December 2019 - 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] [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

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/

3 2

[alsa-devel] [PATCH v1] ASoC: rt5640: Fix NULL dereference on module unload
by Dmitry Osipenko 04 Jan '20

04 Jan '20

The rt5640->jack is NULL if jack is already disabled at the time of driver's module unloading. Unable to handle kernel NULL pointer dereference at virtual address 00000024 ... (rt5640_set_jack [snd_soc_rt5640]) from [<bf86f7ed>] (snd_soc_component_set_jack+0x11/0x1c [snd_soc_core]) (snd_soc_component_set_jack [snd_soc_core]) from [<bf8675cf>] (soc_remove_component+0x1b/0x54 [snd_soc_core]) (soc_remove_component [snd_soc_core]) from [<bf868859>] (soc_cleanup_card_resources+0xad/0x1cc [snd_soc_core]) (soc_cleanup_card_resources [snd_soc_core]) from [<bf86945f>] (snd_soc_unregister_card+0x47/0x78 [snd_soc_core]) (snd_soc_unregister_card [snd_soc_core]) from [<bf8b4013>] (tegra_rt5640_remove+0x13/0x1c [snd_soc_tegra_rt5640]) (tegra_rt5640_remove [snd_soc_tegra_rt5640]) from [<c0516d2f>] (platform_drv_remove+0x17/0x24) (platform_drv_remove) from [<c0515aed>] (device_release_driver_internal+0x95/0x114) (device_release_driver_internal) from [<c0515bd9>] (driver_detach+0x4d/0x90) (driver_detach) from [<c0514d59>] (bus_remove_driver+0x31/0x70) (bus_remove_driver) from [<bf8b4215>] (tegra_rt5640_driver_exit+0x9/0xdf4 [snd_soc_tegra_rt5640]) (tegra_rt5640_driver_exit [snd_soc_tegra_rt5640]) from [<c019336f>] (sys_delete_module+0xe7/0x184) (sys_delete_module) from [<c0101001>] (ret_fast_syscall+0x1/0x28) Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- sound/soc/codecs/rt5640.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index adbae1f36a8a..b245c44cafbc 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2432,16 +2432,22 @@ static void rt5640_disable_jack_detect(struct snd_soc_component *component) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); - disable_irq(rt5640->irq); - rt5640_cancel_work(rt5640); + /* + * soc_remove_component() force-disables jack and thus rt5640->jack + * could be NULL at the time of driver's module unloading. + */ + if (rt5640->jack) { + disable_irq(rt5640->irq); + rt5640_cancel_work(rt5640); - if (rt5640->jack->status & SND_JACK_MICROPHONE) { - rt5640_disable_micbias1_ovcd_irq(component); - rt5640_disable_micbias1_for_ovcd(component); - snd_soc_jack_report(rt5640->jack, 0, SND_JACK_BTN_0); - } + if (rt5640->jack->status & SND_JACK_MICROPHONE) { + rt5640_disable_micbias1_ovcd_irq(component); + rt5640_disable_micbias1_for_ovcd(component); + snd_soc_jack_report(rt5640->jack, 0, SND_JACK_BTN_0); + } - rt5640->jack = NULL; + rt5640->jack = NULL; + } } static int rt5640_set_jack(struct snd_soc_component *component, -- 2.24.0

3 6

[alsa-devel] topology file broadwell.conf
by Jaroslav Kysela 02 Jan '20

02 Jan '20

Hi all, I am working on some (major) cleanups for the topology library and I hit the issue with the topology file broadwell.conf in upstream. It cannot be compiled with the latest alsatplg / libatopology, because the SectionPCMConfig was removed in commit 368e4f2a897b5500573a2ee525f5504c135cfca2 in alsa-lib. Which driver is using this topology in the current linux kernel? Thanks, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 1

[alsa-devel] [PATCH v4 0/2] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 02 Jan '20

02 Jan '20

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the for-next. v4 changes: 1. Fixed wrong signed-off as correct mail suffix. 2. Fixed patch subject that add patch version. v3 changes: 1. https://patchwork.kernel.org/patch/11164283/ and https://patchwork.kernel.org/patch/11164305/ has been merged. 2. Change the name of mtk_wdt_compatible to mtk_wdt_data. 3. Remove toprgu_reset struct and use mtk_wdt_dev instead. 4. Get the value of sw_rst_num from .h file. 5. Adddd mt2712-resets.h for mt2712. 6. Improve commit message. v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on 3. patchwork list: https://patchwork.kernel.org/cover/11164285/ https://patchwork.kernel.org/patch/11164295/ https://patchwork.kernel.org/patch/11164299/ https://patchwork.kernel.org/patch/11164283/ https://patchwork.kernel.org/patch/11164305/ v1 changes: 1. patchwork list: https://patchwork.kernel.org/cover/11164173/ https://patchwork.kernel.org/patch/11164181/ https://patchwork.kernel.org/patch/11164185/ https://patchwork.kernel.org/patch/11164187/ https://patchwork.kernel.org/patch/11164175/ yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../devicetree/bindings/watchdog/mtk-wdt.txt | 10 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 111 +++++++++++++++++- .../reset-controller/mt2712-resets.h | 22 ++++ .../reset-controller/mt8183-resets.h | 15 +++ 5 files changed, 155 insertions(+), 4 deletions(-) create mode 100644 include/dt-bindings/reset-controller/mt2712-resets.h -- 2.18.0

