) + if (pdata->restore_stream == true) { + pcm_data->hw_ptr = 0; schedule_work(&pcm_data->work); + } else sst_byt_stream_resume(byt, pcm_data->stream); break; -- 2.12.2

4 6

[alsa-devel] [PATCH 0/7] constify ALSA usb_device_id.
by Arvind Yadav 06 Aug '17

06 Aug '17

usb_device_id are not supposed to change at runtime. All functions working with usb_device_id provided by <linux/usb.h> work with const usb_device_id. So mark the non-const structs as const. Arvind Yadav (7): [PATCH 1/7] ALSA: 6fire: constify usb_device_id. [PATCH 2/7] ALSA: bcd2000: constify usb_device_id. [PATCH 3/7] ALSA: snd-usb-caiaq: constify usb_device_id. [PATCH 4/7] ALSA: usb-audio: constify usb_device_id. [PATCH 5/7] ALSA: ua101: constify usb_device_id. [PATCH 6/7] ALSA: us122l: constify usb_device_id. [PATCH 7/7] ALSA: usbusx2y: constify usb_device_id. sound/usb/6fire/chip.c | 2 +- sound/usb/bcd2000/bcd2000.c | 2 +- sound/usb/caiaq/device.c | 2 +- sound/usb/card.c | 4 ++-- sound/usb/misc/ua101.c | 2 +- sound/usb/usx2y/us122l.c | 2 +- sound/usb/usx2y/usbusx2y.c | 2 +- 7 files changed, 8 insertions(+), 8 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH] ALSA: hda: Add Cannonlake PCI ID
by Guneshwor Singh 06 Aug '17

06 Aug '17

Cannonlake is next generation Intel platform. This commit adds PCI ID for it. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5ae8ddab6412..f958d8d54d15 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2389,6 +2389,9 @@ static const struct pci_device_id azx_ids[] = { /* Coffelake */ { PCI_DEVICE(0x8086, 0xa348), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Cannonlake */ + { PCI_DEVICE(0x8086, 0x9dc8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, -- 2.13.0

2 1

[alsa-devel] [PATCH] ALSA: ice1712: add const to snd_akm4xxx structures
by Bhumika Goyal 06 Aug '17

06 Aug '17

Declare snd_akm4xxx structures as const as they are only passed to the function snd_ice1712_akm4xxx_init. This argument is of type const, so make the structures const. Done using Coccinelle: @match disable optional_qualifier@ identifier s; position p; @@ static struct snd_akm4xxx s@p={...}; @good1@ identifier match.s; position p; @@ snd_ice1712_akm4xxx_init(...,&s@p,...) @bad@ identifier match.s; position p!={match.p,good1.p}; @@ s@p @depends on !bad disable optional_qualifier@ identifier match.s; @@ static +const struct snd_akm4xxx s={...}; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/ice1712/delta.c | 12 ++++++------ sound/pci/ice1712/ews.c | 6 +++--- sound/pci/ice1712/hoontech.c | 2 +- sound/pci/ice1712/juli.c | 2 +- sound/pci/ice1712/phase.c | 2 +- sound/pci/ice1712/quartet.c | 2 +- sound/pci/ice1712/revo.c | 10 +++++----- 7 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index da5f37b..ab4461d 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -445,7 +445,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco * initialize the chips on M-Audio cards */ -static struct snd_akm4xxx akm_audiophile = { +static const struct snd_akm4xxx akm_audiophile = { .type = SND_AK4528, .num_adcs = 2, .num_dacs = 2, @@ -466,7 +466,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco .mask_flags = 0, }; -static struct snd_akm4xxx akm_delta410 = { +static const struct snd_akm4xxx akm_delta410 = { .type = SND_AK4529, .num_adcs = 2, .num_dacs = 8, @@ -487,7 +487,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco .mask_flags = 0, }; -static struct snd_akm4xxx akm_delta1010lt = { +static const struct snd_akm4xxx akm_delta1010lt = { .type = SND_AK4524, .num_adcs = 8, .num_dacs = 8, @@ -509,7 +509,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco .mask_flags = 0, }; -static struct snd_akm4xxx akm_delta66e = { +static const struct snd_akm4xxx akm_delta66e = { .type = SND_AK4524, .num_adcs = 4, .num_dacs = 4, @@ -532,7 +532,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco }; -static struct snd_akm4xxx akm_delta44 = { +static const struct snd_akm4xxx akm_delta44 = { .type = SND_AK4524, .num_adcs = 4, .num_dacs = 4, @@ -554,7 +554,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco .mask_flags = 0, }; -static struct snd_akm4xxx akm_vx442 = { +static const struct snd_akm4xxx akm_vx442 = { .type = SND_AK4524, .num_adcs = 4, .num_dacs = 4, diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index ec07136..de3603e 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -344,7 +344,7 @@ static void ews88_setup_spdif(struct snd_ice1712 *ice, int rate) /* */ -static struct snd_akm4xxx akm_ews88mt = { +static const struct snd_akm4xxx akm_ews88mt = { .num_adcs = 8, .num_dacs = 8, .type = SND_AK4524, @@ -366,7 +366,7 @@ static void ews88_setup_spdif(struct snd_ice1712 *ice, int rate) .mask_flags = 0, }; -static struct snd_akm4xxx akm_ewx2496 = { +static const struct snd_akm4xxx akm_ewx2496 = { .num_adcs = 2, .num_dacs = 2, .type = SND_AK4524, @@ -387,7 +387,7 @@ static void ews88_setup_spdif(struct snd_ice1712 *ice, int rate) .mask_flags = 0, }; -static struct snd_akm4xxx akm_6fire = { +static const struct snd_akm4xxx akm_6fire = { .num_adcs = 6, .num_dacs = 6, .type = SND_AK4524, diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index a40001c..0bbb5f4 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -269,7 +269,7 @@ static void stdsp24_ak4524_lock(struct snd_akm4xxx *ak, int chip) static int snd_ice1712_value_init(struct snd_ice1712 *ice) { /* Hoontech STDSP24 with modified hardware */ - static struct snd_akm4xxx akm_stdsp24_mv = { + static const struct snd_akm4xxx akm_stdsp24_mv = { .num_adcs = 2, .num_dacs = 2, .type = SND_AK4524, diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 5bb1467..0dbaccf 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -282,7 +282,7 @@ static void juli_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) }; -static struct snd_akm4xxx akm_juli_dac = { +static const struct snd_akm4xxx akm_juli_dac = { .type = SND_AK4358, .num_dacs = 8, /* DAC1 - analog out DAC2 - analog in monitor diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index e9ca89c..94bf8ed 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -102,7 +102,7 @@ struct phase28_spec { #define WM_VOL_MAX (sizeof(wm_vol) - 1) #define WM_VOL_MUTE 0x8000 -static struct snd_akm4xxx akm_phase22 = { +static const struct snd_akm4xxx akm_phase22 = { .type = SND_AK4524, .num_dacs = 2, .num_adcs = 2, diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index f1b3732..d145b5e 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -386,7 +386,7 @@ static void qtet_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) AK_CONTROL(PCM_34_CAPTURE_VOLUME, 2), }; -static struct snd_akm4xxx akm_qtet_dac = { +static const struct snd_akm4xxx akm_qtet_dac = { .type = SND_AK4620, .num_dacs = 4, /* DAC1 - Output 12 */ diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1d81ae6..5cd3225 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -235,7 +235,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, }, }; -static struct snd_akm4xxx akm_revo_front = { +static const struct snd_akm4xxx akm_revo_front = { .type = SND_AK4381, .num_dacs = 2, .ops = { @@ -256,7 +256,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, .mask_flags = 0, }; -static struct snd_akm4xxx akm_revo_surround = { +static const struct snd_akm4xxx akm_revo_surround = { .type = SND_AK4355, .idx_offset = 1, .num_dacs = 6, @@ -278,7 +278,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, .mask_flags = 0, }; -static struct snd_akm4xxx akm_revo51 = { +static const struct snd_akm4xxx akm_revo51 = { .type = SND_AK4358, .num_dacs = 8, .ops = { @@ -299,7 +299,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, .mask_flags = 0, }; -static struct snd_akm4xxx akm_revo51_adc = { +static const struct snd_akm4xxx akm_revo51_adc = { .type = SND_AK5365, .num_adcs = 2, .adc_info = revo51_adc, @@ -346,7 +346,7 @@ static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) AK_DAC("PCM Playback Volume", 2) }; -static struct snd_akm4xxx akm_ap192 = { +static const struct snd_akm4xxx akm_ap192 = { .type = SND_AK4358, .num_dacs = 2, .ops = { -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: ice1712: add const to snd_ak4xxx_private structures
by Bhumika Goyal 06 Aug '17

06 Aug '17

Declare snd_ak4xxx_private structures as const as they are only passed to the function snd_ice1712_akm4xxx_init. This argument is of type const, so make the structures const. Done using Coccinelle: @match disable optional_qualifier@ identifier s; position p; @@ static struct snd_ak4xxx_private s@p={...}; @good1@ identifier match.s; position p; @@ snd_ice1712_akm4xxx_init(...,&s@p,...) @bad@ identifier match.s; position p!={match.p,good1.p}; @@ s@p @depends on !bad disable optional_qualifier@ identifier match.s; @@ static +const struct snd_ak4xxx_private s={...}; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/ice1712/delta.c | 12 ++++++------ sound/pci/ice1712/ews.c | 6 +++--- sound/pci/ice1712/hoontech.c | 2 +- sound/pci/ice1712/phase.c | 2 +- sound/pci/ice1712/revo.c | 10 +++++----- 5 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index da5f37b..4e470f0 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -454,7 +454,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco } }; -static struct snd_ak4xxx_private akm_audiophile_priv = { +static const struct snd_ak4xxx_private akm_audiophile_priv = { .caddr = 2, .cif = 0, .data_mask = ICE1712_DELTA_AP_DOUT, @@ -475,7 +475,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco } }; -static struct snd_ak4xxx_private akm_delta410_priv = { +static const struct snd_ak4xxx_private akm_delta410_priv = { .caddr = 0, .cif = 0, .data_mask = ICE1712_DELTA_AP_DOUT, @@ -497,7 +497,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco } }; -static struct snd_ak4xxx_private akm_delta1010lt_priv = { +static const struct snd_ak4xxx_private akm_delta1010lt_priv = { .caddr = 2, .cif = 0, /* the default level of the CIF pin from AK4524 */ .data_mask = ICE1712_DELTA_1010LT_DOUT, @@ -519,7 +519,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco } }; -static struct snd_ak4xxx_private akm_delta66e_priv = { +static const struct snd_ak4xxx_private akm_delta66e_priv = { .caddr = 2, .cif = 0, /* the default level of the CIF pin from AK4524 */ .data_mask = ICE1712_DELTA_66E_DOUT, @@ -542,7 +542,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco } }; -static struct snd_ak4xxx_private akm_delta44_priv = { +static const struct snd_ak4xxx_private akm_delta44_priv = { .caddr = 2, .cif = 0, /* the default level of the CIF pin from AK4524 */ .data_mask = ICE1712_DELTA_CODEC_SERIAL_DATA, @@ -564,7 +564,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco } }; -static struct snd_ak4xxx_private akm_vx442_priv = { +static const struct snd_ak4xxx_private akm_vx442_priv = { .caddr = 2, .cif = 0, .data_mask = ICE1712_VX442_DOUT, diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index ec07136..5bc14e8 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -354,7 +354,7 @@ static void ews88_setup_spdif(struct snd_ice1712 *ice, int rate) } }; -static struct snd_ak4xxx_private akm_ews88mt_priv = { +static const struct snd_ak4xxx_private akm_ews88mt_priv = { .caddr = 2, .cif = 1, /* CIF high */ .data_mask = ICE1712_EWS88_SERIAL_DATA, @@ -375,7 +375,7 @@ static void ews88_setup_spdif(struct snd_ice1712 *ice, int rate) } }; -static struct snd_ak4xxx_private akm_ewx2496_priv = { +static const struct snd_ak4xxx_private akm_ewx2496_priv = { .caddr = 2, .cif = 1, /* CIF high */ .data_mask = ICE1712_EWS88_SERIAL_DATA, @@ -396,7 +396,7 @@ static void ews88_setup_spdif(struct snd_ice1712 *ice, int rate) } }; -static struct snd_ak4xxx_private akm_6fire_priv = { +static const struct snd_ak4xxx_private akm_6fire_priv = { .caddr = 2, .cif = 1, /* CIF high */ .data_mask = ICE1712_6FIRE_SERIAL_DATA, diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index a40001c..de2fcc9 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -278,7 +278,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice) } }; - static struct snd_ak4xxx_private akm_stdsp24_mv_priv = { + static const struct snd_ak4xxx_private akm_stdsp24_mv_priv = { .caddr = 2, .cif = 1, /* CIF high */ .data_mask = ICE1712_STDSP24_SERIAL_DATA, diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index e9ca89c..4f85482 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -108,7 +108,7 @@ struct phase28_spec { .num_adcs = 2, }; -static struct snd_ak4xxx_private akm_phase22_priv = { +static const struct snd_ak4xxx_private akm_phase22_priv = { .caddr = 2, .cif = 1, .data_mask = 1 << 4, diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1d81ae6..5cc4213 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -244,7 +244,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, .dac_info = revo71_front, }; -static struct snd_ak4xxx_private akm_revo_front_priv = { +static const struct snd_ak4xxx_private akm_revo_front_priv = { .caddr = 1, .cif = 0, .data_mask = VT1724_REVO_CDOUT, @@ -266,7 +266,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, .dac_info = revo71_surround, }; -static struct snd_ak4xxx_private akm_revo_surround_priv = { +static const struct snd_ak4xxx_private akm_revo_surround_priv = { .caddr = 3, .cif = 0, .data_mask = VT1724_REVO_CDOUT, @@ -287,7 +287,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, .dac_info = revo51_dac, }; -static struct snd_ak4xxx_private akm_revo51_priv = { +static const struct snd_ak4xxx_private akm_revo51_priv = { .caddr = 2, .cif = 0, .data_mask = VT1724_REVO_CDOUT, @@ -305,7 +305,7 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, .adc_info = revo51_adc, }; -static struct snd_ak4xxx_private akm_revo51_adc_priv = { +static const struct snd_ak4xxx_private akm_revo51_adc_priv = { .caddr = 2, .cif = 0, .data_mask = VT1724_REVO_CDOUT, @@ -355,7 +355,7 @@ static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) .dac_info = ap192_dac, }; -static struct snd_ak4xxx_private akm_ap192_priv = { +static const struct snd_ak4xxx_private akm_ap192_priv = { .caddr = 2, .cif = 0, .data_mask = VT1724_REVO_CDOUT, -- 1.9.1

2 1

[alsa-devel] [PATCH v06 31/36] uapi linux/patchkey.h: change #error to #warning if file included directly
by Mikko Rapeli 06 Aug '17

06 Aug '17

Would be nice to be able to compile all userspace header files also alone. Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> Cc: alsa-devel(a)alsa-project.org Cc: Andrew Veliath <andrewtv(a)usa.net> Cc: Riccardo Facchetti <fizban(a)tin.it> --- include/uapi/linux/patchkey.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/linux/patchkey.h b/include/uapi/linux/patchkey.h index 1bda0e56bc5e..cee195d6d826 100644 --- a/include/uapi/linux/patchkey.h +++ b/include/uapi/linux/patchkey.h @@ -11,7 +11,7 @@ */ #ifndef _LINUX_PATCHKEY_H_INDIRECT -#error "patchkey.h included directly" +#warning "patchkey.h included directly" #endif #ifndef _UAPI_LINUX_PATCHKEY_H -- 2.13.3

1 0

[alsa-devel] ASUS G752VS: Linux audio/sound problems
by Aljoša 06 Aug '17

06 Aug '17

ASUS G752VS laptop - latest 310 BIOS; OS: Ubuntu 17.04 - fully updated, kernel 4.10.0-30. This Asus laptop has two stereo speakers behind the display on the rear side of the notebook, and a separate subwoofer that sits on the underside. I'm reporting here the following Linux audio/sound issues: 1) Microphone jack doesn't work (it works ok in Windows 10). 2) Headphone jack doesn't work (it works ok in Windows 10). It works somehow when the hdajackretask headphones trick is applied (override pin 0x16 to headpohne). 3) If within sound settings the "Analog Stereo Output" mode is selected, the subwoofer doesn't work and its volume control is grayed out (disabled). If within sound settings the "Analog Surround 4.0 Output" mode is selected, fade control works even if laptop doesn't have both front and rear speakers, and the subwoofer works ok but its volume control is grayed out (disabled). Test laptop sound (2 speakers + subwoofer) without hdajackretask headphones trick (override pin 0x16 to headpohne) applied: Front left and front right both works ok. Rear left and rear right both works ok and sound is coming from the subwoofer both when rear left and rear right test button is clicked. Test laptop sound (2 speakers + subwoofer) with hdajackretask headphones trick (override pin 0x16 to headpohne) applied: Front left and front right both doesn't work. Rear left and rear right both works ok and sound is coming from the subwoofer both when rear left and rear right test button is clicked. 4) Not sure if this is somehow related to audio/sound problems, but the fact is that the right side of the laptop (area around audio jacks) is pretty hot compared to the left side which is completely cold (in Windoes 10 the area around audio jacks is completely cold). Please find attached the 'alsa-info.sh output' and two related images. Best regards, Aljosa.

1 0

[alsa-devel] [PATCH 0/5] ALSA: control: fix issue of concurrent access for TLV operations
by Takashi Sakamoto 05 Aug '17

05 Aug '17

Hi, In a design of ALSA control core, each set of elements have data of Type-Length-Value shape. This is called as TLV data. In ALSA control interface, applications can three types of requests to elements about TLV data; read, write and command. In current implementation of ALSA control core, these three requests are handled with read lock of a counting semaphore. However, this has an issue of concurrent access because in theory write/command requests have sub-effects to change state of the set of elements in a point of TLV data. Read requests and write/command requests are handled exclusively as well as each write/command requests. This patchset is for this purpose. Additionally, this applies code refactoring to TLV ioctl handler and TLV handler for user-defined element set so that they get better shape for readers. Takashi Sakamoto (5): ALSA: control: queue events within locking of controls_rwsem for TLV operation ALSA: control: use counting semaphore as write lock for TLV write/command operations ALSA: control: obsolete user_ctl_lock ALSA: control: code refactoring TLV ioctl handler ALSA: control: code refactoring for TLV request handler to user element set include/sound/core.h | 2 - sound/core/control.c | 252 +++++++++++++++++++++++++++++++-------------------- sound/core/init.c | 1 - 3 files changed, 155 insertions(+), 100 deletions(-) -- 2.11.0

2 7

[alsa-devel] [PATCH] ASoC: Intel: constify pci_device_id.
by Arvind Yadav 04 Aug '17

04 Aug '17

pci_device_id are not supposed to change at runtime. All functions working with pci_device_id provided by <linux/pci.h> work with const pci_device_id. So mark the non-const structs as const. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/soc/intel/atom/sst/sst_pci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c index 3a0b3bf..6906ee6 100644 --- a/sound/soc/intel/atom/sst/sst_pci.c +++ b/sound/soc/intel/atom/sst/sst_pci.c @@ -181,7 +181,7 @@ static void intel_sst_remove(struct pci_dev *pci) } /* PCI Routines */ -static struct pci_device_id intel_sst_ids[] = { +static const struct pci_device_id intel_sst_ids[] = { { PCI_VDEVICE(INTEL, SST_MRFLD_PCI_ID), 0}, { 0, } }; -- 2.7.4

2 1

[alsa-devel] [PATCH RESEND v3 0/3] Add support for es8323 and hdmi audio to the firefly-rk3288
by Romain Perier 04 Aug '17

04 Aug '17

This set of patches enables support for the es8323 codecs on the firefly-rk3288. It adds the required DT definitions to link rockchip-i2s to the es8388 analog output. As the support for HDMI audio is also built-in in the corresponding ASoC machine driver, HDMI audio output is also enabled. Changes in v3: - Synced DT with the one of the Rock2, the properties have been lexically re-ordered in the right order, I have splitted the headphone pinctrl in two: phone_ctl and hp_det Changes in v2: - Adapted DT for the firefly-rk3288 to the ASoC machine driver rk3288-hdmi-analog - Added patch 02/03 to this series - Re-wrote this cover letter Romain Perier (3): SoC: es8328-i2c: Add compatible for ES8323 arm: dts: rockchip: Don't set otp-gpio pinctrl by default in rk3288.dtsi arm: dts: Add support for ES8323 to the Firefly-RK3288 Documentation/devicetree/bindings/sound/es8328.txt | 5 ++- arch/arm/boot/dts/rk3288-evb.dtsi | 4 +++ arch/arm/boot/dts/rk3288-firefly.dtsi | 38 ++++++++++++++++++++++ arch/arm/boot/dts/rk3288-veyron.dtsi | 4 +++ arch/arm/boot/dts/rk3288.dtsi | 4 --- sound/soc/codecs/es8328-i2c.c | 2 ++ 6 files changed, 52 insertions(+), 5 deletions(-) -- 2.11.0

2 13

[alsa-devel] Problem with DoP mode in Chord 2qute USB audio
by Pavel Hofman 03 Aug '17

03 Aug '17

Hello, I am working with owner of Chord 2qute on the following problem (FYI long discussion at http://www.diyaudio.com/forums/pc-based/310578-chord-2qute-dop-challenge.ht…): Regular 176.4/24 PCM plays OK, however 176.4/24 DoP produces random 1-2 sec sound interruptions every several tens of seconds. Tested players - squeezelite, MPD. The soundcard has its own power supply. I googled out several reports of simlar problem with this soundcard, mostly on linux (did not see any problem report for OSX). alsainfo: http://www.fyn13.dk/diy/alsa_output_chord_2qute_4.4.0-62.txt lsusb -v : http://www.fyn13.dk/diy/chord_lsusb.txt hw_params and stream0 files are identical for both 176.4/24 PCM and DoP: /proc/asound/card1/pcm0p/sub0/hw_params: access: RW_INTERLEAVED format: S32_LE subformat: STD channels: 2 rate: 176400 (176400/1) period_size: 1764 buffer_size: 7056 /proc/asound/card1/stream0: Chord Electronics Ltd 2Qute at usb-0000:00:14.0-2, high speed : USB Audio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 224 Momentary freq = 176402 Hz (0x16.0ce0) Feedback Format = 16.16 Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 5 OUT (ASYNC) Rates: 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 Data packet interval: 125 us We checked for XRUNs, none were produced during the interruption in DoP playback. Powerful 4-core CPU Intel Pentium 4405U @ 2.10GHz, idle > 99% of time in DoP playback. However, the IRQs rate of the onboard XHCI controller is 8k/s for both PCM and DoP playback which corresponds to 1 IRQ every 125us frame. The rate drops to almost zero when playback is stopped in the player - clearly it is related to the soundcard. Could this perhaps be a sign of the problem? Please is there a way to lower the rate, like the old removed param nrpacks of the usb-audio driver? I suspect a buggy XMOS firmware. I very much appreciate any suggestion or help, either with fixing in software, or further diagnostics to support potential claim to the manufacturer. Thank you very much. With regards, Pavel.

2 2

[alsa-devel] [PATCH 0/3] ASoC: Intel: kbl: Add Multi-endpoint Playback & Echo Ref support
by Naveen M 03 Aug '17

03 Aug '17

1. Add new FE capabilities to support: a. Parallel playback on 2 ports simultaneously Define new frontends to enable additional playback port. This will be used when platform needs to support multi-playback on different ports. Ex: Headset and Speaker simultaneously b. Echo reference capture capability Certain platforms need reference stream of playback to supplement User-space post-processing algorithms. This pin enables platform driver to provide such a FE. 2. Add DAI links for parallel playback and echo ref 3. Add map for new DAIs for parallel playback and echo ref Naveen M (3): ASoC: Intel: kbl: Add new FEs for Multi-Playback & Echo-Reference ASoC: Intel: kbl: Add DAI links for Multi-Playback & Echo-reference ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref sound/soc/intel/boards/kbl_rt5663_max98927.c | 30 +++++++++++++++++++--- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 30 +++++++++++++++++++--- sound/soc/intel/skylake/skl-pcm.c | 26 +++++++++++++++++++ 3 files changed, 80 insertions(+), 6 deletions(-) -- 1.9.1

3 7

[alsa-devel] [PATCH 0/9] Add Cannonlake Audio DSP support
by Guneshwor Singh 03 Aug '17

03 Aug '17

Cannonlake (CNL) audio is similar to Broxton with a substantial difference in ipc registers. This series enables CNL support by adding required dsp library functions and ipc register programming. CNL has 4 dsp cores, additional core support is also added here. To accommodate the above changes in the current code, existing helper functions, common functions that are reusable and dsp ops are modified. Dharageswari R (1): ASoC: Intel: Skylake: Add num of cores in dsp ops Guneshwor Singh (7): ASoC: Intel: Skylake: Use num_core to allocate instead of macro ASoC: Intel: cnl: Add cnl dsp functions and registers ASoC: Intel: Skylake: Move platform specific init to platform dsp_init() ASoC: Intel: cnl: Unstatify common ipc functions ASoC: Intel: cnl: Add sst library functions for cnl platform ASoC: Intel: cnl: add dsp ops for cannonlake ASoC: Intel: cnl: add pci id for cnl Pardha Saradhi K (1): ASoC: Intel: Skylake: Add dsp cores management sound/soc/intel/skylake/Makefile | 5 +- sound/soc/intel/skylake/bxt-sst.c | 5 +- sound/soc/intel/skylake/cnl-sst-dsp.c | 274 ++++++++++++++++++ sound/soc/intel/skylake/cnl-sst-dsp.h | 112 ++++++++ sound/soc/intel/skylake/cnl-sst.c | 495 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 31 ++ sound/soc/intel/skylake/skl-sst-dsp.c | 2 +- sound/soc/intel/skylake/skl-sst-ipc.c | 6 +- sound/soc/intel/skylake/skl-sst-ipc.h | 12 +- sound/soc/intel/skylake/skl-sst-utils.c | 6 +- sound/soc/intel/skylake/skl-sst.c | 6 +- sound/soc/intel/skylake/skl-topology.c | 26 +- sound/soc/intel/skylake/skl.c | 11 + sound/soc/intel/skylake/skl.h | 1 + 14 files changed, 970 insertions(+), 22 deletions(-) create mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h create mode 100644 sound/soc/intel/skylake/cnl-sst.c -- 2.13.0

3 19

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Use correct nuvoton codec ID
by Naveen M 03 Aug '17

03 Aug '17

Correcting the nau88l25 codec ID to match secondary codec and load the machine driver. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917e..608c147 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -879,12 +879,12 @@ static void skl_remove(struct pci_dev *pci) static struct sst_codecs skl_codecs = { .num_codecs = 1, - .codecs = {"NAU88L25"} + .codecs = {"10508825"} }; static struct sst_codecs kbl_codecs = { .num_codecs = 1, - .codecs = {"NAU88L25"} + .codecs = {"10508825"} }; static struct sst_codecs bxt_codecs = { -- 1.9.1

3 2

[alsa-devel] [PATCH] drm: exynos: Add driver for HDMI audio interface
by Sylwester Nawrocki 03 Aug '17

03 Aug '17

The hdmi-codec interface added in this patch is required to properly support HDMI audio. Currently the audio part of the SoC internal HDMI transmitter is configured with fixed values, which makes HDMI audio working by chance, only on boards equipped with external audio codec connected in parallel with the HDMI audio transmitter I2S input interface. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since RFC version: - fixed hdmi-codec locking issue - added a comment documenting struct hdmi_contex::mutex --- drivers/gpu/drm/exynos/Kconfig | 1 + drivers/gpu/drm/exynos/exynos_hdmi.c | 226 +++++++++++++++++++++++++++++------ 2 files changed, 193 insertions(+), 34 deletions(-) diff --git a/drivers/gpu/drm/exynos/Kconfig b/drivers/gpu/drm/exynos/Kconfig index 1d18534..a6edbb6 100644 --- a/drivers/gpu/drm/exynos/Kconfig +++ b/drivers/gpu/drm/exynos/Kconfig @@ -3,6 +3,7 @@ config DRM_EXYNOS depends on OF && DRM && (ARCH_S3C64XX || ARCH_EXYNOS || ARCH_MULTIPLATFORM) select DRM_KMS_HELPER select VIDEOMODE_HELPERS + select SND_SOC_HDMI_CODEC if SND_SOC help Choose this option if you have a Samsung SoC EXYNOS chipset. If M is selected the module will be called exynosdrm. diff --git a/drivers/gpu/drm/exynos/exynos_hdmi.c b/drivers/gpu/drm/exynos/exynos_hdmi.c index 06bfbe4..64eadf58 100644 --- a/drivers/gpu/drm/exynos/exynos_hdmi.c +++ b/drivers/gpu/drm/exynos/exynos_hdmi.c @@ -40,7 +40,7 @@ #include <linux/component.h> #include <linux/mfd/syscon.h> #include <linux/regmap.h> - +#include <sound/hdmi-codec.h> #include <drm/exynos_drm.h> #include <media/cec-notifier.h> @@ -111,12 +111,19 @@ struct hdmi_driver_data { struct string_array_spec clk_muxes; }; +struct hdmi_audio { + struct platform_device *pdev; + struct hdmi_audio_infoframe infoframe; + unsigned int sample_rate; + unsigned int sample_width; + u8 enable; +}; + struct hdmi_context { struct drm_encoder encoder; struct device *dev; struct drm_device *drm_dev; struct drm_connector connector; - bool powered; bool dvi_mode; struct delayed_work hotplug_work; struct drm_display_mode current_mode; @@ -137,6 +144,11 @@ struct hdmi_context { struct regulator *reg_hdmi_en; struct exynos_drm_clk phy_clk; struct drm_bridge *bridge; + + /* mutex protecting subsequent fields below */ + struct mutex mutex; + struct hdmi_audio audio; + bool powered; }; static inline struct hdmi_context *encoder_to_hdmi(struct drm_encoder *e) @@ -768,6 +780,22 @@ static int hdmi_clk_set_parents(struct hdmi_context *hdata, bool to_phy) return ret; } +static int hdmi_audio_infoframe_apply(struct hdmi_context *hdata) +{ + struct hdmi_audio_infoframe *infoframe = &hdata->audio.infoframe; + u8 buf[HDMI_INFOFRAME_SIZE(AUDIO)]; + int len; + + len = hdmi_audio_infoframe_pack(infoframe, buf, sizeof(buf)); + if (len < 0) + return len; + + hdmi_reg_writeb(hdata, HDMI_AUI_CON, HDMI_AUI_CON_EVERY_VSYNC); + hdmi_reg_write_buf(hdata, HDMI_AUI_HEADER0, buf, len); + + return 0; +} + static void hdmi_reg_infoframes(struct hdmi_context *hdata) { union hdmi_infoframe frm; @@ -805,15 +833,7 @@ static void hdmi_reg_infoframes(struct hdmi_context *hdata) hdmi_reg_write_buf(hdata, HDMI_VSI_DATA(0), buf + 3, ret - 3); } - ret = hdmi_audio_infoframe_init(&frm.audio); - if (!ret) { - frm.audio.channels = 2; - ret = hdmi_audio_infoframe_pack(&frm.audio, buf, sizeof(buf)); - } - if (ret > 0) { - hdmi_reg_writeb(hdata, HDMI_AUI_CON, HDMI_AUI_CON_EVERY_VSYNC); - hdmi_reg_write_buf(hdata, HDMI_AUI_HEADER0, buf, ret); - } + hdmi_audio_infoframe_apply(hdata); } static enum drm_connector_status hdmi_detect(struct drm_connector *connector, @@ -996,23 +1016,18 @@ static void hdmi_reg_acr(struct hdmi_context *hdata, u32 freq) hdmi_reg_writeb(hdata, HDMI_ACR_CON, 4); } -static void hdmi_audio_init(struct hdmi_context *hdata) +static void hdmi_audio_config(struct hdmi_context *hdata) { - u32 sample_rate, bits_per_sample; - u32 data_num, bit_ch, sample_frq; - u32 val; + u32 data_num, sample_freq, val; + u32 bit_ch = 1; - sample_rate = 44100; - bits_per_sample = 16; - switch (bits_per_sample) { + switch (hdata->audio.sample_width) { case 20: data_num = 2; - bit_ch = 1; break; case 24: data_num = 3; - bit_ch = 1; break; default: data_num = 1; @@ -1020,7 +1035,7 @@ static void hdmi_audio_init(struct hdmi_context *hdata) break; } - hdmi_reg_acr(hdata, sample_rate); + hdmi_reg_acr(hdata, hdata->audio.sample_rate); hdmi_reg_writeb(hdata, HDMI_I2S_MUX_CON, HDMI_I2S_IN_DISABLE | HDMI_I2S_AUD_I2S | HDMI_I2S_CUV_I2S_ENABLE @@ -1031,10 +1046,21 @@ static void hdmi_audio_init(struct hdmi_context *hdata) hdmi_reg_writeb(hdata, HDMI_I2S_MUX_CUV, HDMI_I2S_CUV_RL_EN); - sample_frq = (sample_rate == 44100) ? 0 : - (sample_rate == 48000) ? 2 : - (sample_rate == 32000) ? 3 : - (sample_rate == 96000) ? 0xa : 0x0; + switch(hdata->audio.sample_rate) { + case 32000: + sample_freq = 0x3; + break; + case 48000: + sample_freq = 0x2; + break; + case 96000: + sample_freq = 0xa; + break; + case 44100: + default: + sample_freq = 0; + break; + } hdmi_reg_writeb(hdata, HDMI_I2S_CLK_CON, HDMI_I2S_CLK_DIS); hdmi_reg_writeb(hdata, HDMI_I2S_CLK_CON, HDMI_I2S_CLK_EN); @@ -1068,7 +1094,7 @@ static void hdmi_audio_init(struct hdmi_context *hdata) hdmi_reg_writeb(hdata, HDMI_I2S_CH_ST_1, HDMI_I2S_CD_PLAYER); hdmi_reg_writeb(hdata, HDMI_I2S_CH_ST_2, HDMI_I2S_SET_SOURCE_NUM(0)); hdmi_reg_writeb(hdata, HDMI_I2S_CH_ST_3, HDMI_I2S_CLK_ACCUR_LEVEL_2 - | HDMI_I2S_SET_SMP_FREQ(sample_frq)); + | HDMI_I2S_SET_SMP_FREQ(sample_freq)); hdmi_reg_writeb(hdata, HDMI_I2S_CH_ST_4, HDMI_I2S_ORG_SMP_FREQ_44_1 | HDMI_I2S_WORD_LEN_MAX24_24BITS @@ -1077,13 +1103,15 @@ static void hdmi_audio_init(struct hdmi_context *hdata) hdmi_reg_writeb(hdata, HDMI_I2S_CH_ST_CON, HDMI_I2S_CH_STATUS_RELOAD); } -static void hdmi_audio_control(struct hdmi_context *hdata, bool onoff) +static void hdmi_audio_control(struct hdmi_context *hdata) { + bool enable = hdata->audio.enable; + if (hdata->dvi_mode) return; - hdmi_reg_writeb(hdata, HDMI_AUI_CON, onoff ? 2 : 0); - hdmi_reg_writemask(hdata, HDMI_CON_0, onoff ? + hdmi_reg_writeb(hdata, HDMI_AUI_CON, enable ? 2 : 0); + hdmi_reg_writemask(hdata, HDMI_CON_0, enable ? HDMI_ASP_EN : HDMI_ASP_DIS, HDMI_ASP_MASK); } @@ -1403,9 +1431,9 @@ static void hdmi_conf_apply(struct hdmi_context *hdata) { hdmi_start(hdata, false); hdmi_conf_init(hdata); - hdmi_audio_init(hdata); + hdmi_audio_config(hdata); hdmi_mode_apply(hdata); - hdmi_audio_control(hdata, true); + hdmi_audio_control(hdata); } static void hdmi_mode_set(struct drm_encoder *encoder, @@ -1479,17 +1507,24 @@ static void hdmi_enable(struct drm_encoder *encoder) { struct hdmi_context *hdata = encoder_to_hdmi(encoder); + mutex_lock(&hdata->mutex); + hdmiphy_enable(hdata); hdmi_conf_apply(hdata); + + mutex_unlock(&hdata->mutex); } static void hdmi_disable(struct drm_encoder *encoder) { struct hdmi_context *hdata = encoder_to_hdmi(encoder); - if (!hdata->powered) - return; + mutex_lock(&hdata->mutex); + if (!hdata->powered) { + mutex_unlock(&hdata->mutex); + return; + } /* * The SFRs of VP and Mixer are updated by Vertical Sync of * Timing generator which is a part of HDMI so the sequence @@ -1503,6 +1538,8 @@ static void hdmi_disable(struct drm_encoder *encoder) cec_notifier_set_phys_addr(hdata->notifier, CEC_PHYS_ADDR_INVALID); hdmiphy_disable(hdata); + + mutex_unlock(&hdata->mutex); } static const struct drm_encoder_helper_funcs exynos_hdmi_encoder_helper_funcs = { @@ -1516,6 +1553,109 @@ static void hdmi_disable(struct drm_encoder *encoder) .destroy = drm_encoder_cleanup, }; +static void hdmi_audio_shutdown(struct device *dev, void *data) +{ + struct hdmi_context *hdata = dev_get_drvdata(dev); + + mutex_lock(&hdata->mutex); + + hdata->audio.enable = false; + + if (hdata->powered) + hdmi_audio_control(hdata); + + mutex_unlock(&hdata->mutex); +} + +static int hdmi_audio_hw_params(struct device *dev, void *data, + struct hdmi_codec_daifmt *daifmt, + struct hdmi_codec_params *params) +{ + struct hdmi_context *hdata = dev_get_drvdata(dev); + + if (daifmt->fmt != HDMI_I2S || daifmt->bit_clk_inv || + daifmt->frame_clk_inv || daifmt->bit_clk_master || + daifmt->frame_clk_master) { + dev_err(dev, "%s: Bad flags %d %d %d %d\n", __func__, + daifmt->bit_clk_inv, daifmt->frame_clk_inv, + daifmt->bit_clk_master, + daifmt->frame_clk_master); + return -EINVAL; + } + + mutex_lock(&hdata->mutex); + + hdata->audio.sample_rate = params->sample_rate; + hdata->audio.sample_width = params->sample_width; + hdata->audio.infoframe = params->cea; + + if (hdata->powered) { + hdmi_audio_config(hdata); + hdmi_audio_infoframe_apply(hdata); + } + + mutex_unlock(&hdata->mutex); + + return 0; +} + +static int hdmi_audio_digital_mute(struct device *dev, void *data, bool mute) +{ + struct hdmi_context *hdata = dev_get_drvdata(dev); + + mutex_lock(&hdata->mutex); + + hdata->audio.enable = !mute; + + if (hdata->powered) + hdmi_audio_control(hdata); + + mutex_unlock(&hdata->mutex); + + return 0; +} + +static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, + size_t len) +{ + struct hdmi_context *hdata = dev_get_drvdata(dev); + struct drm_connector *connector = &hdata->connector; + + memcpy(buf, connector->eld, min(sizeof(connector->eld), len)); + + return 0; +} + +static const struct hdmi_codec_ops audio_codec_ops = { + .hw_params = hdmi_audio_hw_params, + .audio_shutdown = hdmi_audio_shutdown, + .digital_mute = hdmi_audio_digital_mute, + .get_eld = hdmi_audio_get_eld, +}; + +static int hdmi_register_audio_device(struct hdmi_context *hdata) +{ + struct hdmi_codec_pdata codec_data = { + .ops = &audio_codec_ops, + .max_i2s_channels = 6, + .i2s = 1, + }; + + hdata->audio.pdev = platform_device_register_data( + hdata->dev, HDMI_CODEC_DRV_NAME, PLATFORM_DEVID_AUTO, + &codec_data, sizeof(codec_data)); + + if (IS_ERR(hdata->audio.pdev)) + return PTR_ERR(hdata->audio.pdev); + + return 0; +} + +static void hdmi_unregister_audio_device(struct hdmi_context *hdata) +{ + platform_device_unregister(hdata->audio.pdev); +} + static void hdmi_hotplug_work_func(struct work_struct *work) { struct hdmi_context *hdata; @@ -1700,6 +1840,7 @@ static int hdmi_bind(struct device *dev, struct device *master, void *data) struct drm_device *drm_dev = data; struct hdmi_context *hdata = dev_get_drvdata(dev); struct drm_encoder *encoder = &hdata->encoder; + struct hdmi_audio_infoframe *audio_infoframe = &hdata->audio.infoframe; struct exynos_drm_crtc *exynos_crtc; struct drm_crtc *crtc; int ret, pipe; @@ -1721,6 +1862,12 @@ static int hdmi_bind(struct device *dev, struct device *master, void *data) DRM_DEBUG_KMS("possible_crtcs = 0x%x\n", encoder->possible_crtcs); + hdmi_audio_infoframe_init(audio_infoframe); + audio_infoframe->coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; + audio_infoframe->sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; + audio_infoframe->sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; + audio_infoframe->channels = 2; + drm_encoder_init(drm_dev, encoder, &exynos_hdmi_encoder_funcs, DRM_MODE_ENCODER_TMDS, NULL); @@ -1828,6 +1975,8 @@ static int hdmi_probe(struct platform_device *pdev) hdata->dev = dev; + mutex_init(&hdata->mutex); + ret = hdmi_resources_init(hdata); if (ret) { if (ret != -EPROBE_DEFER) @@ -1887,12 +2036,19 @@ static int hdmi_probe(struct platform_device *pdev) pm_runtime_enable(dev); - ret = component_add(&pdev->dev, &hdmi_component_ops); + ret = hdmi_register_audio_device(hdata); if (ret) goto err_notifier_put; + ret = component_add(&pdev->dev, &hdmi_component_ops); + if (ret) + goto err_unregister_audio; + return ret; +err_unregister_audio: + hdmi_unregister_audio_device(hdata); + err_notifier_put: cec_notifier_put(hdata->notifier); pm_runtime_disable(dev); @@ -1931,6 +2087,8 @@ static int hdmi_remove(struct platform_device *pdev) put_device(&hdata->ddc_adpt->dev); + mutex_destroy(&hdata->mutex); + return 0; } -- 1.9.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8523: Fix array size for bclk_ratios
by Axel Lin 02 Aug '17

