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

[alsa-devel] [PATCH for-5.6 0/2] drm/bridge: dw-hdmi: PCM API updates
by Takashi Iwai 09 Jan '20

09 Jan '20

Hi, this is a patch set for updating ALSA PCM API usages in dw-hdmi driver. I already tried to "fix" the driver some time ago but it was utterly wrong. So this is a combination of the revised patch and another cleanup patch. The first one is to change the buffer allocation mechanism in the driver to the manual allocation of the h/w buffer and the automatic allocation of PCM stream buffers via the new standard API. The significant change is that size of the h/w buffer isn't no longer controlled via ALSA preallocation proc file but rather via the new module option (if any). The second one is a oneliner patch just to remove the superfluous PCM ops. Both need the ALSA PCM core changes in 5.5-rc1, so please apply them on top of 5.5-rc1 or later. Or, just let me know if I should apply them through sound git tree. thanks, Takashi === Takashi Iwai (2): drm/bridge: dw-hdmi: Follow the standard ALSA memalloc way drm/bridge: dw-hdmi: Drop superfluous ioctl PCM ops .../gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c | 49 ++++++++++++---------- 1 file changed, 26 insertions(+), 23 deletions(-) -- 2.16.4

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[alsa-devel] [PATCH 00/10] mfd: Add support for Analog Devices A2B transceiver
by Daniel Mack 08 Jan '20

08 Jan '20

This patch series adds support for Analog Device's AD242x A2B transceivers. https://www.analog.com/media/en/technical-documentation/user-guides/AD242x_… These transceivers are used to form an audio network by connecting the parts in a daisy-chain. On top of audio, the devices expose some other functions such as GPIO, programmable clock outputs and remote-side I2C bus master. The first node in the chain is called the master node, and all other devices are called slave nodes. Up to 15 such devices can be connected this way. The master device responds on two addresses on the I2C bus. The primary one is used to access all registers in the master node itself, the secondary is for accessing remote nodes after prior setup through the master node. In the driver stack, these details are hidden behind specific regmap configs. The driver stack is implemented as MFD core and companion drivers that can be registered as sub-devices in DT. Drivers for these sub-devices can be used for both master and slave nodes, as they just interface with the node's regmap. The master node is responsible for discovering all the slave nodes at probe time, and it needs to take the used audio and routing modes in each of the slave devices into account in order to pre-calculate the bus-timings correctly. Hence, this bus is not hot-pluggable. Transceivers can both receive and provide audio, and streams can be routed from one node to any other, including many others. The tricky bit is how to expose the audio routing in DT in a sane way. The way it is implemented here, the slave nodes specify the number of slots they each consume and generate, and which thereof they forward from one side to the other. This mimics the internal register structure and should allow for even exotic setups. Please let me know what you think and what could be improved. Thanks, Daniel Daniel Mack (10): dt-bindings: mfd: Add documentation for ad242x dt-bindings: i2c: Add documentation for ad242x i2c controllers dt-bindings: gpio: Add documentation for AD242x GPIO controllers dt-bindings: clock: Add documentation for AD242x clock providers dt-bindings: sound: Add documentation for AD242x codecs mfd: Add core driver for AD242x A2B transceivers i2c: Add driver for AD242x bus controller gpio: Add driver for AD242x GPIO controllers clk: Add support for AD242x clock output providers ASoC: Add codec component for AD242x nodes .../bindings/clock/adi,ad242x-clk.yaml | 32 + .../bindings/gpio/adi,ad242x-gpio.yaml | 65 ++ .../bindings/i2c/adi,ad242x-i2c.yaml | 31 + .../bindings/mfd/adi,ad242x-bus.yaml | 29 + .../bindings/mfd/adi,ad242x-master.yaml | 235 +++++++ .../bindings/mfd/adi,ad242x-slave.yaml | 108 ++++ .../bindings/sound/adi,ad242x-codec.yaml | 31 + drivers/clk/Kconfig | 6 + drivers/clk/Makefile | 1 + drivers/clk/clk-ad242x.c | 231 +++++++ drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-ad242x.c | 229 +++++++ drivers/i2c/busses/Kconfig | 10 + drivers/i2c/busses/Makefile | 1 + drivers/i2c/busses/i2c-ad242x.c | 178 +++++ drivers/mfd/Kconfig | 11 + drivers/mfd/Makefile | 1 + drivers/mfd/ad242x-bus.c | 42 ++ drivers/mfd/ad242x-master.c | 611 ++++++++++++++++++ drivers/mfd/ad242x-node.c | 262 ++++++++ drivers/mfd/ad242x-slave.c | 234 +++++++ include/dt-bindings/clock/adi,ad242x.h | 9 + include/linux/mfd/ad242x.h | 400 ++++++++++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad242x.c | 338 ++++++++++ 27 files changed, 3109 insertions(+) create mode 100644 Documentation/devicetree/bindings/clock/adi,ad242x-clk.yaml create mode 100644 Documentation/devicetree/bindings/gpio/adi,ad242x-gpio.yaml create mode 100644 Documentation/devicetree/bindings/i2c/adi,ad242x-i2c.yaml create mode 100644 Documentation/devicetree/bindings/mfd/adi,ad242x-bus.yaml create mode 100644 Documentation/devicetree/bindings/mfd/adi,ad242x-master.yaml create mode 100644 Documentation/devicetree/bindings/mfd/adi,ad242x-slave.yaml create mode 100644 Documentation/devicetree/bindings/sound/adi,ad242x-codec.yaml create mode 100644 drivers/clk/clk-ad242x.c create mode 100644 drivers/gpio/gpio-ad242x.c create mode 100644 drivers/i2c/busses/i2c-ad242x.c create mode 100644 drivers/mfd/ad242x-bus.c create mode 100644 drivers/mfd/ad242x-master.c create mode 100644 drivers/mfd/ad242x-node.c create mode 100644 drivers/mfd/ad242x-slave.c create mode 100644 include/dt-bindings/clock/adi,ad242x.h create mode 100644 include/linux/mfd/ad242x.h create mode 100644 sound/soc/codecs/ad242x.c -- 2.23.0

