January 2020 - Alsa-devel - mailman.alsa-project.org

[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] Right place for defining ACPI GPIO mapping for codec
by Yauhen Kharuzhy 07 Jan '20

07 Jan '20

Hello, I am working now to get sound working at Lenovo Yoga Book tablet. It is Intel CherryTrail-based device and has RT5677 sound codec. The RT5677 codec driver uses two GPIOs to reset and enable the codec, with names 'realtek,reset' and 'realtek,pow-ldo2'. The ACPI definition lacks a _DSD section with GPIO name->CRS ID definition, so I need to manually define this mapping somewhere using existing devm_acpi_dev_add_driver_gpios() method (various devices has various _CRS definitions order and content, so this is cannot be placed into rt5677.c codec driver). The most obvious place for this is ASoC machine driver for my device, but the codec driver is binded to the ACPI device and initializes before machine driver. Can somebody advice how to define such GPIOs mapping for codec in my case? The ACPI definition of sound device is below. Second I2C address definition is for TS3A277E jack detector. The first GPIO in the _CRS is codec reset, the second is pow-ldo2. Device (RTEK) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10EC5677") // _HID: Hardware ID Name (_CID, "10EC5677") // _CID: Compatible ID Name (_DDN, "Realtek IIS Audio Codec") // _DDN: DOS Device Name Name (_SUB, "17AA7005") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 { CLK3 }) Name (CHAN, Package (0x02) { One, 0x0124F800 }) Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { I2cSerialBusV2 (0x002C, ControllerInitiated, 0x000186A0, AddressingMode7Bit, "\\_SB.PCI0.I2C1", 0x00, ResourceConsumer, , Exclusive, ) I2cSerialBusV2 (0x003B, ControllerInitiated, 0x000186A0, AddressingMode7Bit, "\\_SB.PCI0.I2C1", 0x00, ResourceConsumer, , Exclusive, ) GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x0019 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x0012 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x0030 } GpioInt (Edge, ActiveLow, Exclusive, PullNone, 0x0000, "\\_SB.GPO0", 0x00, ResourceConsumer, , ) { // Pin list 0x005B } GpioInt (Edge, ActiveLow, Exclusive, PullNone, 0x0000, "\\_SB.GPO0", 0x00, ResourceConsumer, , ) { // Pin list 0x004D } SpiSerialBusV2 (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x003D0900, ClockPolarityHigh, ClockPhaseSecond, "\\_SB.PCI0.SPI1", 0x00, ResourceConsumer, , Exclusive, ) }) Return (SBUF) /* \_SB_.PCI0.I2C1.RTEK._CRS.SBUF */ } Method (_STA, 0, NotSerialized) // _STA: Status { Return (0x0F) } Method (_DIS, 0, NotSerialized) // _DIS: Disable Device { } } -- Yauhen Kharuzhy

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[alsa-devel] mixer: Fix memory leak for more than 16 file descriptor case
by GitHub pull_request - opened 06 Jan '20

06 Jan '20

alsa-project/alsa-lib pull request #20 was opened from dfries: I saw this when digging through the source code, wrote up a test case, and verified with valgrind. I don't have that many sound cards so I modified mixer.c to always take the conditional. alloca seemed like the least effort way to fix this. Other thoughts were to test pfds != spfds and free, but given how little bytes would be allocated putting on the stack seems reasonable. Request URL : https://github.com/alsa-project/alsa-lib/pull/20 Patch URL : https://github.com/alsa-project/alsa-lib/pull/20.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH v2] ASoC: rt5640: Fix NULL dereference on module unload
by Dmitry Osipenko 06 Jan '20

