May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH 5.15 058/371] sound/oss/dmasound: fix build when drivers are mixed =y/=m
by Greg Kroah-Hartman 08 May '23

08 May '23

From: Randy Dunlap <rdunlap(a)infradead.org> commit 9dd7c46346ca4390f84a7ea9933005eb1b175c15 upstream. When CONFIG_DMASOUND_ATARI=m and CONFIG_DMASOUND_Q40=y (or vice versa), dmasound_core.o can be built without dmasound_deinit() being defined, causing a build error: ERROR: modpost: "dmasound_deinit" [sound/oss/dmasound/dmasound_atari.ko] undefined! Modify dmasound_core.c and dmasound.h so that dmasound_deinit() is always available. The mixed modes (=y/=m) also mean that several variables and structs have to be declared in all cases. Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Suggested-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Link: lore.kernel.org/r/202204032138.EFT9qGEd-lkp@intel.com Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20220405234118.24830-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/oss/dmasound/dmasound.h | 6 ------ sound/oss/dmasound/dmasound_core.c | 24 +----------------------- 2 files changed, 1 insertion(+), 29 deletions(-) --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -88,11 +88,7 @@ static inline int ioctl_return(int __use */ extern int dmasound_init(void); -#ifdef MODULE extern void dmasound_deinit(void); -#else -#define dmasound_deinit() do { } while (0) -#endif /* description of the set-up applies to either hard or soft settings */ @@ -114,9 +110,7 @@ typedef struct { void *(*dma_alloc)(unsigned int, gfp_t); void (*dma_free)(void *, unsigned int); int (*irqinit)(void); -#ifdef MODULE void (*irqcleanup)(void); -#endif void (*init)(void); void (*silence)(void); int (*setFormat)(int); --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -206,12 +206,10 @@ module_param(writeBufSize, int, 0); MODULE_LICENSE("GPL"); -#ifdef MODULE static int sq_unit = -1; static int mixer_unit = -1; static int state_unit = -1; static int irq_installed; -#endif /* MODULE */ /* control over who can modify resources shared between play/record */ static fmode_t shared_resource_owner; @@ -391,9 +389,6 @@ static const struct file_operations mixe static void mixer_init(void) { -#ifndef MODULE - int mixer_unit; -#endif mixer_unit = register_sound_mixer(&mixer_fops, -1); if (mixer_unit < 0) return; @@ -1171,9 +1166,6 @@ static const struct file_operations sq_f static int sq_init(void) { const struct file_operations *fops = &sq_fops; -#ifndef MODULE - int sq_unit; -#endif sq_unit = register_sound_dsp(fops, -1); if (sq_unit < 0) { @@ -1366,9 +1358,6 @@ static const struct file_operations stat static int state_init(void) { -#ifndef MODULE - int state_unit; -#endif state_unit = register_sound_special(&state_fops, SND_DEV_STATUS); if (state_unit < 0) return state_unit ; @@ -1386,10 +1375,9 @@ static int state_init(void) int dmasound_init(void) { int res ; -#ifdef MODULE + if (irq_installed) return -EBUSY; -#endif /* Set up sound queue, /dev/audio and /dev/dsp. */ @@ -1408,9 +1396,7 @@ int dmasound_init(void) printk(KERN_ERR "DMA sound driver: Interrupt initialization failed\n"); return -ENODEV; } -#ifdef MODULE irq_installed = 1; -#endif printk(KERN_INFO "%s DMA sound driver rev %03d installed\n", dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) + @@ -1424,8 +1410,6 @@ int dmasound_init(void) return 0; } -#ifdef MODULE - void dmasound_deinit(void) { if (irq_installed) { @@ -1444,8 +1428,6 @@ void dmasound_deinit(void) unregister_sound_dsp(sq_unit); } -#else /* !MODULE */ - static int dmasound_setup(char *str) { int ints[6], size; @@ -1489,8 +1471,6 @@ static int dmasound_setup(char *str) __setup("dmasound=", dmasound_setup); -#endif /* !MODULE */ - /* * Conversion tables */ @@ -1577,9 +1557,7 @@ char dmasound_alaw2dma8[] = { EXPORT_SYMBOL(dmasound); EXPORT_SYMBOL(dmasound_init); -#ifdef MODULE EXPORT_SYMBOL(dmasound_deinit); -#endif EXPORT_SYMBOL(dmasound_write_sq); EXPORT_SYMBOL(dmasound_catchRadius); #ifdef HAS_8BIT_TABLES

1 0

[PATCH] Sound:last.c: fix code syle trailing white space - modify printk()->netdev_dbg() - Possible unnecessary KERN_INFO removed
by jack1989s＠gmail.com 08 May '23