3 5

[alsa-devel] [PATCH 1/2] ASoC: madera: Enable clocks for input pins when used as a direct clock
by Charles Keepax 31 Dec '19

31 Dec '19

When one of the MCLK pins is used to supply an internal clock directly enable the source clock for that pin. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs47l15.c | 8 ++++--- sound/soc/codecs/cs47l35.c | 6 +++-- sound/soc/codecs/cs47l85.c | 11 +++++---- sound/soc/codecs/cs47l90.c | 11 +++++---- sound/soc/codecs/cs47l92.c | 58 ++++++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/madera.c | 53 ++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/madera.h | 2 ++ 7 files changed, 129 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index ece1276f38eb9..513e2875e4c6e 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -438,11 +438,13 @@ static const struct snd_kcontrol_new cs47l15_aec_loopback_mux[] = { static const struct snd_soc_dapm_widget cs47l15_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, 0, madera_sysclk_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, - MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, MADERA_DSP_CLK_ENA_SHIFT, + 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD1", 20, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index d396a8545d518..ba7aa6b31205c 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -521,11 +521,13 @@ static const struct snd_kcontrol_new cs47l35_aec_loopback_mux[] = { static const struct snd_soc_dapm_widget cs47l35_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, 0, madera_sysclk_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, MADERA_DSP_CLK_ENA_SHIFT, - 0, NULL, 0), + 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD1", 20, 0), diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c index 32fe7ffb75268..f85581a97bdc3 100644 --- a/sound/soc/codecs/cs47l85.c +++ b/sound/soc/codecs/cs47l85.c @@ -790,15 +790,18 @@ static const struct snd_kcontrol_new cs47l85_output_anc_src[] = { static const struct snd_soc_dapm_widget cs47l85_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, 0, madera_sysclk_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("ASYNCCLK", MADERA_ASYNC_CLOCK_1, - MADERA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + MADERA_ASYNC_CLK_ENA_SHIFT, 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", MADERA_OUTPUT_ASYNC_CLOCK, MADERA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, - MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, MADERA_DSP_CLK_ENA_SHIFT, + 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c index 67cac60a859dd..c3b6f4d41005b 100644 --- a/sound/soc/codecs/cs47l90.c +++ b/sound/soc/codecs/cs47l90.c @@ -744,15 +744,18 @@ static const struct snd_kcontrol_new cs47l90_output_anc_src[] = { static const struct snd_soc_dapm_widget cs47l90_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, 0, madera_sysclk_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("ASYNCCLK", MADERA_ASYNC_CLOCK_1, - MADERA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + MADERA_ASYNC_CLK_ENA_SHIFT, 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", MADERA_OUTPUT_ASYNC_CLOCK, MADERA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, - MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, MADERA_DSP_CLK_ENA_SHIFT, + 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index d50f75f3b3e41..663ce2b5605c2 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -163,6 +163,51 @@ static int cs47l92_adsp_power_ev(struct snd_soc_dapm_widget *w, return wm_adsp_early_event(w, kcontrol, event); } +static int cs47l92_outclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs47l92 *cs47l92 = snd_soc_component_get_drvdata(component); + struct madera_priv *priv = &cs47l92->core; + struct madera *madera = priv->madera; + unsigned int val; + int ret; + + ret = regmap_read(madera->regmap, MADERA_OUTPUT_RATE_1, &val); + if (ret) { + dev_err(madera->dev, "Failed to read OUTCLK source: %d\n", ret); + return ret; + } + + val &= MADERA_OUT_CLK_SRC_MASK; + + switch (val) { + case MADERA_OUTCLK_MCLK1: + case MADERA_OUTCLK_MCLK2: + case MADERA_OUTCLK_MCLK3: + val -= (MADERA_OUTCLK_MCLK1 - MADERA_MCLK1); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = clk_prepare_enable(madera->mclk[val].clk); + if (ret) + return ret; + break; + case SND_SOC_DAPM_POST_PMD: + clk_disable_unprepare(madera->mclk[val].clk); + break; + default: + break; + } + default: + break; + } + + return madera_domain_clk_ev(w, kcontrol, event); +} + #define CS47L92_NG_SRC(name, base) \ SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ @@ -615,15 +660,18 @@ static const struct snd_kcontrol_new cs47l92_aec_loopback_mux = static const struct snd_soc_dapm_widget cs47l92_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, 0, madera_sysclk_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("ASYNCCLK", MADERA_ASYNC_CLOCK_1, - MADERA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + MADERA_ASYNC_CLK_ENA_SHIFT, 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", MADERA_OUTPUT_ASYNC_CLOCK, MADERA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, - MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, MADERA_DSP_CLK_ENA_SHIFT, + 0, madera_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD1", 20, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD2", 20, 0), @@ -666,7 +714,7 @@ SND_SOC_DAPM_SUPPLY("ISRC2CLK", SND_SOC_NOPM, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("OUTCLK", SND_SOC_NOPM, MADERA_DOM_GRP_OUT, 0, - madera_domain_clk_ev, + cs47l92_outclk_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("SPDCLK", SND_SOC_NOPM, MADERA_DOM_GRP_SPD, 0, diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 52639811cc525..9b4b4c52b9e4b 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -163,6 +163,48 @@ static const int madera_dsp_bus_error_irqs[MADERA_MAX_ADSP] = { MADERA_IRQ_DSP7_BUS_ERR, }; +int madera_clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct madera_priv *priv = snd_soc_component_get_drvdata(component); + struct madera *madera = priv->madera; + unsigned int val; + int clk_idx; + int ret; + + ret = regmap_read(madera->regmap, w->reg, &val); + if (ret) { + dev_err(madera->dev, "Failed to check clock source: %d\n", ret); + return ret; + } + + switch ((val & MADERA_SYSCLK_SRC_MASK) >> MADERA_SYSCLK_SRC_SHIFT) { + case MADERA_CLK_SRC_MCLK1: + clk_idx = MADERA_MCLK1; + break; + case MADERA_CLK_SRC_MCLK2: + clk_idx = MADERA_MCLK2; + break; + case MADERA_CLK_SRC_MCLK3: + clk_idx = MADERA_MCLK3; + break; + default: + return 0; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return clk_prepare_enable(madera->mclk[clk_idx].clk); + case SND_SOC_DAPM_POST_PMD: + clk_disable_unprepare(madera->mclk[clk_idx].clk); + return 0; + default: + return 0; + } +} +EXPORT_SYMBOL_GPL(madera_clk_ev); + static void madera_spin_sysclk(struct madera_priv *priv) { struct madera *madera = priv->madera; @@ -193,9 +235,16 @@ int madera_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct madera_priv *priv = snd_soc_component_get_drvdata(component); - madera_spin_sysclk(priv); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMD: + madera_spin_sysclk(priv); + break; + default: + break; + } - return 0; + return madera_clk_ev(w, kcontrol, event); } EXPORT_SYMBOL_GPL(madera_sysclk_ev); diff --git a/sound/soc/codecs/madera.h b/sound/soc/codecs/madera.h index 6d8938a3fb647..49a8f68ec43da 100644 --- a/sound/soc/codecs/madera.h +++ b/sound/soc/codecs/madera.h @@ -383,6 +383,8 @@ int madera_eq_coeff_put(struct snd_kcontrol *kcontrol, int madera_lhpf_coeff_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int madera_clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); int madera_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int madera_spk_ev(struct snd_soc_dapm_widget *w, -- 2.11.0

