June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] sound: soc/codecs: tlv320adc3xxx: needs GPIOLIB
by Randy Dunlap 31 Aug '22

31 Aug '22

Fix build errors when CONFIG_GPIOLIB is not enabled: ../sound/soc/codecs/tlv320adc3xxx.c: In function 'adc3xxx_i2c_probe': ../sound/soc/codecs/tlv320adc3xxx.c:1352:28: error: implicit declaration of function 'devm_gpiod_get'; did you mean 'devm_gpio_free'? [-Werror=implicit-function-declaration] 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ../sound/soc/codecs/tlv320adc3xxx.c:1352:57: error: 'GPIOD_OUT_LOW' undeclared (first use in this function); did you mean 'GPIOF_INIT_LOW'? 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); CC lib/dynamic_debug.o ../sound/soc/codecs/tlv320adc3xxx.c:1400:9: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1400 | gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ricard Wanderlof <ricardw(a)axis.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1567,6 +1567,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" depends on I2C + depends on GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs.

1 1

Request for setup of new repositories
by Takashi Sakamoto 27 Aug '22

27 Aug '22

Hi Jaroslav, Iwai-san, Thanks for your maintenance for alsa-project organization in github.com. Currently I'd like to add new three repositories under the organization as a part of my work for ALSA firewire stack. I've been maintaining libhinawa since 2014 and recently realized that current design is not necessarily convenient since it includes two functions; operation to Linux FireWire cdev, and operation of ALSA HwDep cdev. Currently I'm working for new library to split the latter operation. Then I'd like you to setup below repositories: * 'libhitaki' * 'libhitaki-doc' * 'hitaki-rs' The library itself and its Rust binding are utilized by 'snd-firewire-ctl-services'[2], thus it's preferable to register them under 'GObject Introspection' Team. Thanks for your assist for my work. [1] https://github.com/alsa-project/libhinawa [2] https://github.com/alsa-project/snd-firewire-ctl-services Regards Takashi Sakamoto

2 20

[PATCH] ASoC: ops: Don't modify the driver's plaform_max when reading state
by Mark Brown 22 Aug '22

22 Aug '22

Currently snd_soc_info_volsw() will set a platform_max based on the limit the control has if one is not already set. This isn't really great, we shouldn't be modifying the passed in driver data especially in a path like this which may not ever be executed or where we may execute other callbacks before this one. Instead make this function leave the data unchanged, and clarify things a bit by referring to max rather than platform_max within the function. platform_max is now applied as a limit after working out the natural maximum value for the control. This means that platform_max is no longer treated as a direct register value for controls were min is non-zero. The put() callbacks already validate on this basis, and there do not appear to be any in tree users that would be affected. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d39370ddeca..93e72a016b4d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,20 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = uinfo->value.integer.max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; + uinfo->value.integer.max = max; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); -- 2.30.2

2 6

[PATCH] Add quirk to enable Avid Mbox 3 support
by connerknoxpublic＠gmail.com 22 Aug '22