02 Aug '17

ARRAY_SIZE(bclk_ratios) returns 7 for current code, then it cannot catch the error if "no matching BCLK/fs ratio". Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8523.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 6d0a272..b1e4ce7 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -116,7 +116,7 @@ static struct { static struct { int value; int ratio; -} bclk_ratios[WM8523_NUM_RATES] = { +} bclk_ratios[] = { { 2, 32 }, { 3, 64 }, { 4, 128 }, -- 2.9.3

2 3

[alsa-devel] [PATCH] ASoC: wm8524: Don't use dev_err to show supported sample rate
by Axel Lin 02 Aug '17

02 Aug '17

Switch to use dev_dbg instead. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8524.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index e22b76d..b66147e 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -118,7 +118,7 @@ static int wm8524_set_dai_sysclk(struct snd_soc_dai *codec_dai, case 96000: case 176400: case 192000: - dev_err(codec->dev, "Supported sample rate: %dHz\n", + dev_dbg(codec->dev, "Supported sample rate: %dHz\n", val); wm8524->rate_constraint_list[j++] = val; wm8524->rate_constraint.count++; -- 2.9.3

3 2

[alsa-devel] [PATCH] ASoC: rt5663: Seprate the DC offset between headphone and headset
by Oder Chiou 02 Aug '17

02 Aug '17

The patch seprates the DC offset between headphone and headset. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5663.txt | 5 ++- include/sound/rt5663.h | 2 + sound/soc/codecs/rt5663.c | 52 ++++++++++++++++------ 3 files changed, 44 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index f7c7cc5..ff38171 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -14,8 +14,11 @@ Optional properties: - "realtek,dc_offset_l_manual" - "realtek,dc_offset_r_manual" +- "realtek,dc_offset_l_manual_mic" +- "realtek,dc_offset_r_manual_mic" Based on the different PCB layout, add the manual offset value to - compensate the DC offset for each L and R channel. + compensate the DC offset for each L and R channel, and they are different + between headphone and headset. Pins on the device (for linking into audio routes) for RT5663: diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h index be6f1b2..7d00e58 100644 --- a/include/sound/rt5663.h +++ b/include/sound/rt5663.h @@ -14,6 +14,8 @@ struct rt5663_platform_data { unsigned int dc_offset_l_manual; unsigned int dc_offset_r_manual; + unsigned int dc_offset_l_manual_mic; + unsigned int dc_offset_r_manual_mic; }; #endif diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 7bbfb18..7f05682 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1570,9 +1570,43 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) case 2: rt5663->jack_type = SND_JACK_HEADSET; rt5663_enable_push_button_irq(codec, true); + + if (rt5663->pdata.dc_offset_l_manual_mic) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, + rt5663->pdata.dc_offset_l_manual_mic >> + 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3, + rt5663->pdata.dc_offset_l_manual_mic & + 0xffff); + } + + if (rt5663->pdata.dc_offset_r_manual_mic) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5, + rt5663->pdata.dc_offset_r_manual_mic >> + 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6, + rt5663->pdata.dc_offset_r_manual_mic & + 0xffff); + } break; default: rt5663->jack_type = SND_JACK_HEADPHONE; + + if (rt5663->pdata.dc_offset_l_manual) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, + rt5663->pdata.dc_offset_l_manual >> 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3, + rt5663->pdata.dc_offset_l_manual & + 0xffff); + } + + if (rt5663->pdata.dc_offset_r_manual) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5, + rt5663->pdata.dc_offset_r_manual >> 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6, + rt5663->pdata.dc_offset_r_manual & + 0xffff); + } break; } } else { @@ -3133,6 +3167,10 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) &rt5663->pdata.dc_offset_l_manual); device_property_read_u32(dev, "realtek,dc_offset_r_manual", &rt5663->pdata.dc_offset_r_manual); + device_property_read_u32(dev, "realtek,dc_offset_l_manual_mic", + &rt5663->pdata.dc_offset_l_manual_mic); + device_property_read_u32(dev, "realtek,dc_offset_r_manual_mic", + &rt5663->pdata.dc_offset_r_manual_mic); return 0; } @@ -3221,20 +3259,6 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); } - if (rt5663->pdata.dc_offset_l_manual) { - regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, - rt5663->pdata.dc_offset_l_manual >> 16); - regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3, - rt5663->pdata.dc_offset_l_manual & 0xffff); - } - - if (rt5663->pdata.dc_offset_r_manual) { - regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5, - rt5663->pdata.dc_offset_r_manual >> 16); - regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6, - rt5663->pdata.dc_offset_r_manual & 0xffff); - } - /* GPIO1 as IRQ */ regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5663_GP1_PIN_MASK, RT5663_GP1_PIN_IRQ); -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: codec: add DT support to dmic codec
by Arnaud Pouliquen 02 Aug '17

02 Aug '17

These patches add of_table description in dmic codec to support DT probing. As this codec is a generic code to support simple digital microphone, compatibility field is generic. Arnaud Pouliquen (2): ASoC: Add bindings for DMIC codec driver ASoC: codec: add DT support in dmic codec Documentation/devicetree/bindings/sound/dmic.txt | 12 ++++++++++++ sound/soc/codecs/dmic.c | 6 ++++++ 2 files changed, 18 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/dmic.txt -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: rcar: unregister fixed rate on ADG
by Kuninori Morimoto 02 Aug '17

02 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ADG is registering fixed rate clock for audio_clkout, but it had not been unregister clock when removing. Salvator-X board is using ak4613 driver now, and it supports hw_constraints from commit 907cd8809eebc ("ASoC: ak4613: add hw_constraint rule for Sampling Rate"). And this calculation is using input clk. This ak4613 input clock is ADG clkout on Salvator-X. Because ADG had not been unregister clkout when unbinding, it receives fixed rate clk register error when re-binding. Thus, ak4613 can't get correct input clock, and hw_constraints will be failed after re-binding. This means Salvator-X board can't use sound after unbind/bind. This patch solves this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark This patch is for v4.13-rcX sound/soc/sh/rcar/adg.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 197cb3e..5b5389e 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -610,6 +610,13 @@ void rsnd_adg_remove(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; + struct rsnd_adg *adg = priv->adg; + struct clk *clk; + int i; + + for_each_rsnd_clkout(clk, adg, i) + if (adg->clkout[i]) + clk_unregister_fixed_rate(adg->clkout[i]); of_clk_del_provider(np); -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: codecs: msm8916-wcd-digital: Fix dmic support
by srinivas.kandagatla＠linaro.org 02 Aug '17

02 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset fixes dmic support in the codec driver, first and second patch fixes the mclk rate for the codec and the third patch adds missing selection of DMIC in CIC filter source path. Tested on DB410c with Audio Mezz board which has got dmic on it. Changes since v1: - check return values for set_sysclk and clk_set_rate() spoted by Mark. - Added more info in change log. - Fixed multicodec usecase. Srinivas Kandagatla (3): ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk ASoC: qcom: apq8016-sbc: set default mclk rate ASoC: codecs: msm8916-wcd-digital: select DMIC source in CIC filter path sound/soc/codecs/msm8916-wcd-digital.c | 35 ++++++++++++++++++++++++++++++++++ sound/soc/qcom/apq8016_sbc.c | 23 ++++++++++++++++++---- 2 files changed, 54 insertions(+), 4 deletions(-) -- 2.9.3

3 8

[alsa-devel] [PATCH v3 0/3] ASoC: codecs: msm8916-wcd-digital: Fix dmic support
by srinivas.kandagatla＠linaro.org 02 Aug '17

02 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset fixes dmic support in the codec driver, first and second patch fixes the mclk rate for the codec and the third patch adds missing selection of DMIC in CIC filter source path. Tested on DB410c with Audio Mezz board which has got dmic on it. Changes since v2: - added cic selection to dapm route Srinivas Kandagatla (3): ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk ASoC: qcom: apq8016-sbc: set default mclk rate ASoC: codecs: msm8916-wcd-digital: add CIC filter source selection path sound/soc/codecs/msm8916-wcd-digital.c | 35 ++++++++++++++++++++++++++++++++-- sound/soc/qcom/apq8016_sbc.c | 23 ++++++++++++++++++---- 2 files changed, 52 insertions(+), 6 deletions(-) -- 2.9.3

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[alsa-devel] [GIT PULL] ASoC fixes for v4.13
by Mark Brown 02 Aug '17

02 Aug '17

The following changes since commit 16f73eb02d7e1765ccab3d2018e0bd98eb93d973: Linux 4.13-rc3 (2017-07-30 12:40:36 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.13-rc3 for you to fetch changes up to 60668a2dcafcf8aad0860f5a5c93eb2d7438052e: Merge remote-tracking branches 'asoc/fix/rt5663', 'asoc/fix/rt5665', 'asoc/fix/samsung', 'asoc/fix/sgtl5000' and 'asoc/fix/sh' into asoc-linus (2017-08-01 15:17:11 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.13 Quite a few fixes here that have been sent since the merge window, the biggest one is the fix from Tony for some confusion with the device property API which was causing issues with the of-graph card. This is fixed with some changes in the graph API itself as it seemed very likely to be error prone. ---------------------------------------------------------------- Banajit Goswami (1): ASoC: do not close shared backend dailink Bard Liao (2): ASoC: rt5665: fix GPIO6 pin function define ASoC: rt5665: fix wrong register for bclk ratio control Damien Riegel (1): ASoC: codecs: msm8916-analog: fix DIG_CLK_CTL_RXD3_CLK_EN define Fabio Estevam (1): ASoC: sgtl5000: Use snd_soc_kcontrol_codec() Geert Uytterhoeven (1): ASoC: pxa: SND_PXA2XX_SOC should depend on HAS_DMA Gustavo A. R. Silva (1): ASoC: imx-ssi: add check on platform_get_irq return value Harsha Priya N (2): ASoC: Intel: Enabling ASRC for RT5663 codec on kabylake platform ASoC: Intel: Use MCLK instead of BLCK as the sysclock for RT5514 codec on kabylake platform Johan Hovold (2): ASoC: fix pcm-creation regression ASoC: ux500: Restore platform DAI assignments Kuninori Morimoto (1): ASoC: sh: hac: add missing "int ret" Mark Brown (3): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/dpcm', 'asoc/fix/imx', 'asoc/fix/msm8916', 'asoc/fix/multi-pcm', 'asoc/fix/of-graph' and 'asoc/fix/pxa' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5663', 'asoc/fix/rt5665', 'asoc/fix/samsung', 'asoc/fix/sgtl5000' and 'asoc/fix/sh' into asoc-linus Subhransu S. Prusty (1): ASoC: Intel: Skylake: Fix default dma_buffer_size Sylwester Nawrocki (1): ASoC: samsung: odroid: Fix EPLL frequency values Tony Lindgren (1): device property: Fix usecount for of_graph_get_port_parent() Vinod Koul (1): ASoC: Intel: Skylake: Fix missing sentinels in sst_acpi_mach oder_chiou(a)realtek.com (1): ASoC: rt5663: Update the HW default values based on the shipping version drivers/of/property.c | 17 ++++++++-- include/sound/soc.h | 6 ---- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/rt5663.c | 18 +++++------ sound/soc/codecs/rt5665.c | 4 +-- sound/soc/codecs/rt5665.h | 4 +-- sound/soc/codecs/sgtl5000.c | 4 +-- sound/soc/fsl/imx-ssi.c | 4 +++ sound/soc/generic/audio-graph-card.c | 10 +++--- sound/soc/generic/audio-graph-scu-card.c | 15 +++++---- sound/soc/generic/simple-card-utils.c | 8 ++--- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 17 +++------- sound/soc/intel/skylake/skl-messages.c | 8 +++++ sound/soc/intel/skylake/skl.c | 2 ++ sound/soc/pxa/Kconfig | 1 + sound/soc/samsung/odroid.c | 6 ++-- sound/soc/sh/hac.c | 2 ++ sound/soc/soc-core.c | 27 ++-------------- sound/soc/soc-pcm.c | 36 ++++++++-------------- sound/soc/ux500/mop500.c | 4 +++ 20 files changed, 92 insertions(+), 103 deletions(-)

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Re: [alsa-devel] [PATCH] ALSA: hda - Fix speaker output from VAIO VPCL14M1R
by Takashi Iwai 02 Aug '17

02 Aug '17

On Wed, 02 Aug 2017 12:23:48 +0200, Sergei A. Trusov wrote: > > Sony VAIO VPCL14M1R needs the quirk to make the speaker working properly. > > Tested-by: Dmitriy <mexx400(a)yandex.ru> > Cc: <stable(a)vger.kernel.org> > Signed-off-by: Sergei A. Trusov <sergei.a.trusov(a)ya.ru> Applied, thanks. Takashi

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[alsa-devel] [PATCH] streamline TLV320AIC23 drivers
by Jens Rottmann 02 Aug '17

02 Aug '17

The iMX-TLV320AIC23 driver isn't from Freescale, but from a company named Eukrea Electromatique, originally for their own boards. From the code I get the impression it is a bit older, its DT options use a differing naming scheme. Patch it up a bit: - Remove Eukrea naming, i.MX is from Freescale, TLV320AIC23 is from TI, driver was written by Eukrea, but it's DT capable, so it's not exclusive: - Kconfig option title - 'model' option - driver 'compatible' string - Other options just have changed over time, this driver remaining (one of) the last with the old semantics: - 'audio-codec' option (also moved from ssi node) - 'mux-int/ext-port' options - All options stay backwards compatible, the DT binding documents new and old names. CONFIG variable and files have not been renamed, though, so no need to change old defconfigs. Signed-off-by: Jens Rottmann <Jens.Rottmann(a)ADLINKtech.com> --- --- a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -1,16 +1,23 @@ -Audio complex for Eukrea boards with tlv320aic23 codec. +Audio complex for Freescale i.MX boards with TI TLV320AIC23 I2S codecs, +like those from Eukrea Electromatique. Required properties: - - compatible : "eukrea,asoc-tlv320" + - compatible : "fsl,imx-audio-tlv320aic23" or + "eukrea,asoc-tlv320" (deprecated) - - eukrea,model : The user-visible name of this sound complex. + - model : The user-visible name of this sound complex. + - eukrea,model : Dito, deprecated. - ssi-controller : The phandle of the SSI controller. - - fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). + - fsl,mux-int-port : Dito, deprecated. - - fsl,mux-ext-port : The external port of the i.MX audio muxer. + - mux-ext-port : The external port of the i.MX audio muxer. + - fsl,mux-ext-port : Dito, deprecated. + + - audio-codec : The phandle of the audio codec. Note: The AUDMUX port numbering should start at 1, which is consistent with hardware manual. @@ -18,9 +25,10 @@ hardware manual. Example: sound { - compatible = "eukrea,asoc-tlv320"; - eukrea,model = "imx51-eukrea-tlv320aic23"; + compatible = "fsl,imx-audio-tlv320aic23"; + model = "imx51-eukrea-tlv320aic23"; ssi-controller = <&ssi2>; - fsl,mux-int-port = <2>; - fsl,mux-ext-port = <3>; + mux-int-port = <2>; + mux-ext-port = <3>; + audio-codec = <&codec>; }; --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -220,7 +220,7 @@ config SND_SOC_PHYCORE_AC97 and phyCARD boards in AC97 mode config SND_SOC_EUKREA_TLV320 - tristate "Eukrea TLV320" + tristate "SoC Audio support for i.MX boards with TLV320AIC23" depends on ARCH_MXC && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -92,11 +92,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) eukrea_tlv320.dev = &pdev->dev; if (np) { - ret = snd_soc_of_parse_card_name(&eukrea_tlv320, - "eukrea,model"); + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, "model"); + if (ret) /* backwards compatible */ + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, + "eukrea,model"); if (ret) { dev_err(&pdev->dev, - "eukrea,model node missing or invalid.\n"); + "model node missing or invalid.\n"); goto err; } @@ -109,22 +111,28 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) goto err; } - codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!codec_np) /* backwards compatible */ + codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); if (codec_np) eukrea_tlv320_dai.codec_of_node = codec_np; else - dev_err(&pdev->dev, "codec-handle node missing or invalid.\n"); + dev_err(&pdev->dev, "audio-codec node missing or invalid.\n"); - ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) /* backwards compatible */ + ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); if (ret) { dev_err(&pdev->dev, - "fsl,mux-int-port node missing or invalid.\n"); + "mux-int-port node missing or invalid.\n"); return ret; } - ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) /* backwards compatible */ + ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, - "fsl,mux-ext-port node missing or invalid.\n"); + "mux-ext-port node missing or invalid.\n"); return ret; } @@ -213,7 +221,8 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) } static const struct of_device_id imx_tlv320_dt_ids[] = { - { .compatible = "eukrea,asoc-tlv320"}, + { .compatible = "eukrea,asoc-tlv320"}, /* backwards compatible */ + { .compatible = "fsl,imx-audio-tlv320aic23"}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); _

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[alsa-devel] [PATCH v2 0/2] ASoC: hdmi-codec: Allow driver to restrict sample size
by srinivas.kandagatla＠linaro.org 01 Aug '17

01 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patch adds option to allow hdmi-codec drivers to restrict the audio sample size based on the values that are suppored in hardware. First patch is adding that formats option to hdmi_codec_pdata and second patch is a fix in adv7511 codec driver which only supports 16 and 24 bit samples. Changes since v1: - Added Review by Jyri - Updated comment in code suggested by Jyri Srinivas Kandagatla (2): ASoC: hdmi-codec: Allow drivers to restrict sample sizes. drm/bridge: adv7511: restrict audio sample sizes drivers/gpu/drm/bridge/adv7511/adv7511_audio.c | 2 ++ include/sound/hdmi-codec.h | 1 + sound/soc/codecs/hdmi-codec.c | 8 +++++--- 3 files changed, 8 insertions(+), 3 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: codecs: fix wm8524 build error
by Arnd Bergmann 01 Aug '17

01 Aug '17

When CONFIG_OF is disabled, we run into a build error: sound/soc/codecs/wm8524.c:257:21: error: 'wm8524_of_match' undeclared here (not in a function); did you mean 'wm8524_dai'? This removes the unnecessary #ifdef around the match table. Fixes: 007b6a54c305 ("ASoC: codecs: add wm8524 codec driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm8524.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index f42ec664877c..e22b76d64e32 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -206,13 +206,11 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8524 = { }, }; -#ifdef CONFIG_OF static const struct of_device_id wm8524_of_match[] = { { .compatible = "wlf,wm8524" }, { /* sentinel*/ } }; MODULE_DEVICE_TABLE(of, wm8524_of_match); -#endif static int wm8524_codec_probe(struct platform_device *pdev) { -- 2.9.0

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[alsa-devel] [PATCH] ASoC: rt5665: fix wrong register for bclk ratio control
by Bard Liao 01 Aug '17

01 Aug '17

The register of setting back ratio should be RT5665_ADDA_CLK_2 instead of RT5665_ADDA_CLK_1. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index d0a9490..db2e5b9 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4411,12 +4411,12 @@ static int rt5665_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) switch (dai->id) { case RT5665_AIF2_1: case RT5665_AIF2_2: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS2_MASK, RT5665_I2S_BCLK_MS2_64); break; case RT5665_AIF3: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS3_MASK, RT5665_I2S_BCLK_MS3_64); break; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt5663: Add the delay time to correct the calibration
by Oder Chiou 01 Aug '17

01 Aug '17

The patch adds the delay time to correct the calibration. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9704eed..7bbfb18 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3058,6 +3058,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c); regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32); regmap_write(rt5663->regmap, RT5663_DACREF_LDO, 0x3b0b); + msleep(40); regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x0000); regmap_write(rt5663->regmap, RT5663_BYPASS_STO_DAC, 0x000c); regmap_write(rt5663->regmap, RT5663_HP_BIAS, 0xafaa); -- 2.7.4

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[alsa-devel] [PATCH 1/4] ASoC: Intel: Skylake: Reset the controller in probe
by Guneshwor Singh 01 Aug '17

01 Aug '17

Controller can be in reset state by default. Capability structure traversal requires the controller to be out of reset else it results in broken capability parsing. Hence make sure that controller is out of reset before parsing capabilities by doing a full reset. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917ee41cf..a56dfde3928c 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -702,6 +702,8 @@ static int skl_first_init(struct hdac_ext_bus *ebus) return -ENXIO; } + skl_init_chip(bus, true); + snd_hdac_bus_parse_capabilities(bus); if (skl_acquire_irq(ebus, 0) < 0) -- 2.13.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5514: Add the sanity checks of the buffer related address
by Oder Chiou 01 Aug '17

01 Aug '17

The patch add the sanity checks of the buffer related address to make sure the addresses are valid. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 950d1ff..6f513cc 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -173,9 +173,14 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data) rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | buf[3] << 24; + if (rt5514_dsp->buf_rp % 8) + rt5514_dsp->buf_rp = (rt5514_dsp->buf_rp / 8) * 8; + rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - schedule_delayed_work(&rt5514_dsp->copy_work, 0); + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); return IRQ_HANDLED; } -- 2.7.4

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[alsa-devel] [PATCH] device property: Fix usecount for of_graph_get_port_parent()
by Tony Lindgren 01 Aug '17

01 Aug '17

Fix inconsistent use of of_graph_get_port_parent() where asoc_simple_card_parse_graph_dai() does of_node_get() before calling it while other callers do not. We can fix this by not trashing the node passed to of_graph_get_port_parent(). Let's make sure the users have correct refcounts and remove related incorrect of_node_put() calls for of_for_each_phandle as that's done by of_phandle_iterator_next(). Let's fix both issues with a single patch to avoid kobject refcounts getting messed up more if two patches are merged separately. Otherwise strange issues can happen caused by memory corruption caused by too many kobject_del() calls such as: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:747 ... (___might_sleep) (__mutex_lock) (mutex_lock_nested) (kernfs_remove) (kobject_del) (kobject_put) (of_get_next_parent) (of_graph_get_port_parent) (asoc_simple_card_parse_graph_dai [snd_soc_simple_card_utils]) (asoc_graph_card_probe [snd_soc_audio_graph_card]) Fixes: 0ef472a973eb ("of_graph: add of_graph_get_port_parent()") Fixes: 2692c1c63c29 ("ASoC: add audio-graph-card support") Fixes: 1689333f8311 ("ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai()") Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- drivers/of/property.c | 9 +++++++++ sound/soc/generic/audio-graph-card.c | 5 +---- sound/soc/generic/simple-card-utils.c | 5 ----- 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/drivers/of/property.c b/drivers/of/property.c --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -708,6 +708,15 @@ struct device_node *of_graph_get_port_parent(struct device_node *node) { unsigned int depth; + if (!node) + return NULL; + + /* + * Preserve usecount for passed in node as of_get_next_parent() + * will do of_node_put() on it. + */ + of_node_get(node); + /* Walk 3 levels up only if there is 'ports' node. */ for (depth = 3; depth && node; depth--) { node = of_get_next_parent(node); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -224,7 +224,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - of_node_put(it.node); if (ret < 0) return ret; } @@ -239,10 +238,8 @@ static int asoc_graph_get_dais_count(struct device *dev) int count = 0; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) count++; - of_node_put(it.node); - } return count; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -282,11 +282,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, if (!dai_name) return 0; - /* - * of_graph_get_port_parent() will call - * of_node_put(). So, call of_node_get() here - */ - of_node_get(ep); node = of_graph_get_port_parent(ep); /* Get dai->name */ -- 2.13.2

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[alsa-devel] Question about soc_bind_dai_link()
by Kuninori Morimoto 01 Aug '17

01 Aug '17

Hi ALSA ML I noticed a little bit strange operation on soc_bind_dai_link(). To research platform, it is using list_for_each_entry(). After finding platform, it sets rtd->platform, but it doesn't use "break". This means, rtd->platform might be overwrote. Is this on purpose ? or bug ? static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { ... /* find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { platform_of_node = platform->dev->of_node; if (!platform_of_node && platform->dev->parent->of_node) platform_of_node = platform->dev->parent->of_node; if (dai_link->platform_of_node) { if (platform_of_node != dai_link->platform_of_node) continue; } else { if (strcmp(platform->component.name, platform_name)) continue; } => rtd->platform = platform; } ... } Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: max98090 - Expose HP out enable.
by Enric Balletbo i Serra 01 Aug '17

01 Aug '17

From: Hsin-Yu Chao <hychao(a)chromium.org> Expose the headphone output enables to user space. This will allow userspace to ensure that there is stable output before the headphones are turned on, avoiding a pop on insertion. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/codecs/max98090.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6682848..f38c051 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1074,6 +1074,15 @@ static SOC_ENUM_SINGLE_DECL(mixhprsel_mux_enum, static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); +/* HP output enables. */ +static const struct snd_kcontrol_new max98090_hpl_enable = + SOC_DAPM_SINGLE("Switch", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 1, 0); + +static const struct snd_kcontrol_new max98090_hpr_enable = + SOC_DAPM_SINGLE("Switch", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 1, 0); + static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), @@ -1218,10 +1227,10 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), - SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPLEN_SHIFT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPREN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("HP Left Out", SND_SOC_NOPM, 0, 0, + &max98090_hpl_enable), + SND_SOC_DAPM_SWITCH("HP Right Out", SND_SOC_NOPM, 0, 0, + &max98090_hpr_enable), SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, M98090_SPLEN_SHIFT, 0, NULL, 0), @@ -1405,8 +1414,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { * Disable this for lowest power if bypassing * the DAC with an analog signal */ - {"HP Left Out", NULL, "DACL"}, - {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + {"HP Left Out", "Switch", "DACL"}, + {"HP Left Out", "Switch", "MIXHPLSEL Mux"}, {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, @@ -1414,8 +1423,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { * Disable this for lowest power if bypassing * the DAC with an analog signal */ - {"HP Right Out", NULL, "DACR"}, - {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + {"HP Right Out", "Switch", "DACR"}, + {"HP Right Out", "Switch", "MIXHPRSEL Mux"}, {"SPK Left Out", NULL, "Left Speaker Mixer"}, {"SPK Right Out", NULL, "Right Speaker Mixer"}, -- 2.9.3

2 1

[alsa-devel] [PATCH] Set sample rate for Digidesign Mbox 2
by Viktor Radnai 01 Aug '17

01 Aug '17

Hi all, I'm re-submitting this as I didn't get a response on it for about two weeks. Please see the old thread called "Patch for Digidesign Mbox 2 (set sample rate)" for prior discussion on it. Let me know if you need anything else from me to get this applied. Cheers, Vik diff -ur linux-source-4.11-orig/sound/usb/quirks.c linux-source-4.11/sound/usb/quirks.c --- linux-source-4.11-orig/sound/usb/quirks.c 2017-06-17 05:47:27.000000000 +0100 +++ linux-source-4.11/sound/usb/quirks.c 2017-07-18 22:37:58.522858262 +0100 @@ -768,23 +768,32 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev) { - u8 srate[3]; - u8 temp[12]; - /* Choose 48000Hz permanently */ - srate[0] = 0x80; - srate[1] = 0xbb; - srate[2] = 0x00; + u8 srate[3] = { 0x80, 0xbb, 0x00 }; + u8 temp[3]; + int err; /* Send the magic! */ - snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), - 0x01, 0x22, 0x0100, 0x0085, &temp, 0x0003); - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003); - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003); - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003); + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0085, &temp, sizeof(temp))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #1 failed\n"); + + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0085, &srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #2 failed\n"); + + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0086, &srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #3 failed\n"); + + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0003, &srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #4 failed\n"); + return; } @@ -815,16 +824,16 @@ count = 0; bootresponse[0] = MBOX2_BOOT_LOADING; - while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)) { - msleep(500); /* 0.5 second delay */ + do { snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), /* Control magic - load onboard firmware */ - 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012); + UAC_GET_MEM, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x0001, 0x0000, &bootresponse, sizeof(bootresponse)); if (bootresponse[0] == MBOX2_BOOT_READY) break; + msleep(500); /* 0.5 second delay */ dev_dbg(&dev->dev, "device not ready, resending boot sequence...\n"); count++; - } + } while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)); if (bootresponse[0] != MBOX2_BOOT_READY) { dev_err(&dev->dev, "Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse[0]); @@ -847,7 +856,7 @@ mbox2_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 2: 24bit 48kHz"); + dev_info(&dev->dev, "Digidesign Mbox 2 configured"); return 0; /* Successful boot */ } @@ -1111,6 +1120,26 @@ subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static void set_format_mbox2_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + struct usb_device *dev = subs->dev; + u8 srate[3]; + int err; + + srate[0] = subs->cur_rate; + srate[1] = subs->cur_rate >> 8; + srate[2] = subs->cur_rate >> 16; + + /* Set the sample rate for endpoint IN 5 (0x0085) until we succeed */ + while ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0085, srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: set sample rate failed, retrying\n"); + + dev_info(&subs->dev->dev, "mbox2: set sample rate to %dHz\n", subs->cur_rate); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { @@ -1121,6 +1150,9 @@ case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ + set_format_mbox2_quirk(subs, fmt); + break; } } diff -ur linux-source-4.11-orig/sound/usb/quirks-table.h linux-source-4.11/sound/usb/quirks-table.h --- linux-source-4.11-orig/sound/usb/quirks-table.h 2017-06-17 05:47:27.000000000 +0100 +++ linux-source-4.11/sound/usb/quirks-table.h 2017-07-18 17:33:57.620424012 +0100 @@ -3020,12 +3020,13 @@ .attributes = 0x00, .endpoint = 0x03, .ep_attr = USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000, + .rate_min = 44100, .rate_max = 48000, - .nr_rates = 1, + .nr_rates = 2, .rate_table = (unsigned int[]) { - 48000 + 44100, 48000 } } }, @@ -3045,12 +3046,13 @@ .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x85, .ep_attr = USB_ENDPOINT_SYNC_SYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000, + .rate_min = 44100, .rate_max = 48000, - .nr_rates = 1, + .nr_rates = 2, .rate_table = (unsigned int[]) { - 48000 + 44100, 48000 } } },

1 0

[alsa-devel] [PATCH] davinci-evm: extend it to support DM81xx and derivatives
by Vitaly Wool 31 Jul '17

31 Jul '17

Having worked closely with TI DM8148/DM8168 EVMs I came across many derivatives of these boards which didn't use the standard codec but had something like wm897x instead. They all would fit davinci-evm driver nicely if it had support for audio CPU clock divisor and slightly more generic naming. So this patch adds support for reading CPU divisor from a DT and slightly changes names of helper/init functions to be more generic. It also adds support for DaVinci boards that use wm8978 codec. Signed-off-by: Vitaly Wool <vitaly.wool(a)konsulko.com> --- .../bindings/sound/davinci-evm-audio.txt | 6 +- sound/soc/davinci/davinci-evm.c | 66 ++++++++++++++++++++-- 2 files changed, 65 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt index 963e100514c2..e76966a06f22 100644 --- a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -1,7 +1,9 @@ -* Texas Instruments SoC audio setups with TLV320AIC3X Codec +* Texas Instruments SoC audio setups with TLV320AIC3X/WM89XX codecs Required properties: - compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx + "ti,dm8168-evm-audio" : for DM81xx with TLV320AIC3x + "ti,dm8168-evm-audio-wm8978" : for DM81xx with WM897x - ti,model : The user-visible name of this sound complex. - ti,audio-codec : The phandle of the TLV320AIC3x audio codec - ti,mcasp-controller : The phandle of the McASP controller @@ -14,6 +16,8 @@ Optional properties: - ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec. - clocks : Reference to the master clock - clock-names : The clock should be named "mclk" +- ti,cpu-clkdiv : CPU clock divisor (if codec and cpu are clocked at + different rates) - Either codec-clock-rate or the codec-clock reference has to be defined. If the both are defined the driver attempts to set referenced clock to the defined rate and takes the rate from the clock reference. diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7a369e0f2093..aa8e11583a84 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -26,7 +26,8 @@ struct snd_soc_card_drvdata_davinci { struct clk *mclk; - unsigned sysclk; + unsigned int sysclk; + unsigned int clkdiv; }; static int evm_startup(struct snd_pcm_substream *substream) @@ -61,8 +62,11 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_card *soc_card = rtd->card; int ret = 0; - unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) - snd_soc_card_get_drvdata(soc_card))->sysclk; + unsigned int sysclk = ((struct snd_soc_card_drvdata_davinci *) + snd_soc_card_get_drvdata(soc_card))->sysclk; + unsigned int clkdiv = ((struct snd_soc_card_drvdata_davinci *) + snd_soc_card_get_drvdata(soc_card))->clkdiv; + /* set the codec system clock */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); @@ -74,6 +78,13 @@ static int evm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set the CPU system clock divisor */ + if (clkdiv) { + ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, clkdiv); + if (ret < 0) + return ret; + } + return 0; } @@ -84,7 +95,7 @@ static struct snd_soc_ops evm_ops = { }; /* davinci-evm machine dapm widgets */ -static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { +static const struct snd_soc_dapm_widget evm_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), @@ -121,8 +132,8 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) int ret; /* Add davinci-evm specific widgets */ - snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); + snd_soc_dapm_new_controls(&card->dapm, evm_dapm_widgets, + ARRAY_SIZE(evm_dapm_widgets)); if (np) { ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); @@ -142,6 +153,28 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) return 0; } +/* Logic for a non-aic3x (e. g. wm89xx) connected to a davinci-evm */ +static int evm_simple_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct device_node *np = card->dev->of_node; + + /* Add davinci-evm specific widgets */ + snd_soc_dapm_new_controls(&card->dapm, evm_dapm_widgets, + ARRAY_SIZE(evm_dapm_widgets)); + + /* audio routing is optional here */ + if (np) + snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + + /* not connected */ + snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM"); + + return 0; +} + /* davinci-evm digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link dm6446_evm_dai = { .name = "TLV320AIC3X", @@ -340,11 +373,29 @@ static struct snd_soc_dai_link evm_dai_tlv320aic3x = { SND_SOC_DAIFMT_IB_NF, }; +static struct snd_soc_dai_link evm_dai_wm8978 = { + .name = "wm8978", + .stream_name = "wm8978", + .codec_dai_name = "wm8978-hifi", + .ops = &evm_ops, + .init = evm_simple_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, +}; + static const struct of_device_id davinci_evm_dt_ids[] = { { .compatible = "ti,da830-evm-audio", .data = (void *) &evm_dai_tlv320aic3x, }, + { + .compatible = "ti,dm8168-evm-audio", + .data = (void *) &evm_dai_tlv320aic3x, + }, + { + .compatible = "ti,dm8168-evm-audio-wm8978", + .data = (void *) &evm_dai_wm8978, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, davinci_evm_dt_ids); @@ -422,6 +473,9 @@ static int davinci_evm_probe(struct platform_device *pdev) requestd_rate, drvdata->sysclk); } + /* CPU clock divisor may need to be provided for e. g. wm897x */ + of_property_read_u32(np, "ti,cpu-clkdiv", &drvdata->clkdiv); + snd_soc_card_set_drvdata(&evm_soc_card, drvdata); ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card); -- 2.11.0

3 6

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: fix balance of of_node_get/put()
by Kuninori Morimoto 31 Jul '17

31 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> audio-graph-scu-card with CONFIG_OF_DYNAMIC triggers several errors at kernel boot, like the following one OF: ERROR: Bad of_node_put() on /soc/sound@ec500000/ports/port@1 This patch doesn't call unnecessary of_node_put() which was called in other function (= of_graph_get_port_parent(), of_phandle_iterator_next()) Reported-by: Antonio Borneo <borneo.antonio(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index dcd2df3..bff1e6a 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -215,7 +215,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_ep = of_graph_get_remote_endpoint(cpu_ep); rcpu_ep = of_graph_get_remote_endpoint(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(rcpu_ep); @@ -248,11 +247,9 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); + codec_port = of_graph_get_port_parent(codec_ep); /* call put */ - of_node_put(cpu_port); of_node_put(cpu_ep); - of_node_put(codec_ep); of_node_put(codec_port); if (codec) { @@ -304,11 +301,9 @@ static int asoc_graph_get_dais_count(struct device *dev) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); + codec_port = of_graph_get_port_parent(codec_ep); /* call put */ - of_node_put(cpu_port); of_node_put(cpu_ep); - of_node_put(codec_ep); of_node_put(codec_port); if (cpu_ep) -- 1.9.1

1 1

[alsa-devel] snd_ctl_add_enumerated_elem_set and TLV
by Fulup Ar Foll 31 Jul '17

31 Jul '17

Hi, I'm using snd_ctl_add_enumerated_elem_set to create user controls, I do not understand why it set "TLV readable" option at true by default. I have a simple enumeration without TLV. I use snd_ctl_elem_write but do not call snd_ctl_elem_tlv_write. I do no understand how to prevent snd_ctl_elem_info_is_tlv_readable to return true for this control, obviously even if TLV is set as readable, it is empty and reading fail. Example fulup@fulup-desktop:~/Workspace/AlsaPulse> amixer -D hw:USB cget numid=237 numid=237,iface=MIXER,name='Hal-VolRamp-Mode' ; type=ENUMERATED,access=rw---RW-,values=1,items=4 ; Item #0 'None' ; Item #1 'Normal' ; Item #2 'Smooth' ; Item #3 'Emergency' : values=1 amixer: Control hw:USB element TLV read error: No such device or address Question: is there something special to set when a control does not have a TLV ? Fulup PS: if I write a fake TLV everything goes fine, but this is not what I expect for ENUM.

