mirror of
https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
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sound fixes for 6.10-rc1
A collection of small fixes for 6.10-rc1. Most of changes are various device-specific fixes and quirks, while there are a few small changes in ALSA core timer and module / built-in fixes. -----BEGIN PGP SIGNATURE----- iQJCBAABCAAsFiEEIXTw5fNLNI7mMiVaLtJE4w1nLE8FAmZPUagOHHRpd2FpQHN1 c2UuZGUACgkQLtJE4w1nLE9LYg//QurZR7KBAvim5LcsVDLE5pFUjW0v3fz0+vKB /UpoK1EVxc9pqNXzKi8YDiRoKZY8J8krGHd5FV44qhZl2nVJ87hXbHU5b/i29QUu 4xKC1pMmF0ncJ8qMGhzTynyxw0Hr7soCcxz+4ApDzN/pyzc7QTPEaUB1ND7jTB2z bcYgXyFprJQ1RmsV9u2mGhNEv3tYRaQO1GNxr9ktO/I13CCKd7LkGUSxo5UfOFwC bIrpqG35MDzeVrxEfB1UHlyKhULf9fmpUW0OYYS/DMQFptRa+PXEOgzN81wrhNwL sp2k41x4uRtKrB1DFCeweis4m0OHbV0yakkV/3PTdONzJk4PxWoPuGP4uZyoNz3B FwexeSpZICpgGHeS4WuS0RW3SbQ9n/3d33nzpCYrojyxqCuc6UXGPyiq6QHUVtXZ LnOPyeJRIhS52wpELByJmcnf9ev4ImLSnGWUzz/Mf5dFZCVSXKWVvgQ+UcWbZZnj vTp0mTMQUjuVhE0KuRawMx2YSUU7nuRBukFBihjIRSYJYvZETN7WNjMUA3UnpG1d uKXJaTEm2UqlZtsnKkXrWNIpj4EQjoZo0qgx4ZwSYicLgXDJ/WlHvltdo9fJpRh3 u23957ye7wJ4JMikqyhd0Wzh/1UwOs4GTMWTcim6pKXwlkn8TwCB1F/OT/6xqlYZ gScnqBQ= =VeU/ -----END PGP SIGNATURE----- Merge tag 'sound-fix-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "A collection of small fixes for 6.10-rc1. Most of changes are various device-specific fixes and quirks, while there are a few small changes in ALSA core timer and module / built-in fixes" * tag 'sound-fix-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda/realtek: fix mute/micmute LEDs don't work for ProBook 440/460 G11. ALSA: core: Enable proc module when CONFIG_MODULES=y ALSA: core: Fix NULL module pointer assignment at card init ALSA: hda/realtek: Enable headset mic of JP-IK LEAP W502 with ALC897 ASoC: dt-bindings: stm32: Ensure compatible pattern matches whole string ASoC: tas2781: Fix wrong loading calibrated data sequence ASoC: tas2552: Add TX path for capturing AUDIO-OUT data ALSA: usb-audio: Fix for sampling rates support for Mbox3 Documentation: sound: Fix trailing whitespaces ALSA: timer: Set lower bound of start tick time ASoC: codecs: ES8326: solve hp and button detect issue ASoC: rt5645: mic-in detection threshold modification ASoC: Intel: sof_sdw_rt_sdca_jack_common: Use name_prefix for `-sdca` detection
This commit is contained in:
Linus Torvalds
commit
041c9f71a4
@ -72,7 +72,7 @@ patternProperties:
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properties:
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compatible:
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description: Compatible for SAI sub-block A or B.
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pattern: "st,stm32-sai-sub-[ab]"
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pattern: "^st,stm32-sai-sub-[ab]$"
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"#sound-dai-cells":
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const: 0
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@ -15,7 +15,7 @@ problem is broken BIOS, and the rest is the driver implementation.
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This document explains the brief trouble-shooting and debugging
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methods for the HD-audio hardware.
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The HD-audio component consists of two parts: the controller chip and
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The HD-audio component consists of two parts: the controller chip and
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the codec chips on the HD-audio bus. Linux provides a single driver
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for all controllers, snd-hda-intel. Although the driver name contains
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a word of a well-known hardware vendor, it's not specific to it but for
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@ -81,7 +81,7 @@ the wake-up timing. It wakes up a few samples before actually
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processing the data on the buffer. This caused a lot of problems, for
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example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts
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an artificial delay to the wake up timing. This delay is controlled
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via ``bdl_pos_adj`` option.
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via ``bdl_pos_adj`` option.