22 Aug '22

From: Conner Knox <connerknoxpublic(a)gmail.com> Hi all, I believe these are the right emails to send this patch to, but if I've missed something please let me know. This patch enables the Avid Mbox 3 audio interface to be used through ALSA. It locks the device at a 48khz sample rate. Then it mutes and raises the appropriate channels/inputs to allow for expected functionality. Thanks for your time. --- sound/usb/quirks-table.h | 76 ++++++++++ sound/usb/quirks.c | 311 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 387 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..5f43863e5a5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2978,6 +2978,82 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* DIGIDESIGN MBOX 3 */ +{ + USB_DEVICE(0x0dba, 0x5000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 3", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x00, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .attributes = 0x00, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .maxpacksize = 0x009c, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* Tascam US122 MKII - playback-only support */ USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..e7caa7eeb955 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1020,6 +1020,313 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } +static void mbox3_setup_48_24_magic(struct usb_device *dev) +{ + u8 com_buff[4]; + + /* Load 48000Hz rate into buffer */ + com_buff[0] = 0x80; + com_buff[1] = 0xbb; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + + /* Set 48000Hz sample rate */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x0001, &com_buff, 0x0004); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &com_buff, 0x0004); + + /* Set clock source to Internal (as opposed to S/PDIF) */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x01; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x8001, &com_buff, 1); + + /* Mute the hardware loopbacks to start the device in a known state. + * The Mbox 3 is little endian and + * the data written here seems a bit odd. + * 0x8000 (shown in big endian form) is muted + * and volume increases to 0xffff it seems. + * I've yet to observe 0xffff being sent. + * Instead, full volume seems to be 0x0000. + * Per my understanding of 16 bit integers, this is seems strange, but maybe I'm missing something + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue input 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0110, 0x4001, &com_buff, 2); + /* Analogue input 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0111, 0x4001, &com_buff, 2); + /* Analogue input 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0114, 0x4001, &com_buff, 2); + /* Analogue input 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0115, 0x4001, &com_buff, 2); + /* Analogue input 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0118, 0x4001, &com_buff, 2); + /* Analogue input 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0119, 0x4001, &com_buff, 2); + /* Analogue input 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011c, 0x4001, &com_buff, 2); + /* Analogue input 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011d, 0x4001, &com_buff, 2); + + /* Set software sends to output */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue software return 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4001, &com_buff, 2); + /* Analogue software return 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4001, &com_buff, 2); + /* Analogue software return 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4001, &com_buff, 2); + /* Analogue software return 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4001, &com_buff, 2); + /* Analogue software return 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0108, 0x4001, &com_buff, 2); + /* Analogue software return 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0109, 0x4001, &com_buff, 2); + /* Analogue software return 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010c, 0x4001, &com_buff, 2); + /* Analogue software return 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010d, 0x4001, &com_buff, 2); + + /* Return to muting sends */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue fx return left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0120, 0x4001, &com_buff, 2); + /* Analogue fx return right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0121, 0x4001, &com_buff, 2); + + /* Analogue software input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4201, &com_buff, 2); + /* Analogue software input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4201, &com_buff, 2); + /* Analogue software input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4201, &com_buff, 2); + /* Analogue software input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4201, &com_buff, 2); + /* Analogue input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4201, &com_buff, 2); + /* Analogue input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4201, &com_buff, 2); + /* Analogue input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4201, &com_buff, 2); + /* Analogue input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4201, &com_buff, 2); + + /* Toggle allowing host control */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x02; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + + /* Do not dim fx returns */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0002, 0x2001, &com_buff, 1); + + /* Do not set fx returns to mono */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0001, 0x2001, &com_buff, 1); + + /* Mute the S/PDIF hardware loopback + * same odd volume logic here as above + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* S/PDIF hardware input 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0112, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0113, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0116, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0117, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011a, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011b, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011e, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011f, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010a, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010b, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010e, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010f, 0x4001, &com_buff, 2); + /* S/PDIF fx returns left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0122, 0x4001, &com_buff, 2); + /* S/PDIF fx returns right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0123, 0x4001, &com_buff, 2); + + /* Set the dropdown "Effect" to the first option */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0200, 0x4301, &com_buff, 1); + + /* Set the effect duration to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0400, 0x4301, &com_buff, 2); + + /* Set the effect volume and feedback to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* feedback: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0500, 0x4301, &com_buff, 1); + /* volume: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0300, 0x4301, &com_buff, 1); + + /* Set soft button hold duration */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x05; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0005, 0x2001, &com_buff, 1); + + /* Use dim LEDs for button of state */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0004, 0x2001, &com_buff, 1); + + return; +} + +#define MBOX3_DESCRIPTOR_SIZE 464 + +static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + int descriptor_size; + + descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); + + if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { + dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + return -ENODEV; + } + + dev_dbg(&dev->dev, "device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) + dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + + err = usb_reset_configuration(dev); + if (err < 0) + dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); + dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + mbox3_setup_48_24_magic(dev); + dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + + return 0; /* Successful boot */ +} #define MICROBOOK_BUF_SIZE 128 @@ -1324,6 +1631,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x0dba, 0x5000): + /* Digidesign Mbox 3 */ + return snd_usb_mbox3_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ -- 2.35.1

4 8

[PATCH] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 05 Aug '22

05 Aug '22

In commit <c3ecef21c3f26> ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ffc24afb5a7a..f0602077b385 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1054,7 +1054,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