06 Jan '20

The rt5640->jack is NULL if jack is already disabled at the time of driver's module unloading. Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- Changelog: v2: - Addressed review comment from Takashi Iwai by making patch's diff a bit cleaner and simpler. - Addressed review comment from Mark Brown by removing noisy backtrace from the commit's message and moving it after --- in order to keep it discoverable in the ML archive. Backtrace: Unable to handle kernel NULL pointer dereference at virtual address 00000024 ... (rt5640_set_jack [snd_soc_rt5640]) (snd_soc_component_set_jack [snd_soc_core]) (soc_remove_component [snd_soc_core]) (soc_cleanup_card_resources [snd_soc_core]) (snd_soc_unregister_card [snd_soc_core]) (tegra_rt5640_remove [snd_soc_tegra_rt5640]) (platform_drv_remove) (device_release_driver_internal) (driver_detach) (bus_remove_driver) (tegra_rt5640_driver_exit (sys_delete_module) sound/soc/codecs/rt5640.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index adbae1f36a8a..747ca248bf10 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2432,6 +2432,13 @@ static void rt5640_disable_jack_detect(struct snd_soc_component *component) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); + /* + * soc_remove_component() force-disables jack and thus rt5640->jack + * could be NULL at the time of driver's module unloading. + */ + if (!rt5640->jack) + return; + disable_irq(rt5640->irq); rt5640_cancel_work(rt5640); -- 2.24.0

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[alsa-devel] [PATCH] ASoC: Intel: bytcht_es8316: Fix Irbis NB41 netbook quirk
by Hans de Goede 06 Jan '20

06 Jan '20

When a quirk for the Irbis NB41 netbook was added, to override the defaults for this device, I forgot to add/keep the BYT_CHT_ES8316_SSP0 part of the defaults, completely breaking audio on this netbook. This commit adds the BYT_CHT_ES8316_SSP0 flag to the Irbis NB41 netbook quirk, making audio work again. Cc: stable(a)vger.kernel.org Cc: russianneuromancer(a)ya.ru Fixes: aa2ba991c420 ("ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook") Reported-and-tested-by: russianneuromancer(a)ya.ru Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcht_es8316.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 46612331f5ea..54e97455d7f6 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -442,7 +442,8 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"), DMI_MATCH(DMI_PRODUCT_NAME, "NB41"), }, - .driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP + .driver_data = (void *)(BYT_CHT_ES8316_SSP0 + | BYT_CHT_ES8316_INTMIC_IN2_MAP | BYT_CHT_ES8316_JD_INVERTED), }, { /* Teclast X98 Plus II */ -- 2.24.1

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[alsa-devel] [PATCH v3] ASoC: Intel: sof_rt5682: Ignore the speaker amp when there isn't one.
by Sam McNally 06 Jan '20

06 Jan '20

Some members of the Google_Hatch family include a rt5682 jack codec, but no speaker amplifier. This uses the same driver (sof_rt5682) as a combination of rt5682 jack codec and max98357a speaker amplifier. Within the sof_rt5682 driver, these cases are not currently distinguishable, relying on a DMI quirk to decide the configuration. This causes an incorrect configuration when only the rt5682 is present on a Google_Hatch device. For CML, the jack codec is used as the primary key when matching, with a possible speaker amplifier described in quirk_data. The two cases of interest are the second and third 10EC5682 entries in snd_soc_acpi_intel_cml_machines[]. The second entry matches the combination of rt5682 and max98357a, resulting in the quirk_data field in the snd_soc_acpi_mach being non-null, pointing at max98357a_spk_codecs, the snd_soc_acpi_codecs for the matched speaker amplifier. The third entry matches just the rt5682, resulting in a null quirk_data. The sof_rt5682 driver's DMI data matching identifies that a speaker amplifier is present for all Google_Hatch family devices. Detect cases where there is no speaker amplifier by checking for a null quirk_data in the snd_soc_acpi_mach and remove the speaker amplifier bit in that case. Signed-off-by: Sam McNally <sammc(a)chromium.org> --- Changes in v3: - Rewrote commit message to refer to correct snd_soc_acpi_intel_cml_machines[] entries and better describe the change Changes in v2: - Added details about the relevant ACPI matches to the description sound/soc/intel/boards/sof_rt5682.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ad8a2b4bc709..8a13231dee15 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -603,6 +603,14 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); + mach = (&pdev->dev)->platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -663,7 +671,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682, -- 2.24.1.735.g03f4e72817-goog

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[alsa-devel] [PATCH] ASoC: bdw-rt5677: add spi driver compile switches
by Curtis Malainey 06 Jan '20