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When ``bdl_pos_adj`` is a negative value (as default), it's assigned to
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an appropriate value depending on the controller chip. For Intel
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@ -144,7 +144,7 @@ see a regression wrt the sound quality (stuttering, etc) or a lock-up
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in the recent kernel, try to pass ``enable_msi=0`` option to disable
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MSI. If it works, you can add the known bad device to the blacklist
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defined in hda_intel.c. In such a case, please report and give the
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patch back to the upstream developer.
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patch back to the upstream developer.
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HD-Audio Codec
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@ -375,7 +375,7 @@ HD-Audio Reconfiguration
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------------------------
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This is an experimental feature to allow you re-configure the HD-audio
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codec dynamically without reloading the driver. The following sysfs
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files are available under each codec-hwdep device directory (e.g.
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files are available under each codec-hwdep device directory (e.g.
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/sys/class/sound/hwC0D0):
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vendor_id
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@ -433,7 +433,7 @@ re-configure based on that state, run like below:
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::
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# echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs
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# echo 1 > /sys/class/sound/hwC0D0/reconfig
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# echo 1 > /sys/class/sound/hwC0D0/reconfig
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Hint Strings
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@ -494,7 +494,7 @@ indep_hp (bool)
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mixer control, if available
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add_stereo_mix_input (bool)
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add the stereo mix (analog-loopback mix) to the input mux if
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available
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available
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add_jack_modes (bool)
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add "xxx Jack Mode" enum controls to each I/O jack for allowing to
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change the headphone amp and mic bias VREF capabilities
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@ -504,7 +504,7 @@ power_save_node (bool)
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stream states
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power_down_unused (bool)
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power down the unused widgets, a subset of power_save_node, and
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will be dropped in future
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will be dropped in future
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add_hp_mic (bool)
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add the headphone to capture source if possible
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hp_mic_detect (bool)
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@ -603,7 +603,7 @@ present.
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The patch module option is specific to each card instance, and you
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need to give one file name for each instance, separated by commas.
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For example, if you have two cards, one for an on-board analog and one
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For example, if you have two cards, one for an on-board analog and one
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for an HDMI video board, you may pass patch option like below:
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::
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@ -2,7 +2,7 @@
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//
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// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier
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//
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// Copyright (C) 2022 - 2023 Texas Instruments Incorporated
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// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
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// https://www.ti.com
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//
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// The TAS2781 driver implements a flexible and configurable
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@ -13,8 +13,8 @@
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// Author: Kevin Lu <kevin-lu@ti.com>
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//
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#ifndef __TASDEVICE_DSP_H__
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#define __TASDEVICE_DSP_H__
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#ifndef __TAS2781_DSP_H__
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#define __TAS2781_DSP_H__
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#define MAIN_ALL_DEVICES 0x0d
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#define MAIN_DEVICE_A 0x01
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@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context);
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int tasdevice_select_tuningprm_cfg(void *context, int prm,
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int cfg_no, int rca_conf_no);
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int tasdevice_prmg_load(void *context, int prm_no);