2 2

[alsa-devel] [PATCH] ASoC: fix balance of of_node_get()/of_node_put()
by Antonio Borneo 30 Jul '17

30 Jul '17

On Hikey target board, enabling CONFIG_OF_DYNAMIC triggers several errors at kernel boot, like the following one OF: ERROR: Bad of_node_put() on /soc/i2s@f7118000/ports/port@0/endpoint each followed by stack dump. Fixed by: - removing of_node_put() in the body of of_for_each_phandle(){}, since already provided at each iteration. Add it in case the loop is break out; - adding of_node_get() before calling of_graph_get_port_parent() or asoc_graph_card_dai_link_of(). Tested with kernel v4.13-rc1 with hikey_defconfig taken from https://git.linaro.org/people/john.stultz/android-dev.git branch dev/hikey-mainline-WIP Signed-off-by: Antonio Borneo <borneo.antonio(a)gmail.com> --- To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> To: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: Wei Xu <xuwei5(a)hisilicon.com> Cc: John Stultz <john.stultz(a)linaro.org> Cc: linux-arm-kernel(a)lists.infradead.org --- sound/soc/generic/audio-graph-card.c | 14 +++++++++----- sound/soc/generic/simple-card-utils.c | 5 +++++ sound/soc/soc-core.c | 5 +++++ 3 files changed, 19 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c0f08a6..7574c5f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -228,10 +228,16 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) */ of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + /* + * asoc_graph_card_dai_link_of() will call + * of_node_put(). So, call of_node_get() here + */ + of_node_get(it.node); ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - of_node_put(it.node); - if (ret < 0) + if (ret < 0) { + of_node_put(it.node); return ret; + } } return asoc_simple_card_parse_card_name(card, NULL); @@ -244,10 +250,8 @@ static int asoc_graph_get_dais_count(struct device *dev) int count = 0; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) count++; - of_node_put(it.node); - } return count; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 26d64fa..144954b 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -250,6 +250,11 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) if (ret != -ENOTSUPP) return ret; + /* + * of_graph_get_port_parent() will call + * of_node_put(). So, call of_node_get() here + */ + of_node_get(ep); node = of_graph_get_port_parent(ep); /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a..a0f39de 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4087,6 +4087,11 @@ int snd_soc_get_dai_id(struct device_node *ep) struct device_node *node; int ret; + /* + * of_graph_get_port_parent() will call + * of_node_put(). So, call of_node_get() here + */ + of_node_get(ep); node = of_graph_get_port_parent(ep); /* -- 1.9.1

3 9

[alsa-devel] Fwd: Fwd: Re: Basic questions
by tsepp2001＠t-online.de 30 Jul '17

30 Jul '17

Hi Clemens, thank you. In blocking-mode it works. Then i will follow the non-blocking mode. I modified first the callback: void playback_callback (snd_pcm_sframes_t nframes) { snd_pcm_sframes_t avail; avail = snd_pcm_avail_update(handle); while (avail >= period_size) { snd_pcm_writei(handle, buffer_alsa , nframes); avail = snd_pcm_avail_update(handle); std::cout << " Avail: " << avail << std::endl; } } ... called it at the end: playback_callback(period_size); and I get (for "avail"): Frames: 6894888 buffer_size (frames): 2048 buffer_time (frames): 46439 period_size (frames): 256 period_time (usec): 5804 periods (count): 8 Avail: 1282 Avail: 1027 Avail: 772 Avail: 517 Avail: 261 Avail: 6 the same mistake... Regards Thomas tsepp2001(a)t-online.de wrote: > Am I right to use async mode No. It is not supported by all devices, and almost impossible to use correctly. > to get lowest latency times Latency depends on the buffer size, and how fast your code can react to the notification that some space is available. > and using pause/play/stop? You probably want to use non-blocking mode. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] Gina 3g and Echomixer issue
by Britt Yazel 30 Jul '17

30 Jul '17

Hey guys, I work in an auditory neuroscience research lab, and we are using a PCI Gina 3g to communicate with a Fireface 800 over ADAT. My issue is this, 1) each time I reboot the system the digital mode is set to Coaxial digital, rather than ADAT, and I must set it after each reboot. Is there a way to make this setting persist through reboot, or perhaps a parameter that can be set when the kernel module is loaded to force ADAT? 2) The first time I switch from Coaxial digital to ADAT upon a reboot, I consistently get a loud squeeling sound coming from my studio monitors. I don't think this is the intended behaviuor, and it makes me worried when switching into ADAT mode. Granted, after switching the first time to ADAT, I can switch between coaxial digital and ADAT without the squeeling happeneing, it is just that first time after the reboot. Thanks for the help guys. The Gina 3g, even though it is an old audio device, is still our most reliable sound card. We have tried many others with horrible success, and most other ADAT capable PCI/PCIe sound cards are far too expensive, and don't have robust Linux support. However, if you guys have a good reccomendation of an ADAT capable PCIe sound card with full Linux support, please let me know. -Britt

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[alsa-devel] Basic questions
by tsepp2001＠t-online.de 30 Jul '17

30 Jul '17

Hi everybody, I am new to ALSA-API and would be glad, if someone give me an advice to the following questions. Up to now I used Rtaudio for cross compiling the player for windows and Linux. But know I need to go more into ALSA... - Am I right to use async mode to get lowest latency times and using pause/play/stop? - I used the following piece of code. But don't get any output. The callback output shows "avail" with very short count of frames... I write a whole file (raw) to buffer, called buffer_alsa with 6894888 frames. What is going wrong? I get this output: Frames: 6894888 buffer_size (frames): 2048 buffer_time (frames): 46439 period_size (frames): 256 period_time (usec): 5804 periods (count): 8 Avail: 1540 Avail: 1285 Avail: 1030 Avail: 775 Avail: 520 Avail: 265 Avail: 10 Avail: 2 Avail: 2 Avail: 3 Avail: 2 ... The code follows ... Best regards Thomas ----------------------------- void MyCallback(snd_async_handler_t *pcm_callback) { snd_pcm_t *handle = snd_async_handler_get_pcm(pcm_callback); snd_pcm_sframes_t avail; //std::cout << "Period_size: " << period_size << std::endl; avail = snd_pcm_avail_update(handle); while (avail >= period_size) { snd_pcm_writei(handle, buffer_alsa , period_size); avail = snd_pcm_avail_update(handle); std::cout << " Avail: " << avail << std::endl; } } //---------------------------------------------------- snd_pcm_hw_params_alloca(&params); if ((err = snd_pcm_hw_params_any(handle, params)) < 0) std::cout << " Error (reading configuration): " << snd_strerror(err) << std::endl; /* Set parameters */ if ((err = snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) //printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(err)); std::cout << " Error (can't set interleaved mode): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_format(handle, params, FORMAT)) < 0) //printf("ERROR: Can't set format. %s\n", snd_strerror(err)); std::cout << " Error (can't set format): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_channels(handle, params, data.channels)) < 0) //printf("ERROR: Can't set channels number. %s\n", snd_strerror(err)); std::cout << " Error (can't set channels number): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_rate_near(handle, params, &fs, 0)) < 0) //printf("ERROR: Can't set rate. %s\n", snd_strerror(err)); std::cout << " Error (can't set rate): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size)) < 0) std::cout << " Error (can't set buffer size): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_period_size_near (handle, params, &period_size, NULL)) < 0) std::cout << " Error (can't set period size): " << snd_strerror(err) << std::endl; /* Write parameters */ if ((err = snd_pcm_hw_params(handle, params)) < 0) //printf("ERROR: Can't set hardware parameters. %s\n", snd_strerror(err)); std::cout << " Error (can't set harware params): " << snd_strerror(err) << std::endl; //---------------------- ------------------------------ // Get Parameters snd_pcm_hw_params_get_buffer_size(params, &buffer_sz); std::cout << " buffer_size (frames): " << buffer_sz << std::endl; snd_pcm_hw_params_get_buffer_time(params, &buffer_tm, NULL); std::cout << " buffer_time (frames): " << buffer_tm << std::endl; snd_pcm_hw_params_get_period_size(params, &aframes, 0); std::cout << " period_size (frames): " << aframes << std::endl; snd_pcm_hw_params_get_period_time(params, &data.channels, NULL); // tmp Zeit in us std::cout << " period_time (usec): " << data.channels << std::endl; snd_pcm_hw_params_get_periods(params, &piods, 0); std::cout << " periods (count): " << piods << std::endl; // ---------------------------------------------------------------------------- //snd_pcm_hw_params_free (params); // --------------------- Hardware-Params End ------------------------------ snd_pcm_sw_params_t *sw_params; if ((err = snd_pcm_sw_params_malloc (&sw_params)) < 0) { std::cout << " Error (can't allocate sw params): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_sw_params_current(handle, sw_params)) < 0) { std::cout << "Error (can't return curr sw config): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, buffer_size)) < 0) // 0U oder 4096 ? { // This parameter controls the wake up point. If the count of available samples is equal or greater than this value, then the application will be activated. std::cout << "Error (can't set start thresh mode): " << snd_strerror(err) << std::endl; return; } /* if ((err = snd_pcm_sw_params_set_silence_threshold( handle, sw_params, 0 )) < 0) { std::cout << " Error (can't set silence thres): " << snd_strerror(err) << std::endl; return; } */ if ((err = snd_pcm_sw_params_set_avail_min (handle, sw_params, period_size)) < 0) { std::cout << " Error (can't set min avail count): " << snd_strerror(err) << std::endl; return; } // here are two options for a fix //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); /* snd_pcm_uframes_t val; if ((err = snd_pcm_sw_params_get_boundary(sw_params, &val)) < 0) { std::cout << " Error (can't get boundary): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_sw_params_set_silence_size(handle, sw_params, val)) < 0) { std::cout << " Error (can't set silence): " << snd_strerror(err) << std::endl; return; } */ if ((err = snd_pcm_sw_params(handle, sw_params)) < 0) { std::cout << " Error (can't set sw params): " << snd_strerror(err) << std::endl; return; } snd_pcm_sw_params_free(sw_params); // ------------------------- SW-Params End --------------------------------- if ((err = snd_pcm_prepare(handle)) < 0) { std::cout << " Error (can't prepare audio): " << snd_strerror(err) << std::endl; return; } snd_async_handler_t *pcm_callback; if ((err = snd_async_add_pcm_handler(&pcm_callback, handle, MyCallback, NULL)) < 0) { std::cout << " Error (can't add pcm handler): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_writei(handle, buffer_alsa, 2 * period_size) == -EPIPE)) { std::cout << " Error: XRUN" << std::endl; err = snd_pcm_recover(handle, err, 0); if ((err = snd_pcm_prepare (handle)) < 0) { std::cout << " Error (can't prepare audio): " << snd_strerror(err) << std::endl; return; } } if ((err = snd_pcm_start(handle)) < 0) { std::cout << " Error (can't start audio): " << snd_strerror(err) << std::endl; return; } void MyCallback(snd_async_handler_t *pcm_callback);

2 1

[alsa-devel] [PATCH] ASoC: soc-pcm: Remove unused 'debugfs_dpcm_state' entry
by Fabio Estevam 29 Jul '17

29 Jul '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> 'debugfs_dpcm_state' member from structure snd_soc_pcm_runtime is never used at all, so it is safe to remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- include/sound/soc.h | 1 - sound/soc/soc-pcm.c | 5 ++--- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8f9376f..9d6e1db 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1233,7 +1233,6 @@ struct snd_soc_pcm_runtime { struct delayed_work delayed_work; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; - struct dentry *debugfs_dpcm_state; #endif unsigned int num; /* 0-based and monotonic increasing */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7d3859e..24148de 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -3000,8 +3000,7 @@ void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) return; } - rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444, - rtd->debugfs_dpcm_root, - rtd, &dpcm_state_fops); + debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, + rtd, &dpcm_state_fops); } #endif -- 2.7.4

1 0

Re: [alsa-devel] [PATCH] ASoC: rt5514: Use the local variable to get the data from SPI device
by Mark Brown 28 Jul '17

28 Jul '17

On Mon, Jul 24, 2017 at 03:06:36AM +0000, Oder Chiou wrote: > > On Fri, Jul 21, 2017 at 04:00:47PM +0800, Oder Chiou wrote: > > > The patch uses the local variable to get the data from SPI device to > > > prevent to get all zero data. > > How do you end up with all zero data from reading directly into rxbuf? > While the "runtime->dma_area" put into the rxbuf directly on the Intel > platform, it seemed that had the limitation to let the memory cannot be > written the data using the driver of the SPI controller. We verified the > original code on the platforms of Nvidia and Rockchip, and these are > working fine. Do we understand why that is? Note that IIRC you're not really supposed to DMA to or from the stack, there are some platforms with restrictions there so it's possible you're just moving the problem around here if rxbuf can be on the stack.

1 0

[alsa-devel] [PATCH 0/5] ALSA: constify snd_timer_hardware structures
by Bhumika Goyal 28 Jul '17

28 Jul '17

Declare snd_timer_hardware structures as const. Bhumika Goyal (5): ALSA: sound/core: constify snd_timer_hardware structures ALSA: isa: constify snd_timer_hardware structures ALSA: pci: constify snd_timer_hardware structures ALSA: sparc: constify snd_timer_hardware structure ALSA: opl3: constify snd_timer_hardware structures sound/core/hrtimer.c | 2 +- sound/core/pcm_timer.c | 2 +- sound/core/timer.c | 2 +- sound/drivers/opl3/opl3_lib.c | 4 ++-- sound/isa/ad1816a/ad1816a_lib.c | 2 +- sound/isa/gus/gus_timer.c | 4 ++-- sound/isa/wss/wss_lib.c | 2 +- sound/pci/emu10k1/timer.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/sparc/cs4231.c | 2 +- 10 files changed, 12 insertions(+), 12 deletions(-) -- 1.9.1

1 6

[alsa-devel] [PATCH] ASoC: Intel: Reset hw_ptr on resume trigger
by Cheng-Yi Chiang 28 Jul '17

28 Jul '17

From: "U. Artie Eoff" <ullysses.a.eoff(a)intel.com> Reset the hw_ptr before queuing the restore_stream_context work to eradicate a nasty white audio noise on resume. Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 4765ad474544..e0db7070cd42 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -187,8 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - if (pdata->restore_stream == true) + if (pdata->restore_stream == true) { + pcm_data->hw_ptr = 0; schedule_work(&pcm_data->work); + } else sst_byt_stream_resume(byt, pcm_data->stream); break; -- 2.12.2

2 1

[alsa-devel] [v2 0/3] ASoC: Intel: Fixes in Kabylake machine driver for clear playback and recording
by Harsha Priya 28 Jul '17

28 Jul '17

This patch series includes fixes for clean playback and recording in kabylake machine driver. It fixes the below issues in the corresponding patches and the details of each fix is provided in the patch's commit message. 1. Fix cracking noise in rt5663 headphones 2. Fix pop noise of dmic recording 3. Enable 4 slot IV feedback for maxim amps Harsha Priya (3): ASoC: Intel: Fix cracking noise in rt5663 headphones for kabylake platform ASoC: Intel: Fix pop noise of dmic recording in kabylake platform using rt5514 ASoC: Intel: Enabling 4 slot IV feedback for max98927 on Kabylake platform .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 28 +++++++++++----------- 1 file changed, 14 insertions(+), 14 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: codecs: add wm8524 codec driver
by Mihai Serban 28 Jul '17

28 Jul '17

WM8524 is a 24-bit 192KHz stereo digital/analog converter (DAC) with integral charge pump and a simple hardware control interface. Product information can be found at: https://www.cirrus.com/products/wm8524/ Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/wm8524.txt | 16 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8524.c | 265 +++++++++++++++++++++ 4 files changed, 288 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8524.txt create mode 100644 sound/soc/codecs/wm8524.c diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt new file mode 100644 index 0000000..20c6200 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8524.txt @@ -0,0 +1,16 @@ +WM8524 audio CODEC + +This device does not use I2C or SPI but a simple Hardware Control Interface. + +Required properties: + + - compatible : "wlf,wm8524" + + - wlf,mute-gpios: a GPIO spec for the MUTE pin. + +Example: + +codec: wm8524@0 { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 024ddc9..010811e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -174,6 +174,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8523 if I2C + select SND_SOC_WM8524 if GPIOLIB select SND_SOC_WM8580 if I2C select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8727 @@ -974,6 +975,10 @@ config SND_SOC_WM8523 tristate "Wolfson Microelectronics WM8523 DAC" depends on I2C +config SND_SOC_WM8524 + tristate "Wolfson Microelectronics WM8524 DAC" + depends on GPIOLIB + config SND_SOC_WM8580 tristate "Wolfson Microelectronics WM8580 and WM8581 CODECs" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a8a4b07..12dc2f7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -183,6 +183,7 @@ snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8523-objs := wm8523.o +snd-soc-wm8524-objs := wm8524.o snd-soc-wm8580-objs := wm8580.o snd-soc-wm8711-objs := wm8711.o snd-soc-wm8727-objs := wm8727.o @@ -416,6 +417,7 @@ obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o +obj-$(CONFIG_SND_SOC_WM8524) += snd-soc-wm8524.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c new file mode 100644 index 0000000..f42ec66 --- /dev/null +++ b/sound/soc/codecs/wm8524.c @@ -0,0 +1,265 @@ +/* + * wm8524.c -- WM8524 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2017 NXP + * + * Based on WM8523 ALSA SoC Audio driver written by Mark Brown + * + * 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/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +#define WM8524_NUM_RATES 7 + +/* codec private data */ +struct wm8524_priv { + struct gpio_desc *mute; + unsigned int sysclk; + unsigned int rate_constraint_list[WM8524_NUM_RATES]; + struct snd_pcm_hw_constraint_list rate_constraint; +}; + + +static const struct snd_soc_dapm_widget wm8524_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LINEVOUTL"), +SND_SOC_DAPM_OUTPUT("LINEVOUTR"), +}; + +static const struct snd_soc_dapm_route wm8524_dapm_routes[] = { + { "LINEVOUTL", NULL, "DAC" }, + { "LINEVOUTR", NULL, "DAC" }, +}; + +static const struct { + int value; + int ratio; +} lrclk_ratios[WM8524_NUM_RATES] = { + { 1, 128 }, + { 2, 192 }, + { 3, 256 }, + { 4, 384 }, + { 5, 512 }, + { 6, 768 }, + { 7, 1152 }, +}; + +static int wm8524_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8524->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8524->rate_constraint); + + gpiod_set_value_cansleep(wm8524->mute, 1); + + return 0; +} + +static void wm8524_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + + gpiod_set_value_cansleep(wm8524->mute, 0); +} + +static int wm8524_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i, j = 0; + + wm8524->sysclk = freq; + + wm8524->rate_constraint.count = 0; + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + val = freq / lrclk_ratios[i].ratio; + /* Check that it's a standard rate since core can't + * cope with others and having the odd rates confuses + * constraint matching. + */ + switch (val) { + case 8000: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + case 176400: + case 192000: + dev_err(codec->dev, "Supported sample rate: %dHz\n", + val); + wm8524->rate_constraint_list[j++] = val; + wm8524->rate_constraint.count++; + break; + default: + dev_dbg(codec->dev, "Skipping sample rate: %dHz\n", + val); + } + } + + /* Need at least one supported rate... */ + if (wm8524->rate_constraint.count == 0) + return -EINVAL; + + return 0; +} + +static int wm8524_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK | + SND_SOC_DAIFMT_MASTER_MASK); + + if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS)) { + dev_err(codec_dai->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(dai->codec); + + if (wm8524->mute) + gpiod_set_value_cansleep(wm8524->mute, mute); + + return 0; +} + +#define WM8524_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops wm8524_dai_ops = { + .startup = wm8524_startup, + .shutdown = wm8524_shutdown, + .set_sysclk = wm8524_set_dai_sysclk, + .set_fmt = wm8524_set_fmt, + .mute_stream = wm8524_mute_stream, +}; + +static struct snd_soc_dai_driver wm8524_dai = { + .name = "wm8524-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8524_RATES, + .formats = WM8524_FORMATS, + }, + .ops = &wm8524_dai_ops, +}; + +static int wm8524_probe(struct snd_soc_codec *codec) +{ + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + + wm8524->rate_constraint.list = &wm8524->rate_constraint_list[0]; + wm8524->rate_constraint.count = + ARRAY_SIZE(wm8524->rate_constraint_list); + + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_dev_wm8524 = { + .probe = wm8524_probe, + + .component_driver = { + .dapm_widgets = wm8524_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8524_dapm_widgets), + .dapm_routes = wm8524_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8524_dapm_routes), + }, +}; + +#ifdef CONFIG_OF +static const struct of_device_id wm8524_of_match[] = { + { .compatible = "wlf,wm8524" }, + { /* sentinel*/ } +}; +MODULE_DEVICE_TABLE(of, wm8524_of_match); +#endif + +static int wm8524_codec_probe(struct platform_device *pdev) +{ + struct wm8524_priv *wm8524; + int ret; + + wm8524 = devm_kzalloc(&pdev->dev, sizeof(struct wm8524_priv), + GFP_KERNEL); + if (wm8524 == NULL) + return -ENOMEM; + + platform_set_drvdata(pdev, wm8524); + + wm8524->mute = devm_gpiod_get(&pdev->dev, "wlf,mute", GPIOD_OUT_LOW); + if (IS_ERR(wm8524->mute)) { + ret = PTR_ERR(wm8524->mute); + dev_err(&pdev->dev, "Failed to get mute line: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_wm8524, &wm8524_dai, 1); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); + snd_soc_unregister_platform(&pdev->dev); + } + + return ret; +} + +static int wm8524_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver wm8524_codec_driver = { + .probe = wm8524_codec_probe, + .remove = wm8524_codec_remove, + .driver = { + .name = "wm8524-codec", + .of_match_table = wm8524_of_match, + }, +}; +module_platform_driver(wm8524_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8524 driver"); +MODULE_AUTHOR("Mihai Serban <mihai.serban(a)nxp.com>"); +MODULE_ALIAS("platform:wm8524-codec"); +MODULE_LICENSE("GPL"); -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: Intel: Add Kabylake machine driver for RT5663
by Kevin Cheng 28 Jul '17

28 Jul '17

Add kabylake machine driver & corresponding driver entry in machine table for rt5663 codec. For this particular machine, the RT5663 is a headset codec connected on ssp1. there is no dmic and speaker connected. Modify the machine driver in kbl_rt5663_max98927 to reuse the common codes. V2: adjust the code order. Kevin Cheng (2): ASoC: Intel: Add Kabylake machine driver for RT5663 ASoC: Intel: Add Kabylake RT5663 machine driver entry sound/soc/intel/boards/kbl_rt5663_max98927.c | 246 +++++++++++++++++++++++---- sound/soc/intel/skylake/skl.c | 5 + 2 files changed, 214 insertions(+), 37 deletions(-) -- 2.12.2

3 4

[alsa-devel] [PATCH 0/2] ASoC: Intel: Add Kabylake machine driver for RT5663
by Kevin Cheng 28 Jul '17

28 Jul '17

Add kabylake machine driver & corresponding driver entry in machine table for rt5663 codec. For this particular machine, the RT5663 is a headset codec connected on ssp1. there is no dmic and speaker connected. Modify the machine driver to kbl_rt5663_max98927 to reuse the common codes. Kevin Cheng (2): ASoC: Intel: Add Kabylake machine driver for RT5663 ASoC: Intel: Add Kabylake RT5663 machine driver entry sound/soc/intel/boards/kbl_rt5663_max98927.c | 246 +++++++++++++++++++++++---- sound/soc/intel/skylake/skl.c | 5 + 2 files changed, 214 insertions(+), 37 deletions(-) -- 1.9.1

2 3

[alsa-devel] [RFT 1/2] ASoC: samsung: Fix possible double iounmap on s3c24xx driver probe failure
by Krzysztof Kozlowski 28 Jul '17

28 Jul '17

Commit 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API") moved ioremap() call from s3c-i2s-v2.c:s3c_i2sv2_probe() to s3c2412-i2s.c:s3c2412_iis_dev_probe() and converted it to devm- resource managed interface. However the error path in first of them - s3c_i2sv2_probe() - was not updated. If getting a iis clock in s3c_i2sv2_probe() failed, the address space would be unmapped there. This could lead to: 1. double iounmap() later from devm-interface of s3c2412_iis_dev_probe()), 2. accessing the memory by other functions in s3c2412-i2s.c unit. Anyway, the owner of this mapped region should be s3c2412-i2s.c because it starts the mapping. Affected are drivers for S3C24xx family although issue was not reproduced. Fixes: 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API") Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Not marking as Cc-stable because this is theoretical problem, not reproduced and also not tested. Please, kindly test on S3C24xx hardware. --- sound/soc/samsung/s3c-i2s-v2.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index ca522a95160b..9b28046eea8e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -634,7 +634,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, i2s->iis_pclk = clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); - iounmap(i2s->regs); return -ENOENT; } -- 2.11.0

3 5

[alsa-devel] [RFT v2 0/3] ASoC: samsung: s3c24xx: Fixes for error paths
by Krzysztof Kozlowski 28 Jul '17

28 Jul '17

Hi, Changes since v1: 1. Add patch 2/3. 2. Drop assignment of iis_cclk=iis_pclk to have balance with clk disables error and remove paths (pointed by Arvind). Best regards, Krzysztof Krzysztof Kozlowski (3): ASoC: samsung: Fix possible double iounmap on s3c24xx driver probe failure ASoC: samsung: Add missing prepare for iis clock of s3c24xx ASoC: samsung: Add proper error paths to s3c24xx I2S driver sound/soc/samsung/s3c-i2s-v2.c | 12 ++++++++++-- sound/soc/samsung/s3c-i2s-v2.h | 7 +++++++ sound/soc/samsung/s3c2412-i2s.c | 13 +++++++++---- 3 files changed, 26 insertions(+), 6 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rsnd: merge snd_pcm_ops::open and snd_soc_dai_ops::startup
by Kuninori Morimoto 28 Jul '17

28 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd driver has snd_pcm_ops::open and snd_soc_dai_ops::startup, but, these are called from soc_pcm_open in same time. This patch merge these into rsnd_soc_dai_startup() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 80 +++++++++++++++++++----------------------------- 1 file changed, 31 insertions(+), 49 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8dcf1a8..6fbbcbf 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -844,12 +844,28 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, ir, &ic); } -static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime, - struct snd_soc_dai *dai) +static struct snd_pcm_hardware rsnd_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint; + struct snd_pcm_runtime *runtime = substream->runtime; unsigned int max_channels = rsnd_rdai_channels_get(rdai); + int ret; int i; /* @@ -866,34 +882,26 @@ static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime, constraint->count = i + 1; } + snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, constraint); + snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + /* * Sampling Rate / Channel Limitation * It depends on Clock Master Mode */ - if (!rsnd_rdai_is_clk_master(rdai)) - return; - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rsnd_soc_hw_rule_rate, dai, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rsnd_soc_hw_rule_channels, dai, - SNDRV_PCM_HW_PARAM_RATE, -1); -} - -static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); - struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); - int ret; - - /* rsnd_io_to_runtime() is not yet enabled here */ - rsnd_soc_hw_constraint(substream->runtime, dai); + if (rsnd_rdai_is_clk_master(rdai)) { + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + rsnd_soc_hw_rule_rate, dai, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + rsnd_soc_hw_rule_channels, dai, + SNDRV_PCM_HW_PARAM_RATE, -1); + } /* * call rsnd_dai_call without spinlock @@ -1105,31 +1113,6 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) /* * pcm ops */ -static struct snd_pcm_hardware rsnd_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .buffer_bytes_max = 64 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 32, - .fifo_size = 256, -}; - -static int rsnd_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - - return ret; -} - static int rsnd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -1159,7 +1142,6 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream) } static struct snd_pcm_ops rsnd_pcm_ops = { - .open = rsnd_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = rsnd_hw_params, .hw_free = snd_pcm_lib_free_pages, -- 1.9.1

2 1

[alsa-devel] [PATCH v3 0/3] ASoC: Intel: Fixes in Kabylake machine driver for clear playback and recording
by Harsha Priya 28 Jul '17

28 Jul '17

This patch series includes fixes for clean playback and recording in kabylake machine driver. It fixes the below issues in the corresponding patches and the details of each fix is provided in the patch's commit message. 1. Fix cracking noise in rt5663 headphones 2. Fix pop noise of dmic recording 3. Enable 4 slot IV feedback for maxim amps Harsha Priya (3): ASoC: Intel: Fix cracking noise in rt5663 headphones for kabylake platform ASoC: Intel: Fix pop noise of dmic recording in kabylake platform using rt5514 ASoC: Intel: Enabling 4 slot IV feedback for max98927 on Kabylake platform .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 28 +++++++++++----------- 1 file changed, 14 insertions(+), 14 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: Intel: Add Kabylake machine driver for RT5663
by Kevin Cheng 28 Jul '17

28 Jul '17

Add kabylake machine driver & corresponding driver entry in machine table for rt5663 codec. For this particular machine, the RT5663 is a headset codec connected on ssp1. there is no dmic and speaker connected. Modify the machine driver in kbl_rt5663_max98927 to reuse the common codes. V2: adjust the code order. Kevin Cheng (2): ASoC: Intel: Add Kabylake machine driver for RT5663 ASoC: Intel: Add Kabylake RT5663 machine driver entry sound/soc/intel/boards/kbl_rt5663_max98927.c | 246 +++++++++++++++++++++++---- sound/soc/intel/skylake/skl.c | 5 + 2 files changed, 214 insertions(+), 37 deletions(-) -- 2.12.2

2 3

[alsa-devel] [PATCH v1 0/3] add jack detection to MSM8916 analog
by Damien Riegel 27 Jul '17

27 Jul '17

This series adds support for the MBHC (Multi-Button Headset Control) of the PM8916's analog codec. This IP is capable to identify up to five buttons on a headset, but for now only the jack detection is supported. A complete implementation exists in the Android kernel, but it's quite complex and I'd rather go step by step. First patch fixes a define that is squatting a bit used by the MBHC. Second patch is the actual implementation of the jack detection, and third patch plugs the jack detection in the sound card driver. Damien Riegel (3): ASoC: codecs: msm8916-analog: fix DIG_CLK_CTL_RXD3_CLK_EN define ASoC: codecs: msm8916-analog: support jack detection ASoC: qcom: apq8016-sbc: enable jack detection sound/soc/codecs/msm8916-wcd-analog.c | 105 +++++++++++++++++++++++++++++++++- sound/soc/codecs/msm8916-wcd-analog.h | 18 ++++++ sound/soc/qcom/apq8016_sbc.c | 16 ++++++ 3 files changed, 138 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/msm8916-wcd-analog.h -- 2.13.3

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[alsa-devel] [PATCH v3 0/2] Add support for TI TAS6424 digital amplifier
by Michael Stecklein 27 Jul '17

27 Jul '17

Hello all, This two-patch series adds the dt bindings, driver, and other related changes for the TAS6424 digital amplifier. I am upstreaming this driver on behalf of Andreas. Regards, Michael Stecklein <m-stecklein(a)ti.com> Changes since v2: - fixed binding as per Rob Herring's comments on v2 Changes since v1: - rebased on 4.13-rc1 Original v1 patches: - https://patchwork.kernel.org/patch/9795235/ - https://patchwork.kernel.org/patch/9795237/ Andreas Dannenberg (1): ASoC: add support for TAS6424 digital amplifier Michael Stecklein (1): dt-bindings: sound: add bindings for TAS6424 .../devicetree/bindings/sound/ti,tas6424.txt | 28 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas6424.c | 755 +++++++++++++++++++++ sound/soc/codecs/tas6424.h | 127 ++++ 5 files changed, 920 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas6424.txt create mode 100644 sound/soc/codecs/tas6424.c create mode 100644 sound/soc/codecs/tas6424.h -- 2.7.4

3 6

[alsa-devel] [PATCH v3 00/17] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 27 Jul '17

27 Jul '17

The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Summary of main changes since V1: - Removed child DT nodes and of_compatibles to make the DT flat - Fixed review comments on irqchip driver - Used new field names for IRQ1_STATUS_33 register - Moved the gpio driver bindings into the main MFD binding doc - Some code improvements to the pinctrl driver Richard Fitzgerald (17): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add DT bindings for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add DT bindings for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 Documentation/devicetree/bindings/mfd/madera.txt | 102 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 99 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 67 + MAINTAINERS | 23 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 301 + drivers/mfd/Kconfig | 46 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1688 ++++ drivers/mfd/cs47l85-tables.c | 3169 +++++++ drivers/mfd/cs47l90-tables.c | 2830 +++++++ drivers/mfd/madera-core.c | 632 ++ drivers/mfd/madera-i2c.c | 131 + drivers/mfd/madera-spi.c | 132 + drivers/mfd/madera.h | 52 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1074 +++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 27 + include/linux/irqchip/irq-madera-pdata.h | 23 + include/linux/irqchip/irq-madera.h | 96 + include/linux/mfd/madera/core.h | 180 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 8832 ++++++++++++++++++++ include/sound/madera-pdata.h | 61 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1747 ++++ sound/soc/codecs/cs47l85.c | 2706 ++++++ sound/soc/codecs/cs47l90.c | 2645 ++++++ sound/soc/codecs/madera.c | 4430 ++++++++++ sound/soc/codecs/madera.h | 470 ++ 46 files changed, 32268 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: hdmi-codec: Allow driver to restrict sample size
by srinivas.kandagatla＠linaro.org 27 Jul '17