06 Jan '20

If the SPI driver is not compiled in then the RT5677_SPI driver will not be included as well which will cause the bdw-rt5677 driver to fail to probe since the DSP components are managed by the RT5677_SPI driver. The solution is to remove them when the driver is not present as part of the build. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> CC: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 2af8e5a62da84..d5ded2c545d2c 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -74,11 +74,13 @@ static const struct snd_soc_dapm_route bdw_rt5677_map[] = { /* CODEC BE connections */ {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI) {"DSP Capture", NULL, "DSP Buffer"}, /* DSP Clock Connections */ { "DSP Buffer", NULL, "SSP0 CODEC IN" }, { "SSP0 CODEC IN", NULL, "DSPTX" }, +#endif }; static const struct snd_kcontrol_new bdw_rt5677_controls[] = { @@ -319,6 +321,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { SND_SOC_DAILINK_REG(fe, dummy, platform), }, +#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI) /* Non-DPCM links */ { .name = "Codec DSP", @@ -326,6 +329,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { .ops = &bdw_rt5677_dsp_ops, SND_SOC_DAILINK_REG(dsp), }, +#endif /* Back End DAI links */ { -- 2.24.1.735.g03f4e72817-goog

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[alsa-devel] [GIT PULL] ASoC fixes for v5.5
by Mark Brown 06 Jan '20

06 Jan '20

The following changes since commit 556672d75ff486e0b6786056da624131679e0576: ASoC: wm8962: fix lambda value (2019-12-16 11:47:28 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.5-rc5 for you to fetch changes up to ff3b57417012fcc963ec281f5705bed837e4b1ac: ASoC: Intel: boards: Fix compile-testing RT1011/RT5682 (2020-01-03 00:56:47 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.5 More fixes that have been collected, nothing super remarkable here - the few core fixes are mainly error handling related as are many of the driver fixes. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: Intel: boards: Fix compile-testing RT1011/RT5682 Chuhong Yuan (1): ASoC: fsl_audmix: add missed pm_runtime_disable Colin Ian King (1): ASoC: SOF: imx8: fix memory allocation failure check on priv->pd_dev Daniel Baluta (2): ASoC: soc-core: Set dpcm_playback / dpcm_capture ASoC: SOF: imx8: Fix dsp_box offset Dragos Tarcatu (1): ASoC: topology: Prevent use-after-free in snd_soc_get_pcm_runtime() Kai Vehmanen (1): ASoC: SOF: fix fault at driver unload after failed probe Olivier Moysan (3): ASoC: stm32: spdifrx: fix inconsistent lock state ASoC: stm32: spdifrx: fix race condition in irq handler ASoC: stm32: spdifrx: fix input pin state management Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda: hda-dai: fix oops on hda_link .hw_free Takashi Iwai (1): ASoC: core: Fix access to uninitialized list heads sound/soc/fsl/fsl_audmix.c | 9 ++++++- sound/soc/intel/boards/cml_rt1011_rt5682.c | 1 - sound/soc/soc-core.c | 14 ++++++----- sound/soc/soc-topology.c | 6 ++--- sound/soc/sof/imx/imx8.c | 5 +++- sound/soc/sof/intel/hda-dai.c | 11 ++++++-- sound/soc/sof/ipc.c | 3 +++ sound/soc/stm/stm32_spdifrx.c | 40 +++++++++++++++++++----------- 8 files changed, 61 insertions(+), 28 deletions(-)

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[alsa-devel] [PATCH] slimbus: Use the correct style for SPDX License Identifier
by Nishad Kamdar 06 Jan '20

06 Jan '20

This patch corrects the SPDX License Identifier style in header file related to SLIMbus driver. For C header files Documentation/process/license-rules.rst mandates C-like comments (opposed to C source files where C++ style should be used). Changes made by using a script provided by Joe Perches here: https://lkml.org/lkml/2019/2/7/46. Suggested-by: Joe Perches <joe(a)perches.com> Signed-off-by: Nishad Kamdar <nishadkamdar(a)gmail.com> --- drivers/slimbus/slimbus.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index b2f013bfe42e..c73035915f1d 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 +/* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2011-2017, The Linux Foundation */ -- 2.17.1

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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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