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int tasdevice_prmg_calibdata_load(void *context, int prm_no);
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void tasdevice_tuning_switch(void *context, int state);
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int tas2781_load_calibration(void *context, char *file_name,
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unsigned short i);
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@ -50,7 +50,7 @@ MODULE_PARM_DESC(slots, "Module names assigned to the slots.");
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static int module_slot_match(struct module *module, int idx)
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{
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int match = 1;
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#ifdef MODULE
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#ifdef CONFIG_MODULES
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const char *s1, *s2;
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if (!module || !*module->name || !slots[idx])
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@ -77,7 +77,7 @@ static int module_slot_match(struct module *module, int idx)
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if (!c1)
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break;
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}
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#endif /* MODULE */
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#endif /* CONFIG_MODULES */
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return match;
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}
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@ -311,10 +311,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent,
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}
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card->dev = parent;
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card->number = idx;
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#ifdef MODULE
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WARN_ON(!module);
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WARN_ON(IS_MODULE(CONFIG_SND) && !module);
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card->module = module;
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#endif
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INIT_LIST_HEAD(&card->devices);
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init_rwsem(&card->controls_rwsem);
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rwlock_init(&card->ctl_files_rwlock);
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@ -969,7 +967,7 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer)
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#endif
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#ifdef MODULE
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#ifdef CONFIG_MODULES
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static void snd_card_module_info_read(struct snd_info_entry *entry,
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struct snd_info_buffer *buffer)
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{
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@ -997,7 +995,7 @@ int __init snd_card_info_init(void)
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if (snd_info_register(entry) < 0)
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return -ENOMEM; /* freed in error path */
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#ifdef MODULE
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#ifdef CONFIG_MODULES
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entry = snd_info_create_module_entry(THIS_MODULE, "modules", NULL);
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if (!entry)
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return -ENOMEM;
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@ -544,6 +544,14 @@ static int snd_timer_start1(struct snd_timer_instance *timeri,
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SNDRV_TIMER_IFLG_START))
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return -EBUSY;
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/* check the actual time for the start tick;
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* bail out as error if it's way too low (< 100us)
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*/
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if (start) {
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if ((u64)snd_timer_hw_resolution(timer) * ticks < 100000)
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return -EINVAL;
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}
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if (start)
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timeri->ticks = timeri->cticks = ticks;
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else if (!timeri->cticks)
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@ -10194,8 +10194,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c89, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c8a, "HP EliteBook 630", ALC236_FIXUP_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c8c, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c8d, "HP ProBook 440 G11", ALC236_FIXUP_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c8e, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c90, "HP EliteBook 640", ALC236_FIXUP_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c91, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED),
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SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
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@ -12028,6 +12031,7 @@ enum {
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ALC897_FIXUP_LENOVO_HEADSET_MODE,
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ALC897_FIXUP_HEADSET_MIC_PIN2,
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ALC897_FIXUP_UNIS_H3C_X500S,
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ALC897_FIXUP_HEADSET_MIC_PIN3,
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};
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static const struct hda_fixup alc662_fixups[] = {
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@ -12474,10 +12478,18 @@ static const struct hda_fixup alc662_fixups[] = {
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{}
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},
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},