27 Jul '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patch adds option to allow hdmi-codec drivers to restrict the audio sample size based on the values that are suppored in hardware. First patch is adding that formats option to hdmi_codec_pdata and second patch is a fix in adv7511 codec driver which only supports 16 and 24 bit samples. Srinivas Kandagatla (2): ASoC: hdmi-codec: Allow drivers to restrict sample sizes. drm/bridge: adv7511: restrict audio sample sizes drivers/gpu/drm/bridge/adv7511/adv7511_audio.c | 2 ++ include/sound/hdmi-codec.h | 1 + sound/soc/codecs/hdmi-codec.c | 3 +++ 3 files changed, 6 insertions(+) -- 2.11.0

3 4

[alsa-devel] [PATCHv2 0/6] Motorola Droid 4 Audio Support
by Sebastian Reichel 27 Jul '17

27 Jul '17

Hi, This adds audio support to Motorola Droid 4. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (actually untested, but probably needs some quirks) Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (5): dt-bindings: sound: add motorola,cpcap-audio-codec ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio ARM: dts: omap4-droid4: add soundcard Tony Lindgren (1): ALSA: pcm: Export soc_dpcm_runtime_update .../bindings/sound/motorola,cpcap-audio-codec.txt | 19 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 7 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1607 ++++++++++++++++++++ sound/soc/soc-pcm.c | 2 + 7 files changed, 1726 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.13.2

4 19

Re: [alsa-devel] [PATCH v1 5/6] ASoC: codecs: msm8916-wcd-analog: add MBHC support
by Damien Riegel 27 Jul '17

27 Jul '17

[Sorry for breaking the thread, but I don't have the initial message to reply to.] I tried your patchset and faced some issues regarding removal detection: the driver detects headphone and microphone removal as soon as I press a button on the headset. evtest logs: [Headset insertion] Event: time 10207.841157, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 1 Event: time 10207.841157, -------------- SYN_REPORT ------------ Event: time 10208.039105, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 0 Event: time 10208.039105, -------------- SYN_REPORT ------------ Event: time 10208.075519, type 5 (EV_SW), code 2 (SW_HEADPHONE_INSERT), value 1 Event: time 10208.075519, type 5 (EV_SW), code 4 (SW_MICROPHONE_INSERT), value 1 Event: time 10208.075519, -------------- SYN_REPORT ------------ [Pressing Volume up Button] Event: time 10210.590403, type 1 (EV_KEY), code 115 (KEY_VOLUMEUP), value 1 Event: time 10210.590403, type 5 (EV_SW), code 2 (SW_HEADPHONE_INSERT), value 0 Event: time 10210.590403, type 5 (EV_SW), code 4 (SW_MICROPHONE_INSERT), value 0 Event: time 10210.590403, -------------- SYN_REPORT ------------ Event: time 10210.719560, type 1 (EV_KEY), code 115 (KEY_VOLUMEUP), value 0 Event: time 10210.719560, -------------- SYN_REPORT ------------ [Unplugging] Event: time 10221.926156, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 1 Event: time 10221.926156, -------------- SYN_REPORT ------------ Event: time 10222.026466, type 1 (EV_KEY), code 226 (KEY_MEDIA), value 0 Event: time 10222.026466, -------------- SYN_REPORT ------------ Also, the KEY_MEDIA is wrongly reported but I don't see that every time, I think it depends on how fast I plug in the headset. Anyway, I think that can easily be filtered out by testing the presence of the microphone. Regards, -- Damien

2 1

[alsa-devel] [PATCH] ASoC: samsung: s3c2412: cleanups / fixes for preparation of clocks.
by Arvind Yadav 27 Jul '17

27 Jul '17

-Use devm_clk_get() to make cleanup paths more simple. -clk_prepare_enable() can fail here and we must check its return value. -Add s3c_i2sv2_remove cleanup function of s3c_i2sv2_probe. -No need to iounmap. Here, mapping done by devm_ioremap_resource. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/soc/samsung/s3c-i2s-v2.c | 19 ++++++++++++++++--- sound/soc/samsung/s3c-i2s-v2.h | 3 +++ sound/soc/samsung/s3c2412-i2s.c | 10 ++++++++-- 3 files changed, 27 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 8f42dea..ab4899f 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -625,20 +625,24 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, { struct device *dev = dai->dev; unsigned int iismod; + int ret; i2s->dev = dev; /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->iis_pclk = clk_get(dev, "iis"); + i2s->iis_pclk = devm_clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); - iounmap(i2s->regs); return -ENOENT; } - clk_enable(i2s->iis_pclk); + ret = clk_prepare_enable(i2s->iis_pclk); + if (ret) { + dev_err(dev, "failed to prepare iis_clock\n"); + return ret; + } /* Mark ourselves as in TXRX mode so we can run through our cleanup * process without warnings. */ @@ -652,6 +656,15 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); +int s3c_i2sv2_remove(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + clk_disable_unprepare(i2s->iis_pclk); + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_remove); + #ifdef CONFIG_PM static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) { diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index 182d805..d6844b9 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -91,6 +91,9 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, struct s3c_i2sv2_info *i2s, unsigned long base); + +extern int s3c_i2sv2_remove(struct snd_soc_dai *dai); + /** * s3c_i2sv2_register_component - register component and dai with soc core * @dev: DAI device diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 0a47182..adfbd52d 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -65,13 +65,18 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); + s3c_i2sv2_remove(dai); return PTR_ERR(s3c2412_i2s.iis_cclk); } /* Set MPLL as the source for IIS CLK */ - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_prepare_enable(s3c2412_i2s.iis_cclk); + clk_set_parent(s3c2412_i2s.iis_cclk, devm_clk_get(dai->dev, "mpll")); + ret = clk_prepare_enable(s3c2412_i2s.iis_cclk); + if (ret) { + s3c_i2sv2_remove(dai); + return ret; + } s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; @@ -85,6 +90,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { clk_disable_unprepare(s3c2412_i2s.iis_cclk); + s3c_i2sv2_remove(dai); return 0; } -- 2.7.4

2 2

[alsa-devel] [PATCH][resend] ASoC: rsnd: merge snd_pcm_ops::open and snd_soc_dai_ops::startup
by Kuninori Morimoto 27 Jul '17

27 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd driver has snd_pcm_ops::open and snd_soc_dai_ops::startup, but, these are called from soc_pcm_open in same time. This patch merge these into rsnd_soc_dai_startup() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 80 +++++++++++++++++++----------------------------- 1 file changed, 31 insertions(+), 49 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8dcf1a8..6fbbcbf 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -844,12 +844,28 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, ir, &ic); } -static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime, - struct snd_soc_dai *dai) +static struct snd_pcm_hardware rsnd_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint; + struct snd_pcm_runtime *runtime = substream->runtime; unsigned int max_channels = rsnd_rdai_channels_get(rdai); + int ret; int i; /* @@ -866,34 +882,26 @@ static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime, constraint->count = i + 1; } + snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, constraint); + snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + /* * Sampling Rate / Channel Limitation * It depends on Clock Master Mode */ - if (!rsnd_rdai_is_clk_master(rdai)) - return; - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rsnd_soc_hw_rule_rate, dai, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rsnd_soc_hw_rule_channels, dai, - SNDRV_PCM_HW_PARAM_RATE, -1); -} - -static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); - struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); - int ret; - - /* rsnd_io_to_runtime() is not yet enabled here */ - rsnd_soc_hw_constraint(substream->runtime, dai); + if (rsnd_rdai_is_clk_master(rdai)) { + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + rsnd_soc_hw_rule_rate, dai, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + rsnd_soc_hw_rule_channels, dai, + SNDRV_PCM_HW_PARAM_RATE, -1); + } /* * call rsnd_dai_call without spinlock @@ -1105,31 +1113,6 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) /* * pcm ops */ -static struct snd_pcm_hardware rsnd_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .buffer_bytes_max = 64 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 32, - .fifo_size = 256, -}; - -static int rsnd_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - - return ret; -} - static int rsnd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -1159,7 +1142,6 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream) } static struct snd_pcm_ops rsnd_pcm_ops = { - .open = rsnd_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = rsnd_hw_params, .hw_free = snd_pcm_lib_free_pages, -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: Intel: Fix Kabylake machine driver for pops and clicks
by Harsha Priya 26 Jul '17

26 Jul '17

This patch adds fixes for the following issues in Kabylake machine driver 1. crackling noise in rt5663 headset codec 2. enabling 4 slot IV feedback for max98927 speaker amp codec 3. pop noise while recording from rt5514 dmic codec Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 28 ++++++++++++---------- 1 file changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 3fe4a08..4c51965 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -319,7 +319,9 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); ret = snd_soc_dai_set_sysclk(codec_dai, RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); @@ -349,27 +351,27 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, 0, - RT5514_PLL1_S_BCLK, RT5514_AIF1_BCLK_FREQ, - RT5514_AIF1_SYSCLK_FREQ); + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { - dev_err(rtd->dev, "set bclk err: %d\n", ret); + dev_err(rtd->dev, "set sysclk err: %d\n", ret); return ret; } + } - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_PLL1, RT5514_AIF1_SYSCLK_FREQ, - SND_SOC_CLOCK_IN); + 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(rtd->dev, "set sclk err: %d\n", ret); + dev_err(rtd->dev, "DEV0 TDM slot err:%d\n", ret); return ret; } } - if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME) || - !strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF0, 3, 8, 16); + + 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(rtd->dev, "set TDM slot err:%d\n", ret); + dev_err(rtd->dev, "DEV1 TDM slot err:%d\n", ret); return ret; } } -- 1.9.1

4 3

[alsa-devel] intel sound device not detected
by fulgor fulgor 26 Jul '17

26 Jul '17

Hello I read a post from you in a mailling list I use manjaro with latest kernel but sound device is not detected could you help ? thanks fulgorit

1 0

[alsa-devel] [PATCH 00/11] Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Arvind Yadav (11): [PATCH 01/11] ASoC: samsung: s3c2412: Handle return value of clk_prepare_enable. [PATCH 02/11] ASoC: samsung: s3c24xx: Handle return value of clk_prepare_enable. [PATCH 03/11] ASoC: samsung: pcm: Handle return value of clk_prepare_enable. [PATCH 04/11] ASoC: samsung: i2s: Handle return value of clk_prepare_enable. [PATCH 05/11] ASoC: samsung: spdif: Handle return value of clk_prepare_enable. [PATCH 06/11] ASoC: mxs-saif: Handle return value of clk_prepare_enable/clk_prepare. [PATCH 07/11] ASoC: jz4740: Handle return value of clk_prepare_enable. [PATCH 08/11] ASoC: sun4i-spdif: Handle return value of clk_prepare_enable. [PATCH 09/11] ASoC: atmel: ac97c: Handle return value of clk_prepare_enable. [PATCH 10/11] ASoC: pxa: Handle return value of clk_prepare_enable. [PATCH 11/11] ASoC: pxa-ssp: Handle return value of clk_prepare_enable. sound/atmel/ac97c.c | 8 +++++--- sound/soc/jz4740/jz4740-i2s.c | 21 +++++++++++++++++---- sound/soc/mxs/mxs-saif.c | 13 ++++++++++--- sound/soc/pxa/pxa-ssp.c | 23 +++++++++++++++++------ sound/soc/pxa/pxa2xx-i2s.c | 6 +++++- sound/soc/samsung/i2s.c | 19 +++++++++++++++---- sound/soc/samsung/pcm.c | 8 ++++++-- sound/soc/samsung/s3c2412-i2s.c | 4 +++- sound/soc/samsung/s3c24xx-i2s.c | 11 +++++++++-- sound/soc/samsung/spdif.c | 8 ++++++-- sound/soc/sunxi/sun4i-spdif.c | 11 ++++++++--- 11 files changed, 101 insertions(+), 31 deletions(-) -- 1.9.1

3 22

[alsa-devel] [PATCH v2 01/11] ASoC: samsung: s3c2412: Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- Chnage in v2 : Error handling for things done in s3c_i2sv2_probe(). sound/soc/samsung/s3c2412-i2s.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 0a47182..0b96927 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -65,13 +65,16 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - return PTR_ERR(s3c2412_i2s.iis_cclk); + ret = PTR_ERR(s3c2412_i2s.iis_cclk); + goto err; } /* Set MPLL as the source for IIS CLK */ clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_prepare_enable(s3c2412_i2s.iis_cclk); + ret = clk_prepare_enable(s3c2412_i2s.iis_cclk); + if (ret) + goto err; s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; @@ -80,6 +83,11 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) S3C_GPIO_PULL_NONE); return 0; + +err: + clk_disable(s3c2412_i2s.iis_pclk); + clk_put(s3c2412_i2s.iis_pclk); + return ret; } static int s3c2412_i2s_remove(struct snd_soc_dai *dai) -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: rt5665: add MONOVOL Playback Volume control
by Bard Liao 26 Jul '17

26 Jul '17

MONOVOL Playback Volume is a playback volume control on "MONOVOL MIX". Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 7e64b25..54ace3b 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1402,6 +1402,9 @@ static const struct snd_kcontrol_new rt5665_snd_controls[] = { RT5665_L_VOL_SFT, 15, 1, snd_soc_get_volsw, rt5665_mono_vol_put, mono_vol_tlv), + SOC_SINGLE_TLV("MONOVOL Playback Volume", RT5665_MONO_OUT, + RT5665_L_VOL_SFT, 39, 1, out_vol_tlv), + /* Output Volume */ SOC_DOUBLE_TLV("OUT Playback Volume", RT5665_LOUT, RT5665_L_VOL_SFT, RT5665_R_VOL_SFT, 39, 1, out_vol_tlv), -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: sh: hac: add missing "int ret"
by Kuninori Morimoto 26 Jul '17

26 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit b047e1cce8 ("ASoC: ac97: Support multi-platform AC'97") modified hac_soc_platform_probe(), but "int ret" was missed. This patch adds missing "int ret", otherwise, we will get linux/sound/soc/sh/hac.c: In function 'hac_soc_platform_probe': linux/sound/soc/sh/hac.c:318: error: 'ret' undeclared (first use in this function) linux/sound/soc/sh/hac.c:318: error: (Each undeclared identifier is reported only once linux/sound/soc/sh/hac.c:318: error: for each function it appears in.) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/hac.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 84c5103..624aaf5 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -315,6 +315,8 @@ static int hac_hw_params(struct snd_pcm_substream *substream, static int hac_soc_platform_probe(struct platform_device *pdev) { + int ret; + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); if (ret != 0) return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: don't use private_value on rsnd_kctrl_new()
by Kuninori Morimoto 26 Jul '17

26 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd kctrl is using both .private_value (for rsnd_kctrl_cfg) and .private_data (for rsnd_mod) on snd_kcontrol. But only 1 private data (= rsnd_kctrl_cfg) can be enough if it has rsnd_mod pointer. This patch doesn't use private_value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 14 ++++++-------- sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index d3381f3..c345623 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1151,11 +1151,10 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream) /* * snd_kcontrol */ -#define kcontrol_to_cfg(kctrl) ((struct rsnd_kctrl_cfg *)kctrl->private_value) static int rsnd_kctrl_info(struct snd_kcontrol *kctrl, struct snd_ctl_elem_info *uinfo) { - struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl); if (cfg->texts) { uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -1181,7 +1180,7 @@ static int rsnd_kctrl_info(struct snd_kcontrol *kctrl, static int rsnd_kctrl_get(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *uc) { - struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl); int i; for (i = 0; i < cfg->size; i++) @@ -1196,8 +1195,7 @@ static int rsnd_kctrl_get(struct snd_kcontrol *kctrl, static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *uc) { - struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl); int i, change = 0; if (!cfg->accept(cfg->io)) @@ -1214,7 +1212,7 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, } if (change && cfg->update) - cfg->update(cfg->io, mod); + cfg->update(cfg->io, cfg->mod); return change; } @@ -1266,14 +1264,13 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, .index = rtd->num, .get = rsnd_kctrl_get, .put = rsnd_kctrl_put, - .private_value = (unsigned long)cfg, }; int ret; if (size > RSND_MAX_CHANNELS) return -EINVAL; - kctrl = snd_ctl_new1(&knew, mod); + kctrl = snd_ctl_new1(&knew, cfg); if (!kctrl) return -ENOMEM; @@ -1289,6 +1286,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->card = card; cfg->kctrl = kctrl; cfg->io = io; + cfg->mod = mod; return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 99c5761..c5de71f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -614,6 +614,7 @@ struct rsnd_kctrl_cfg { struct rsnd_dai_stream *io; struct snd_card *card; struct snd_kcontrol *kctrl; + struct rsnd_mod *mod; }; #define RSND_MAX_CHANNELS 8 -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Add the I2S ASRC support
by Oder Chiou 26 Jul '17

26 Jul '17

The patch adds the I2S ASRC support. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 19 +++++++++++++++++++ sound/soc/codecs/rt5514.h | 7 +++++++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a105c84..882fdc4 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -63,6 +63,7 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_SRC_CTRL, 0x44000eee}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, + {RT5514_ASRC_IN_CTRL1, 0x00000003}, }; static const struct reg_default rt5514_reg[] = { @@ -471,6 +472,15 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int rt5514_i2s_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + return (rt5514->sysclk > rt5514->lrck * 384); +} + static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1L"), @@ -541,6 +551,10 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { RT5514_POW_PLL1_LDO_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL1", RT5514_PWR_ANA2, RT5514_POW_PLL1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ASRC AD1", 1, RT5514_CLK_CTRL2, + RT5514_CLK_AD0_ASRC_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ASRC AD2", 1, RT5514_CLK_CTRL2, + RT5514_CLK_AD1_ASRC_EN_BIT, 0, NULL, 0), /* ADC Mux */ SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, @@ -638,6 +652,7 @@ static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, { "Stereo1 ADC MIX", NULL, "adc stereo1 filter" }, { "adc stereo1 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll }, + { "adc stereo1 filter", NULL, "ASRC AD1", rt5514_i2s_use_asrc }, { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, @@ -654,6 +669,7 @@ static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { { "Stereo2 ADC MIX", NULL, "Stereo2 ADC MIXR" }, { "Stereo2 ADC MIX", NULL, "adc stereo2 filter" }, { "adc stereo2 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll }, + { "adc stereo2 filter", NULL, "ASRC AD2", rt5514_i2s_use_asrc }, { "AIF1TX", NULL, "Stereo1 ADC MIX"}, { "AIF1TX", NULL, "Stereo2 ADC MIX"}, @@ -706,6 +722,9 @@ static int rt5514_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_I2S_DL_MASK, val_len); + regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL1, + RT5514_CLK_AD_ANA1_SEL_MASK, + (pre_div + 1) << RT5514_CLK_AD_ANA1_SEL_SFT); regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL2, RT5514_CLK_SYS_DIV_OUT_MASK | RT5514_SEL_ADC_OSR_MASK, pre_div << RT5514_CLK_SYS_DIV_OUT_SFT | diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 75024a4..2dc40e6 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -37,6 +37,7 @@ #define RT5514_PLL3_CALIB_CTRL5 0x2124 #define RT5514_DELAY_BUF_CTRL1 0x2140 #define RT5514_DELAY_BUF_CTRL3 0x2148 +#define RT5514_ASRC_IN_CTRL1 0x2180 #define RT5514_DOWNFILTER0_CTRL1 0x2190 #define RT5514_DOWNFILTER0_CTRL2 0x2194 #define RT5514_DOWNFILTER0_CTRL3 0x2198 @@ -197,8 +198,14 @@ #define RT5514_CLK_AD0_EN_BIT 23 #define RT5514_CLK_DMIC_OUT_SEL_MASK (0x7 << 8) #define RT5514_CLK_DMIC_OUT_SEL_SFT 8 +#define RT5514_CLK_AD_ANA1_SEL_MASK (0xf << 0) +#define RT5514_CLK_AD_ANA1_SEL_SFT 0 /* RT5514_CLK_CTRL2 (0x2108) */ +#define RT5514_CLK_AD1_ASRC_EN (0x1 << 17) +#define RT5514_CLK_AD1_ASRC_EN_BIT 17 +#define RT5514_CLK_AD0_ASRC_EN (0x1 << 16) +#define RT5514_CLK_AD0_ASRC_EN_BIT 16 #define RT5514_CLK_SYS_DIV_OUT_MASK (0x7 << 8) #define RT5514_CLK_SYS_DIV_OUT_SFT 8 #define RT5514_SEL_ADC_OSR_MASK (0x7 << 4) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] constify codecs acpi_device_id.
by Arvind Yadav 26 Jul '17

26 Jul '17

acpi_device_id are not supposed to change at runtime. All functions working with acpi_device_id provided by <acpi/acpi_bus.h> work with const acpi_device_id. So mark the non-const structs as const. Arvind Yadav (4): [PATCH 1/4] ASoC: rt5514: constify acpi_device_id. [PATCH 2/4] ASoC: rt5663: constify acpi_device_id. [PATCH 3/4] ASoC: rt5659: constify acpi_device_id. [PATCH 4/4] ASoC: rt5665: constify acpi_device_id. sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH] ASoC: rt5514: Use the IS_ENABLED to supports the module build
by Oder Chiou 26 Jul '17

26 Jul '17

The patch uses the macro IS_ENABLED to make sure that the SPI function also supports the module build. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 882fdc4..7fffa7c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -31,7 +31,7 @@ #include "rl6231.h" #include "rt5514.h" -#if defined(CONFIG_SND_SOC_RT5514_SPI) +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) #include "rt5514-spi.h" #endif @@ -312,7 +312,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); if (fw) { -#if defined(CONFIG_SND_SOC_RT5514_SPI) +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_write(0x4ff60000, fw->data, ((fw->size/8)+1)*8); #else @@ -325,7 +325,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, request_firmware(&fw, RT5514_FIRMWARE2, codec->dev); if (fw) { -#if defined(CONFIG_SND_SOC_RT5514_SPI) +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_write(0x4ffc0000, fw->data, ((fw->size/8)+1)*8); #else -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5670: fix wrong name of sto2 adc 2 mux
by Bard Liao 26 Jul '17

26 Jul '17

The kcontrol name of Stereo2 ADC 2 Mux is not matched with what we used in dapm widget. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0376667..2866f9b 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1175,7 +1175,7 @@ static const struct snd_kcontrol_new rt5670_sto_adc_2_mux = static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l_mux = +static const struct snd_kcontrol_new rt5670_sto2_adc_2_mux = SOC_DAPM_ENUM("Stereo2 ADC 2 Mux", rt5670_stereo2_adc2_enum); -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: remove cache_bypass from snd_soc_codec
by Kuninori Morimoto 26 Jul '17

26 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_codec .cache_bypass related operation code has been removed. Let's remove remaining code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 - sound/soc/codecs/max98927.c | 1 - 2 files changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index dfc6dae..436fa0e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -900,7 +900,6 @@ struct snd_soc_codec { struct list_head list; /* runtime */ - unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int cache_init:1; /* codec cache has been initialized */ /* codec IO */ diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b5ee294..b0380b5 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -577,7 +577,6 @@ static int max98927_probe(struct snd_soc_codec *codec) max98927->codec = codec; codec->control_data = max98927->regmap; - codec->cache_bypass = 1; /* Software Reset */ regmap_write(max98927->regmap, -- 1.9.1

2 1

[alsa-devel] [PATCH 08/11] ASoC: sun4i-spdif: Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/soc/sunxi/sun4i-spdif.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index eaefd07..4a316c0 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -458,11 +458,16 @@ static int sun4i_spdif_runtime_suspend(struct device *dev) static int sun4i_spdif_runtime_resume(struct device *dev) { struct sun4i_spdif_dev *host = dev_get_drvdata(dev); + int ret; - clk_prepare_enable(host->spdif_clk); - clk_prepare_enable(host->apb_clk); + ret = clk_prepare_enable(host->spdif_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->apb_clk); + if (ret) + clk_disable_unprepare(host->spdif_clk); - return 0; + return ret; } static int sun4i_spdif_probe(struct platform_device *pdev) -- 1.9.1

2 1

[alsa-devel] [PATCH 09/11] ASoC: atmel: ac97c: Handle return value of clk_prepare_enable.
by Arvind Yadav 26 Jul '17

26 Jul '17

clk_prepare_enable() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/atmel/ac97c.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 9d2c9d9..52b0522 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -783,7 +783,9 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_prepare_enable(pclk); + retval = clk_prepare_enable(pclk); + if (retval) + return retval; retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -897,9 +899,9 @@ static int atmel_ac97c_resume(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; + int ret = clk_prepare_enable(chip->pclk); - clk_prepare_enable(chip->pclk); - return 0; + return ret; } static SIMPLE_DEV_PM_OPS(atmel_ac97c_pm, atmel_ac97c_suspend, atmel_ac97c_resume); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix missing sentinels in sst_acpi_mach
by Vinod Koul 26 Jul '17

26 Jul '17

Couple of instances of sst_acpi_mach were having missing sentinels so add them up Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917ee41cf..9e3f8c04dd32 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -941,6 +941,7 @@ static struct sst_acpi_mach sst_bxtp_devdata[] = { .machine_quirk = sst_acpi_codec_list, .quirk_data = &bxt_codecs, }, + {} }; static struct sst_acpi_mach sst_kbl_devdata[] = { @@ -991,6 +992,7 @@ static struct sst_acpi_mach sst_glk_devdata[] = { .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", }, + {} }; /* PCI IDs */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: board: Fix missing sentinel for bxt_board_id
by Pradeep Tewani 26 Jul '17

26 Jul '17

Without a sentinel in the array of platform devices, we can get a panic so add a sentinel. Full stack dump below: [ 231.564705] BUG: KASAN: global-out-of-bounds in platform_match+0xb0/0x146 [ 231.565500] Read of size 1 at addr ffffffff8380a1a0 by task swapper/0/1 [ 231.566280] [ 231.566594] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.12.0-00289-g152771f #1 [ 231.567526] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.9.3-20161025_171302-gandalf 04/01/2014 [ 231.568730] Call Trace: [ 231.569128] dump_stack+0xf8/0x14a [ 231.569615] print_address_description+0x57/0x1e4 [ 231.570216] ? driver_probe_device+0x814/0x814 [ 231.570798] kasan_report+0x1cb/0x1eb [ 231.571302] ? platform_match+0xb0/0x146 [ 231.571833] __asan_load1+0x45/0x47 [ 231.572324] platform_match+0xb0/0x146 [ 231.572843] ? platform_uevent+0x73/0x73 [ 231.573370] __driver_attach+0x47/0x16a [ 231.573890] bus_for_each_dev+0x11a/0x15c [ 231.574422] ? bus_remove_file+0x57/0x57 [ 231.574951] ? do_raw_spin_unlock+0xc3/0xe0 [ 231.575501] driver_attach+0x2b/0x2e [ 231.576002] bus_add_driver+0x25e/0x3c5 [ 231.576520] ? set_debug_rodata+0x20/0x20 [ 231.584776] driver_register+0x13e/0x19d [ 231.585305] __platform_driver_register+0x6c/0x71 [ 231.585908] ? skl_driver_init+0x20/0x20 [ 231.586436] broxton_audio_init+0x17/0x19 [ 231.586976] do_one_initcall+0x11a/0x221 [ 231.587499] ? start_kernel+0x525/0x525 [ 231.588020] ? lock_downgrade+0x281/0x2ec [ 231.588557] ? trace_hardirqs_on_caller+0x223/0x267 [ 231.589169] ? _raw_spin_unlock_irqrestore+0x49/0x55 [ 231.589791] ? set_debug_rodata+0x20/0x20 [ 231.590324] kernel_init_freeable+0x1fd/0x2b0 [ 231.590891] ? rest_init+0x17c/0x17c [ 231.591384] kernel_init+0x11/0x157 [ 231.591876] ? rest_init+0x17c/0x17c [ 231.592371] ret_from_fork+0x2a/0x40 [ 231.592865] [ 231.593166] The buggy address belongs to the variable: [ 231.593806] bxt_board_ids+0x40/0x9e0 Fixes: 152771fbc3 ("ASoC: Intel: board: Add Geminilake platform support") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Pradeep Tewani <pradeep.d.tewani(a)intel.com> --- sound/soc/intel/boards/bxt_rt298.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index a4b5bcb..7843104 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -595,6 +595,7 @@ static const struct platform_device_id bxt_board_ids[] = { (unsigned long)&broxton_rt298 }, { .name = "glk_alc298s_i2s", .driver_data = (unsigned long)&geminilake_rt298 }, + {} }; static struct platform_driver broxton_audio = { -- 2.7.4

3 3

[alsa-devel] [PATCH 1/2] ASoC: rt5670: merge ADC L/R Mux
by Bard Liao 26 Jul '17

26 Jul '17

Some ADC L/R Muxes share the some control bit. So, merge them. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 42 ++++++++++++++++-------------------------- 1 file changed, 16 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0ec7985..2866f9b 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1151,20 +1151,15 @@ static const char * const rt5670_stereo_adc1_src[] = { static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); -static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = - SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); - -static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = - SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); +static const struct snd_kcontrol_new rt5670_sto_adc_1_mux = + SOC_DAPM_ENUM("Stereo1 ADC 1 Mux", rt5670_stereo1_adc1_enum); static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = - SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); +static const struct snd_kcontrol_new rt5670_sto2_adc_1_mux = + SOC_DAPM_ENUM("Stereo2 ADC 1 Mux", rt5670_stereo2_adc1_enum); -static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = - SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); /* MX-27 MX-26 [11] */ static const char * const rt5670_stereo_adc2_src[] = { @@ -1174,20 +1169,15 @@ static const char * const rt5670_stereo_adc2_src[] = { static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = - SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); - -static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = - SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); +static const struct snd_kcontrol_new rt5670_sto_adc_2_mux = + SOC_DAPM_ENUM("Stereo1 ADC 2 Mux", rt5670_stereo1_adc2_enum); static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = - SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); +static const struct snd_kcontrol_new rt5670_sto2_adc_2_mux = + SOC_DAPM_ENUM("Stereo2 ADC 2 Mux", rt5670_stereo2_adc2_enum); -static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = - SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); /* MX-27 MX26 [10] */ static const char * const rt5670_stereo_adc_src[] = { @@ -1642,23 +1632,23 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto1_dmic_mux), SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_l2_mux), + &rt5670_sto_adc_2_mux), SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_r2_mux), + &rt5670_sto_adc_2_mux), SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_l1_mux), + &rt5670_sto_adc_1_mux), SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_r1_mux), + &rt5670_sto_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto2_dmic_mux), SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_l2_mux), + &rt5670_sto2_adc_2_mux), SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_r2_mux), + &rt5670_sto2_adc_2_mux), SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_l1_mux), + &rt5670_sto2_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_r1_mux), + &rt5670_sto2_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto2_adc_lr_mux), SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, -- 2.7.4

2 2

[alsa-devel] [PATCH v4 0/9] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 26 Jul '17

26 Jul '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. v4: - initialized the clk_init_data structure, - switched to of_clk_parent_fill instead of manually setting parent_names, - removed include of clkdev.h in favor of slab.h, - removed unnecessary second cast in operation, - used clamp instead of two conditions, - removed dependency on clock-output-names, the name is gotten from DT node, - merged all three drivers for audio PLLs (FRAC, PMC, PAD) into one, v3: - added a flag per generated clock to know which one is allowed to modify audio_pll rate, - set the flag in setup function depending on the compatible and the clock ID, v2: - split big patch containing dt-binding, pll drivers and classd modifications, - removed unused AUDIO_PLL_*FOUT* defines from clk-audio-pll-pmc, - added conditions for audio pll rate setting restriction for SSC and I2S, - renamed all mentions of ACLK to GCLK in classD driver, Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop dt-bindings: clk: at91: add audio plls to the compatible list clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock Documentation/devicetree/bindings/clock/at91-clock.txt | 10 +- Documentation/devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 +- arch/arm/boot/dts/sama5d2.dtsi | 39 +- arch/arm/mach-at91/Kconfig | 4 +- drivers/clk/at91/Makefile | 1 +- drivers/clk/at91/clk-audio-pll.c | 531 ++++++++- drivers/clk/at91/clk-generated.c | 101 +- include/linux/clk/at91_pmc.h | 25 +- sound/soc/atmel/atmel-classd.c | 47 +- 10 files changed, 724 insertions(+), 59 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll.c base-commit: 047a0f692a9354d730dad30f25e1ebd8437b32b1 -- git-series 0.9.1

3 20

[alsa-devel] [PATCH v3 0/9] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 26 Jul '17

26 Jul '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop clk: at91: add audio plls to the compatible list in DT binding clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock .../devicetree/bindings/clock/at91-clock.txt | 10 + .../devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 ++ arch/arm/boot/dts/sama5d2.dtsi | 39 +++- arch/arm/mach-at91/Kconfig | 4 + drivers/clk/at91/Makefile | 2 + drivers/clk/at91/clk-audio-pll-pad.c | 206 ++++++++++++++++++ drivers/clk/at91/clk-audio-pll-pmc.c | 174 +++++++++++++++ drivers/clk/at91/clk-audio-pll.c | 239 +++++++++++++++++++++ drivers/clk/at91/clk-generated.c | 101 +++++++-- include/linux/clk/at91_pmc.h | 25 +++ sound/soc/atmel/atmel-classd.c | 47 ++-- 12 files changed, 813 insertions(+), 59 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll-pad.c create mode 100644 drivers/clk/at91/clk-audio-pll-pmc.c create mode 100644 drivers/clk/at91/clk-audio-pll.c -- 2.11.0

4 19

[alsa-devel] [PATCH] ALSA: hda - Add mute led support for HP ProBook 440 G4
by Kai-Heng Feng 26 Jul '17

26 Jul '17

Mic mute led does not work on HP ProBook 440 G4. We can use CXT_FIXUP_MUTE_LED_GPIO fixup to support it. BugLink: https://bugs.launchpad.net/bugs/1705586 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e8253737c47a..615a69a32970 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -931,6 +931,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.13.3

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 25 Jul '17

25 Jul '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 I don't own a A33 device which uses the i2s block for the audio codec so if someone could test against that it would be much appreciated. I'm also wondering if there is a preferred way of setting the lrclk size in the dts?..currently it is set to the sample width but for example the pcm5102a wants it to be 32 bits whatever the sample rate. Thanks in advance, CK v2 changes compared to v1 are: - massive refactoring to remove duplicate code making use of regmap_fields. - extending the clock divisors. - removed code that should be delivered when we support 20/24bits --- Marcus Cooper (2): ASoC: sun4i-i2s: Add more quirks for newer SoCs ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 469 ++++++++++++++++++--- 2 files changed, 407 insertions(+), 64 deletions(-) -- 2.13.3

3 6

[alsa-devel] [ANNOUNCE] Audio Mini Summit 2017 at Prague, Oct 27
by Takashi Iwai 25 Jul '17

25 Jul '17

Hi, this is an announcement of Audio Mini Summit 2017. It's an annual meeting we regularly have along with major Linux conferences, and this year, we'll hold the meeting at Prague (Czech Republic) on Friday, October 27, right after GStreamer Summit, ELCE and Open Source Summit Europe. We planned a whole-day meeting, and will likely have an evening dinner event, too. Please note that the meeting will be held in a conference room of SUSE Prague office, not in Hilton hotel where ELCE/OSSE is held. It's located near Hilton, just a couple of blocks away. The address of SUSE office is: Corso II, Křižíkova 148/34, 18600 Praha 8 https://goo.gl/maps/anET1yh3XXu Also, since the conference room is hosted by SUSE, it's an open event at this time; that is, you can attend only this meeting without registration for other Linux Foundation events in that week, too. If you are interested in the event, please fill up the registration form in Google doc spreadsheet below: https://docs.google.com/spreadsheets/d/1a-RaYtTVVtgh_cyRN5GwATiixppGCMEwqw6… Just add yourself in the attendee list from the web page. And, now we are gathering the topics to discuss, too. If you have any interesting topics to be discussed in the meeting, feel free to post to alsa-devel ML. I'm going to write up later. thanks, Takashi

2 1

Re: [alsa-devel] [PATCH] ASoC: rt5514: Use the local variable to get the data from SPI device
by Hsin-yu Chao 25 Jul '17

25 Jul '17

The original cause seems to be in spi driver. I tested setting enable_dma flag to false in spi-pxa2xx driver and that fixes the problem. Will investigate in spi driver side. Thanks! Hsin-yu On Mon, Jul 24, 2017 at 11:06 AM, Oder Chiou <oder_chiou(a)realtek.com> wrote: > > On Fri, Jul 21, 2017 at 04:00:47PM +0800, Oder Chiou wrote: > > > The patch uses the local variable to get the data from SPI device to > > > prevent to get all zero data. > > > > How do you end up with all zero data from reading directly into rxbuf? > > > While the "runtime->dma_area" put into the rxbuf directly on the Intel > platform, it seemed that had the limitation to let the memory cannot be > written the data using the driver of the SPI controller. We verified the > original code on the platforms of Nvidia and Rockchip, and these are > working fine. >

1 0

[alsa-devel] [PATCH 0/3] Motorola Droid 4 Audio Support
by Sebastian Reichel 25 Jul '17

25 Jul '17

Hi, I got working sound on Droid 4 with mainline \o/. The codec is currently missing support for detecting if something has been plugged into the 3.5mm connector, since that seems to require some closed source firmware and needs further investigation. I think this can be added later. Tested: - Playing via [HiFi] using Speaker, Earpiece, Headphone - Playing via [Voice] using Speaker, Earpiece, Headphone - Recording via [Voice] using Mic - Volume Controls - Checking that DAPM graph look correct -- Sebastian Sebastian Reichel (3): ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../bindings/sound/motorola,cpcap-audio-codec.txt | 19 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 6 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 76 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1422 ++++++++++++++++++++ 6 files changed, 1528 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.13.2

