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

[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

4 4

[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

2 12

[alsa-devel] [PATCH 00/68] ALSA: Constifications, take #2
by Takashi Iwai 05 Jan '20

05 Jan '20

Hi, this is the next series of patches for adding more const prefix allover the places. They do just nothing but adding cost. Takashi === Takashi Iwai (68): ALSA: Allow const arrays for legacy resource management helpers ALSA: aoa: More constifications ALSA: hda/realtek - More constifications ALSA: dummy: More constifications ALSA: pcm: More constifications ALSA: seq: More constifications ALSA: vx: More constifications ALSA: intel8x0: More constifications ALSA: hda: More constifications ALSA: ac97: More constifications ALSA: usb-audio: More constifications ALSA: caiaq: More constifications ALSA: au88x0: More constifications ALSA: emu10k1: More constifications ALSA: riptide: More constifications ALSA: ice1712: More constifications ALSA: ppc: More constifications ALSA: hdsp: More constifications ALSA: hdspm: More constifications ALSA: rme9652: More constifications ALSA: emux: More constifications ALSA: usx2y: More constifications ALSA: pcxhr: More constifications ALSA: mixart: More constifications ALSA: ymfpci: More constifications ALSA: bcd2000: More constifications ALSA: nm256: More constifications ALSA: korg1212: More constifications ALSA: cs46xx: More constifications ALSA: ca0106: More constifications ALSA: ctxfi: More constifications ALSA: asihpi: More constifications ALSA: atiixp: More constifications ALSA: es1938: More constifications ALSA: es1968: More constifications ALSA: cmipci: More constifications ALSA: sparc: More constifications ALSA: info: More constifications ALSA: jack: More constification ALSA: oss: More constifications ALSA: rawmidi: More constification ALSA: opl3: More constifications ALSA: cmi8328: More constifications ALSA: cs423x: More constification ALSA: sb: More constifications ALSA: cmi8330: More constifications ALSA: es1688: More constifications ALSA: es18xx: More constifications ALSA: gus: More constifications ALSA: opti9xx: More constifications ALSA: wss: More constifications ALSA: sc6000: More constification ALSA: mts64: More constifications ALSA: aw2: More constifications ALSA: lx6464es: More constifications ALSA: arm: More constification ALSA: opl4: More constification ALSA: serial-u16550: More constification ALSA: ak4531: More constification ALSA: azt3328: More constification ALSA: cs4281: More constification ALSA: echoaudio: More constification ALSA: ens137x: More constification ALSA: fm801: More constification ALSA: via82xx: More constification ALSA: pdaudiocf: More constification ALSA: spi: More constification ALSA: sh: Fix compile warning wrt const include/sound/initval.h | 6 +- sound/aoa/codecs/onyx.c | 4 +- sound/aoa/codecs/tas-basstreble.h | 4 +- sound/aoa/codecs/tas-gain-table.h | 2 +- sound/aoa/soundbus/i2sbus/core.c | 2 +- sound/arm/aaci.c | 2 +- sound/core/info.c | 4 +- sound/core/jack.c | 2 +- sound/core/oss/mixer_oss.c | 2 +- sound/core/oss/pcm_plugin.c | 2 +- sound/core/pcm.c | 12 ++-- sound/core/pcm_lib.c | 2 +- sound/core/pcm_misc.c | 5 +- sound/core/rawmidi.c | 2 +- sound/core/seq/oss/seq_oss_init.c | 4 +- sound/core/seq/seq_midi_emul.c | 6 +- sound/core/seq/seq_midi_event.c | 6 +- sound/drivers/dummy.c | 24 +++---- sound/drivers/mts64.c | 4 +- sound/drivers/opl3/opl3_drums.c | 2 +- sound/drivers/opl3/opl3_midi.c | 6 +- sound/drivers/opl4/opl4_synth.c | 2 +- sound/drivers/serial-u16550.c | 2 +- sound/drivers/vx/vx_cmd.c | 2 +- sound/drivers/vx/vx_core.c | 12 ++-- sound/drivers/vx/vx_hwdep.c | 2 +- sound/drivers/vx/vx_mixer.c | 2 +- sound/hda/hdac_device.c | 4 +- sound/hda/hdmi_chmap.c | 2 +- sound/hda/intel-nhlt.c | 2 +- sound/isa/cmi8328.c | 24 +++---- sound/isa/cmi8330.c | 8 +-- sound/isa/cs423x/cs4236_lib.c | 2 +- sound/isa/es1688/es1688.c | 6 +- sound/isa/es1688/es1688_lib.c | 4 +- sound/isa/es18xx.c | 10 +-- sound/isa/gus/gus_io.c | 2 +- sound/isa/gus/gus_main.c | 4 +- sound/isa/gus/gus_volume.c | 9 +-- sound/isa/gus/gusclassic.c | 6 +- sound/isa/gus/gusextreme.c | 10 +-- sound/isa/gus/gusmax.c | 6 +- sound/isa/gus/interwave.c | 10 +-- sound/isa/opti9xx/miro.c | 18 ++--- sound/isa/opti9xx/opti92x-ad1848.c | 18 ++--- sound/isa/sb/emu8000.c | 16 ++--- sound/isa/sb/jazz16.c | 10 +-- sound/isa/sb/sb16.c | 8 +-- sound/isa/sb/sb8.c | 2 +- sound/isa/sb/sb_mixer.c | 38 +++++----- sound/isa/sc6000.c | 4 +- sound/isa/wss/wss_lib.c | 6 +- sound/pci/ac97/ac97_codec.c | 10 +-- sound/pci/ac97/ac97_patch.c | 26 +++---- sound/pci/ac97/ac97_pcm.c | 4 +- sound/pci/ak4531_codec.c | 2 +- sound/pci/asihpi/asihpi.c | 4 +- sound/pci/asihpi/hpimsgx.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0_core.c | 2 +- sound/pci/au88x0/au88x0_eq.c | 22 +++--- sound/pci/au88x0/au88x0_eqdata.c | 18 ++--- sound/pci/au88x0/au88x0_pcm.c | 4 +- sound/pci/aw2/aw2-tsl.c | 4 +- sound/pci/azt3328.c | 2 +- sound/pci/ca0106/ca0106_main.c | 4 +- sound/pci/ca0106/ca0106_mixer.c | 14 ++-- sound/pci/cmipci.c | 8 +-- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos.c | 4 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 6 +- sound/pci/ctxfi/ctdaio.c | 4 +- sound/pci/ctxfi/ctresource.c | 2 +- sound/pci/echoaudio/echoaudio.c | 2 +- sound/pci/emu10k1/emu10k1_main.c | 12 ++-- sound/pci/emu10k1/emufx.c | 14 ++-- sound/pci/emu10k1/emumixer.c | 28 ++++---- sound/pci/emu10k1/emuproc.c | 16 ++--- sound/pci/emu10k1/io.c | 4 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 8 ++- sound/pci/es1968.c | 4 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_codec.c | 6 +- sound/pci/hda/hda_eld.c | 6 +- sound/pci/hda/hda_intel.c | 8 +-- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0132.c | 18 ++--- sound/pci/hda/patch_realtek.c | 118 ++++++++++++++++---------------- sound/pci/ice1712/aureon.c | 8 +-- sound/pci/ice1712/juli.c | 6 +- sound/pci/ice1712/maya44.c | 10 +-- sound/pci/ice1712/phase.c | 4 +- sound/pci/ice1712/pontis.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/ice1712/prodigy_hifi.c | 16 ++--- sound/pci/ice1712/psc724.c | 2 +- sound/pci/ice1712/quartet.c | 8 +-- sound/pci/ice1712/se.c | 8 +-- sound/pci/ice1712/vt1720_mobo.c | 4 +- sound/pci/ice1712/wtm.c | 6 +- sound/pci/intel8x0.c | 28 ++++---- sound/pci/intel8x0m.c | 12 ++-- sound/pci/korg1212/korg1212.c | 8 +-- sound/pci/lx6464es/lx_core.c | 4 +- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/mixart/mixart_mixer.c | 4 +- sound/pci/nm256/nm256.c | 4 +- sound/pci/nm256/nm256_coef.c | 4 +- sound/pci/pcxhr/pcxhr.c | 2 +- sound/pci/pcxhr/pcxhr_core.c | 6 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- sound/pci/riptide/riptide.c | 60 ++++++++-------- sound/pci/rme9652/hdsp.c | 14 ++-- sound/pci/rme9652/hdspm.c | 66 +++++++++--------- sound/pci/rme9652/rme9652.c | 10 +-- sound/pci/via82xx.c | 2 +- sound/pci/vx222/vx222_ops.c | 4 +- sound/pci/ymfpci/ymfpci_main.c | 8 +-- sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 2 +- sound/pcmcia/vx/vxp_ops.c | 2 +- sound/ppc/beep.c | 2 +- sound/ppc/pmac.c | 8 +-- sound/ppc/pmac.h | 2 +- sound/ppc/tumbler.c | 20 +++--- sound/ppc/tumbler_volume.h | 12 ++-- sound/sh/aica.c | 4 +- sound/sparc/cs4231.c | 4 +- sound/sparc/dbri.c | 2 +- sound/spi/at73c213.c | 2 +- sound/synth/emux/emux_nrpn.c | 4 +- sound/synth/emux/emux_synth.c | 10 +-- sound/synth/emux/soundfont.c | 8 +-- sound/usb/bcd2000/bcd2000.c | 4 +- sound/usb/caiaq/audio.c | 2 +- sound/usb/caiaq/control.c | 20 +++--- sound/usb/caiaq/input.c | 10 +-- sound/usb/mixer.c | 60 ++++++++-------- sound/usb/mixer_maps.c | 56 +++++++-------- sound/usb/mixer_quirks.c | 6 +- sound/usb/mixer_scarlett.c | 14 ++-- sound/usb/proc.c | 2 +- sound/usb/stream.c | 4 +- sound/usb/usx2y/usX2Yhwdep.c | 2 +- sound/usb/usx2y/usbusx2yaudio.c | 6 +- sound/usb/validate.c | 4 +- 150 files changed, 677 insertions(+), 669 deletions(-) -- 2.16.4