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[ALC897_FIXUP_HEADSET_MIC_PIN3] = {
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.type = HDA_FIXUP_PINS,
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.v.pins = (const struct hda_pintbl[]) {
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{ 0x19, 0x03a11050 }, /* use as headset mic */
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{ }
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},
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},
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};
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static const struct snd_pci_quirk alc662_fixup_tbl[] = {
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SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
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SND_PCI_QUIRK(0x1019, 0x9859, "JP-IK LEAP W502", ALC897_FIXUP_HEADSET_MIC_PIN3),
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SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC),
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SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC),
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SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
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@ -829,8 +829,8 @@ static void es8326_jack_detect_handler(struct work_struct *work)
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/* mute adc when mic path switch */
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regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44);
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regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66);
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es8326->hp = 0;
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}
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es8326->hp = 0;
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regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
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regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a);
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regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03);
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@ -981,7 +981,7 @@ static int es8326_resume(struct snd_soc_component *component)
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regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0);
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usleep_range(10000, 15000);
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regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9);
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regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb);
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regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xd8);
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/* set headphone default type and detect pin */
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regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83);
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regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05);
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@ -1018,7 +1018,7 @@ static int es8326_resume(struct snd_soc_component *component)
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regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F);
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/* select vdda as micbias source */
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regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x23);
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regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x03);
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/* set dac dsmclip = 1 */
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regmap_write(es8326->regmap, ES8326_DAC_DSM, 0x08);
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regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15);
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@ -81,6 +81,7 @@ static const struct reg_sequence init_list[] = {
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static const struct reg_sequence rt5650_init_list[] = {
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{0xf6, 0x0100},
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{RT5645_PWR_ANLG1, 0x02},
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{RT5645_IL_CMD3, 0x0018},
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};
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static const struct reg_default rt5645_reg[] = {
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||||
|
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@ -2,7 +2,8 @@
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||||
/*
|
||||
* tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
|
||||
*
|
||||
* Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
|
||||
* Copyright (C) 2014 - 2024 Texas Instruments Incorporated -
|
||||
* https://www.ti.com
|
||||
*
|
||||
* Author: Dan Murphy <dmurphy@ti.com>
|
||||
*/
|
||||
@ -119,12 +120,14 @@ static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] =
|
||||
&tas2552_input_mux_control),
|
||||
|
||||
SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
|
||||
SND_SOC_DAPM_AIF_OUT("ASI OUT", "DAC Capture", 0, SND_SOC_NOPM, 0, 0),
|
||||
SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
|
||||
SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0),
|
||||
SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0),
|
||||
SND_SOC_DAPM_POST("Post Event", tas2552_post_event),
|
||||
|
||||
SND_SOC_DAPM_OUTPUT("OUT")
|
||||
SND_SOC_DAPM_OUTPUT("OUT"),
|
||||
SND_SOC_DAPM_INPUT("DMIC")
|
||||
};
|
||||
|
||||
static const struct snd_soc_dapm_route tas2552_audio_map[] = {
|
||||
@ -134,6 +137,7 @@ static const struct snd_soc_dapm_route tas2552_audio_map[] = {
|
||||
{"ClassD", NULL, "Input selection"},
|
||||
{"OUT", NULL, "ClassD"},
|
||||
{"ClassD", NULL, "PLL"},
|
||||
{"ASI OUT", NULL, "DMIC"}
|
||||
};
|
||||
|
||||
#ifdef CONFIG_PM
|
||||
@ -538,6 +542,13 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
|
||||
.rates = SNDRV_PCM_RATE_8000_192000,
|
||||
.formats = TAS2552_FORMATS,
|
||||
},
|
||||
.capture = {
|
||||
.stream_name = "Capture",
|
||||
.channels_min = 2,
|
||||
.channels_max = 2,
|
||||
.rates = SNDRV_PCM_RATE_8000_192000,
|
||||
.formats = TAS2552_FORMATS,
|
||||
},
|
||||