5 16

[alsa-devel] [PATCH] linux/kernel.h: move DIV_ROUND_DOWN_ULL() macro
by Masahiro Yamada 25 Jul '17

25 Jul '17

This macro is useful to avoid link error on 32-bit systems. We have the same definition in two drivers, so move it to include/linux/kernel.h While we are here, refactor DIV_ROUND_UP_ULL() by using DIV_ROUND_DOWN_ULL(). Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- drivers/mtd/nand/denali.c | 3 --- include/linux/kernel.h | 7 +++++-- sound/soc/codecs/pcm512x.c | 3 --- 3 files changed, 5 insertions(+), 8 deletions(-) diff --git a/drivers/mtd/nand/denali.c b/drivers/mtd/nand/denali.c index d723be352148..3087b0ba7b7f 100644 --- a/drivers/mtd/nand/denali.c +++ b/drivers/mtd/nand/denali.c @@ -980,9 +980,6 @@ static int denali_erase(struct mtd_info *mtd, int page) return irq_status & INTR__ERASE_COMP ? 0 : NAND_STATUS_FAIL; } -#define DIV_ROUND_DOWN_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) - static int denali_setup_data_interface(struct mtd_info *mtd, int chipnr, const struct nand_data_interface *conf) { diff --git a/include/linux/kernel.h b/include/linux/kernel.h index 96381f2b0e09..2a254ae6f7f7 100644 --- a/include/linux/kernel.h +++ b/include/linux/kernel.h @@ -78,8 +78,11 @@ #define FIELD_SIZEOF(t, f) (sizeof(((t*)0)->f)) #define DIV_ROUND_UP __KERNEL_DIV_ROUND_UP -#define DIV_ROUND_UP_ULL(ll,d) \ - ({ unsigned long long _tmp = (ll)+(d)-1; do_div(_tmp, d); _tmp; }) + +#define DIV_ROUND_DOWN_ULL(ll, d) \ + ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) + +#define DIV_ROUND_UP_ULL(ll, d) DIV_ROUND_DOWN_ULL((ll) + (d) - 1, (d)) #if BITS_PER_LONG == 32 # define DIV_ROUND_UP_SECTOR_T(ll,d) DIV_ROUND_UP_ULL(ll, d) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 72b19e62f626..ad3bfb894ee0 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -30,9 +30,6 @@ #include "pcm512x.h" -#define DIV_ROUND_DOWN_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) - #define PCM512x_NUM_SUPPLIES 3 static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { "AVDD", -- 2.7.4

1 0

[alsa-devel] I use manjaro but no device detected
by fulgor fulgor 22 Jul '17

22 Jul '17

hello, I have no sound The sound device is not detected I have logging informations http://www.alsa-project.org/db/?f=6fb68d0201f421b41fe65677fde2ec5b1c9d4429 thanks for all -- Lionel Van Winnendaele

1 0

[alsa-devel] [PATCH v2] sound: Convert to using %pOF instead of full_name
by Rob Herring 21 Jul '17

21 Jul '17

Now that we have a custom printf format specifier, convert users of full_name to use %pOF instead. This is preparation to remove storing of the full path string for each node. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org --- v2: - Add printf specifier change for fsl_dma.c sound/aoa/codecs/tas.c | 4 ++-- sound/soc/fsl/fsl_dma.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 16 ++++++++-------- 3 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 733b6365dad6..15c05755d270 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -905,8 +905,8 @@ static int tas_i2c_probe(struct i2c_client *client, goto fail; } printk(KERN_DEBUG - "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - (unsigned int)client->addr, node->full_name); + "snd-aoa-codec-tas: tas found, addr 0x%02x on %pOF\n", + (unsigned int)client->addr, node); return 0; fail: mutex_destroy(&tas->mtx); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ccadefceeff2..ed8ea002902d 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -897,8 +897,8 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) ret = of_address_to_resource(ssi_np, 0, &res); if (ret) { - dev_err(&pdev->dev, "could not determine resources for %s\n", - ssi_np->full_name); + dev_err(&pdev->dev, "could not determine resources for %pOF\n", + ssi_np); of_node_put(ssi_np); return ret; } diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da341d61..392d5eef356d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -268,13 +268,13 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, ret = of_property_read_u32(child, "fsl,audmux-port", &port); if (ret) { - dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n", - child->full_name); + dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%pOF\"\n", + child); continue; } if (!of_property_read_bool(child, "fsl,port-config")) { - dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n", - child->full_name); + dev_warn(&pdev->dev, "child node \"%pOF\" does not have property fsl,port-config\n", + child); continue; } @@ -292,15 +292,15 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, } if (ret != -EOVERFLOW) { - dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", - i, child->full_name); + dev_err(&pdev->dev, "Failed to read u32 at index %d of child %pOF\n", + i, child); continue; } if (audmux_type == IMX31_AUDMUX) { if (i % 2) { - dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n", - child->full_name); + dev_err(&pdev->dev, "One pdcr value is missing in child node %pOF\n", + child); continue; } imx_audmux_v2_configure_port(port, ptcr, pdcr); -- 2.11.0

2 1

[alsa-devel] [PATCH v2] sound: fsl_dma: remove dma_object path member
by Rob Herring 21 Jul '17

21 Jul '17

dma_object.path is unused, so rather than fix it to work with DT full_name changes, just remove it. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org --- v2: - Move printf specifier change to correct patch sound/soc/fsl/fsl_dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ed8ea002902d..b38dd328c783 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -63,7 +63,6 @@ struct dma_object { struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; - char path[1]; }; /* @@ -903,14 +902,13 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) return ret; } - dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + dma = kzalloc(sizeof(*dma), GFP_KERNEL); if (!dma) { dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } - strcpy(dma->path, np->full_name); dma->dai.ops = &fsl_dma_ops; dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Use snd_soc_kcontrol_codec()
by Fabio Estevam 21 Jul '17

21 Jul '17

Since commit a72952672005 ("ASoC: sgtl5000: add avc support") the following kernel crash happens after running a 'reboot' command: ALSA: Storing mixer settings... [ 20.031604] Unable to handle kernel paging request at virtual address fffffffe [ 20.039268] pgd = de2a0000 [ 20.041999] [fffffffe] *pgd=8fffd861, *pte=00000000, *ppte=00000000 [ 20.048387] Internal error: Oops: 80000007 [#1] SMP ARM The function that takes a kcontrol parameter and returns the codec that registered the control is snd_soc_kcontrol_codec(), so use the correct function to fix the problem. Fixes: a72952672005 ("ASoC: sgtl5000: add avc support") Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8f6814c..80f6d1d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -409,7 +409,7 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, static int avc_get_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db, i; u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); @@ -442,7 +442,7 @@ static int avc_get_threshold(struct snd_kcontrol *kcontrol, static int avc_put_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db; u16 reg; -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: samsung: odroid: Fix EPLL frequency values
by Sylwester Nawrocki 21 Jul '17

21 Jul '17

To prevent incorrect setting of the EPLL the clock frequency values are changed to exact values as possible to obtain on the EPLL output with given PLL coefficients. This patch is required after recent change of the EPLL rate table by patch "clk: samsung: exynos5420: The EPLL rate table corrections". Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- It would be good to apply this patch for 4.13-rc, I plan to get applied corresponding clk patch also to v4.13-rc. --- sound/soc/samsung/odroid.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 0c0b00e..0834319 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -42,17 +42,17 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 32000: case 64000: - pll_freq = 131072000U; + pll_freq = 131072006U; break; case 44100: case 88200: case 176400: - pll_freq = 180633600U; + pll_freq = 180633609U; break; case 48000: case 96000: case 192000: - pll_freq = 196608000U; + pll_freq = 196608001U; break; default: return -EINVAL; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Use the local variable to get the data from SPI device
by Oder Chiou 21 Jul '17

21 Jul '17

The patch uses the local variable to get the data from SPI device to prevent to get all zero data. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 950d1ff..e693204 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -289,7 +289,7 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) { u8 spi_cmd = RT5514_SPI_CMD_BURST_READ; int status; - u8 write_buf[8]; + u8 write_buf[8], read_buf[RT5514_SPI_BUF_LEN]; unsigned int i, end, offset = 0; struct spi_message message; @@ -319,7 +319,7 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) spi_message_add_tail(&x[1], &message); x[2].len = end; - x[2].rx_buf = rxbuf + offset; + x[2].rx_buf = read_buf; spi_message_add_tail(&x[2], &message); status = spi_sync(rt5514_spi, &message); @@ -327,6 +327,8 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) if (status) return false; + memcpy(rxbuf + offset, read_buf, end); + offset += RT5514_SPI_BUF_LEN; } @@ -365,14 +367,9 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) { u8 spi_cmd = RT5514_SPI_CMD_BURST_WRITE; - u8 *write_buf; + u8 write_buf[RT5514_SPI_BUF_LEN + 6]; unsigned int i, end, offset = 0; - write_buf = kmalloc(RT5514_SPI_BUF_LEN + 6, GFP_KERNEL); - - if (write_buf == NULL) - return -ENOMEM; - while (offset < len) { if (offset + RT5514_SPI_BUF_LEN <= len) end = RT5514_SPI_BUF_LEN; @@ -403,8 +400,6 @@ int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) offset += RT5514_SPI_BUF_LEN; } - kfree(write_buf); - return 0; } EXPORT_SYMBOL_GPL(rt5514_spi_burst_write); -- 2.7.4

2 1

[alsa-devel] Boss BR 80 Problem
by Benjamin 21 Jul '17

21 Jul '17

Hello Alsa-Developpers! Thank you for your excellent work. It's great how many devices have worked flawlessly over the years of me using Linux;) There seems to be a problem with the Boss Micro Br 80 that me and others are experiencing: https://linuxmusicians.com/viewtopic.php?f=6&t=17248 It's recognised but will only produce errors. It is reporting working with older Kernels, so maybe someone could look into that! Thank you, Ben

1 0

[alsa-devel] No Loopback for ALC225/ALC295
by Kailang 21 Jul '17

21 Jul '17

Hi Takashi, There are also no loopback mixer for ALC225 / ALC295 codecs. Attach is the update patch. Many Thanks. BR, Kailang

2 1

[alsa-devel] ALC225 headset mode update
by Kailang 21 Jul '17

21 Jul '17

Hi Takashi, Dell also create new layout for ALC225. Headset mode policy need to change to match new layout. Attach patch was the update. Many thanks. BR, Kailang

2 1

[alsa-devel] [Test Request] Component level pcm_new/pcm_free v2
by Kuninori Morimoto 21 Jul '17

21 Jul '17

Hi Mark, Johan, Brian These are v2 of Component level pcm_new/free patches. I think current ALSA SoC need to cleanup and move to new style. Then, current all CPU/Codec/Platform features needs to be merged into "component" (and remove CPU/Codec/Platform categorize). My previous pcm_new/free patch had wrong operation, thus Johan, Brian got damage from it. I'm so sorry about that. Current ALSA SoC has rtd->platform, but it should be removed in new style, I think. So, this patch adds new "component list" for rtd (= I named it as rtdcom, maybe we want to have more nice naming). This list includes all components (= CPU/Codec/Platform), not only platform component. I'm thinking that rtd->platform related operation will be replaced like below - if(rtd->platform->xxx) - rtd->platform->xxx(yyy); + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; /* rtd connected CPU/Codec/Platform */ + if (component->xxx) + component->xxx(yyy); + } Here, xxxx is pcm_new/free on this patch. If my understanding was correct, this v2 pcm_new/free works well for you, but I don't know. So, can you please test these patches ? These are based on mark/fix/multi-pcm Kuninori Morimoto (3): ASoC: soc-core: add snd_soc_rtdcom_xxx() ASoC: use snd_soc_rtdcom_add() and convert to consistent operation ASoC: add Component level pcm_new/pcm_free v2 Best regards --- Kuninori Morimoto

1 4

[alsa-devel] [PATCH v3 00/12] AC97 device/driver model revamp
by Robert Jarzmik 20 Jul '17

20 Jul '17

Hi Lars, Mark, Charles, Lee, This is a resend of the v2, with patch 1 added (forgotten in v2). Patch 12 was slightly modified after module unloading tests, and Dmitry acks were added. This serie has been stalled for almost one year, as I wasn't feeling that well lately. I'm quite better now, so let's get back to it. The following sentences is a what was already said in v2. As it's been so long since the last review, I basically withdrew all my former objections and took into account all remarks regardless of my own judgement. This is especially the case with Lars's review on the bus code, where I tried to address all his comments. For this next round, let's sum up what happened : - I rebased against latest Linus's tree => as Lars had made ports to regmap of several impacted wm97xx drivers, all the better for me. - ALSA: ac97: add an ac97 bus => this addresses Lars's comments => history : https://patchwork.kernel.org/patch/9398229/ => there is one remark which was left out, the drop of the table ac97_controller->codecs. The reason is that there are traversals required in the bus scan and controller removal, and the code is way simpler with a table, and the data structure very compact. - Input: wm97xx: split out touchscreen registering => Charles's comment taken into account => history: http://https://patchwork.kernel.org/patch/9398223/ Have a happy review. -- Robert Robert Jarzmik (12): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++++++----- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 282 +++++++++++++++++ include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 118 +++++++ include/sound/ac97/compat.h | 21 ++ include/sound/ac97/controller.h | 85 ++++++ include/sound/ac97/regs.h | 262 ++++++++++++++++ include/sound/ac97_codec.h | 239 +-------------- include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 10 + sound/ac97/bus.c | 526 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 105 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 37 ++- sound/arm/pxa2xx-ac97.c | 35 ++- sound/soc/Kconfig | 4 + sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 59 ++-- sound/soc/codecs/wm9712.c | 37 ++- sound/soc/codecs/wm9713.c | 39 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 46 ++- 30 files changed, 1866 insertions(+), 414 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.11.0

4 22

[alsa-devel] [PATCH v3] ASoC: sgtl5000: add avc support
by Richard Leitner 20 Jul '17

20 Jul '17

The sgtl5000 features a automatic volume control block (AVC), which reduces loud signals and amplifies low level signals for easier listening. This patch adds support for this AVC block to the driver. Apart from the "AVC Switch" control which enables the block following controls for the configuration of AVC are added: + AVC Threshold Volume: threshold where audio is compressed when the measured level is above or expanded when below + AVC Max Gain Volume: maximum gain which can be applied when the measured audio level is below threshold + AVC Hard Limiter Switch: when enabled the signal is limited to the programmed threshold. + AVC Integrator Response: response time of the integrator The AVC block is enabled and configured using the DAP_AVC_CTRL and DAP_AVC_THRESHOLD registers. Following 2 checkpatch.pl strict checks are ignored because the indentation style is different for the struct snd_kcontrol_new definition: patch:145: CHECK: Alignment should match open parenthesis patch:148: CHECK: Alignment should match open parenthesis Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- CHANGES v3: - rename "AVC Enable Switch" to "AVC Switch" - add AVC configuration controls CHANGES v2: - added trailing "Switch" as covered in ControlNames.txt --- sound/soc/codecs/sgtl5000.c | 87 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 87 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a2702e..5d4b69f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; +/* AVC: Threshold dB -> register: pre-calculated values */ +static const u16 avc_thr_db2reg[97] = { + 0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068, + 0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F, + 0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414, + 0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172, + 0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083, + 0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E, + 0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010, + 0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005, + 0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002, + 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000}; + /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -382,6 +396,63 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, return 0; } +/* + * custom function to get AVC threshold + * + * The threshold dB is calculated by rearranging the calculation from the + * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> + * dB = ( fls(register_value) - 14.347 ) * 6.02 + * + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_get_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db, i; + u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); + + /* register value 0 => -96dB */ + if (!reg) { + ucontrol->value.integer.value[0] = 96; + ucontrol->value.integer.value[1] = 96; + return 0; + } + + /* get dB from register value (rounded down) */ + for (i = 0; avc_thr_db2reg[i] > reg; i++) + ; + db = i; + + ucontrol->value.integer.value[0] = db; + ucontrol->value.integer.value[1] = db; + + return 0; +} + +/* + * custom function to put AVC threshold + * + * The register value is calculated by following formula: + * register_value = 10^(dB/20) * 0.636 * 2^15 + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_put_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db; + u16 reg; + + db = clamp((int)ucontrol->value.integer.value[0], 0, 96); + reg = avc_thr_db2reg[db]; + snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg); + + return 0; +} + static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ @@ -396,6 +467,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); /* tlv for lineout volume, 31 steps of .5db each */ static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0); +/* tlv for dap avc max gain, 0db, 6db, 12db */ +static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0); + +/* tlv for dap avc threshold, */ +static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600); + static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { /* SOC_DOUBLE_S8_TLV with invert */ { @@ -434,6 +511,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0x1f, 1, lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + + /* Automatic Volume Control (DAP AVC) */ + SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), + SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), + SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0, + avc_max_gain), + SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0), + SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, + 0, 96, 0, avc_get_threshold, avc_put_threshold, + avc_threshold), }; /* mute the codec used by alsa core */ -- 2.1.4

3 6

[alsa-devel] [PATCH v2 0/3]ASoC: Intel: Skylake: Enable hdmi for GLK
by Pradeep Tewani 20 Jul '17

20 Jul '17

GLK vendor nid is different from rest of the skylake platforms. Add vendor nid as part of driver data and use it. Also enable support for GLK platform. Add the patch to remove the .owner initialization in bxt driver as well Changes in v2: Removed the .owner field in driver Replaced the strcmp with strncmp for bxt_rt298 probe Pradeep Tewani (3): ASoC: Intel: board: Remove .owner initialization in bxt_rt298 driver ASoC: hdac_hdmi: Add the vendor nid for Geminilake HDMI ASoC: Intel: board: Add Geminilake platform support sound/soc/codecs/hdac_hdmi.c | 39 ++++++++++++++--- sound/soc/intel/boards/bxt_rt298.c | 85 ++++++++++++++++++++++++++++++++++++-- 2 files changed, 115 insertions(+), 9 deletions(-) -- 2.7.4

3 6

[alsa-devel] [PATCH 0/2] ASoC: Intel: Skylake: Enable hdmi for GLK
by Pradeep Tewani 20 Jul '17

20 Jul '17

GLK vendor nid is different from rest of the skylake platforms. Add vendor nid as part of driver data and use it. Also enable support for GLK platform. Pradeep Tewani (2): ASoC: Intel: board: Add Geminilake platform support ASoC: hdac_hdmi: Add the vendor nid for Geminilake HDMI sound/soc/codecs/hdac_hdmi.c | 39 +++++++++++++++--- sound/soc/intel/boards/bxt_rt298.c | 82 ++++++++++++++++++++++++++++++++++++-- 2 files changed, 113 insertions(+), 8 deletions(-) -- 2.7.4

3 4

[alsa-devel] [PATCH v2 0/3] ASoC: rt5677: Few clean ups to the driver
by Andy Shevchenko 20 Jul '17

20 Jul '17

This series removes legacy and unused platform data along with fixing code which compiler complains about. In v2: - fix compilation error (kbuild bot) - add couple more patches to fix the code according to compile complains Andy Shevchenko (3): ASoC: rt5677: Hide platform data in the module sources ASoC: rt5677: Refactor code to avoid comparison unsigned >= 0 ASoC: rt5677: Remove never used variables include/sound/rt5677.h | 45 -------------------------- sound/soc/codecs/rt5677.c | 81 ++++++++++++++++++++++------------------------- sound/soc/codecs/rt5677.h | 30 +++++++++++++++++- 3 files changed, 67 insertions(+), 89 deletions(-) delete mode 100644 include/sound/rt5677.h -- 2.11.0

3 9

[alsa-devel] [PATCH] ASoC: rt5514: Support the TDM docking mode
by Oder Chiou 20 Jul '17

20 Jul '17

The TDM ducking mode can share the bus to other devices simultaneously. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 33 ++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5514.h | 12 ++++++++++++ 2 files changed, 44 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 4ca73bf..a105c84 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -871,11 +871,38 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, { struct snd_soc_codec *codec = dai->codec; struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); - unsigned int val = 0; + unsigned int val = 0, val2 = 0; if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; + switch (tx_mask) { + case 0x3: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0x30: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + case 0xf: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0xf0: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + default: + break; + } + + + switch (slots) { case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; @@ -921,6 +948,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | RT5514_TDM_MODE2, val); + regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL2, + RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH_MASK | + RT5514_TDM_DOCKING_START_MASK, val2); + return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212..75024a4 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -164,6 +164,18 @@ #define RT5514_I2S_DL_24 (0x2 << 0) #define RT5514_I2S_DL_8 (0x3 << 0) +/* RT5514_I2S_CTRL2 (0x2014) */ +#define RT5514_TDM_DOCKING_MODE (0x1 << 31) +#define RT5514_TDM_DOCKING_MODE_SFT 31 +#define RT5514_TDM_DOCKING_VALID_CH_MASK (0x1 << 29) +#define RT5514_TDM_DOCKING_VALID_CH_SFT 29 +#define RT5514_TDM_DOCKING_VALID_CH2 (0x0 << 29) +#define RT5514_TDM_DOCKING_VALID_CH4 (0x1 << 29) +#define RT5514_TDM_DOCKING_START_MASK (0x1 << 28) +#define RT5514_TDM_DOCKING_START_SFT 28 +#define RT5514_TDM_DOCKING_START_SLOT0 (0x0 << 28) +#define RT5514_TDM_DOCKING_START_SLOT4 (0x1 << 28) + /* RT5514_DIG_SOURCE_CTRL (0x20a4) */ #define RT5514_AD1_DMIC_INPUT_SEL (0x1 << 1) #define RT5514_AD1_DMIC_INPUT_SEL_SFT 1 -- 2.7.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5665: fix GPIO6 pin function define
by Bard Liao 20 Jul '17

20 Jul '17

The GPIO6 pin function select value was wrong. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index 1db5c6a..d95249c 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1692,8 +1692,8 @@ #define RT5665_GP6_PIN_MASK (0x3 << 5) #define RT5665_GP6_PIN_SFT 5 #define RT5665_GP6_PIN_GPIO6 (0x0 << 5) -#define RT5665_GP6_PIN_BCLK3 (0x0 << 5) -#define RT5665_GP6_PIN_PDM_SCL (0x1 << 5) +#define RT5665_GP6_PIN_BCLK3 (0x1 << 5) +#define RT5665_GP6_PIN_PDM_SCL (0x2 << 5) #define RT5665_GP7_PIN_MASK (0x3 << 3) #define RT5665_GP7_PIN_SFT 3 #define RT5665_GP7_PIN_GPIO7 (0x0 << 3) -- 2.7.4

2 5

[alsa-devel] [PATCH v2] ASoC: rt5514: Support the TDM docking mode
by Oder Chiou 20 Jul '17

20 Jul '17

The TDM docking mode can share the bus to other devices simultaneously. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 33 ++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5514.h | 12 ++++++++++++ 2 files changed, 44 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 4ca73bf..a105c84 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -871,11 +871,38 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, { struct snd_soc_codec *codec = dai->codec; struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); - unsigned int val = 0; + unsigned int val = 0, val2 = 0; if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; + switch (tx_mask) { + case 0x3: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0x30: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + case 0xf: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0xf0: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + default: + break; + } + + + switch (slots) { case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; @@ -921,6 +948,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | RT5514_TDM_MODE2, val); + regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL2, + RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH_MASK | + RT5514_TDM_DOCKING_START_MASK, val2); + return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212..75024a4 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -164,6 +164,18 @@ #define RT5514_I2S_DL_24 (0x2 << 0) #define RT5514_I2S_DL_8 (0x3 << 0) +/* RT5514_I2S_CTRL2 (0x2014) */ +#define RT5514_TDM_DOCKING_MODE (0x1 << 31) +#define RT5514_TDM_DOCKING_MODE_SFT 31 +#define RT5514_TDM_DOCKING_VALID_CH_MASK (0x1 << 29) +#define RT5514_TDM_DOCKING_VALID_CH_SFT 29 +#define RT5514_TDM_DOCKING_VALID_CH2 (0x0 << 29) +#define RT5514_TDM_DOCKING_VALID_CH4 (0x1 << 29) +#define RT5514_TDM_DOCKING_START_MASK (0x1 << 28) +#define RT5514_TDM_DOCKING_START_SFT 28 +#define RT5514_TDM_DOCKING_START_SLOT0 (0x0 << 28) +#define RT5514_TDM_DOCKING_START_SLOT4 (0x1 << 28) + /* RT5514_DIG_SOURCE_CTRL (0x20a4) */ #define RT5514_AD1_DMIC_INPUT_SEL (0x1 << 1) #define RT5514_AD1_DMIC_INPUT_SEL_SFT 1 -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: rt274: correct comment style
by Bard Liao 19 Jul '17

19 Jul '17

There was a comment style issue in the driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt274.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 81e29ba..fb683ff 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1159,7 +1159,7 @@ static int rt274_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt274->regmap, 0x6f, 0x3000, 0x2000); /* HP DC Calibration */ regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0x0); - //Set NID=58h.Index 00h [15]= 1b; + /* Set NID=58h.Index 00h [15]= 1b; */ regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888); msleep(500); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Correct the mixer switch setting and remove redundant routing path
by Oder Chiou 19 Jul '17

19 Jul '17

The patch corrects the mixer siwtch setting that was the mistake in the previous commitment. And remove the redundant routing paths. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index d324817..38c4027 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -59,7 +59,7 @@ struct rt5663_priv { }; static const struct reg_sequence rt5663_patch_list[] = { - { 0x002a, 0x8080 }, + { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, }; @@ -482,7 +482,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0023, 0x0039 }, { 0x0026, 0xc0c0 }, { 0x0029, 0x8080 }, - { 0x002a, 0x8080 }, + { 0x002a, 0x8020 }, { 0x002c, 0x000c }, { 0x002d, 0x0000 }, { 0x0040, 0x0808 }, @@ -1967,8 +1967,8 @@ static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = { }; static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = { - SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, - RT5663_M_DAC_L1_STO_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, + RT5663_M_DAC_R1_STO_R_SHIFT, 1, 1), }; /* Out Switch */ @@ -2358,11 +2358,9 @@ static const struct snd_soc_dapm_route rt5663_dapm_routes[] = { { "DAC R1", NULL, "ADDA MIXR" }, { "STO1 DAC MIXL", "DAC L Switch", "DAC L1" }, - { "STO1 DAC MIXL", "DAC R Switch", "DAC R1" }, { "STO1 DAC MIXL", NULL, "STO1 DAC L Power" }, { "STO1 DAC MIXL", NULL, "STO1 DAC Filter" }, { "STO1 DAC MIXR", "DAC R Switch", "DAC R1" }, - { "STO1 DAC MIXR", "DAC L Switch", "DAC L1" }, { "STO1 DAC MIXR", NULL, "STO1 DAC R Power" }, { "STO1 DAC MIXR", NULL, "STO1 DAC Filter" }, -- 2.7.4

2 1

Re: [alsa-devel] [PATCH] ASoC: rt274: add rt274 codec driver
by Mark Brown 19 Jul '17

19 Jul '17

On Wed, Jul 19, 2017 at 02:08:58AM +0000, Bard Liao wrote: > I have sent a patch for it. But I was wondering why checkpatch.pl can't > find the style issue? No idea, sorry.

1 0

[alsa-devel] Patch for Digidesign Mbox 2 (set sample rate)
by Viktor Radnai 19 Jul '17

19 Jul '17

Hi there, I have recently bought a Digidesign Mbox 2 Mini and found the lack of support for 44100Hz sample rate in the driver limiting (the sample rate was hard-set to 48000Hz by the original author even though the device supports it, because he couldn't get the frame rate switching to work reliably). I had a go at hacking the driver and came up with the following patch which appears to work on the MBox 2 Mini: https://pastebin.com/kzzTDj9T I have built and tested this with Linux 4.8.15 without any realtime patches. I am planning to pick up a "full size" Mbox 2 as well, in the meantime if anyone on the list owns one I would appreciate if you could test this and let me know the result. There is a possibility that it will not work without the "magic" init sequence that I have removed because it does not look like it's needed on the Mini. Although I would be a bit surprised because the two devices are apparently identical to the point that the Mini allows the use of it's physically nonexistent features. Once someone could confirm that this works, I would like to get it merged. This is my first attempt at submitting a patch for any Linux device driver so please let me know if I need to do anything else :) Cheers, Vik -- My other sig is hilarious

3 6

[alsa-devel] Alsa library plugin license
by Jean-Francois Dockes 19 Jul '17

19 Jul '17

Hello, I was requested to study the possibility to implement an alsa library pcm plugin doing specific audio processing. I had a look around, but I could not find an explicit answer to this question: is a closed source alsa library plugin possible ? My guess would be that it has to be GPL'd, but I have to ask. Cheers, jf

2 1

[alsa-devel] [PATCH] sound: fsl_dma: remove dma_object path member
by Rob Herring 19 Jul '17

19 Jul '17

dma_object.path is unused, so rather than fix it to work with DT full_name changes, just remove it. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org --- sound/soc/fsl/fsl_dma.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ccadefceeff2..b38dd328c783 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -63,7 +63,6 @@ struct dma_object { struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; - char path[1]; }; /* @@ -897,20 +896,19 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) ret = of_address_to_resource(ssi_np, 0, &res); if (ret) { - dev_err(&pdev->dev, "could not determine resources for %s\n", - ssi_np->full_name); + dev_err(&pdev->dev, "could not determine resources for %pOF\n", + ssi_np); of_node_put(ssi_np); return ret; } - dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + dma = kzalloc(sizeof(*dma), GFP_KERNEL); if (!dma) { dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } - strcpy(dma->path, np->full_name); dma->dai.ops = &fsl_dma_ops; dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; -- 2.11.0

2 1

[alsa-devel] [PATCH] sound: Convert to using %pOF instead of full_name
by Rob Herring 19 Jul '17

19 Jul '17

Now that we have a custom printf format specifier, convert users of full_name to use %pOF instead. This is preparation to remove storing of the full path string for each node. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org --- sound/aoa/codecs/tas.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 16 ++++++++-------- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 733b6365dad6..15c05755d270 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -905,8 +905,8 @@ static int tas_i2c_probe(struct i2c_client *client, goto fail; } printk(KERN_DEBUG - "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - (unsigned int)client->addr, node->full_name); + "snd-aoa-codec-tas: tas found, addr 0x%02x on %pOF\n", + (unsigned int)client->addr, node); return 0; fail: mutex_destroy(&tas->mtx); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da341d61..392d5eef356d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -268,13 +268,13 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, ret = of_property_read_u32(child, "fsl,audmux-port", &port); if (ret) { - dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n", - child->full_name); + dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%pOF\"\n", + child); continue; } if (!of_property_read_bool(child, "fsl,port-config")) { - dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n", - child->full_name); + dev_warn(&pdev->dev, "child node \"%pOF\" does not have property fsl,port-config\n", + child); continue; } @@ -292,15 +292,15 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, } if (ret != -EOVERFLOW) { - dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", - i, child->full_name); + dev_err(&pdev->dev, "Failed to read u32 at index %d of child %pOF\n", + i, child); continue; } if (audmux_type == IMX31_AUDMUX) { if (i % 2) { - dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n", - child->full_name); + dev_err(&pdev->dev, "One pdcr value is missing in child node %pOF\n", + child); continue; } imx_audmux_v2_configure_port(port, ptcr, pdcr); -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: add AVC configuration controls
by Richard Leitner 18 Jul '17

18 Jul '17

For the configuration of the automatic volume control block (AVC), which reduces loud signals and amplifies low level signals for easier listening, following controls are added: + AVC Threshold Volume: threshold where audio is compressed when the measured level is above or expanded when below + AVC Max Gain Volume: maximum gain which can be applied when the measured audio level is below threshold + AVC Hard Limiter Switch: when enabled the signal is limited to the programmed threshold. + AVC Integrator Response These controls use the DAP_AVC_CTRL and DAP_AVC_THRESHOLD registers for configuration. This patch depends on "ASoC: sgtl5000: add avc enable control" Following 2 checkpatch.pl strict checks are ignored because the indentation style is different for the struct snd_kcontrol_new definition: patch:144: CHECK: Alignment should match open parenthesis patch:147: CHECK: Alignment should match open parenthesis Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- sound/soc/codecs/sgtl5000.c | 85 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0477f2e..2bb3235 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; +/* AVC: Threshold dB -> register: pre-calculated values */ +static const u16 avc_thr_db2reg[97] = { + 0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068, + 0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F, + 0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414, + 0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172, + 0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083, + 0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E, + 0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010, + 0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005, + 0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002, + 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000}; + /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -382,6 +396,63 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, return 0; } +/* + * custom function to get AVC threshold + * + * The threshold dB is calculated by rearranging the calculation from the + * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> + * dB = ( fls(register_value) - 14.347 ) * 6.02 + * + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_get_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db, i; + u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); + + /* register value 0 => -96dB */ + if (!reg) { + ucontrol->value.integer.value[0] = 96; + ucontrol->value.integer.value[1] = 96; + return 0; + } + + /* get dB from register value (rounded down) */ + for (i = 0; avc_thr_db2reg[i] > reg; i++) + ; + db = i; + + ucontrol->value.integer.value[0] = db; + ucontrol->value.integer.value[1] = db; + + return 0; +} + +/* + * custom function to put AVC threshold + * + * The register value is calculated by following formula: + * register_value = 10^(dB/20) * 0.636 * 2^15 + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_put_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db; + u16 reg; + + db = clamp((int)ucontrol->value.integer.value[0], 0, 96); + reg = avc_thr_db2reg[db]; + snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg); + + return 0; +} + static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ @@ -396,6 +467,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); /* tlv for lineout volume, 31 steps of .5db each */ static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0); +/* tlv for dap avc max gain, 0db, 6db, 12db */ +static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0); + +/* tlv for dap avc threshold, */ +static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600); + static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { /* SOC_DOUBLE_S8_TLV with invert */ { @@ -435,7 +512,15 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + /* Automatic Volume Control (DAP AVC) */ SOC_SINGLE("AVC Enable Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), + SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), + SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0, + avc_max_gain), + SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0), + SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, + 0, 96, 0, avc_get_threshold, avc_put_threshold, + avc_threshold), }; /* mute the codec used by alsa core */ -- 2.1.4

3 2

[alsa-devel] [PATCH] ALSA: hda: constify pci_device_id.
by Arvind Yadav 18 Jul '17

18 Jul '17

pci_device_id are not supposed to change at runtime. All functions working with pci_device_id provided by <linux/pci.h> work with const pci_device_id. So mark the non-const structs as const. File size before: text data bss dec hex filename 2610 304 8 2922 b6a sound/hda/hdac_i915.o File size After adding 'const': text data bss dec hex filename 2674 240 8 2922 b6a sound/hda/hdac_i915.o Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/hda/hdac_i915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 0659bf3..038a180 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -336,7 +336,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); /* check whether intel graphics is present */ static bool i915_gfx_present(void) { - static struct pci_device_id ids[] = { + static const struct pci_device_id ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_BASE_CLASS_DISPLAY << 16, .class_mask = 0xff << 16 }, -- 2.7.4

2 1

[alsa-devel] [PATCH v2 1/7] ALSA: als100: fix format string overflow warning
by Arnd Bergmann 18 Jul '17

18 Jul '17

The compiler sees that the format string might overflow for the longname: sound/isa/als100.c: In function 'snd_als100_pnp_detect': sound/isa/als100.c:225:27: error: ', dma ' directive writing 6 bytes into a region of size between 0 and 64 [-Werror=format-overflow=] sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/isa/als100.c:225:3: note: 'sprintf' output between 24 and 113 bytes into a destination of size 80 sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", Open-coding "shortname" here gets us below the limit, and using snprintf() is a good idea too. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/isa/als100.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/isa/als100.c b/sound/isa/als100.c index bc9ea306ee02..6a2c5b48f3ae 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -222,15 +222,16 @@ static int snd_card_als100_probe(int dev, if (pid->driver_data == SB_HW_DT019X) { strcpy(card->driver, "DT-019X"); strcpy(card->shortname, "Diamond Tech. DT-019X"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev]); + snprintf(card->longname, sizeof(card->longname), + "Diamond Tech. DT-019X, %s at 0x%lx, irq %d, dma %d", + chip->name, chip->port, irq[dev], dma8[dev]); } else { strcpy(card->driver, "ALS100"); strcpy(card->shortname, "Avance Logic ALS100"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev], dma16[dev]); + snprintf(card->longname, sizeof(card->longname), + "Avance Logic ALS100, %s at 0x%lx, irq %d, dma %d&%d", + chip->name, chip->port, irq[dev], dma8[dev], + dma16[dev]); } if ((error = snd_sb16dsp_pcm(chip, 0)) < 0) { -- 2.9.0

2 7

[alsa-devel] [PATCH] ASoC: rt5663: Modify the default value for ASRC function
by Oder Chiou 18 Jul '17

18 Jul '17

The patch modifies the default value for ASRC function. It could prevent the pop noise and recording no sound with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ac6dd25..c46d47f 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -60,6 +60,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8080 }, + { 0x0086, 0x0028 }, }; static const struct reg_default rt5663_v2_reg[] = { @@ -509,7 +510,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0082, 0x0000 }, { 0x0083, 0x0000 }, { 0x0084, 0x0000 }, - { 0x0086, 0x0008 }, + { 0x0086, 0x0028 }, { 0x0087, 0x0000 }, { 0x008a, 0x0000 }, { 0x008b, 0x0000 }, -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: rt274: add rt274 codec driver
by Bard Liao 18 Jul '17