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[alsa-devel] [PATCH 0/7] ALSA: hda: Enable HDAudio compress
by Cezary Rojewski 07 Jan '20

07 Jan '20

This set of patches provides new interfaces - page allocation - and runtime flow adjustments - PM support - for compress operations. For HDA part, work has been done to account for compress streams when servicing IRQs, setting up BDLs and assigning DMAs. End goal is to make room for one of DSP debug features: data probing. It takes advantage of compress streams when extracting data from running audio pipeline. Initial review and development of probes can be found under: https://github.com/thesofproject/linux/pull/1276 with this very set of patches being separated and reviewed on: https://github.com/thesofproject/linux/pull/1571 Cezary Rojewski (7): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: compress: Add pm_runtime support ASoC: SOF: Intel: Account for compress streams when servicing IRQs include/sound/compress_driver.h | 40 ++++++++++++++++++------ include/sound/hdaudio.h | 2 ++ include/sound/hdaudio_ext.h | 2 ++ sound/core/compress_offload.c | 42 ++++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 46 +++++++++++++++++++++++++--- sound/hda/hdac_controller.c | 4 +-- sound/hda/hdac_stream.c | 52 ++++++++++++++++++++------------ sound/soc/soc-compress.c | 29 +++++++++++++++++- sound/soc/sof/intel/hda-stream.c | 26 ++++++++++++++-- 9 files changed, 205 insertions(+), 38 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH alsa-utils 00/11] alsa-info.sh: Improve output and fix file upload issues
by David Ward 07 Jan '20

07 Jan '20

David Ward (11): alsa-info.sh: Consolidate PCI device output alsa-info.sh: Read from /proc/modules and sort the result alsa-info.sh: Simplify iteration over cards when calling amixer alsa-info.sh: Use existing function to print ALSA configuration files alsa-info.sh: Exit script after writing information to stdout alsa-info.sh: Replace gauge with infobox for upload dialog alsa-info.sh: Remove progress spinner during upload without dialog alsa-info.sh: Condense nested commands for file upload alsa-info.sh: Condense nested commands for formatting upload result alsa-info.sh: Perform test for wget earlier alsa-info.sh: Warn after actual upload failure; do not ping server alsa-info/alsa-info.sh | 251 ++++++++++++++++++------------------------------- 1 file changed, 90 insertions(+), 161 deletions(-) -- 1.8.3.1

2 12

[alsa-devel] [PATCH] pinctrl: lochnagar: select GPIOLIB
by Arnd Bergmann 07 Jan '20

07 Jan '20

In a rare randconfig build I came across one configuration that does not enable CONFIG_GPIOLIB, which is needed by lochnagar: ERROR: "devm_gpiochip_add_data" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! ERROR: "gpiochip_generic_free" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! ERROR: "gpiochip_generic_request" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! ERROR: "gpiochip_get_data" [drivers/pinctrl/cirrus/pinctrl-lochnagar.ko] undefined! Add another 'select' like all other pinctrl drivers have. Fixes: 0548448b719a ("pinctrl: lochnagar: Add support for the Cirrus Logic Lochnagar") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I wonder if GPIOLIB should just become mandatory when enabling the pinctrl subsystem, or if there are still good reasons for leaving it disabled on any machine that uses CONFIG_PINCTRL. --- drivers/pinctrl/cirrus/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/pinctrl/cirrus/Kconfig b/drivers/pinctrl/cirrus/Kconfig index f1806fd781a0..530426a74f75 100644 --- a/drivers/pinctrl/cirrus/Kconfig +++ b/drivers/pinctrl/cirrus/Kconfig @@ -2,6 +2,7 @@ config PINCTRL_LOCHNAGAR tristate "Cirrus Logic Lochnagar pinctrl driver" depends on MFD_LOCHNAGAR + select GPIOLIB select PINMUX select PINCONF select GENERIC_PINCONF -- 2.20.0