1 68

[alsa-devel] [PATCH] ALSA: usb-audio: Use lower hex numbers for IDs
by Takashi Iwai 05 Jan '20

05 Jan '20

For consistency reason, make all hex numbers with lower alphabets for USB ID entries. It improves grep-ability and reduces careless mistakes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 8 ++++---- sound/usb/midi.c | 4 ++-- sound/usb/quirks.c | 14 +++++++------- 3 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db71305f6..5e7c4c7beff2 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -292,10 +292,10 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, struct audioformat *fp) { switch (chip->usb_id) { - case USB_ID(0x0E41, 0x4241): /* Line6 Helix */ - case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ - case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ - case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0e41, 0x4241): /* Line6 Helix */ + case USB_ID(0x0e41, 0x4242): /* Line6 Helix Rack */ + case USB_ID(0x0e41, 0x4244): /* Line6 Helix LT */ + case USB_ID(0x0e41, 0x4246): /* Line6 HX-Stomp */ /* supported rates: 48Khz */ kfree(fp->rate_table); fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0ec77d0..392e5fda680c 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1408,8 +1408,8 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi, /* * Some devices only work with 9 bytes packet size: */ - case USB_ID(0x0644, 0x800E): /* Tascam US-122L */ - case USB_ID(0x0644, 0x800F): /* Tascam US-144 */ + case USB_ID(0x0644, 0x800e): /* Tascam US-122L */ + case USB_ID(0x0644, 0x800f): /* Tascam US-144 */ ep->max_transfer = 9; break; } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 82184036437b..11159fcfdcdf 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1393,22 +1393,22 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) { /* devices which do not support reading the sample rate. */ switch (chip->usb_id) { - case USB_ID(0x041E, 0x4080): /* Creative Live Cam VF0610 */ - case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ + case USB_ID(0x041e, 0x4080): /* Creative Live Cam VF0610 */ + case USB_ID(0x04d8, 0xfeea): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ - case USB_ID(0x05A3, 0x9420): /* ELP HD USB Camera */ + case USB_ID(0x05a3, 0x9420): /* ELP HD USB Camera */ case USB_ID(0x05a7, 0x1020): /* Bose Companion 5 */ - case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x074d, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x1395, 0x740a): /* Sennheiser DECT */ case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ - case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ return true; } /* devices of these vendors don't support reading rate, either */ switch (USB_ID_VENDOR(chip->usb_id)) { - case 0x045E: /* MS Lifecam */ - case 0x047F: /* Plantronics */ + case 0x045e: /* MS Lifecam */ + case 0x047f: /* Plantronics */ case 0x1de7: /* Phoenix Audio */ return true; } -- 2.16.4

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[alsa-devel] [PATCH] ALSA: hda/realtek - Apply mic mute LED quirk for Dell E7xx laptops, too
by Takashi Iwai 05 Jan '20

05 Jan '20

Dell E7xx laptops have also mic mute LED that is driven by the dell-laptop platform driver. Bind it with the capture control as already done for other models. A caveat is that the fixup hook for the mic mute LED has to be applied at last, otherwise it results in the invalid override of the callback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205529 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 40e30ea86663..03da83bcd6b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5844,6 +5844,7 @@ enum { ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, ALC288_FIXUP_DISABLE_AAMIX, + ALC292_FIXUP_DELL_E7X_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, @@ -6536,12 +6537,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE }, - [ALC292_FIXUP_DELL_E7X] = { + [ALC292_FIXUP_DELL_E7X_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, .chained = true, .chain_id = ALC292_FIXUP_DISABLE_AAMIX }, + [ALC292_FIXUP_DELL_E7X] = { + .type = HDA_FIXUP_FUNC, + .v.func = snd_hda_gen_fixup_micmute_led, + /* micmute fixup must be applied at last */ + .chained_before = true, + .chain_id = ALC292_FIXUP_DELL_E7X_AAMIX, + }, [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { -- 2.16.4

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