18 Jul '17

RT274 is a HD-A/SOC dual mode codec. This is the initial codec driver of SOC mode. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt274.txt | 33 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt274.c | 1228 +++++++++++++++++++++ sound/soc/codecs/rt274.h | 217 ++++ 5 files changed, 1487 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt274.txt create mode 100644 sound/soc/codecs/rt274.c create mode 100644 sound/soc/codecs/rt274.h diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt new file mode 100644 index 0000000..e9a6178 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt274.txt @@ -0,0 +1,33 @@ +RT274 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt274". + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + + +Pins on the device (for linking into audio routes) for RT274: + + * DMIC1 Pin + * DMIC2 Pin + * MIC + * LINE1 + * LINE2 + * HPO Pin + * SPDIF + * LINE3 + +Example: + +codec: rt274@1c { + compatible = "realtek,rt274"; + reg = <0x1c>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6c78b0b..024ddc9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -114,6 +114,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C select SND_SOC_RT5514 if I2C @@ -716,11 +717,17 @@ config SND_SOC_RL6231 config SND_SOC_RL6347A tristate + default y if SND_SOC_RT274=y default y if SND_SOC_RT286=y default y if SND_SOC_RT298=y + default m if SND_SOC_RT274=m default m if SND_SOC_RT286=m default m if SND_SOC_RT298=m +config SND_SOC_RT274 + tristate + depends on I2C + config SND_SOC_RT286 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1755a54..a8a4b07 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -113,6 +113,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o +snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o @@ -349,6 +350,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o +obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c new file mode 100644 index 0000000..81e29ba --- /dev/null +++ b/sound/soc/codecs/rt274.c @@ -0,0 +1,1228 @@ +/* + * rt274.c -- RT274 ALSA SoC audio codec driver + * + * Copyright 2017 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/dmi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> + +#include "rl6347a.h" +#include "rt274.h" + +#define RT274_VENDOR_ID 0x10ec0274 + +struct rt274_priv { + struct reg_default *index_cache; + int index_cache_size; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + int clk_id; + int fs; + bool master; +}; + +static const struct reg_default rt274_index_def[] = { + { 0x00, 0x1004 }, + { 0x01, 0xaaaa }, + { 0x02, 0x88aa }, + { 0x03, 0x0002 }, + { 0x04, 0xaa09 }, + { 0x05, 0x0700 }, + { 0x06, 0x6110 }, + { 0x07, 0x0200 }, + { 0x08, 0xa807 }, + { 0x09, 0x0021 }, + { 0x0a, 0x7770 }, + { 0x0b, 0x7770 }, + { 0x0c, 0x002b }, + { 0x0d, 0x2420 }, + { 0x0e, 0x65c0 }, + { 0x0f, 0x7770 }, + { 0x10, 0x0420 }, + { 0x11, 0x7418 }, + { 0x12, 0x6bd0 }, + { 0x13, 0x645f }, + { 0x14, 0x0400 }, + { 0x15, 0x8ccc }, + { 0x16, 0x4c50 }, + { 0x17, 0xff00 }, + { 0x18, 0x0003 }, + { 0x19, 0x2c11 }, + { 0x1a, 0x830b }, + { 0x1b, 0x4e4b }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + { 0x1e, 0x0000 }, + { 0x1f, 0x0000 }, + { 0x20, 0x51ff }, + { 0x21, 0x8000 }, + { 0x22, 0x8f00 }, + { 0x23, 0x88f4 }, + { 0x24, 0x0000 }, + { 0x25, 0x0000 }, + { 0x26, 0x0000 }, + { 0x27, 0x0000 }, + { 0x28, 0x0000 }, + { 0x29, 0x3000 }, + { 0x2a, 0x0000 }, + { 0x2b, 0x0000 }, + { 0x2c, 0x0f00 }, + { 0x2d, 0x100f }, + { 0x2e, 0x2902 }, + { 0x2f, 0xe280 }, + { 0x30, 0x1000 }, + { 0x31, 0x8400 }, + { 0x32, 0x5aaa }, + { 0x33, 0x8420 }, + { 0x34, 0xa20c }, + { 0x35, 0x096a }, + { 0x36, 0x5757 }, + { 0x37, 0xfe05 }, + { 0x38, 0x4901 }, + { 0x39, 0x110a }, + { 0x3a, 0x0010 }, + { 0x3b, 0x60d9 }, + { 0x3c, 0xf214 }, + { 0x3d, 0xc2ba }, + { 0x3e, 0xa928 }, + { 0x3f, 0x0000 }, + { 0x40, 0x9800 }, + { 0x41, 0x0000 }, + { 0x42, 0x2000 }, + { 0x43, 0x3d90 }, + { 0x44, 0x4900 }, + { 0x45, 0x5289 }, + { 0x46, 0x0004 }, + { 0x47, 0xa47a }, + { 0x48, 0xd049 }, + { 0x49, 0x0049 }, + { 0x4a, 0xa83b }, + { 0x4b, 0x0777 }, + { 0x4c, 0x065c }, + { 0x4d, 0x7fff }, + { 0x4e, 0x7fff }, + { 0x4f, 0x0000 }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0xbf5f }, + { 0x53, 0x3320 }, + { 0x54, 0xcc00 }, + { 0x55, 0x0000 }, + { 0x56, 0x3f00 }, + { 0x57, 0x0000 }, + { 0x58, 0x0000 }, + { 0x59, 0x0000 }, + { 0x5a, 0x1300 }, + { 0x5b, 0x005f }, + { 0x5c, 0x0000 }, + { 0x5d, 0x1001 }, + { 0x5e, 0x1000 }, + { 0x5f, 0x0000 }, + { 0x60, 0x5554 }, + { 0x61, 0xffc0 }, + { 0x62, 0xa000 }, + { 0x63, 0xd010 }, + { 0x64, 0x0000 }, + { 0x65, 0x3fb1 }, + { 0x66, 0x1881 }, + { 0x67, 0xc810 }, + { 0x68, 0x2000 }, + { 0x69, 0xfff0 }, + { 0x6a, 0x0300 }, + { 0x6b, 0x5060 }, + { 0x6c, 0x0000 }, + { 0x6d, 0x0000 }, + { 0x6e, 0x0c25 }, + { 0x6f, 0x0c0b }, + { 0x70, 0x8000 }, + { 0x71, 0x4008 }, + { 0x72, 0x0000 }, + { 0x73, 0x0800 }, + { 0x74, 0xa28f }, + { 0x75, 0xa050 }, + { 0x76, 0x7fe8 }, + { 0x77, 0xdb8c }, + { 0x78, 0x0000 }, + { 0x79, 0x0000 }, + { 0x7a, 0x2a96 }, + { 0x7b, 0x800f }, + { 0x7c, 0x0200 }, + { 0x7d, 0x1600 }, + { 0x7e, 0x0000 }, + { 0x7f, 0x0000 }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt274_index_def) + +static const struct reg_default rt274_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x00000057 }, + { 0x0023a000, 0x00000057 }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x00000097 }, + { 0x00936000, 0x00000097 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000400 }, + { 0x00b37200, 0x00000400 }, + { 0x00b37300, 0x00000400 }, + { 0x00c37000, 0x00000400 }, + { 0x00c37100, 0x00000400 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01937000, 0x00000000 }, + { 0x01970500, 0x00000400 }, + { 0x02050000, 0x0000001b }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000001 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x01970700, 0x00000020 }, + { 0x00830000, 0x00000097 }, + { 0x00930000, 0x00000097 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt274_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT274_GET_HP_SENSE: + case RT274_GET_MIC_SENSE: + case RT274_PROC_COEF: + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0): + return true; + default: + return false; + } + + +} + +static bool rt274_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT274_GET_HP_SENSE: + case RT274_GET_MIC_SENSE: + case RT274_SET_AUDIO_POWER: + case RT274_SET_HPO_POWER: + case RT274_SET_DMIC1_POWER: + case RT274_LOUT_MUX: + case RT274_HPO_MUX: + case RT274_ADC0_MUX: + case RT274_ADC1_MUX: + case RT274_SET_MIC: + case RT274_SET_PIN_HPO: + case RT274_SET_PIN_LOUT3: + case RT274_SET_PIN_DMIC1: + case RT274_SET_AMP_GAIN_HPO: + case RT274_SET_DMIC2_DEFAULT: + case RT274_DAC0L_GAIN: + case RT274_DAC0R_GAIN: + case RT274_DAC1L_GAIN: + case RT274_DAC1R_GAIN: + case RT274_ADCL_GAIN: + case RT274_ADCR_GAIN: + case RT274_MIC_GAIN: + case RT274_HPOL_GAIN: + case RT274_HPOR_GAIN: + case RT274_LOUTL_GAIN: + case RT274_LOUTR_GAIN: + case RT274_DAC_FORMAT: + case RT274_ADC_FORMAT: + case RT274_COEF_INDEX: + case RT274_PROC_COEF: + case RT274_SET_AMP_GAIN_ADC_IN1: + case RT274_SET_AMP_GAIN_ADC_IN2: + case RT274_SET_POWER(RT274_DAC_OUT0): + case RT274_SET_POWER(RT274_DAC_OUT1): + case RT274_SET_POWER(RT274_ADC_IN1): + case RT274_SET_POWER(RT274_ADC_IN2): + case RT274_SET_POWER(RT274_DMIC2): + case RT274_SET_POWER(RT274_MIC): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0): + return true; + default: + return false; + } +} + +#ifdef CONFIG_PM +static void rt274_index_sync(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt274->index_cache[i].reg, + rt274->index_cache[i].def); + } +} +#endif + +static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) +{ + unsigned int buf; + + *hp = false; + *mic = false; + + if (!rt274->codec) + return -EINVAL; + + regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + *mic = buf & 0x80000000; + + pr_debug("*hp = %d *mic = %d\n", *hp, *mic); + + return 0; +} + +static void rt274_jack_detect_work(struct work_struct *work) +{ + struct rt274_priv *rt274 = + container_of(work, struct rt274_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + if (rt274_jack_detect(rt274, &hp, &mic) < 0) + return; + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt274->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +static irqreturn_t rt274_irq(int irq, void *data); + +static int rt274_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, void *data) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + if (jack == NULL) { + /* Disable jack detection */ + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_EN, RT274_IRQ_DIS); + + return 0; + } + rt274->jack = jack; + + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_EN, RT274_IRQ_EN); + + /* Send an initial report */ + rt274_irq(0, rt274); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt274_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT274_DAC0L_GAIN, + RT274_DAC0R_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", RT274_DAC1L_GAIN, + RT274_DAC1R_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT274_ADCL_GAIN, + RT274_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT274_ADCL_GAIN, + RT274_ADCR_GAIN, RT274_MUTE_SFT, 1, 1), + SOC_SINGLE_TLV("AMIC Volume", RT274_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), +}; + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOL_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOR_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new loutl_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTL_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new loutr_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTR_GAIN, + RT274_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt274_adc_src[] = { + "Mic", "Line1", "Line2", "Dmic" +}; + +static SOC_ENUM_SINGLE_DECL( + rt274_adc0_enum, RT274_ADC0_MUX, RT274_ADC_SEL_SFT, + rt274_adc_src); + +static const struct snd_kcontrol_new rt274_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt274_adc0_enum); + +static SOC_ENUM_SINGLE_DECL( + rt274_adc1_enum, RT274_ADC1_MUX, RT274_ADC_SEL_SFT, + rt274_adc_src); + +static const struct snd_kcontrol_new rt274_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt274_adc1_enum); + +static const char * const rt274_dac_src[] = { + "DAC OUT0", "DAC OUT1" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt274_hpo_enum, RT274_HPO_MUX, + 0, rt274_dac_src); + +static const struct snd_kcontrol_new rt274_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt274_hpo_enum); + +/* Line out source */ +static SOC_ENUM_SINGLE_DECL(rt274_lout_enum, RT274_LOUT_MUX, + 0, rt274_dac_src); + +static const struct snd_kcontrol_new rt274_lout_mux = +SOC_DAPM_ENUM("LOUT source", rt274_lout_enum); + +static const struct snd_soc_dapm_widget rt274_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + + /* DMIC */ + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, RT274_SET_STREAMID_ADC1, 4, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, RT274_SET_STREAMID_ADC2, 4, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("ADC 0 Mux", SND_SOC_NOPM, 0, 0, + &rt274_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", SND_SOC_NOPM, 0, 0, + &rt274_adc1_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RXL", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF1RXR", "AIF1 Playback", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TXL", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TXR", "AIF1 Capture", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RXL", "AIF1 Playback", 2, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RXR", "AIF1 Playback", 3, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TXL", "AIF1 Capture", 2, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TXR", "AIF1 Capture", 3, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, RT274_SET_STREAMID_DAC0, 4, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, RT274_SET_STREAMID_DAC1, 4, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt274_hpo_mux), + SND_SOC_DAPM_MUX("LOUT Mux", SND_SOC_NOPM, 0, 0, &rt274_lout_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT274_SET_PIN_HPO, + RT274_SET_PIN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LOUT Power", RT274_SET_PIN_LOUT3, + RT274_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_PGA("DAC OUT0", SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("DAC OUT1", SND_SOC_NOPM, 0, 0, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH("LOUT L", SND_SOC_NOPM, 0, 0, + &loutl_enable_control), + SND_SOC_DAPM_SWITCH("LOUT R", SND_SOC_NOPM, 0, 0, + &loutr_enable_control), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), + SND_SOC_DAPM_OUTPUT("LINE3"), +}; + +static const struct snd_soc_dapm_route rt274_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + + {"ADC 0 Mux", "Mic", "MIC"}, + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Line1", "LINE1"}, + {"ADC 0 Mux", "Line2", "LINE2"}, + {"ADC 1 Mux", "Mic", "MIC"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Line1", "LINE1"}, + {"ADC 1 Mux", "Line2", "LINE2"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TXL", NULL, "ADC 0"}, + {"AIF1TXR", NULL, "ADC 0"}, + {"AIF2TXL", NULL, "ADC 1"}, + {"AIF2TXR", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RXL"}, + {"DAC 0", NULL, "AIF1RXR"}, + {"DAC 1", NULL, "AIF2RXL"}, + {"DAC 1", NULL, "AIF2RXR"}, + + {"DAC OUT0", NULL, "DAC 0"}, + + {"DAC OUT1", NULL, "DAC 1"}, + + {"LOUT Mux", "DAC OUT0", "DAC OUT0"}, + {"LOUT Mux", "DAC OUT1", "DAC OUT1"}, + + {"LOUT L", "Switch", "LOUT Mux"}, + {"LOUT R", "Switch", "LOUT Mux"}, + {"LOUT L", NULL, "LOUT Power"}, + {"LOUT R", NULL, "LOUT Power"}, + + {"LINE3", NULL, "LOUT L"}, + {"LINE3", NULL, "LOUT R"}, + + {"HPO Mux", "DAC OUT0", "DAC OUT0"}, + {"HPO Mux", "DAC OUT1", "DAC OUT1"}, + + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt274_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code = 0, c_len_code = 0; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt274->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt274->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt274->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + c_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + c_len_code = 3; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + c_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + c_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + c_len_code = 0; + break; + default: + return -EINVAL; + } + + if (rt274->master) + c_len_code = 0x3; + + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0xc018, d_len_code << 3 | c_len_code << 14); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt274_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_M); + rt274->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_S); + rt274->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_LJ); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMA); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMB); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + switch (source) { + case RT274_PLL2_S_MCLK: + snd_soc_update_bits(codec, RT274_PLL2_CTRL, + RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_MCLK); + break; + default: + dev_warn(codec->dev, "invalid pll source, use BCLK\n"); + case RT274_PLL2_S_BCLK: + snd_soc_update_bits(codec, RT274_PLL2_CTRL, + RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK); + break; + } + + if (source == RT274_PLL2_S_BCLK) { + snd_soc_update_bits(codec, RT274_MCLK_CTRL, + (0x3 << 12), (0x3 << 12)); + switch (rt274->fs) { + case 50: + snd_soc_write(codec, 0x7a, 0xaab6); + snd_soc_write(codec, 0x7b, 0x0301); + snd_soc_write(codec, 0x7c, 0x04fe); + break; + case 64: + snd_soc_write(codec, 0x7a, 0xaa96); + snd_soc_write(codec, 0x7b, 0x8003); + snd_soc_write(codec, 0x7c, 0x081e); + break; + case 128: + snd_soc_write(codec, 0x7a, 0xaa96); + snd_soc_write(codec, 0x7b, 0x8003); + snd_soc_write(codec, 0x7c, 0x080e); + break; + default: + dev_warn(codec->dev, "invalid freq_in, assume 4.8M\n"); + case 100: + snd_soc_write(codec, 0x7a, 0xaab6); + snd_soc_write(codec, 0x7b, 0x0301); + snd_soc_write(codec, 0x7c, 0x047e); + break; + } + } + + return 0; +} + +static int rt274_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + unsigned int clk_src, mclk_en; + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + switch (clk_id) { + case RT274_SCLK_S_MCLK: + mclk_en = RT274_MCLK_MODE_EN; + clk_src = RT274_CLK_SRC_MCLK; + break; + case RT274_SCLK_S_PLL1: + mclk_en = RT274_MCLK_MODE_DIS; + clk_src = RT274_CLK_SRC_MCLK; + break; + case RT274_SCLK_S_PLL2: + mclk_en = RT274_MCLK_MODE_EN; + clk_src = RT274_CLK_SRC_PLL2; + break; + default: + mclk_en = RT274_MCLK_MODE_DIS; + clk_src = RT274_CLK_SRC_MCLK; + dev_warn(codec->dev, "invalid sysclk source, use PLL1\n"); + break; + } + snd_soc_update_bits(codec, RT274_MCLK_CTRL, + RT274_MCLK_MODE_MASK, mclk_en); + snd_soc_update_bits(codec, RT274_CLK_CTRL, + RT274_CLK_SRC_MASK, clk_src); + + switch (freq) { + case 19200000: + if (clk_id == RT274_SCLK_S_MCLK) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT274_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (clk_id == RT274_SCLK_S_MCLK) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT274_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT274_MCLK_CTRL, 0x1fcf, 0x0008); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT274_MCLK_CTRL, 0x1fcf, 0x1543); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt274->sys_clk = freq; + rt274->clk_id = clk_id; + + return 0; +} + +static int rt274_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + rt274->fs = ratio; + if ((ratio / 50) == 0) + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt274_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) + +{ + struct snd_soc_codec *codec = dai->codec; + + if (rx_mask || tx_mask) { + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_EN); + } else { + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_DIS); + return 0; + } + + switch (slots) { + case 4: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_4CH); + break; + case 2: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_2CH); + break; + default: + dev_err(codec->dev, + "Support 2 or 4 slots TDM only\n"); + return -EINVAL; + } + + return 0; +} + +static int rt274_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == + snd_soc_codec_get_bias_level(codec)) { + snd_soc_write(codec, + RT274_SET_AUDIO_POWER, AC_PWRST_D0); + } + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT274_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + + return 0; +} + +static irqreturn_t rt274_irq(int irq, void *data) +{ + struct rt274_priv *rt274 = data; + bool hp = false; + bool mic = false; + int ret, status = 0; + + /* Clear IRQ */ + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_CLR, RT274_IRQ_CLR); + + ret = rt274_jack_detect(rt274, &hp, &mic); + + if (ret == 0) { + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt274->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt274->i2c->dev, 300); + } + + return IRQ_HANDLED; +} + +static int rt274_probe(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + rt274->codec = codec; + + if (rt274->i2c->irq) { + INIT_DELAYED_WORK(&rt274->jack_detect_work, + rt274_jack_detect_work); + schedule_delayed_work(&rt274->jack_detect_work, + msecs_to_jiffies(1250)); + } + + return 0; +} + +static int rt274_remove(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt274->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt274_suspend(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt274->regmap, true); + regcache_mark_dirty(rt274->regmap); + + return 0; +} + +static int rt274_resume(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt274->regmap, false); + rt274_index_sync(codec); + regcache_sync(rt274->regmap); + + return 0; +} +#else +#define rt274_suspend NULL +#define rt274_resume NULL +#endif + +#define RT274_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT274_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt274_aif_dai_ops = { + .hw_params = rt274_hw_params, + .set_fmt = rt274_set_dai_fmt, + .set_sysclk = rt274_set_dai_sysclk, + .set_pll = rt274_set_dai_pll, + .set_bclk_ratio = rt274_set_bclk_ratio, + .set_tdm_slot = rt274_set_tdm_slot, +}; + +static struct snd_soc_dai_driver rt274_dai[] = { + { + .name = "rt274-aif1", + .id = RT274_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT274_STEREO_RATES, + .formats = RT274_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT274_STEREO_RATES, + .formats = RT274_FORMATS, + }, + .ops = &rt274_aif_dai_ops, + .symmetric_rates = 1, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt274 = { + .probe = rt274_probe, + .remove = rt274_remove, + .suspend = rt274_suspend, + .resume = rt274_resume, + .set_bias_level = rt274_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt274_snd_controls, + .num_controls = ARRAY_SIZE(rt274_snd_controls), + .dapm_widgets = rt274_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt274_dapm_widgets), + .dapm_routes = rt274_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt274_dapm_routes), + }, + .set_jack = rt274_mic_detect, +}; + +static const struct regmap_config rt274_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x05bfffff, + .volatile_reg = rt274_volatile_register, + .readable_reg = rt274_readable_register, + .reg_write = rl6347a_hw_write, + .reg_read = rl6347a_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt274_reg, + .num_reg_defaults = ARRAY_SIZE(rt274_reg), +}; + +#ifdef CONFIG_OF +static const struct of_device_id rt274_of_match[] = { + {.compatible = "realtek,rt274"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt274_of_match); +#endif + +static const struct i2c_device_id rt274_i2c_id[] = { + {"rt274", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt274_i2c_id); + +static const struct acpi_device_id rt274_acpi_match[] = { + { "10EC0274", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt274_acpi_match); + +static int rt274_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt274_priv *rt274; + + int ret; + unsigned int val; + + rt274 = devm_kzalloc(&i2c->dev, sizeof(*rt274), + GFP_KERNEL); + if (rt274 == NULL) + return -ENOMEM; + + rt274->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt274_regmap); + if (IS_ERR(rt274->regmap)) { + ret = PTR_ERR(rt274->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt274->regmap, + RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (val != RT274_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt274\n", val); + return -ENODEV; + } + + rt274->index_cache = devm_kmemdup(&i2c->dev, rt274_index_def, + sizeof(rt274_index_def), GFP_KERNEL); + if (!rt274->index_cache) + return -ENOMEM; + + rt274->index_cache_size = INDEX_CACHE_SIZE; + rt274->i2c = i2c; + i2c_set_clientdata(i2c, rt274); + + /* reset codec */ + regmap_write(rt274->regmap, RT274_RESET, 0); + regmap_update_bits(rt274->regmap, 0x1a, 0x4000, 0x4000); + + /* Set Pad PDB is floating */ + regmap_update_bits(rt274->regmap, RT274_PAD_CTRL12, 0x3, 0x0); + regmap_write(rt274->regmap, RT274_COEF5b_INDEX, 0x01); + regmap_write(rt274->regmap, RT274_COEF5b_COEF, 0x8540); + regmap_update_bits(rt274->regmap, 0x6f, 0x0100, 0x0100); + /* Combo jack auto detect */ + regmap_write(rt274->regmap, 0x4a, 0x201b); + /* Aux mode off */ + regmap_update_bits(rt274->regmap, 0x6f, 0x3000, 0x2000); + /* HP DC Calibration */ + regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0x0); + //Set NID=58h.Index 00h [15]= 1b; + regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); + regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888); + msleep(500); + regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0xb); + regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); + regmap_write(rt274->regmap, RT274_COEF58_COEF, 0x3888); + /* Set pin widget */ + regmap_write(rt274->regmap, RT274_SET_PIN_HPO, 0x40); + regmap_write(rt274->regmap, RT274_SET_PIN_LOUT3, 0x40); + regmap_write(rt274->regmap, RT274_SET_MIC, 0x20); + regmap_write(rt274->regmap, RT274_SET_PIN_DMIC1, 0x20); + + regmap_update_bits(rt274->regmap, RT274_I2S_CTRL2, 0xc004, 0x4004); + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_GPI2_SEL_MASK, RT274_GPI2_SEL_DMIC_CLK); + + /* jack detection */ + regmap_write(rt274->regmap, RT274_UNSOLICITED_HP_OUT, 0x81); + regmap_write(rt274->regmap, RT274_UNSOLICITED_MIC, 0x82); + + if (rt274->i2c->irq) { + ret = request_threaded_irq(rt274->i2c->irq, NULL, rt274_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt274", rt274); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt274, + rt274_dai, ARRAY_SIZE(rt274_dai)); + + return ret; +} + +static int rt274_i2c_remove(struct i2c_client *i2c) +{ + struct rt274_priv *rt274 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt274); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt274_i2c_driver = { + .driver = { + .name = "rt274", + .acpi_match_table = ACPI_PTR(rt274_acpi_match), +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(rt274_of_match), +#endif + }, + .probe = rt274_i2c_probe, + .remove = rt274_i2c_remove, + .id_table = rt274_i2c_id, +}; + +module_i2c_driver(rt274_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT274 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt274.h b/sound/soc/codecs/rt274.h new file mode 100644 index 0000000..4fd1bcb --- /dev/null +++ b/sound/soc/codecs/rt274.h @@ -0,0 +1,217 @@ +/* + * rt274.h -- RT274 ALSA SoC audio driver + * + * Copyright 2016 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.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 __RT274_H__ +#define __RT274_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT274_AUDIO_FUNCTION_GROUP 0x01 +#define RT274_DAC_OUT0 0x02 +#define RT274_DAC_OUT1 0x03 +#define RT274_ADC_IN2 0x08 +#define RT274_ADC_IN1 0x09 +#define RT274_DIG_CVT 0x0a +#define RT274_DMIC1 0x12 +#define RT274_DMIC2 0x13 +#define RT274_MIC 0x19 +#define RT274_LINE1 0x1a +#define RT274_LINE2 0x1b +#define RT274_LINE3 0x16 +#define RT274_SPDIF 0x1e +#define RT274_VENDOR_REGISTERS 0x20 +#define RT274_HP_OUT 0x21 +#define RT274_MIXER_IN1 0x22 +#define RT274_MIXER_IN2 0x23 +#define RT274_INLINE_CMD 0x55 + +#define RT274_SET_PIN_SFT 6 +#define RT274_SET_PIN_ENABLE 0x40 +#define RT274_SET_PIN_DISABLE 0 +#define RT274_SET_EAPD_HIGH 0x2 +#define RT274_SET_EAPD_LOW 0 + +#define RT274_MUTE_SFT 7 + +/* Verb commands */ +#define RT274_RESET\ + VERB_CMD(AC_VERB_SET_CODEC_RESET, RT274_AUDIO_FUNCTION_GROUP, 0) +#define RT274_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT274_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT274_SET_AUDIO_POWER RT274_SET_POWER(RT274_AUDIO_FUNCTION_GROUP) +#define RT274_SET_HPO_POWER RT274_SET_POWER(RT274_HP_OUT) +#define RT274_SET_DMIC1_POWER RT274_SET_POWER(RT274_DMIC1) +#define RT274_LOUT_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_LINE3, 0) +#define RT274_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_HP_OUT, 0) +#define RT274_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN1, 0) +#define RT274_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN2, 0) +#define RT274_SET_MIC\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_MIC, 0) +#define RT274_SET_PIN_LOUT3\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_LINE3, 0) +#define RT274_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0) +#define RT274_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0) +#define RT274_SET_PIN_SPDIF\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_SPDIF, 0) +#define RT274_SET_PIN_DIG_CVT\ + VERB_CMD(AC_VERB_SET_DIGI_CONVERT_1, RT274_DIG_CVT, 0) +#define RT274_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0) +#define RT274_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0) +#define RT274_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0) +#define RT274_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_HP_OUT, 0) +#define RT274_GET_MIC_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_MIC, 0) +#define RT274_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_DMIC2, 0) +#define RT274_SET_SPDIF_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_SPDIF, 0) +#define RT274_DAC0L_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0xa000) +#define RT274_DAC0R_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0x9000) +#define RT274_DAC1L_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0xa000) +#define RT274_DAC1R_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0x9000) +#define RT274_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x6000) +#define RT274_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x5000) +#define RT274_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_MIC, 0x7000) +#define RT274_LOUTL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0xa000) +#define RT274_LOUTR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0x9000) +#define RT274_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0xa000) +#define RT274_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0x9000) +#define RT274_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_DAC_OUT0, 0) +#define RT274_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_ADC_IN1, 0) +#define RT274_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT274_VENDOR_REGISTERS, 0) +#define RT274_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT274_VENDOR_REGISTERS, 0) +#define RT274_UNSOLICITED_INLINE_CMD\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_INLINE_CMD, 0) +#define RT274_UNSOLICITED_HP_OUT\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_HP_OUT, 0) +#define RT274_UNSOLICITED_MIC\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_MIC, 0) +#define RT274_COEF58_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x58, 0) +#define RT274_COEF58_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, 0x58, 0) +#define RT274_COEF5b_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x5b, 0) +#define RT274_COEF5b_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, 0x5b, 0) +#define RT274_SET_STREAMID_DAC0\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT0, 0) +#define RT274_SET_STREAMID_DAC1\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT1, 0) +#define RT274_SET_STREAMID_ADC1\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN1, 0) +#define RT274_SET_STREAMID_ADC2\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN2, 0) + +/* Index registers */ +#define RT274_EAPD_GPIO_IRQ_CTRL 0x10 +#define RT274_PAD_CTRL12 0x35 +#define RT274_I2S_CTRL1 0x63 +#define RT274_I2S_CTRL2 0x64 +#define RT274_MCLK_CTRL 0x71 +#define RT274_CLK_CTRL 0x72 +#define RT274_PLL2_CTRL 0x7b + + +/* EAPD GPIO IRQ control (Index 0x10) */ +#define RT274_IRQ_DIS (0x0 << 13) +#define RT274_IRQ_EN (0x1 << 13) +#define RT274_IRQ_CLR (0x1 << 12) +#define RT274_GPI2_SEL_MASK (0x3 << 7) +#define RT274_GPI2_SEL_GPIO2 (0x0 << 7) +#define RT274_GPI2_SEL_I2S (0x1 << 7) +#define RT274_GPI2_SEL_DMIC_CLK (0x2 << 7) +#define RT274_GPI2_SEL_CBJ (0x3 << 7) + +/* Front I2S_Interface control 1 (Index 0x63) */ +#define RT274_I2S_MODE_MASK (0x1 << 11) +#define RT274_I2S_MODE_S (0x0 << 11) +#define RT274_I2S_MODE_M (0x1 << 11) +#define RT274_TDM_DIS (0x0 << 10) +#define RT274_TDM_EN (0x1 << 10) +#define RT274_TDM_CH_NUM (0x1 << 7) +#define RT274_TDM_2CH (0x0 << 7) +#define RT274_TDM_4CH (0x1 << 7) +#define RT274_I2S_FMT_MASK (0x3 << 8) +#define RT274_I2S_FMT_I2S (0x0 << 8) +#define RT274_I2S_FMT_LJ (0x1 << 8) +#define RT274_I2S_FMT_PCMA (0x2 << 8) +#define RT274_I2S_FMT_PCMB (0x3 << 8) + +/* MCLK clock domain control (Index 0x71) */ +#define RT274_MCLK_MODE_MASK (0x1 << 14) +#define RT274_MCLK_MODE_DIS (0x0 << 14) +#define RT274_MCLK_MODE_EN (0x1 << 14) + +/* Clock control (Index 0x72) */ +#define RT274_CLK_SRC_MASK (0x7 << 3) +#define RT274_CLK_SRC_MCLK (0x0 << 3) +#define RT274_CLK_SRC_PLL2 (0x3 << 3) + +/* PLL2 control (Index 0x7b) */ +#define RT274_PLL2_SRC_MASK (0x1 << 13) +#define RT274_PLL2_SRC_MCLK (0x0 << 13) +#define RT274_PLL2_SRC_BCLK (0x1 << 13) + +/* HP-OUT (0x21) */ +#define RT274_M_HP_MUX_SFT 14 +#define RT274_HP_SEL_MASK 0x1 +#define RT274_HP_SEL_SFT 0 +#define RT274_HP_SEL_F 0 +#define RT274_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT274_ADC_SEL_MASK 0x7 +#define RT274_ADC_SEL_SFT 0 +#define RT274_ADC_SEL_MIC 0 +#define RT274_ADC_SEL_LINE1 1 +#define RT274_ADC_SEL_LINE2 2 +#define RT274_ADC_SEL_DMIC 3 + +#define RT274_SCLK_S_MCLK 0 +#define RT274_SCLK_S_PLL1 1 +#define RT274_SCLK_S_PLL2 2 + +#define RT274_PLL2_S_MCLK 0 +#define RT274_PLL2_S_BCLK 1 + +enum { + RT274_AIF1, + RT274_AIFS, +}; + +#endif /* __RT274_H__ */ + -- 2.7.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: fix pcm-creation regression
by Johan Hovold 18 Jul '17

18 Jul '17

This reverts commit 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free"), which started calling the pcm_new callback for every component in a *card* when creating a new pcm, something which does not seem to make any sense. This specifically led to memory leaks in systems with more than one platform component and where DMA memory is allocated in the platform-driver callback. For example, when both mcasp devices are being used on an am335x board, DMA memory would be allocated twice for every DAI link during probe. When CONFIG_SND_VERBOSE_PROCFS was set this fortunately also led to warnings such as: WARNING: CPU: 0 PID: 565 at ../fs/proc/generic.c:346 proc_register+0x110/0x154 proc_dir_entry 'sub0/prealloc' already registered Since there seems to be no users of the new component callbacks, and the current implementation introduced a regression, let's revert the offending commit for now. Fixes: 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free") Cc: stable <stable(a)vger.kernel.org> # 4.10 Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Brian Norris <briannorris(a)chromium.org> Signed-off-by: Johan Hovold <johan(a)kernel.org> --- include/sound/soc.h | 6 ------ sound/soc/soc-core.c | 25 ------------------------- sound/soc/soc-pcm.c | 32 +++++++++----------------------- 3 files changed, 9 insertions(+), 54 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9c94b97c17f8..c4a8b1947566 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -795,10 +795,6 @@ struct snd_soc_component_driver { int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - /* pcm creation and destruction */ - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); - /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, @@ -874,8 +870,6 @@ struct snd_soc_component { void (*remove)(struct snd_soc_component *); int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); /* machine specific init */ int (*init)(struct snd_soc_component *component); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a01944..c240e13ba057 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3171,8 +3171,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; component->suspend = component->driver->suspend; component->resume = component->driver->resume; - component->pcm_new = component->driver->pcm_new; - component->pcm_free = component->driver->pcm_free; dapm = &component->dapm; dapm->dev = dev; @@ -3360,25 +3358,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } -static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_new) - return platform->driver->pcm_new(rtd); - else - return 0; -} - -static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_free) - platform->driver->pcm_free(pcm); -} - /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -3402,10 +3381,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->pcm_new) - platform->component.pcm_new = snd_soc_platform_drv_pcm_new; - if (platform_drv->pcm_free) - platform->component.pcm_free = snd_soc_platform_drv_pcm_free; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcc5ece08668..553f7a76743c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2628,25 +2628,12 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } -static void soc_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_component *component; - - list_for_each_entry(component, &rtd->card->component_dev_list, - card_list) { - if (component->pcm_free) - component->pcm_free(pcm); - } -} - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; @@ -2756,18 +2743,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); - list_for_each_entry(component, &rtd->card->component_dev_list, card_list) { - if (component->pcm_new) { - ret = component->pcm_new(rtd); - if (ret < 0) { - dev_err(component->dev, - "ASoC: pcm constructor failed: %d\n", - ret); - return ret; - } + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + dev_err(platform->dev, + "ASoC: pcm constructor failed: %d\n", + ret); + return ret; } } - pcm->private_free = soc_pcm_free; + + pcm->private_free = platform->driver->pcm_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, -- 2.13.2

3 12

[alsa-devel] [PATCH 20/22] sound: pci: avoid string overflow warnings
by Arnd Bergmann 18 Jul '17