.ops = &tas2552_speaker_dai_ops,
|
||||
},
|
||||
};
|
||||
|
||||
@ -2151,6 +2151,24 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv,
|
||||
return ret;
|
||||
}
|
||||
|
||||
static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i)
|
||||
{
|
||||
struct tasdevice_calibration *cal;
|
||||
struct tasdevice_fw *cal_fmw;
|
||||
|
||||
cal_fmw = priv->tasdevice[i].cali_data_fmw;
|
||||
|
||||
/* No calibrated data for current devices, playback will go ahead. */
|
||||
if (!cal_fmw)
|
||||
return;
|
||||
|
||||
cal = cal_fmw->calibrations;
|
||||
if (cal)
|
||||
return;
|
||||
|
||||
load_calib_data(priv, &cal->dev_data);
|
||||
}
|
||||
|
||||
int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
|
||||
int cfg_no, int rca_conf_no)
|
||||
{
|
||||
@ -2210,21 +2228,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
|
||||
for (i = 0; i < tas_priv->ndev; i++) {
|
||||
if (tas_priv->tasdevice[i].is_loaderr == true)
|
||||
continue;
|
||||
else if (tas_priv->tasdevice[i].is_loaderr == false
|
||||
&& tas_priv->tasdevice[i].is_loading == true) {
|
||||
struct tasdevice_fw *cal_fmw =
|
||||
tas_priv->tasdevice[i].cali_data_fmw;
|
||||
|
||||
if (cal_fmw) {
|
||||
struct tasdevice_calibration
|
||||
*cal = cal_fmw->calibrations;
|
||||
|
||||
if (cal)
|
||||
load_calib_data(tas_priv,
|
||||
&(cal->dev_data));
|
||||
}
|
||||
if (tas_priv->tasdevice[i].is_loaderr == false &&
|
||||
tas_priv->tasdevice[i].is_loading == true)
|
||||
tas_priv->tasdevice[i].cur_prog = prm_no;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
@ -2245,11 +2251,15 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
|
||||
tasdevice_load_data(tas_priv, &(conf->dev_data));
|
||||
for (i = 0; i < tas_priv->ndev; i++) {
|
||||
if (tas_priv->tasdevice[i].is_loaderr == true) {
|
||||
status |= 1 << (i + 4);
|
||||
status |= BIT(i + 4);
|
||||
continue;
|
||||
} else if (tas_priv->tasdevice[i].is_loaderr == false
|
||||
&& tas_priv->tasdevice[i].is_loading == true)
|
||||
}
|
||||
|
||||
if (tas_priv->tasdevice[i].is_loaderr == false &&
|
||||
tas_priv->tasdevice[i].is_loading == true) {
|
||||
tasdev_load_calibrated_data(tas_priv, i);
|
||||
tas_priv->tasdevice[i].cur_conf = cfg_no;
|
||||
}
|
||||
}
|
||||
} else
|
||||
dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n",
|
||||
@ -2308,65 +2318,6 @@ int tasdevice_prmg_load(void *context, int prm_no)
|
||||
}
|
||||
EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB);
|
||||
|
||||
int tasdevice_prmg_calibdata_load(void *context, int prm_no)
|
||||
{
|
||||
struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context;
|
||||
struct tasdevice_fw *tas_fmw = tas_priv->fmw;
|
||||
struct tasdevice_prog *program;
|
||||
int prog_status = 0;
|
||||
int i;
|
||||
|
||||
if (!tas_fmw) {
|
||||
dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__);
|
||||
goto out;
|
||||
}
|
||||
|
||||
if (prm_no >= tas_fmw->nr_programs) {
|
||||
dev_err(tas_priv->dev,
|
||||
"%s: prm(%d) is not in range of Programs %u\n",
|
||||
__func__, prm_no, tas_fmw->nr_programs);
|
||||
goto out;
|
||||
}
|
||||
|
||||
for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) {
|
||||
if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) {
|
||||
tas_priv->tasdevice[i].cur_conf = -1;
|
||||
tas_priv->tasdevice[i].is_loading = true;
|
||||
prog_status++;
|
||||
}
|
||||
tas_priv->tasdevice[i].is_loaderr = false;
|
||||
}
|
||||
|
||||
if (prog_status) {
|
||||
program = &(tas_fmw->programs[prm_no]);
|
||||
tasdevice_load_data(tas_priv, &(program->dev_data));
|
||||
for (i = 0; i < tas_priv->ndev; i++) {
|
||||
if (tas_priv->tasdevice[i].is_loaderr == true)
|
||||
continue;
|
||||
else if (tas_priv->tasdevice[i].is_loaderr == false
|
||||
&& tas_priv->tasdevice[i].is_loading == true) {
|
||||
struct tasdevice_fw *cal_fmw =
|
||||
tas_priv->tasdevice[i].cali_data_fmw;
|
||||
|
||||
if (cal_fmw) {
|
||||
struct tasdevice_calibration *cal =
|
||||
cal_fmw->calibrations;
|
||||
|
||||
if (cal)
|
||||
load_calib_data(tas_priv,
|
||||
&(cal->dev_data));
|
||||
}
|
||||
tas_priv->tasdevice[i].cur_prog = prm_no;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
out:
|
||||
return prog_status;
|
||||
}
|
||||
EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load,
|
||||
SND_SOC_TAS2781_FMWLIB);
|
||||
|
||||
void tasdevice_tuning_switch(void *context, int state)
|
||||
{
|
||||
struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context;
|
||||
|
||||
@ -2,7 +2,7 @@
|
||||
//
|
||||
// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier
|
||||
//
|
||||
// Copyright (C) 2022 - 2023 Texas Instruments Incorporated
|
||||
// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
|
||||
// https://www.ti.com
|
||||
//
|
||||
// The TAS2563/TAS2781 driver implements a flexible and configurable
|
||||
@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw,
|
||||
__func__, tas_priv->cal_binaryname[i]);
|
||||
}
|
||||
|
||||
tasdevice_prmg_calibdata_load(tas_priv, 0);
|
||||
tasdevice_prmg_load(tas_priv, 0);
|
||||
tas_priv->cur_prog = 0;
|
||||
out:
|
||||
if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) {
|
||||
|
||||
@ -109,7 +109,7 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *d
|
||||
return -ENOMEM;
|
||||
|
||||
for (i = 0; i < ARRAY_SIZE(need_sdca_suffix); i++) {
|
||||
if (strstr(codec_dai->name, need_sdca_suffix[i])) {
|
||||
if (strstr(component->name_prefix, need_sdca_suffix[i])) {
|
||||
/* Add -sdca suffix for existing UCMs */
|
||||
card->components = devm_kasprintf(card->dev, GFP_KERNEL,
|
||||
"%s-sdca", card->components);
|
||||
|
||||
@ -1740,7 +1740,7 @@ static void mbox3_set_format_quirk(struct snd_usb_substream *subs,
|
||||
u32 current_rate;
|
||||
|
||||
// Get current rate from card and check if changing it is needed
|
||||
snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
|
||||
snd_usb_ctl_msg(subs->dev, usb_rcvctrlpipe(subs->dev, 0),
|
||||
0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
|
||||
current_rate = le32_to_cpu(buff4);
|
||||
dev_dbg(&subs->dev->dev,
|
||||
@ -1765,7 +1765,7 @@ static void mbox3_set_format_quirk(struct snd_usb_substream *subs,
|
||||
|
||||
// Check whether the change was successful
|
||||
buff4 = 0;
|
||||
snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
|
||||
snd_usb_ctl_msg(subs->dev, usb_rcvctrlpipe(subs->dev, 0),
|
||||
0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
|
||||
if (new_rate != le32_to_cpu(buff4))
|
||||
dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate");
|
||||
|
||||
Loading…
Reference in New Issue
Block a user