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[alsa-devel] [PATCH] ASoC: cs47l92: Simplify error handling code in 'cs47l92_probe()'
by Christophe JAILLET 31 Dec '19

31 Dec '19

If 'madera_init_bus_error_irq()' fails, 'wm_adsp2_remove(&cs47l92->core.adsp[0])' will be called twice. Once in the 'if' block, and once in the error handling path. This is harmless, but one of this call can be axed. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/cs47l92.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index d50f75f3b3e4..536b7d35d6b2 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -1959,10 +1959,8 @@ static int cs47l92_probe(struct platform_device *pdev) goto error_dsp_irq; ret = madera_init_bus_error_irq(&cs47l92->core, 0, wm_adsp2_bus_error); - if (ret != 0) { - wm_adsp2_remove(&cs47l92->core.adsp[0]); + if (ret != 0) goto error_adsp; - } madera_init_fll(madera, 1, MADERA_FLL1_CONTROL_1 - 1, &cs47l92->fll[0]); -- 2.20.1

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[alsa-devel] [PATCH 6/6] ASoC: amd MMAP_INTERLEAVED Support
by Ravulapati Vishnu vardhan rao 31 Dec '19

31 Dec '19

ACP-I2S device support MMAP_INTERLEAVED. Added support for the same. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 151daa9..5c3ec3c 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -21,6 +21,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | @@ -41,7 +42,8 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, -- 2.7.4

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