18 Jul '17

With gcc-7, we get various warnings about a possible string overflow: sound/pci/rme9652/hdspm.c: In function 'snd_hdspm_create_alsa_devices': sound/pci/rme9652/hdspm.c:2123:17: error: ' MIDIoverMADI' directive writing 13 bytes into a region of size between 1 and 32 [-Werror=format-overflow=] sound/pci/pcxhr/pcxhr.c: In function 'pcxhr_probe': sound/pci/pcxhr/pcxhr.c:1647:28: error: ' [PCM #' directive writing 7 bytes into a region of size between 1 and 32 [-Werror=format-overflow=] sound/pci/mixart/mixart.c: In function 'snd_mixart_probe': sound/pci/mixart/mixart.c:1353:28: error: ' [PCM #' directive writing 7 bytes into a region of size between 1 and 32 [-Werror=format-overflow=] sprintf(card->shortname, "%s [PCM #%d]", mgr->shortname, i); ^~~~~~~~~~~~~~ sound/pci/mixart/mixart.c:1353:28: note: using the range [-2147483648, 2147483647] for directive argument sound/pci/mixart/mixart.c:1353:3: note: 'sprintf' output between 10 and 51 bytes into a destination of size 32 sprintf(card->shortname, "%s [PCM #%d]", mgr->shortname, i); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/mixart/mixart.c:1354:27: error: ' [PCM #' directive writing 7 bytes into a region of size between 1 and 80 [-Werror=format-overflow=] sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); ^~~~~~~~~~~~~~ sound/pci/mixart/mixart.c:1354:27: note: using the range [-2147483648, 2147483647] for directive argument sound/pci/mixart/mixart.c:1354:3: note: 'sprintf' output between 10 and 99 bytes into a destination of size 80 I have checked these all and found that the driver-private shortname strings for mixart and pcxhr are longer than necessary, and making them shorter will be safe while also making it clear that no overflow can happen when they get passed as a substring into the card shortname. For hdspm, we have a local buffer of the same size as its substring. In this case, making the buffer a little longer is safe as the functions that take it as an argument all use length checking and the strings we pass into it are actually short enough. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/mixart/mixart.h | 4 ++-- sound/pci/pcxhr/pcxhr.h | 4 ++-- sound/pci/rme9652/hdspm.c | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h index 426743871540..c8309e327663 100644 --- a/sound/pci/mixart/mixart.h +++ b/sound/pci/mixart/mixart.h @@ -75,8 +75,8 @@ struct mixart_mgr { struct mem_area mem[2]; /* share the name */ - char shortname[32]; /* short name of this soundcard */ - char longname[80]; /* name of this soundcard */ + char shortname[16]; /* short name of this soundcard */ + char longname[40]; /* name of this soundcard */ /* one and only blocking message or notification may be pending */ u32 pending_event; diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 9e39e509a3ef..4909a43ce3d9 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -75,8 +75,8 @@ struct pcxhr_mgr { unsigned long port[3]; /* share the name */ - char shortname[32]; /* short name of this soundcard */ - char longname[96]; /* name of this soundcard */ + char shortname[16]; /* short name of this soundcard */ + char longname[40]; /* name of this soundcard */ struct pcxhr_rmh *prmh; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 254c3d040118..a1cbf5938a0e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2061,7 +2061,7 @@ static int snd_hdspm_create_midi(struct snd_card *card, struct hdspm *hdspm, int id) { int err; - char buf[32]; + char buf[64]; hdspm->midi[id].id = id; hdspm->midi[id].hdspm = hdspm; -- 2.9.0

2 2

[alsa-devel] Reset to clean state after snd_ctl_add_integer_elem_set ?
by Fulup Ar Foll 18 Jul '17

18 Jul '17

I'm using snd_ctl_add_boolean_elem_set and snd_ctl_add_integer_elem_set to create some user kernel controls. While the operation work find I did not find any simple command to erase all those controls. Even "/usr/sbin/alsactl --init" does not remove those added controls. The only way I find is to prevent /usr/sbin/alsactlto happen during reboot to really reset the ALSA driver to an initial state, which is really not pratical when developing/debugging. Question: is there a simple method to return to a clean and fresh state after snd_ctl_add_xxxx without rebooting the machine ? Thank you Fulup

2 2

[alsa-devel] [PATCH v1] ALSA: fm801: Initialize chip after IRQ handler is registered
by Andy Shevchenko 18 Jul '17

18 Jul '17

The commit b56fa687e02b ("ALSA: fm801: detect FM-only card earlier") rearranged initialization calls, i.e. it makes snd_fm801_chip_init() to be called before we register interrupt handler and set PCI bus mastering. Somehow it prevents FM801-AU to work properly. Thus, partially revert initialization order changed by commit mentioned above. Fixes: b56fa687e02b ("ALSA: fm801: detect FM-only card earlier") Reported-by: Émeric MASCHINO <emeric.maschino(a)gmail.com> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/fm801.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 2e402ece4c86..8e6b04b39dcc 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1235,8 +1235,6 @@ static int snd_fm801_create(struct snd_card *card, } } - snd_fm801_chip_init(chip); - if ((chip->tea575x_tuner & TUNER_ONLY) == 0) { if (devm_request_irq(&pci->dev, pci->irq, snd_fm801_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { @@ -1248,6 +1246,8 @@ static int snd_fm801_create(struct snd_card *card, pci_set_master(pci); } + snd_fm801_chip_init(chip); + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); return err; -- 2.11.0

3 5

[alsa-devel] [PATCH v2] ASoC: rt5663: Modify the default value for ASRC function
by Oder Chiou 18 Jul '17

18 Jul '17

The patch modifies the default value for ASRC function. It could prevent the pop noise and recording no sound with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index b262b6f..d324817 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -60,6 +60,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8080 }, + { 0x0086, 0x0028 }, }; static const struct reg_default rt5663_v2_reg[] = { @@ -509,7 +510,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0082, 0x0000 }, { 0x0083, 0x0000 }, { 0x0084, 0x0000 }, - { 0x0086, 0x0008 }, + { 0x0086, 0x0028 }, { 0x0087, 0x0000 }, { 0x008a, 0x0000 }, { 0x008b, 0x0000 }, -- 2.7.4

1 0

[alsa-devel] [PATCH v2 0/2] Add support for TI TAS6424 digital amplifier
by Michael Stecklein 17 Jul '17

17 Jul '17

Hello all, This two-patch series adds the dt bindings, driver, and other related changes for the TAS6424 digital amplifier. I am upstreaming this driver on behalf of Andreas. Regards, Michael Stecklein <m-stecklein(a)ti.com> Changes since v1: - rebased on 4.13-rc1 Original v1 patches: - https://patchwork.kernel.org/patch/9795235/ - https://patchwork.kernel.org/patch/9795237/ Andreas Dannenberg (1): ASoC: add support for TAS6424 digital amplifier Michael Stecklein (1): dt-bindings: sound: add bindings for TAS6424 .../devicetree/bindings/sound/ti,tas6424.txt | 31 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas6424.c | 755 +++++++++++++++++++++ sound/soc/codecs/tas6424.h | 127 ++++ 5 files changed, 923 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas6424.txt create mode 100644 sound/soc/codecs/tas6424.c create mode 100644 sound/soc/codecs/tas6424.h -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: rsnd: remove unnecessary static in rsnd_ssiu_probe()
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

Remove unnecessary static on local variable ops. Such variable is initialized before being used, on every execution path throughout the function. The static has no benefit and, removing it reduces the code size. This issue was detected using Coccinelle and the following semantic patch: @bad exists@ position p; identifier x; type T; @@ static T x@p; ... x = <+...x...+> @@ identifier x; expression e; type T; position p != bad.p; @@ -static T x@p; ... when != x when strict ?x = e; In the following log you can see the difference in the code size. Also, there is a significant difference in the bss segment. This log is the output of the size command, before and after the code change: before: text data bss dec hex filename 3211 680 64 3955 f73 sound/soc/sh/rcar/ssiu.o after: text data bss dec hex filename 3207 592 0 3799 ed7 sound/soc/sh/rcar/ssiu.o Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/sh/rcar/ssiu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index bed2c9c..4d94875 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -250,7 +250,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssiu *ssiu; - static struct rsnd_mod_ops *ops; + struct rsnd_mod_ops *ops; int i, nr, ret; /* same number to SSI */ -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: samsung: Remove non-existing CONFIG_CPU_S3C2413
by Krzysztof Kozlowski 17 Jul '17

17 Jul '17

There is no CONFIG_CPU_S3C2413 so get rid of it. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/samsung/s3c-i2s-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 8f42deaa184b..ca522a95160b 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -27,7 +27,7 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ +#if defined(CONFIG_CPU_S3C2412) \ || defined(CONFIG_ARCH_S3C64XX) || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: fsi: constify dev_pm_ops structure" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: fsi: constify dev_pm_ops structure has been applied to the asoc tree at git://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 4b2983da22d30c53d778b2e5035dc347dc4b0037 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sun, 16 Jul 2017 21:12:21 -0500 Subject: [PATCH] ASoC: fsi: constify dev_pm_ops structure This dev_pm_ops structure is only stored in the pm field of a device_driver structure. This field is declared const, so dev_pm_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7c4bdd82bb95..005b21506556 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2109,7 +2109,7 @@ static int fsi_resume(struct device *dev) return 0; } -static struct dev_pm_ops fsi_pm_ops = { +static const struct dev_pm_ops fsi_pm_ops = { .suspend = fsi_suspend, .resume = fsi_resume, }; -- 2.13.2

1 0

[alsa-devel] Applied "ASoC: psc-ac97: constify dev_pm_ops structure" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: psc-ac97: constify dev_pm_ops structure has been applied to the asoc tree at git://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 da61a11823395c87db0471d8d54055fa5a961688 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sun, 16 Jul 2017 21:12:39 -0500 Subject: [PATCH] ASoC: psc-ac97: constify dev_pm_ops structure This dev_pm_ops structure is only stored in the pm field of a device_driver structure. This field is declared const, so dev_pm_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/au1x/psc-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index bb53c7059005..a2050ae5a3fe 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -474,7 +474,7 @@ static int au1xpsc_ac97_drvresume(struct device *dev) return 0; } -static struct dev_pm_ops au1xpscac97_pmops = { +static const struct dev_pm_ops au1xpscac97_pmops = { .suspend = au1xpsc_ac97_drvsuspend, .resume = au1xpsc_ac97_drvresume, }; -- 2.13.2

1 0

[alsa-devel] Applied "ASoC: au1x: psc-i2s: constify dev_pm_ops structure" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: au1x: psc-i2s: constify dev_pm_ops structure has been applied to the asoc tree at git://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 90982dcdda4f0163bc5e53f49301e155c2cd8547 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sun, 16 Jul 2017 21:12:49 -0500 Subject: [PATCH] ASoC: au1x: psc-i2s: constify dev_pm_ops structure This dev_pm_ops structure is only stored in the pm field of a device_driver structure. This field is declared const, so dev_pm_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/au1x/psc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 0bf9d62b91a0..e6eec081eaae 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -392,7 +392,7 @@ static int au1xpsc_i2s_drvresume(struct device *dev) return 0; } -static struct dev_pm_ops au1xpsci2s_pmops = { +static const struct dev_pm_ops au1xpsci2s_pmops = { .suspend = au1xpsc_i2s_drvsuspend, .resume = au1xpsc_i2s_drvresume, }; -- 2.13.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix type in debug message
by Colin King 17 Jul '17

17 Jul '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to typo in debug message, dst_instacne should be dst_instance Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl-messages.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index fb2f1f603f3c..c5b4e0a9946c 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -950,7 +950,7 @@ static void skl_dump_bind_info(struct skl_sst *ctx, struct skl_module_cfg { dev_dbg(ctx->dev, "%s: src module_id = %d src_instance=%d\n", __func__, src_module->id.module_id, src_module->id.pvt_id); - dev_dbg(ctx->dev, "%s: dst_module=%d dst_instacne=%d\n", __func__, + dev_dbg(ctx->dev, "%s: dst_module=%d dst_instance=%d\n", __func__, dst_module->id.module_id, dst_module->id.pvt_id); dev_dbg(ctx->dev, "src_module state = %d dst module state = %d\n", -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: Intel: Skylake: constify snd_soc_dai_ops structures
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

These structures are only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/intel/skylake/skl-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 0ebea34a..b1f2dba 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -628,7 +628,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops skl_pcm_dai_ops = { +static const struct snd_soc_dai_ops skl_pcm_dai_ops = { .startup = skl_pcm_open, .shutdown = skl_pcm_close, .prepare = skl_pcm_prepare, @@ -637,15 +637,15 @@ static struct snd_soc_dai_ops skl_pcm_dai_ops = { .trigger = skl_pcm_trigger, }; -static struct snd_soc_dai_ops skl_dmic_dai_ops = { +static const struct snd_soc_dai_ops skl_dmic_dai_ops = { .hw_params = skl_be_hw_params, }; -static struct snd_soc_dai_ops skl_be_ssp_dai_ops = { +static const struct snd_soc_dai_ops skl_be_ssp_dai_ops = { .hw_params = skl_be_hw_params, }; -static struct snd_soc_dai_ops skl_link_dai_ops = { +static const struct snd_soc_dai_ops skl_link_dai_ops = { .prepare = skl_link_pcm_prepare, .hw_params = skl_link_hw_params, .hw_free = skl_link_hw_free, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: Atom: constify snd_soc_dai_ops structures
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

These structures are only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index b082b31..49c7b88 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -471,7 +471,7 @@ static void sst_disable_ssp(struct snd_pcm_substream *substream, } } -static struct snd_soc_dai_ops sst_media_dai_ops = { +static const struct snd_soc_dai_ops sst_media_dai_ops = { .startup = sst_media_open, .shutdown = sst_media_close, .prepare = sst_media_prepare, @@ -480,11 +480,11 @@ static struct snd_soc_dai_ops sst_media_dai_ops = { .mute_stream = sst_media_digital_mute, }; -static struct snd_soc_dai_ops sst_compr_dai_ops = { +static const struct snd_soc_dai_ops sst_compr_dai_ops = { .mute_stream = sst_media_digital_mute, }; -static struct snd_soc_dai_ops sst_be_dai_ops = { +static const struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, .hw_params = sst_be_hw_params, .set_fmt = sst_set_format, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: rt5665: force using PLL if MCLK is not suitable
by Bard Liao 17 Jul '17

17 Jul '17

Idealy PLL will be set in machine driver if MCLK doesn't meet the requirement of codec. To let the codec driver be more general, we can use a common PLL setting once sysclk/pll doesn't set properly in machine driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 370ed54d..d3103ef 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4098,9 +4098,12 @@ static int rt5665_hw_params(struct snd_pcm_substream *substream, rt5665->lrck[dai->id] = params_rate(params); pre_div = rl6231_get_clk_info(rt5665->sysclk, rt5665->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", - rt5665->lrck[dai->id], dai->id); - return -EINVAL; + dev_warn(codec->dev, "Force using PLL"); + snd_soc_codec_set_pll(codec, 0, RT5665_PLL1_S_MCLK, + rt5665->sysclk, rt5665->lrck[dai->id] * 512); + snd_soc_codec_set_sysclk(codec, RT5665_SCLK_S_PLL1, 0, + rt5665->lrck[dai->id] * 512, 0); + pre_div = 1; } frame_size = snd_soc_params_to_frame_size(params); if (frame_size < 0) { -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: do not close shared backend dailink
by Patrick Lai 17 Jul '17

17 Jul '17

From: Banajit Goswami <bgoswami(a)codeaurora.org> Multiple frontend dailinks may be connected to a backend dailink at the same time. When one of frontend dailinks is closed, the associated backend dailink should not be closed if it is connected to other active frontend dailinks. Change ensures that backend dailink is closed only after all connected frontend dailinks are closed. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index efc5831..3d2c69f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -181,6 +181,10 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n", be->dai_link->name, event, dir); + if ((event == SND_SOC_DAPM_STREAM_STOP) && + (be->dpcm[dir].users >= 1)) + continue; + snd_soc_dapm_stream_event(be, dir, event); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,\na Linux Foundation Collaborative Project

2 1

[alsa-devel] Applied "ASoC: rsnd: add missing of_node_put" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: rsnd: add missing of_node_put has been applied to the asoc tree at git://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 53ba2aa314164f7fcd3acf9594313811256a52c4 Mon Sep 17 00:00:00 2001 From: Julia Lawall <Julia.Lawall(a)lip6.fr> Date: Sat, 15 Jul 2017 09:19:07 +0200 Subject: [PATCH] ASoC: rsnd: add missing of_node_put for_each_child_of_node performs an of_node_get on each iteration, so a jump out of the loop requires an of_node_put. The semantic patch that fixes this problem is as follows (http://coccinelle.lip6.fr): // <smpl> @@ local idexpression n; expression e,e1; identifier l; @@ for_each_child_of_node(e1,n) { ... ( of_node_put(n); | e = n | + of_node_put(n); ? goto l; ) ... } ... l: ... when != n // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ctu.c | 5 ++++- sound/soc/sh/rcar/dvc.c | 5 ++++- sound/soc/sh/rcar/mix.c | 5 ++++- sound/soc/sh/rcar/src.c | 6 +++++- sound/soc/sh/rcar/ssi.c | 6 +++++- 5 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 4ba8f2fe7a4c..e7f53f44165d 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -394,13 +394,16 @@ int rsnd_ctu_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_ctu_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops, clk, rsnd_mod_get_status, RSND_MOD_CTU, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_ctu_probe_done; + } i++; } diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 99d2d9459e75..1743ade3cc55 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -380,13 +380,16 @@ int rsnd_dvc_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_dvc_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops, clk, rsnd_mod_get_status, RSND_MOD_DVC, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_dvc_probe_done; + } i++; } diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 195fc7bb22af..6c4826c189a4 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -168,13 +168,16 @@ int rsnd_mix_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_mix_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops, clk, rsnd_mod_get_status, RSND_MOD_MIX, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_mix_probe_done; + } i++; } diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 7aa239e28491..8e9a1de0ec25 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -581,20 +581,24 @@ int rsnd_src_probe(struct rsnd_priv *priv) src->irq = irq_of_parse_and_map(np, 0); if (!src->irq) { ret = -EINVAL; + of_node_put(np); goto rsnd_src_probe_done; } clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_src_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(src), &rsnd_src_ops, clk, rsnd_mod_get_status, RSND_MOD_SRC, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_src_probe_done; + } skip: i++; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 46feddd78ee2..49cdc5e007b3 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1079,6 +1079,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_ssi_probe_done; } @@ -1091,6 +1092,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) ssi->irq = irq_of_parse_and_map(np, 0); if (!ssi->irq) { ret = -EINVAL; + of_node_put(np); goto rsnd_ssi_probe_done; } @@ -1101,8 +1103,10 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk, rsnd_ssi_get_status, RSND_MOD_SSI, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_ssi_probe_done; + } i++; } -- 2.13.2

1 0

[alsa-devel] [PATCH 1/8] ASoC: sun4i-codec: Remove unneeded gpiod NULL check
by Fabio Estevam 17 Jul '17

17 Jul '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> The gpiod API checks for NULL descriptors, so there is no need to duplicate the check in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/sunxi/sun4i-codec.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 1500699..4c37231 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1171,9 +1171,8 @@ static int sun4i_codec_spk_event(struct snd_soc_dapm_widget *w, { struct sun4i_codec *scodec = snd_soc_card_get_drvdata(w->dapm->card); - if (scodec->gpio_pa) - gpiod_set_value_cansleep(scodec->gpio_pa, - !!SND_SOC_DAPM_EVENT_ON(event)); + gpiod_set_value_cansleep(scodec->gpio_pa, + !!SND_SOC_DAPM_EVENT_ON(event)); return 0; } -- 2.7.4

2 15

[alsa-devel] [PATCH v2] ASoC: rt5514: Support the DSP recording continuously after the hotwording triggered
by Oder Chiou 17 Jul '17

17 Jul '17

The patch uses the IRQ to copy the PCM data to userspace continuously after the hotwording triggered from DSP. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 139 ++++++++++++++++++++++-------------------- sound/soc/codecs/rt5514-spi.h | 7 +-- 2 files changed, 76 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 7ed62e8..950d1ff 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -43,9 +43,7 @@ struct rt5514_dsp { struct mutex dma_lock; struct snd_pcm_substream *substream; unsigned int buf_base, buf_limit, buf_rp; - size_t buf_size; - size_t dma_offset; - size_t dsp_offset; + size_t buf_size, get_size, dma_offset; }; static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { @@ -80,6 +78,8 @@ static void rt5514_spi_copy_work(struct work_struct *work) container_of(work, struct rt5514_dsp, copy_work.work); struct snd_pcm_runtime *runtime; size_t period_bytes, truncated_bytes = 0; + unsigned int cur_wp, remain_data; + u8 buf[8]; mutex_lock(&rt5514_dsp->dma_lock); if (!rt5514_dsp->substream) { @@ -90,8 +90,24 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); - if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) - period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset; + if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) { + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, + sizeof(buf)); + cur_wp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + if (cur_wp >= rt5514_dsp->buf_rp) + remain_data = (cur_wp - rt5514_dsp->buf_rp); + else + remain_data = + (rt5514_dsp->buf_limit - rt5514_dsp->buf_rp) + + (cur_wp - rt5514_dsp->buf_base); + + if (remain_data < period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + } if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { rt5514_spi_burst_read(rt5514_dsp->buf_rp, @@ -112,24 +128,58 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime->dma_area + rt5514_dsp->dma_offset + truncated_bytes, period_bytes - truncated_bytes); - rt5514_dsp->buf_rp = rt5514_dsp->buf_base + - period_bytes - truncated_bytes; + rt5514_dsp->buf_rp = rt5514_dsp->buf_base + period_bytes - + truncated_bytes; } + rt5514_dsp->get_size += period_bytes; rt5514_dsp->dma_offset += period_bytes; if (rt5514_dsp->dma_offset >= runtime->dma_bytes) rt5514_dsp->dma_offset = 0; - rt5514_dsp->dsp_offset += period_bytes; - snd_pcm_period_elapsed(rt5514_dsp->substream); - if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size) - schedule_delayed_work(&rt5514_dsp->copy_work, 5); + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + done: mutex_unlock(&rt5514_dsp->dma_lock); } +static irqreturn_t rt5514_spi_irq(int irq, void *data) +{ + struct rt5514_dsp *rt5514_dsp = data; + u8 buf[8]; + + rt5514_dsp->get_size = 0; + rt5514_dsp->dma_offset = 0; + + /** + * The address area x1800XXXX is the register address, and it cannot + * support spi burst read perfectly. So we use the spi burst read + * individually to make sure the data correctly. + */ + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; + + schedule_delayed_work(&rt5514_dsp->copy_work, 0); + + return IRQ_HANDLED; +} + /* PCM for streaming audio from the DSP buffer */ static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) { @@ -170,59 +220,6 @@ static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_vmalloc_buffer(substream); } -static int rt5514_spi_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct rt5514_dsp *rt5514_dsp = - snd_soc_platform_get_drvdata(rtd->platform); - u8 buf[8]; - - rt5514_dsp->dma_offset = 0; - rt5514_dsp->dsp_offset = 0; - - /** - * The address area x1800XXXX is the register address, and it cannot - * support spi burst read perfectly. So we use the spi burst read - * individually to make sure the data correctly. - */ - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - return 0; -} - -static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct rt5514_dsp *rt5514_dsp = - snd_soc_platform_get_drvdata(rtd->platform); - - if (cmd == SNDRV_PCM_TRIGGER_START) { - if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && - rt5514_dsp->buf_rp && rt5514_dsp->buf_size) - schedule_delayed_work(&rt5514_dsp->copy_work, 0); - } - - return 0; -} - static snd_pcm_uframes_t rt5514_spi_pcm_pointer( struct snd_pcm_substream *substream) { @@ -238,8 +235,6 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = { .open = rt5514_spi_pcm_open, .hw_params = rt5514_spi_hw_params, .hw_free = rt5514_spi_hw_free, - .trigger = rt5514_spi_trigger, - .prepare = rt5514_spi_prepare, .pointer = rt5514_spi_pcm_pointer, .mmap = snd_pcm_lib_mmap_vmalloc, .page = snd_pcm_lib_get_vmalloc_page, @@ -248,6 +243,7 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = { static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) { struct rt5514_dsp *rt5514_dsp; + int ret; rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), GFP_KERNEL); @@ -257,6 +253,17 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); snd_soc_platform_set_drvdata(platform, rt5514_dsp); + if (rt5514_spi->irq) { + ret = devm_request_threaded_irq(&rt5514_spi->dev, + rt5514_spi->irq, NULL, rt5514_spi_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5514-spi", + rt5514_dsp); + if (ret) + dev_err(&rt5514_spi->dev, + "%s Failed to reguest IRQ: %d\n", __func__, + ret); + } + return 0; } diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h index f69b1cd..a6434ee 100644 --- a/sound/soc/codecs/rt5514-spi.h +++ b/sound/soc/codecs/rt5514-spi.h @@ -17,10 +17,9 @@ */ #define RT5514_SPI_BUF_LEN 240 -#define RT5514_BUFFER_VOICE_BASE 0x18001034 -#define RT5514_BUFFER_VOICE_LIMIT 0x18001038 -#define RT5514_BUFFER_VOICE_RP 0x1800103c -#define RT5514_BUFFER_VOICE_SIZE 0x18001040 +#define RT5514_BUFFER_VOICE_BASE 0x18000200 +#define RT5514_BUFFER_VOICE_LIMIT 0x18000204 +#define RT5514_BUFFER_VOICE_WP 0x1800020c /* SPI Command */ enum { -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: tegra: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/tegra/tegra30_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index b2b279c..0b176ea 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -275,7 +275,7 @@ static int tegra30_i2s_probe(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops tegra30_i2s_dai_ops = { +static const struct snd_soc_dai_ops tegra30_i2s_dai_ops = { .set_fmt = tegra30_i2s_set_fmt, .hw_params = tegra30_i2s_hw_params, .trigger = tegra30_i2s_trigger, -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoC: sun8i-codec: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/sunxi/sun8i-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 5723c34..253ae0b 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -341,7 +341,7 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { "AIF1 Slot 0 Right"}, }; -static struct snd_soc_dai_ops sun8i_codec_dai_ops = { +static const struct snd_soc_dai_ops sun8i_codec_dai_ops = { .hw_params = sun8i_codec_hw_params, .set_fmt = sun8i_set_fmt, }; -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoC: fsl_spdif: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 1ff467c..7e6cc4d 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -626,7 +626,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops fsl_spdif_dai_ops = { +static const struct snd_soc_dai_ops fsl_spdif_dai_ops = { .startup = fsl_spdif_startup, .hw_params = fsl_spdif_hw_params, .trigger = fsl_spdif_trigger, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_esai: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 809a069..cef79a1 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -620,7 +620,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static struct snd_soc_dai_ops fsl_esai_dai_ops = { +static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .startup = fsl_esai_startup, .shutdown = fsl_esai_shutdown, .trigger = fsl_esai_trigger, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_asrc: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/fsl/fsl_asrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 8cfffa7..806d399 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -542,7 +542,7 @@ static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static struct snd_soc_dai_ops fsl_asrc_dai_ops = { +static const struct snd_soc_dai_ops fsl_asrc_dai_ops = { .hw_params = fsl_asrc_dai_hw_params, .hw_free = fsl_asrc_dai_hw_free, .trigger = fsl_asrc_dai_trigger, -- 2.5.0

4 7

[alsa-devel] [PATCH] ASoC: hisilicon: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/hisilicon/hi6210-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index b193d3b..0c8f86d 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -517,7 +517,7 @@ static int hi6210_i2s_dai_probe(struct snd_soc_dai *dai) } -static struct snd_soc_dai_ops hi6210_i2s_dai_ops = { +static const struct snd_soc_dai_ops hi6210_i2s_dai_ops = { .trigger = hi6210_i2s_trigger, .hw_params = hi6210_i2s_hw_params, .set_fmt = hi6210_i2s_set_fmt, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: mediatek: mt2701: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index bc5d4db..10dcd69 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -595,7 +595,7 @@ static const struct snd_soc_dai_ops mt2701_afe_i2s_ops = { }; /* MRG BE DAIs */ -static struct snd_soc_dai_ops mt2701_btmrg_ops = { +static const struct snd_soc_dai_ops mt2701_btmrg_ops = { .startup = mt2701_btmrg_startup, .shutdown = mt2701_btmrg_shutdown, .hw_params = mt2701_btmrg_hw_params, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: mmp-sspa: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/pxa/mmp-sspa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 9cc3501..64b85e3 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -380,7 +380,7 @@ static int mmp_sspa_probe(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops mmp_sspa_dai_ops = { +static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { .startup = mmp_sspa_startup, .shutdown = mmp_sspa_shutdown, .trigger = mmp_sspa_trigger, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: zx-i2s: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/zte/zx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 8bbad1d..9a05659 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -357,7 +357,7 @@ static void zx_i2s_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(zx_i2s->dai_pclk); } -static struct snd_soc_dai_ops zx_i2s_dai_ops = { +static const struct snd_soc_dai_ops zx_i2s_dai_ops = { .trigger = zx_i2s_trigger, .hw_params = zx_i2s_hw_params, .set_fmt = zx_i2s_set_fmt, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: zx-tdm: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/zte/zx-tdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index bd632cc..dc95527 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -309,7 +309,7 @@ static void zx_tdm_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(zx_tdm->dai_wclk); } -static struct snd_soc_dai_ops zx_tdm_dai_ops = { +static const struct snd_soc_dai_ops zx_tdm_dai_ops = { .trigger = zx_tdm_trigger, .hw_params = zx_tdm_hw_params, .set_fmt = zx_tdm_set_fmt, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: zte: spdif: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/zte/zx-spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c index 9fa6463..b143f9f 100644 --- a/sound/soc/zte/zx-spdif.c +++ b/sound/soc/zte/zx-spdif.c @@ -264,7 +264,7 @@ static void zx_spdif_shutdown(struct snd_pcm_substream *substream, (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE \ | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops zx_spdif_dai_ops = { +static const struct snd_soc_dai_ops zx_spdif_dai_ops = { .trigger = zx_spdif_trigger, .startup = zx_spdif_startup, .shutdown = zx_spdif_shutdown, -- 2.5.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic32x4: Add support for tlv320aic32x6
by Dan Murphy 17 Jul '17

17 Jul '17

Add support for the tlv320aic32x6 family of devices. These devices share the same register set as the tlv320aic32x4 and this driver can be re-used. TLV320aic32x6 data sheets are here: tlv320aic3256 - http://www.ti.com/lit/gpn/tlv320aic3256 tlv320aic3206 - http://www.ti.com/lit/gpn/tlv320aic3206 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tlv320aic32x4.txt | 4 +++- sound/soc/codecs/tlv320aic32x4-i2c.c | 2 ++ sound/soc/codecs/tlv320aic32x4-spi.c | 2 ++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt index 5e2741af27be..9651fc7503a4 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -3,7 +3,9 @@ Texas Instruments - tlv320aic32x4 Codec module The tlv320aic32x4 serial control bus communicates through I2C protocols Required properties: - - compatible: Should be "ti,tlv320aic32x4" + - compatible - "string" - One of: + "ti,tlv320aic32x4" TLV320AIC3204 + "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 - reg: I2C slave address - supply-*: Required supply regulators are: "iov" - digital IO power supply diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c index 59606cf3008f..385fa2e9525a 100644 --- a/sound/soc/codecs/tlv320aic32x4-i2c.c +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -47,12 +47,14 @@ static int aic32x4_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id aic32x4_i2c_id[] = { { "tlv320aic32x4", 0 }, + { "tlv320aic32x6", 1 }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); static const struct of_device_id aic32x4_of_id[] = { { .compatible = "ti,tlv320aic32x4", }, + { .compatible = "ti,tlv320aic32x6", }, { /* senitel */ } }; MODULE_DEVICE_TABLE(of, aic32x4_of_id); diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c index 724fcdd491b2..07d78ae51e05 100644 --- a/sound/soc/codecs/tlv320aic32x4-spi.c +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -48,12 +48,14 @@ static int aic32x4_spi_remove(struct spi_device *spi) static const struct spi_device_id aic32x4_spi_id[] = { { "tlv320aic32x4", 0 }, + { "tlv320aic32x6", 1 }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(spi, aic32x4_spi_id); static const struct of_device_id aic32x4_of_id[] = { { .compatible = "ti,tlv320aic32x4", }, + { .compatible = "ti,tlv320aic32x6", }, { /* senitel */ } }; MODULE_DEVICE_TABLE(of, aic32x4_of_id); -- 2.13.2.516.ge629a7d28

2 3

[alsa-devel] [PATCH 00/12] constify snd_soc_dai_ops structures
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This patchset aims to constify all snd_soc_dai_ops structures that are only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Thanks! Gustavo A. R. Silva (12): ASoC: cs42l42: constify snd_soc_dai_ops structure ASoC: rt5663: constify snd_soc_dai_ops structure ASoC: rt5616: constify snd_soc_dai_ops structure ASoC: msm8916-wcd-analog: constify snd_soc_dai_ops structure ASoC: max98926: constify snd_soc_dai_ops structure ASoC: rt5514: constify snd_soc_dai_ops structure ASoC: codecs: msm8916-wcd-digital: constify snd_soc_dai_ops structure ASoC: max9867: constify snd_soc_dai_ops structure ASoC: rk3036: constify snd_soc_dai_ops structure ASoC: hdac_hdmi: constify snd_soc_dai_ops structure ASoC: dwc: constify snd_soc_dai_ops structure ASoC: tas5720: constify snd_soc_dai_ops structure sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/dwc/dwc-i2s.c | 2 +- 12 files changed, 12 insertions(+), 12 deletions(-) -- 2.5.0

2 24

[alsa-devel] Applied "ASoC: rsnd: remove unnecessary static in rsnd_ssiu_probe()" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: rsnd: remove unnecessary static in rsnd_ssiu_probe() has been applied to the asoc tree at git://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 32e1b60d8f91be932c4a624d4153224895b65ec7 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Mon, 17 Jul 2017 10:30:20 -0500 Subject: [PATCH] ASoC: rsnd: remove unnecessary static in rsnd_ssiu_probe() Remove unnecessary static on local variable ops. Such variable is initialized before being used, on every execution path throughout the function. The static has no benefit and, removing it reduces the code size. This issue was detected using Coccinelle and the following semantic patch: @bad exists@ position p; identifier x; type T; @@ static T x@p; ... x = <+...x...+> @@ identifier x; expression e; type T; position p != bad.p; @@ -static T x@p; ... when != x when strict ?x = e; In the following log you can see the difference in the code size. Also, there is a significant difference in the bss segment. This log is the output of the size command, before and after the code change: before: text data bss dec hex filename 3211 680 64 3955 f73 sound/soc/sh/rcar/ssiu.o after: text data bss dec hex filename 3207 592 0 3799 ed7 sound/soc/sh/rcar/ssiu.o Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index bed2c9c0004b..4d948757d300 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -250,7 +250,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssiu *ssiu; - static struct rsnd_mod_ops *ops; + struct rsnd_mod_ops *ops; int i, nr, ret; /* same number to SSI */ -- 2.13.2

1 0

[alsa-devel] [PATCH 0/3] ASoC: codecs: msm8916-wcd-digital: Fix dmic support
by srinivas.kandagatla＠linaro.org 17 Jul '17

17 Jul '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset fixes dmic support in the codec driver, first and second patch fixes the mclk rate for the codec and the third patch adds few missing register writes to select dmic. Tested on DB410c with Audio Mezz board which has got dmic on it. Srinivas Kandagatla (3): ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk ASoC: qcom: apq8016-sbc: set default mclk rate ASoC: codecs: msm8916-wcd-digital: fix dmic support sound/soc/codecs/msm8916-wcd-digital.c | 36 ++++++++++++++++++++++++++++++++++ sound/soc/qcom/apq8016_sbc.c | 6 ++++++ 2 files changed, 42 insertions(+) -- 2.11.0

3 7

[alsa-devel] ALC298 headset mode update
by Kailang 17 Jul '17

17 Jul '17

Hi Takashi, Dell new platform with new hardware layout. Old headset mode policy will cause headset mode check type has a chance to check fail. So, we update new headset mode to match new hardware layout. New headset mode policy also could suitable to old hardware layout. This patch was tested by Ubuntu QA team. BR, Kailang

2 1

Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

Hi Mark, Joe, On 07/14/2017 06:25 AM, Mark Brown wrote: > On Fri, Jul 14, 2017 at 04:08:21AM -0700, Joe Perches wrote: >> On Fri, 2017-07-14 at 12:02 +0100, Mark Brown wrote: >>> On Thu, Jul 13, 2017 at 11:18:11AM -0700, Joe Perches wrote: >>>> I don't it's better. >>>> It's not that confusing if the 0/n patch cover letter is cc'd >>>> to all the appropriate mailing lists and all the [1..n]/n >>>> patches are sent with in-reply-to of the cover letter and >>>> send to the maintainers and appropriate mailing lists. >>> With large serieses like Gustavo is sending the CC list can easily hit >>> the points where mailing lists start blocking it, and the individual >>> pathces really do need to go to the relevant people so they have sight >>> of them. >> I agree and that's what I wrote. > The set of people who should have sight of the patches is wider than > just the maintainers. I'm running get_maintainer.pl in the following way in order to get all the supporters, maintainers and lists: $ scripts/get_maintainer.pl --nokeywords --nogit --nogit-fallback <file> Thank you -- Gustavo A. R. Silva

1 0

[alsa-devel] [PATCH 0/3] constify dev_pm_ops structures
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

This patchset aims to constify all dev_pm_ops structures that only stored in the pm field of a device_driver structure. This field is declared const, so dev_pm_ops structures that have this property can be declared as const also. Thanks! Gustavo A. R. Silva (3): ASoC: fsi: constify dev_pm_ops structure ASoC: psc-ac97: constify dev_pm_ops structure ASoC: au1x: psc-i2s: constify dev_pm_ops structure sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/sh/fsi.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) -- 2.5.0

1 3

[alsa-devel] [PATCH] ALSA: hda - Add hdmi id for GeForce GT 1030
by Jan Andres 16 Jul '17

16 Jul '17

HDMI audio gets routed to the wrong HDMI/DP ports on the GT 1030. Fix this by adding a patch entry referring to patch_nvhdmi(), like we do for other NVidia cards. Signed-off-by: Jan Andres <jandres(a)gmx.net> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d549f35f39d3..2a583c95edf6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3772,6 +3772,7 @@ HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0080, "GPU 80 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0081, "GPU 81 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0082, "GPU 82 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), -- 2.13.3

2 1

Re: [alsa-devel] [PATCH - alsa-lib 1/1] snd_user_file: avoid use wordexp
by Takashi Iwai 15 Jul '17