2 2

[PATCH] ASoC: fsl_asrc_dma: fully initialize structs
by Sascha Hauer 05 Aug '22

05 Aug '22

The driver uses two statically ininitialized struct dma_slave_config, but only one of them is initialized to zero. Initialize config_be to zero as well to make sure that no fields are filled with random values. Let the compiler do this instead of explicitly calling memset() which makes it easier to read. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_asrc_dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cba..fec56dbb64e60 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -138,7 +138,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct dma_chan *tmp_chan = NULL, *be_chan = NULL; struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; - struct dma_slave_config config_fe, config_be; + struct dma_slave_config config_fe = {}, config_be = {}; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -181,7 +181,6 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, return -EINVAL; } - memset(&config_fe, 0, sizeof(config_fe)); ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); if (ret) { dev_err(dev, "failed to prepare DMA config for Front-End\n"); -- 2.30.2

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[PATCH][next] ASoC: q6asm: use kcalloc() instead of kzalloc()
by Gustavo A. R. Silva 28 Jul '22

28 Jul '22

Use 2-factor multiplication argument form kcalloc() instead of kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 9251d8548965..195780f75d05 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -513,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, return 0; } - buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC); + buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC); if (!buf) { spin_unlock_irqrestore(&ac->lock, flags); return -ENOMEM; -- 2.27.0

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Question about DPCM locking
by Kuninori Morimoto 26 Jul '22

26 Jul '22

Hi Takashi May I ask about this patch ? ASoC: soc-pcm: Fix and cleanup DPCM locking b7898396f4bbe160f546d0c5e9fa17cca9a7d153 Q1. It exchanges many function parameters - func(substream, xxx) + func(rtd, substream, xxx) ^^^^ I guess the purpose is it want to call snd_soc_dpcm_mutex_assert_held() inside the func ? But it looks very verbose. we can do like this, but am I misunderstanding ? func(substream, xxx) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_assert_held(rtd); ... } Q2. It added new __soc_pcm_close(). But soc_pcm_close() is using soc_pcm_clean() instead of __soc_pcm_close(). Is this just a mistake ? static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_lock(rtd); - soc_pcm_clean(substream, 0); + __soc_pcm_close(substream); snd_soc_dpcm_mutex_unlock(rtd); return 0; } Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH 0/4] ASoC: Intel: Mark BE DAIs as nonatomic for hsw and
by Cezary Rojewski 15 Jul '22

15 Jul '22

Address the warning: "Codec: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic" by marking BE DAI as nonatomic. Aligns with what is already done for FE DAIs. This patchset iterates the change over all HSW and BDW related machine board drivers. Cezary Rojewski (4): ASoC: Intel: hsw_rt5640: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt286: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5650: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5677: Mark BE DAI as nonatomic sound/soc/intel/boards/bdw-rt5650.c | 1 + sound/soc/intel/boards/bdw-rt5677.c | 1 + sound/soc/intel/boards/bdw_rt286.c | 1 + sound/soc/intel/boards/hsw_rt5640.c | 1 + 4 files changed, 4 insertions(+) -- 2.25.1

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[PATCH v8 00/14] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 15 Jul '22

15 Jul '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. changes since v7: - Use private_data rather than private_value to save control info - Clean up alsa control memory allocation/deallocation - Remove unnecessary whitespace - Get subsystem id from codec, rather than saving it separately changes since v6: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v5: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v4: - Fully remove tlv remnants from control add apis - Remove unnecessary debug - Rename variable to be more generic - Remove redundent length check from read/write control apis - Use SNDRV_CTL_ELEM_IFACE_CARD for firmware load controls - Make kcontrol add/remove synchronous - Load firmware asynchronous when loading via control - Used cached controls when reloading firmware; only delete controls when removing the driver itself - Improve kcontrol remove - Fix control write + notify - Cleanup of unnecessary code - Fix race condition when loading firmware before playback - Ensure errors are properly propogated - Fix include for Module parameters Stefan Binding (13): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 4 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 883 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 39 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 3 + sound/pci/hda/hda_cs_dsp_ctl.c | 240 +++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 39 ++ sound/pci/hda/patch_realtek.c | 26 +- 12 files changed, 1242 insertions(+), 5 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

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