4 6

[alsa-devel] [PATCH] ASoC: rt5677: add SPI_MASTER dependency
by Arnd Bergmann 07 Jan '20

07 Jan '20

When CONFIG_SPI is disabled, the newly added code for the DSP firmware loading fails to link: ERROR: "rt5677_spi_hotword_detected" [sound/soc/codecs/snd-soc-rt5677.ko] undefined! ERROR: "rt5677_spi_write" [sound/soc/codecs/snd-soc-rt5677.ko] undefined! Add a dependency to prevent this configuration. Note: the does not work with the DT probing, as there is no binding for the SPI half of the driver, but nothing seems to be using that with the mainline kernel anyway. Fixes: 461c623270e4 ("ASoC: rt5677: Load firmware via SPI using delayed work") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/mediatek/Kconfig | 2 +- sound/soc/tegra/Kconfig | 2 +- 4 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 71b7286d14f2..8e9c461a84f8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1047,6 +1047,7 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate + depends on SPI_MASTER select REGMAP_I2C select REGMAP_IRQ diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index ef20316e83d1..da7f9111f3d3 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -34,6 +34,7 @@ if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C + depends on SPI_MASTER depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on GPIOLIB || COMPILE_TEST depends on X86_INTEL_LPSS || COMPILE_TEST diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index a656d2014127..4699787c93ed 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -97,7 +97,7 @@ config SND_SOC_MT8173_RT5650_RT5514 config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" - depends on SND_SOC_MT8173 && I2C + depends on SND_SOC_MT8173 && I2C && SPI_MASTER select SND_SOC_RT5645 select SND_SOC_RT5677 select SND_SOC_HDMI_CODEC diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index addadc827b91..df36e84c0116 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -122,7 +122,7 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on SND_SOC_TEGRA && I2C && GPIOLIB && SPI_MASTER select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra -- 2.20.0

4 4

[alsa-devel] [PATCH v5 00/11] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 07 Jan '20

07 Jan '20

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. gpio controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. Thanks, srini Changes since v4: - updated microvolt bindings as suggested by Rob. - slimbus optional property patch already applied. Srinivas Kandagatla (11): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: gpio: wcd934x: Add bindings for gpio gpio: wcd934x: Add support to wcd934x gpio controller ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga .../bindings/gpio/qcom,wcd934x-gpio.yaml | 47 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 175 + drivers/gpio/Kconfig | 8 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-wcd934x.c | 104 + drivers/mfd/Kconfig | 12 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 306 + include/linux/mfd/wcd934x/registers.h | 531 ++ include/linux/mfd/wcd934x/wcd934x.h | 31 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5084 +++++++++++++++++ sound/soc/qcom/sdm845.c | 86 +- 15 files changed, 6401 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/qcom,wcd934x-gpio.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/gpio/gpio-wcd934x.c create mode 100644 drivers/mfd/wcd934x.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Add support for Presonus Studio 1810c
by mickflemm＠gmail.com 06 Jan '20

06 Jan '20

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

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[alsa-devel] [PATCH AUTOSEL 5.4 001/187] ASoC: rt5682: fix i2c arbitration lost issue
by Sasha Levin 05 Jan '20

05 Jan '20

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

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Re: [alsa-devel] [PATCH 2/3] ASoC: SOF: Intel: Fix CFL and CML FW nocodec binary names.
by Bernie Innocenti 04 Jan '20

04 Jan '20

Greetings Liam and Pierre, > @@ -149,7 +149,7 @@ static const struct sof_dev_desc cml_desc = { > .chip_info = &cnl_chip_info, > .default_fw_path = "intel/sof", > .default_tplg_path = "intel/sof-tplg", >- .nocodec_fw_filename = "sof-cnl.ri", >+ .nocodec_fw_filename = "sof-cml.ri", Where is this new firmware file? This patch has flown into Fedora kernels 5.4.5-301.fc31 and 5.5.0-0.rc3.git0.1.fc32, and now I'm getting this error when the module loads on a Thinkpad X1 Gen 7: sof-audio-pci 0000:00:1f.3: Direct firmware load for intel/sof/sof-cml.ri failed with error -2 The other .ri files come with alsa-firmware, but sof-cml.ri is absent in the latest upstream release (alsa-firmware-1.2.1). The only reference I could find is this symlink, but it's not clear to me how and when it will get into distributions: https://github.com/thesofproject/linux-firmware/commit/beff732e6642bd77dac5… -- _ // Bernie Innocenti \X/ https://codewiz.org/

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