15 Jul '17

On Wed, 12 Jul 2017 09:53:08 +0200, Natanael Copa wrote: > > As suggested in POSIX[1], wordexp might execute the shell. If the libc > implementation does so, it will break the firefox sandbox which does > not allow exec. This happened on Alpine Linux with musl libc[2]. > > Since we cannot guarantee that the system wordexp implementation does > not execute shell, we cannot really use it, and need to implement the > ~/ expansion ourselves. > > Generally, wordexp is a large attack vector and it is better to avoid it > since only tilde expansion is needed. > > [1]: http://pubs.opengroup.org/onlinepubs/9699919799/functions/wordexp.html#tag_… > [2]: http://bugs.alpinelinux.org/issues/7454#note-2 > > Signed-off-by: Natanael Copa <ncopa(a)alpinelinux.org> Well, this may bring a functional regression, so I don't want to introduce this unconditionally. Instead, we may introduce configure option to explicitly control the usage of wordexp(), and add your extension to the fallback code, for example. If the security issue really matters, we can leave it disabled as default, too. thanks, Takashi > > diff --git a/configure.ac b/configure.ac > index 26e5d125..076c6bab 100644 > --- a/configure.ac > +++ b/configure.ac > @@ -304,7 +304,7 @@ fi > AC_SUBST(ALSA_DEPLIBS) > > dnl Check for headers > -AC_CHECK_HEADERS([wordexp.h endian.h sys/endian.h sys/shm.h]) > +AC_CHECK_HEADERS([endian.h sys/endian.h sys/shm.h]) > > dnl Check for resmgr support... > AC_MSG_CHECKING(for resmgr support) > diff --git a/src/userfile.c b/src/userfile.c > index 72779da4..0e3f5fae 100644 > --- a/src/userfile.c > +++ b/src/userfile.c > @@ -21,6 +21,11 @@ > #include <config.h> > #include <string.h> > #include <errno.h> > +#include <sys/types.h> > +#include <unistd.h> > +#include <pwd.h> > +#include <stdio.h> > +#include <stdlib.h> > > /** > * \brief Get the full file name > @@ -28,46 +33,58 @@ > * \param result The pointer to store the resultant file name > * \return 0 if successful, or a negative error code > * > - * Parses the given file name with POSIX-Shell-like expansion and > - * stores the first matchine one. The returned string is strdup'ed. > + * Parses the given file name with POSIX-Shell-like expansion for ~/. > + * The returned string is strdup'ed. > */ > > -#ifdef HAVE_WORDEXP_H > -#include <wordexp.h> > #include <assert.h> > int snd_user_file(const char *file, char **result) > { > - wordexp_t we; > int err; > - > + size_t len; > + char *buf = NULL; > + > assert(file && result); > - err = wordexp(file, &we, WRDE_NOCMD); > - switch (err) { > - case WRDE_NOSPACE: > - wordfree(&we); > - return -ENOMEM; > - case 0: > - if (we.we_wordc == 1) > - break; > - wordfree(&we); > - /* fall thru */ > - default: > - return -EINVAL; > + *result = NULL; > + > + /* expand ~/ if needed */ > + if (file[0] == '~' && file[1] == '/') { > + const char *home = getenv("HOME"); > + if (home == NULL) { > + struct passwd pwent, *p = NULL; > + uid_t id = getuid(); > + size_t bufsize = 1024; > + > + buf = malloc(bufsize); > + if (buf == NULL) > + goto out; > + > + while ((err = getpwuid_r(id, &pwent, buf, bufsize, &p)) == ERANGE) { > + char *newbuf; > + bufsize += 1024; > + if (bufsize < 1024) > + break; > + newbuf = realloc(buf, bufsize); > + if (newbuf == NULL) > + goto out; > + buf = newbuf; > + } > + home = err ? "" : pwent.pw_dir; > + } > + len = strlen(home) + strlen(&file[2]) + 2; > + *result = malloc(len); > + if (*result) > + snprintf(*result, len, "%s/%s", home, &file[2]); > + } else { > + *result = strdup(file); > } > - *result = strdup(we.we_wordv[0]); > - wordfree(&we); > + > +out: > + if (buf) > + free(buf); > + > if (*result == NULL) > return -ENOMEM; > return 0; > } > > -#else /* !HAVE_WORDEXP_H */ > -/* just copy the string - would be nicer to expand by ourselves, though... */ > -int snd_user_file(const char *file, char **result) > -{ > - *result = strdup(file); > - if (! *result) > - return -ENOMEM; > - return 0; > -} > -#endif /* HAVE_WORDEXP_H */ > -- > 2.13.2 >

2 4

[alsa-devel] [PATCH - alsa-lib 1/2] cleanup: fix poll.h includes
by Natanael Copa 15 Jul '17

15 Jul '17

According POSIX[1] and linux manpage[2] the include is poll.h, not sys/poll.h. This fixes the he following compiler warning when build with musl libc: /usr/include/sys/poll.h:1:2: warning: #warning redirecting incorrect #include <sys/poll.h> to <poll.h> [-Wcpp] #warning redirecting incorrect #include <sys/poll.h> to <poll.h> ^~~~~~~ Signed-off-by: Natanael Copa <ncopa(a)alpinelinux.org> diff --git a/aserver/aserver.c b/aserver/aserver.c index ac20706b..46f731a4 100644 --- a/aserver/aserver.c +++ b/aserver/aserver.c @@ -20,7 +20,7 @@ #include <sys/shm.h> #include <sys/socket.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/un.h> #include <sys/uio.h> #include <stdio.h> diff --git a/include/asoundlib-head.h b/include/asoundlib-head.h index 1ec611e5..21e32c6b 100644 --- a/include/asoundlib-head.h +++ b/include/asoundlib-head.h @@ -35,6 +35,6 @@ #include <string.h> #include <fcntl.h> #include <assert.h> -#include <sys/poll.h> +#include <poll.h> #include <errno.h> #include <stdarg.h> diff --git a/include/local.h b/include/local.h index 317f2e32..6a43a473 100644 --- a/include/local.h +++ b/include/local.h @@ -47,7 +47,7 @@ #error Header defining endianness not defined #endif #include <stdarg.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/types.h> #include <errno.h> #if defined(__linux__) diff --git a/src/control/control.c b/src/control/control.c index 134ba4c8..6439b294 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -90,7 +90,7 @@ against the original design. #include <string.h> #include <fcntl.h> #include <signal.h> -#include <sys/poll.h> +#include <poll.h> #include <stdbool.h> #include "control_local.h" diff --git a/src/control/control_shm.c b/src/control/control_shm.c index bd07d4af..9a2e268b 100644 --- a/src/control/control_shm.c +++ b/src/control/control_shm.c @@ -27,7 +27,7 @@ #include <fcntl.h> #include <sys/shm.h> #include <sys/socket.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/un.h> #include <sys/uio.h> #include <sys/mman.h> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 200b10c2..2b4ce8ec 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -651,7 +651,7 @@ playback devices. #include <stdarg.h> #include <signal.h> #include <ctype.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/mman.h> #include <limits.h> #include "pcm_local.h" diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 9fd376d8..393083f5 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -30,7 +30,7 @@ #include <grp.h> #include <sys/ioctl.h> #include <sys/mman.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/shm.h> #include <sys/sem.h> #include <sys/wait.h> diff --git a/src/pcm/pcm_mmap.c b/src/pcm/pcm_mmap.c index 1948289c..4cf220a4 100644 --- a/src/pcm/pcm_mmap.c +++ b/src/pcm/pcm_mmap.c @@ -22,7 +22,7 @@ #include <stdio.h> #include <malloc.h> #include <string.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/mman.h> #ifdef HAVE_SYS_SHM_H #include <sys/shm.h> diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c index 5d8aaf21..21a57fc6 100644 --- a/src/pcm/pcm_share.c +++ b/src/pcm/pcm_share.c @@ -34,7 +34,7 @@ #include <signal.h> #include <math.h> #include <sys/socket.h> -#include <sys/poll.h> +#include <poll.h> #include <pthread.h> #include "pcm_local.h" diff --git a/src/pcm/pcm_shm.c b/src/pcm/pcm_shm.c index a815ac6b..4ee958c1 100644 --- a/src/pcm/pcm_shm.c +++ b/src/pcm/pcm_shm.c @@ -36,7 +36,7 @@ #include <sys/ioctl.h> #include <sys/shm.h> #include <sys/socket.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/un.h> #include <sys/mman.h> #include <netinet/in.h> diff --git a/src/seq/seq.c b/src/seq/seq.c index b206e2f8..d5ed1c6a 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -777,7 +777,7 @@ void event_filter(snd_seq_t *seq, snd_seq_event_t *ev) */ -#include <sys/poll.h> +#include <poll.h> #include "seq_local.h" /**************************************************************************** diff --git a/src/shmarea.c b/src/shmarea.c index 9843aa8b..eaa71f00 100644 --- a/src/shmarea.c +++ b/src/shmarea.c @@ -27,7 +27,7 @@ #include <malloc.h> #include <string.h> #include <errno.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/mman.h> #include <sys/shm.h> #include "list.h" -- 2.13.2

2 9

[alsa-devel] [PATCH - pulse 1/1] pulse: prevent double-free when pulse_hw_constraint returns error
by ? ? 15 Jul '17

15 Jul '17

From: Kui Wang <wangkuisuper(a)hotmail.com> When pulse_hw_constraint returns error, snd_pcm_ioplug_delete() is called. It will then call pulse_close() where "snd_pcm_pulse_t *pcm" will be free. Then if goto the "error" label, the "snd_pcm_pulse_t *pcm" will be double-free. To prevent this, just jump over the code which might cause double-free. Signed-off-by: Kui Wang <wangkuisuper(a)hotmail.com> diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 5cb3452..a8983c6 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -1143,7 +1143,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) err = pulse_hw_constraint(pcm); if (err < 0) { snd_pcm_ioplug_delete(&pcm->io); - goto error; + goto error2; } *pcmp = pcm->io.pcm; @@ -1156,6 +1156,7 @@ error: free(pcm->device); free(pcm); +error2: if (fallback_name) return snd_pcm_open_fallback(pcmp, root, fallback_name, name, stream, mode); -- 2.13.2.windows.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add missing NVIDIA GPU codec IDs to patch table
by Daniel Dadap 15 Jul '17

15 Jul '17

Add codec IDs for several recently released, pending, and historical NVIDIA GPU audio controllers to the patch table, to allow the correct patch functions to be selected for them. Signed-off-by: Daniel Dadap <ddadap(a)nvidia.com> Reviewed-by: Andy Ritger <aritger(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 76c85f08bea6..0328561f20a1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3731,11 +3731,15 @@ HDA_CODEC_ENTRY(0x1002aa01, "R6xx HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x10951390, "SiI1390 HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x10951392, "SiI1392 HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x17e80047, "Chrontel HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x10de0001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de0002, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), HDA_CODEC_ENTRY(0x10de0003, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), +HDA_CODEC_ENTRY(0x10de0004, "GPU 04 HDMI", patch_nvhdmi_8ch_7x), HDA_CODEC_ENTRY(0x10de0005, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), HDA_CODEC_ENTRY(0x10de0006, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), HDA_CODEC_ENTRY(0x10de0007, "MCP79/7A HDMI", patch_nvhdmi_8ch_7x), +HDA_CODEC_ENTRY(0x10de0008, "GPU 08 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0009, "GPU 09 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de000a, "GPU 0a HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de000b, "GPU 0b HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de000c, "MCP89 HDMI", patch_nvhdmi), @@ -3762,17 +3766,40 @@ HDA_CODEC_ENTRY(0x10de0041, "GPU 41 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0042, "GPU 42 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0043, "GPU 43 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0044, "GPU 44 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0045, "GPU 45 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0050, "GPU 50 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0051, "GPU 51 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0052, "GPU 52 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0060, "GPU 60 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0061, "GPU 61 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0062, "GPU 62 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0067, "MCP67 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0073, "GPU 73 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0074, "GPU 74 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0076, "GPU 76 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de007b, "GPU 7b HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de007c, "GPU 7c HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de007e, "GPU 7e HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0080, "GPU 80 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0081, "GPU 81 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0082, "GPU 82 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0084, "GPU 84 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0090, "GPU 90 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0091, "GPU 91 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0092, "GPU 92 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0093, "GPU 93 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0094, "GPU 94 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: rsnd: add missing of_node_put
by Julia Lawall 15 Jul '17

15 Jul '17

for_each_child_of_node performs an of_node_get on each iteration, so a jump out of the loop requires an of_node_put. The semantic patch that fixes this problem is as follows (http://coccinelle.lip6.fr): // <smpl> @@ local idexpression n; expression e,e1; identifier l; @@ for_each_child_of_node(e1,n) { ... ( of_node_put(n); | e = n | + of_node_put(n); ? goto l; ) ... } ... l: ... when != n // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/sh/rcar/ctu.c | 5 ++++- sound/soc/sh/rcar/dvc.c | 5 ++++- sound/soc/sh/rcar/mix.c | 5 ++++- sound/soc/sh/rcar/src.c | 6 +++++- sound/soc/sh/rcar/ssi.c | 6 +++++- 5 files changed, 22 insertions(+), 5 deletions(-) diff -u -p a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -168,13 +168,16 @@ int rsnd_mix_probe(struct rsnd_priv *pri clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_mix_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops, clk, rsnd_mod_get_status, RSND_MOD_MIX, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_mix_probe_done; + } i++; } diff -u -p a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -394,13 +394,16 @@ int rsnd_ctu_probe(struct rsnd_priv *pri clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_ctu_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops, clk, rsnd_mod_get_status, RSND_MOD_CTU, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_ctu_probe_done; + } i++; } diff -u -p a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1079,6 +1079,7 @@ int rsnd_ssi_probe(struct rsnd_priv *pri clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_ssi_probe_done; } @@ -1091,6 +1092,7 @@ int rsnd_ssi_probe(struct rsnd_priv *pri ssi->irq = irq_of_parse_and_map(np, 0); if (!ssi->irq) { ret = -EINVAL; + of_node_put(np); goto rsnd_ssi_probe_done; } @@ -1101,8 +1103,10 @@ int rsnd_ssi_probe(struct rsnd_priv *pri ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk, rsnd_ssi_get_status, RSND_MOD_SSI, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_ssi_probe_done; + } i++; } diff -u -p a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -581,20 +581,24 @@ int rsnd_src_probe(struct rsnd_priv *pri src->irq = irq_of_parse_and_map(np, 0); if (!src->irq) { ret = -EINVAL; + of_node_put(np); goto rsnd_src_probe_done; } clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_src_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(src), &rsnd_src_ops, clk, rsnd_mod_get_status, RSND_MOD_SRC, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_src_probe_done; + } skip: i++; diff -u -p a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -380,13 +380,16 @@ int rsnd_dvc_probe(struct rsnd_priv *pri clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_dvc_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops, clk, rsnd_mod_get_status, RSND_MOD_DVC, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_dvc_probe_done; + } i++; }

1 0

[alsa-devel] [PATCH v2 00/18] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 14 Jul '17

14 Jul '17

The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Summary of main changes since V1: - dt bindings moved to separate patches - pdata struct descriptions fixed to be kerneldoc format - now re-uses the arizona micsupp and ldo1 regulators - various fixes and improvements to the pinctrl driver - gpio driver can now interoperate with the pinctrl driver - some changes to the way Kconfig options are selected Richard Fitzgerald (18): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs dt-bindings: mfd: Add bindings for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs dt-bindings: pinctrl: Add bindings for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs dt-bindings: gpio: Add bindings for GPIO on Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs dt-bindings: sound: Add bindings for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 .../devicetree/bindings/gpio/gpio-madera.txt | 27 + Documentation/devicetree/bindings/mfd/madera.txt | 96 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 102 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 66 + MAINTAINERS | 24 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 349 + drivers/mfd/Kconfig | 50 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1688 ++++ drivers/mfd/cs47l85-tables.c | 3169 +++++++ drivers/mfd/cs47l90-tables.c | 2830 +++++++ drivers/mfd/madera-core.c | 635 ++ drivers/mfd/madera-i2c.c | 131 + drivers/mfd/madera-spi.c | 132 + drivers/mfd/madera.h | 52 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1106 +++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 18 + include/linux/irqchip/irq-madera-pdata.h | 19 + include/linux/irqchip/irq-madera.h | 96 + include/linux/mfd/madera/core.h | 180 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 8832 ++++++++++++++++++++ include/sound/madera-pdata.h | 58 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1747 ++++ sound/soc/codecs/cs47l85.c | 2706 ++++++ sound/soc/codecs/cs47l90.c | 2645 ++++++ sound/soc/codecs/madera.c | 4430 ++++++++++ sound/soc/codecs/madera.h | 470 ++ 47 files changed, 32363 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/gpio-madera.txt create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 1.9.1

8 49

Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: constify snd_soc_dai_ops structure
by Joe Perches 14 Jul '17

14 Jul '17

On Fri, 2017-07-14 at 12:02 +0100, Mark Brown wrote: > On Thu, Jul 13, 2017 at 11:18:11AM -0700, Joe Perches wrote: > > > I don't it's better. > > It's not that confusing if the 0/n patch cover letter is cc'd > > to all the appropriate mailing lists and all the [1..n]/n > > patches are sent with in-reply-to of the cover letter and > > send to the maintainers and appropriate mailing lists. > > With large serieses like Gustavo is sending the CC list can easily hit > the points where mailing lists start blocking it, and the individual > pathces really do need to go to the relevant people so they have sight > of them. I agree and that's what I wrote.

1 0

[alsa-devel] [PATCH 0/2] Patches for salsa-lib
by Felix Hädicke 14 Jul '17

14 Jul '17

Hi, Here are two little patches for salsa-lib. I would be happy if these could be included in the next release. And I saw that in the released salsa-lib 0.1.6 tarball, the generated headers asoundlib.h, version.h and recipe.h are included. I guess this is by mistake? This caused unexpected behaviour for me, because when performing an out-of-tree build, these are used instead of the generated headers in the build directory. Regards, Felix -- 2.13.2

2 9

Re: [alsa-devel] [PATCH] ASoC: rt5514: Support the DSP recording continuously after the hotwording triggered
by Mark Brown 13 Jul '17

13 Jul '17

On Thu, Jul 13, 2017 at 02:01:51AM +0000, Oder Chiou wrote: > > > +static irqreturn_t rt5514_spi_irq(int irq, void *data) { > > > + struct rt5514_dsp *rt5514_dsp = data; > > > + > > > + schedule_work(&rt5514_dsp->start_work); > > > + > > > + return IRQ_HANDLED; > > > +} > > Why not just use a threaded interrupt handler for the start_work? > > That'd get you into process context. > Because of the SPI related function the work thread should be manipulated > by the interrupt handler function. If the SPI related function is > implemented to the interrupt handler directly, it will cause the runtime > error while the interrupt handler is triggered. Why will this cause an error?

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Add hdmi id for a Geminilake variant
by Subhransu S. Prusty 12 Jul '17

12 Jul '17

Few GLK platform variants report a different vendor id. Add it. Also add the missing check for GLK in is_haswell_plus(). Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v1 -> v2: Fix a typo in commit message update the commit message to add glk check in is_haswell_plus() sound/pci/hda/patch_hdmi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 76c85f08bea6..d549f35f39d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -53,9 +53,11 @@ #define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809) #define is_broxton(codec) ((codec)->core.vendor_id == 0x8086280a) #define is_kabylake(codec) ((codec)->core.vendor_id == 0x8086280b) +#define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ + ((codec)->core.vendor_id == 0x80862800)) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) + || is_kabylake(codec)) || is_geminilake(codec) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) @@ -3790,6 +3792,7 @@ static int patch_via_hdmi(struct hda_codec *codec) HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), +HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: hda - Add hdmi id for a Geminilake variant
by Subhransu S. Prusty 12 Jul '17

12 Jul '17

Few GLK platform variants report a different vendor nid. Add it. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 76c85f08bea6..d549f35f39d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -53,9 +53,11 @@ #define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809) #define is_broxton(codec) ((codec)->core.vendor_id == 0x8086280a) #define is_kabylake(codec) ((codec)->core.vendor_id == 0x8086280b) +#define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ + ((codec)->core.vendor_id == 0x80862800)) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) + || is_kabylake(codec)) || is_geminilake(codec) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) @@ -3790,6 +3792,7 @@ static int patch_via_hdmi(struct hda_codec *codec) HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), +HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rt5514: Support the DSP recording continuously after the hotwording triggered
by Oder Chiou 12 Jul '17

12 Jul '17

The patch uses the IRQ to copy the PCM data to userspace continuously after the hotwording triggered from DSP. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 147 +++++++++++++++++++++++------------------- sound/soc/codecs/rt5514-spi.h | 7 +- 2 files changed, 84 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 7ed62e8..5884850 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -39,13 +39,12 @@ static struct spi_device *rt5514_spi; struct rt5514_dsp { struct device *dev; + struct work_struct start_work; struct delayed_work copy_work; struct mutex dma_lock; struct snd_pcm_substream *substream; unsigned int buf_base, buf_limit, buf_rp; - size_t buf_size; - size_t dma_offset; - size_t dsp_offset; + size_t buf_size, get_size, dma_offset; }; static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { @@ -80,6 +79,8 @@ static void rt5514_spi_copy_work(struct work_struct *work) container_of(work, struct rt5514_dsp, copy_work.work); struct snd_pcm_runtime *runtime; size_t period_bytes, truncated_bytes = 0; + unsigned int cur_wp, remain_data; + u8 buf[8]; mutex_lock(&rt5514_dsp->dma_lock); if (!rt5514_dsp->substream) { @@ -90,8 +91,22 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); - if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) - period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset; + if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) { + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, + sizeof(buf)); + cur_wp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + remain_data = (cur_wp >= rt5514_dsp->buf_rp) ? + (cur_wp - rt5514_dsp->buf_rp) : + ((rt5514_dsp->buf_limit - rt5514_dsp->buf_rp) + + (cur_wp - rt5514_dsp->buf_base)); + + if (remain_data < period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + } if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { rt5514_spi_burst_read(rt5514_dsp->buf_rp, @@ -112,24 +127,57 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime->dma_area + rt5514_dsp->dma_offset + truncated_bytes, period_bytes - truncated_bytes); - rt5514_dsp->buf_rp = rt5514_dsp->buf_base + - period_bytes - truncated_bytes; + rt5514_dsp->buf_rp = rt5514_dsp->buf_base + period_bytes - + truncated_bytes; } + rt5514_dsp->get_size += period_bytes; rt5514_dsp->dma_offset += period_bytes; if (rt5514_dsp->dma_offset >= runtime->dma_bytes) rt5514_dsp->dma_offset = 0; - rt5514_dsp->dsp_offset += period_bytes; - snd_pcm_period_elapsed(rt5514_dsp->substream); - if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size) - schedule_delayed_work(&rt5514_dsp->copy_work, 5); + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + done: mutex_unlock(&rt5514_dsp->dma_lock); } +static void rt5514_spi_start_work(struct work_struct *work) +{ + struct rt5514_dsp *rt5514_dsp = + container_of(work, struct rt5514_dsp, start_work); + u8 buf[8]; + + rt5514_dsp->get_size = 0; + rt5514_dsp->dma_offset = 0; + + /** + * The address area x1800XXXX is the register address, and it cannot + * support spi burst read perfectly. So we use the spi burst read + * individually to make sure the data correctly. + */ + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; + + schedule_delayed_work(&rt5514_dsp->copy_work, 0); +} + /* PCM for streaming audio from the DSP buffer */ static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) { @@ -170,59 +218,6 @@ static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_vmalloc_buffer(substream); } -static int rt5514_spi_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct rt5514_dsp *rt5514_dsp = - snd_soc_platform_get_drvdata(rtd->platform); - u8 buf[8]; - - rt5514_dsp->dma_offset = 0; - rt5514_dsp->dsp_offset = 0; - - /** - * The address area x1800XXXX is the register address, and it cannot - * support spi burst read perfectly. So we use the spi burst read - * individually to make sure the data correctly. - */ - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - return 0; -} - -static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct rt5514_dsp *rt5514_dsp = - snd_soc_platform_get_drvdata(rtd->platform); - - if (cmd == SNDRV_PCM_TRIGGER_START) { - if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && - rt5514_dsp->buf_rp && rt5514_dsp->buf_size) - schedule_delayed_work(&rt5514_dsp->copy_work, 0); - } - - return 0; -} - static snd_pcm_uframes_t rt5514_spi_pcm_pointer( struct snd_pcm_substream *substream) { @@ -238,25 +233,45 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = { .open = rt5514_spi_pcm_open, .hw_params = rt5514_spi_hw_params, .hw_free = rt5514_spi_hw_free, - .trigger = rt5514_spi_trigger, - .prepare = rt5514_spi_prepare, .pointer = rt5514_spi_pcm_pointer, .mmap = snd_pcm_lib_mmap_vmalloc, .page = snd_pcm_lib_get_vmalloc_page, }; +static irqreturn_t rt5514_spi_irq(int irq, void *data) +{ + struct rt5514_dsp *rt5514_dsp = data; + + schedule_work(&rt5514_dsp->start_work); + + return IRQ_HANDLED; +} + static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) { struct rt5514_dsp *rt5514_dsp; + int ret; rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), GFP_KERNEL); rt5514_dsp->dev = &rt5514_spi->dev; mutex_init(&rt5514_dsp->dma_lock); + INIT_WORK(&rt5514_dsp->start_work, rt5514_spi_start_work); INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); snd_soc_platform_set_drvdata(platform, rt5514_dsp); + if (rt5514_spi->irq) { + ret = devm_request_threaded_irq(&rt5514_spi->dev, + rt5514_spi->irq, NULL, rt5514_spi_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5514-spi", + rt5514_dsp); + if (ret) + dev_err(&rt5514_spi->dev, + "%s Failed to reguest IRQ: %d\n", __func__, + ret); + } + return 0; } diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h index f69b1cd..a6434ee 100644 --- a/sound/soc/codecs/rt5514-spi.h +++ b/sound/soc/codecs/rt5514-spi.h @@ -17,10 +17,9 @@ */ #define RT5514_SPI_BUF_LEN 240 -#define RT5514_BUFFER_VOICE_BASE 0x18001034 -#define RT5514_BUFFER_VOICE_LIMIT 0x18001038 -#define RT5514_BUFFER_VOICE_RP 0x1800103c -#define RT5514_BUFFER_VOICE_SIZE 0x18001040 +#define RT5514_BUFFER_VOICE_BASE 0x18000200 +#define RT5514_BUFFER_VOICE_LIMIT 0x18000204 +#define RT5514_BUFFER_VOICE_WP 0x1800020c /* SPI Command */ enum { -- 2.7.4

2 1

[alsa-devel] Slow/piecewise send of sysex?
by Ville Oikarinen 12 Jul '17

12 Jul '17

Hello. I just created this question as a ticket https://github.com/opensrc/alsa/issues/24 but was instructed to ask here. So: I'm using amidi --send-hex to send data to a Commodore 64 via a CLab MIDI interface. The problem is that even with 100 bytes the probability of a buffer overrun is about 30%. I didn't find a way to throttle the send speed. I also tried sending the data in smaller chunks, but I cannot get amidi to send anything after the first chunk. I suspect it refuses to send anything if the data doesn't start with F0 i.e. sysex dump start. If I am right about that assumption, as a workaround I could encapsulate each chunk as a full sysex dump, but that requires extra logic on the receiving side and it also wastes bandwidth, so I don't really wish to do that. Primarily I would prefer a way to throttle the speed, but if I cannot have that, I would like to be able to send the data a few bytes at a time in a simple shell loop. - Ville Oikarinen

2 2

[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Support more resolution rates
by Jaechul Lee 12 Jul '17

12 Jul '17

This driver can support more frequencies over 96KHz. There are no reasons to limit the frequency range below 96KHz. If codecs/amps or something else can't support higher resolution rates, the constraints would be set rates properly because each drivers have its own limits. I added the 'pcm_rates' field to the dai_data to be set rates by the compatibilities. As a result, rates will be set each devices respectively. For example of exynos5433, rates will be set from 8KHz to 192KHz. Signed-off-by: Jaechul Lee <jcsing.lee(a)samsung.com> --- v2: - changed the name of variable to pcm_rates. - removed duplicated code. - modified commit message. --- sound/soc/samsung/i2s.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d4ccc5..c9f87f7bae99 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -50,6 +50,7 @@ struct samsung_i2s_variant_regs { struct samsung_i2s_dai_data { u32 quirks; + unsigned int pcm_rates; const struct samsung_i2s_variant_regs *i2s_variant_regs; }; @@ -1076,13 +1077,13 @@ static const struct snd_soc_component_driver samsung_i2s_component = { .name = "samsung-i2s", }; -#define SAMSUNG_I2S_RATES SNDRV_PCM_RATE_8000_96000 - #define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) +static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, + const struct samsung_i2s_dai_data *i2s_dai_data, + bool sec) { struct i2s_dai *i2s; @@ -1101,13 +1102,13 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.resume = i2s_resume; i2s->i2s_dai_drv.playback.channels_min = 1; i2s->i2s_dai_drv.playback.channels_max = 2; - i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES; + i2s->i2s_dai_drv.playback.rates = i2s_dai_data->pcm_rates; i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; if (!sec) { i2s->i2s_dai_drv.capture.channels_min = 1; i2s->i2s_dai_drv.capture.channels_max = 2; - i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; + i2s->i2s_dai_drv.capture.rates = i2s_dai_data->pcm_rates; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; } return i2s; @@ -1242,7 +1243,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) i2s_dai_data = (struct samsung_i2s_dai_data *) platform_get_device_id(pdev)->driver_data; - pri_dai = i2s_alloc_dai(pdev, false); + pri_dai = i2s_alloc_dai(pdev, i2s_dai_data, false); if (!pri_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_pri\n"); return -ENOMEM; @@ -1316,7 +1317,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) goto err_disable_clk; if (quirks & QUIRK_SEC_DAI) { - sec_dai = i2s_alloc_dai(pdev, true); + sec_dai = i2s_alloc_dai(pdev, i2s_dai_data, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); ret = -ENOMEM; @@ -1452,29 +1453,34 @@ static const struct samsung_i2s_variant_regs i2sv5_i2s1_regs = { static const struct samsung_i2s_dai_data i2sv3_dai_type = { .quirks = QUIRK_NO_MUXPSR, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv5_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_IDMA, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv6_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_TDM | QUIRK_SUPPORTS_IDMA, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv6_regs, }; static const struct samsung_i2s_dai_data i2sv7_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_TDM, + .pcm_rates = SNDRV_PCM_RATE_8000_192000, .i2s_variant_regs = &i2sv7_regs, }; static const struct samsung_i2s_dai_data i2sv5_dai_type_i2s1 = { .quirks = QUIRK_PRI_6CHAN | QUIRK_NEED_RSTCLR, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv5_i2s1_regs, }; -- 2.11.0

5 4

[alsa-devel] [PATCH - alsa-lib 1/1] snd_user_file: avoid use wordexp
by Natanael Copa 12 Jul '17

12 Jul '17

As suggested in POSIX[1], wordexp might execute the shell. If the libc implementation does so, it will break the firefox sandbox which does not allow exec. This happened on Alpine Linux with musl libc[2]. Since we cannot guarantee that the system wordexp implementation does not execute shell, we cannot really use it, and need to implement the ~/ expansion ourselves. Generally, wordexp is a large attack vector and it is better to avoid it since only tilde expansion is needed. [1]: http://pubs.opengroup.org/onlinepubs/9699919799/functions/wordexp.html#tag_… [2]: http://bugs.alpinelinux.org/issues/7454#note-2 Signed-off-by: Natanael Copa <ncopa(a)alpinelinux.org> diff --git a/configure.ac b/configure.ac index 26e5d125..076c6bab 100644 --- a/configure.ac +++ b/configure.ac @@ -304,7 +304,7 @@ fi AC_SUBST(ALSA_DEPLIBS) dnl Check for headers -AC_CHECK_HEADERS([wordexp.h endian.h sys/endian.h sys/shm.h]) +AC_CHECK_HEADERS([endian.h sys/endian.h sys/shm.h]) dnl Check for resmgr support... AC_MSG_CHECKING(for resmgr support) diff --git a/src/userfile.c b/src/userfile.c index 72779da4..0e3f5fae 100644 --- a/src/userfile.c +++ b/src/userfile.c @@ -21,6 +21,11 @@ #include <config.h> #include <string.h> #include <errno.h> +#include <sys/types.h> +#include <unistd.h> +#include <pwd.h> +#include <stdio.h> +#include <stdlib.h> /** * \brief Get the full file name @@ -28,46 +33,58 @@ * \param result The pointer to store the resultant file name * \return 0 if successful, or a negative error code * - * Parses the given file name with POSIX-Shell-like expansion and - * stores the first matchine one. The returned string is strdup'ed. + * Parses the given file name with POSIX-Shell-like expansion for ~/. + * The returned string is strdup'ed. */ -#ifdef HAVE_WORDEXP_H -#include <wordexp.h> #include <assert.h> int snd_user_file(const char *file, char **result) { - wordexp_t we; int err; - + size_t len; + char *buf = NULL; + assert(file && result); - err = wordexp(file, &we, WRDE_NOCMD); - switch (err) { - case WRDE_NOSPACE: - wordfree(&we); - return -ENOMEM; - case 0: - if (we.we_wordc == 1) - break; - wordfree(&we); - /* fall thru */ - default: - return -EINVAL; + *result = NULL; + + /* expand ~/ if needed */ + if (file[0] == '~' && file[1] == '/') { + const char *home = getenv("HOME"); + if (home == NULL) { + struct passwd pwent, *p = NULL; + uid_t id = getuid(); + size_t bufsize = 1024; + + buf = malloc(bufsize); + if (buf == NULL) + goto out; + + while ((err = getpwuid_r(id, &pwent, buf, bufsize, &p)) == ERANGE) { + char *newbuf; + bufsize += 1024; + if (bufsize < 1024) + break; + newbuf = realloc(buf, bufsize); + if (newbuf == NULL) + goto out; + buf = newbuf; + } + home = err ? "" : pwent.pw_dir; + } + len = strlen(home) + strlen(&file[2]) + 2; + *result = malloc(len); + if (*result) + snprintf(*result, len, "%s/%s", home, &file[2]); + } else { + *result = strdup(file); } - *result = strdup(we.we_wordv[0]); - wordfree(&we); + +out: + if (buf) + free(buf); + if (*result == NULL) return -ENOMEM; return 0; } -#else /* !HAVE_WORDEXP_H */ -/* just copy the string - would be nicer to expand by ourselves, though... */ -int snd_user_file(const char *file, char **result) -{ - *result = strdup(file); - if (! *result) - return -ENOMEM; - return 0; -} -#endif /* HAVE_WORDEXP_H */ -- 2.13.2

1 0

[alsa-devel] Clock Drift between hardware sound card and snd_aloop
by Rob Nertney 12 Jul '17

12 Jul '17

Hi all, I have successfully built a sound capture card that runs at a given rate off its own clock. I receive the stream and do some post-processing on it, and I want to present this modified data to the user as a capture pcm. I initially thought that snd_aloop would be the best idea, as I can pipe my post-processed data directly into it. However, the driver is apparently calculating exact timing of what "should" be the rate based off of CPU ticks/jiffies and the requested period/sample rate/etc in the requested output stream. In my actual hardware I see cycle slips. Underruns and overruns based off of minor adjustments to the card's rate. Is snd_aloop the appropriate driver to be using? Or should I create a new card, similar to snd_aloop, which is more specifically interrupted (and associated pointer locations, etc. modified) from the hardware card as well? It seems like I should be able to synchronize snd_aloop's timing with my hardware clock, but I haven't found anywhere in the API to achieve this. How else can I avoid this clock drift? Thanks, Rob

2 1

[alsa-devel] [PATCH] WIP:asoc:qdsp6: fix semicolon.cocci warnings
by Julia Lawall 11 Jul '17

11 Jul '17

Remove unneeded semicolons. Generated by: scripts/coccinelle/misc/semicolon.cocci Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- tree: https://github.com/rrnayak/linux genpd-performance-state head: 3f1a0555478c46892595728fcb4a2c7ae7e1f7da commit: b4f8ada6fb36369cf0e45421dde636d79e845554 [48/114] WIP:asoc:qdsp6: add adm driver :::::: branch date: 2 hours ago :::::: commit date: 8 weeks ago q6adm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) --- a/sound/soc/qcom/qdsp6v2/q6adm.c +++ b/sound/soc/qcom/qdsp6v2/q6adm.c @@ -806,7 +806,7 @@ int adm_arrange_mch_map(struct adm_cmd_d default: goto non_mch_path; break; - }; + } if ((open->dev_num_channel > 2) && multi_ch_maps[idx].set_channel_map) memcpy(open->dev_channel_mapping, @@ -1019,10 +1019,10 @@ int adm_matrix_map(int path, struct rout route->hdr.pkt_size = cmd_size; route->hdr.src_svc = 0; route->hdr.src_domain = APR_DOMAIN_APPS; - route->hdr.src_port = 0; /* Ignored */; + route->hdr.src_port = 0; /* Ignored */ route->hdr.dest_svc = APR_SVC_ADM; route->hdr.dest_domain = APR_DOMAIN_ADSP; - route->hdr.dest_port = 0; /* Ignored */; + route->hdr.dest_port = 0; /* Ignored */ route->hdr.token = 0; if (path == ADM_PATH_COMPRESSED_RX) { pr_debug("%s: ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x%x\n",

1 0

[alsa-devel] [PATCH] ASoC: pxa: SND_PXA2XX_SOC should depend on HAS_DMA
by Geert Uytterhoeven 11 Jul '17

11 Jul '17

If NO_DMA=y: ERROR: "bad_dma_ops" [sound/soc/pxa/snd-soc-pxa2xx.ko] undefined! ERROR: "bad_dma_ops" [sound/arm/snd-pxa2xx-lib.ko] undefined! ERROR: "dma_common_mmap" [sound/arm/snd-pxa2xx-lib.ko] undefined! Add a dependency on HAS_DMA to fix this. Fixes: 73d7ee2e831f106c ("ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc0549..484ab3c2ad672fc8 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,7 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA || COMPILE_TEST + depends on HAS_DMA select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to -- 2.7.4

3 2