08 May '23

From: Federico Di Lembo <jack1989s(a)gmail.com> Fixing coding style. Signed-off-by: Federico Di Lembo <jack1989s(a)gmail.com> --- sound/last.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/last.c b/sound/last.c index f0bb98780e70..b8cfd6ded95d 100644 --- a/sound/last.c +++ b/sound/last.c @@ -11,18 +11,18 @@ static int __init alsa_sound_last_init(void) { struct snd_card *card; int idx, ok = 0; - - printk(KERN_INFO "ALSA device list:\n"); + + netdev_dbg("ALSA device list:\n"); for (idx = 0; idx < SNDRV_CARDS; idx++) { card = snd_card_ref(idx); if (card) { - printk(KERN_INFO " #%i: %s\n", idx, card->longname); + netdev_dbg(" #%i: %s\n", idx, card->longname); snd_card_unref(card); ok++; } } if (ok == 0) - printk(KERN_INFO " No soundcards found.\n"); + netdev_dbg(" No soundcards found.\n"); return 0; } -- 2.34.1

1 0

[PATCH] ALSA: emu10k1: minor E-MU naming fixups
by Oswald Buddenhagen 08 May '23

08 May '23

- Fix mixer source port names. These will require some users to re-adjust their mixer settings, which seems acceptable: - The S/PDIF port is on the main 1010 card, not the 0202 daughter card - The 1616m CardBus card has all inputs on the dock, so there is no point in specifying it - Conversely, the 1010 card has "dispersed" inputs, so say where the ADAT port is, consistently with the S/PDIF port - The 1616m CardBus card is actually named E-MU 02 (due to the headphone output jack it has) - Fix capitalization of "E-MU" Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emu10k1_main.c | 16 +++++++------- sound/pci/emu10k1/emumixer.c | 36 ++++++++++++++++---------------- 2 files changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 3abdaf1b9624..5d9d772e3a13 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1310,7 +1310,7 @@ static const struct snd_emu_chip_details emu_chip_details[] = { /* Does NOT support sync daughter card (obviously). */ /* Tested by James(a)superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, - .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", + .driver = "Audigy2", .name = "E-MU 02 CardBus [MAEM8950]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1323,7 +1323,7 @@ static const struct snd_emu_chip_details emu_chip_details[] = { * MicroDock[M] to make it an E-MU 1616[m]. */ /* Does NOT support sync daughter card. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, - .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM8960]", + .driver = "Audigy2", .name = "E-MU 1010b PCI [MAEM8960]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1337,48 +1337,48 @@ static const struct snd_emu_chip_details emu_chip_details[] = { * still work. */ /* Does NOT support sync daughter card. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102, - .driver = "Audigy2", .name = "E-mu 1010 PCIe [MAEM8986]", + .driver = "Audigy2", .name = "E-MU 1010 PCIe [MAEM8986]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */ /* Tested by James(a)superbug.co.uk 8th July 2005. */ /* This is MAEM8810 "Hana", 0202 is MAEM8820 "Hamoa" */ /* Attach 0202 daughter card to make it an E-MU 1212m, OR an * AudioDock[M] to make it an E-MU 1820[m]. */ /* Supports sync daughter card. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, - .driver = "Audigy2", .name = "E-mu 1010 [MAEM8810]", + .driver = "Audigy2", .name = "E-MU 1010 [MAEM8810]", .id = "EMU1010", .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010}, /* EMU 1010 old revision */ /* This is MAEM8852 "HanaLiteLite" */ /* Supports sync daughter card. */ /* Tested by oswald.buddenhagen(a)gmx.de Mar 2023. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, - .driver = "Audigy2", .name = "E-mu 0404b PCI [MAEM8852]", + .driver = "Audigy2", .name = "E-MU 0404b PCI [MAEM8852]", .id = "EMU0404", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 new revision */ /* This is MAEM8850 "HanaLite" */ /* Supports sync daughter card. */ /* Tested by James(a)superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, - .driver = "Audigy2", .name = "E-mu 0404 [MAEM8850]", + .driver = "Audigy2", .name = "E-MU 0404 [MAEM8850]", .id = "EMU0404", .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* EMU0404 PCIe */ /* Does NOT support sync daughter card. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102, - .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]", + .driver = "Audigy2", .name = "E-MU 0404 PCIe [MAEM8984]", .id = "EMU0404", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1645,7 +1645,7 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102, - .driver = "EMU10K1", .name = "E-mu APS [PC545]", + .driver = "EMU10K1", .name = "E-MU APS [PC545]", .id = "APS", .emu10k1_chip = 1, .ecard = 1} , diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 8fce3413f4ae..a101fe7598c4 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -81,16 +81,16 @@ static const char * const emu1010_src_texts[] = { "Dock ADC3 Right", "0202 ADC Left", "0202 ADC Right", - "0202 SPDIF Left", - "0202 SPDIF Right", - "ADAT 0", - "ADAT 1", - "ADAT 2", - "ADAT 3", - "ADAT 4", - "ADAT 5", - "ADAT 6", - "ADAT 7", + "1010 SPDIF Left", + "1010 SPDIF Right", + "1010 ADAT 0", + "1010 ADAT 1", + "1010 ADAT 2", + "1010 ADAT 3", + "1010 ADAT 4", + "1010 ADAT 5", + "1010 ADAT 6", + "1010 ADAT 7", "DSP 0", "DSP 1", "DSP 2", @@ -129,14 +129,14 @@ static const char * const emu1010_src_texts[] = { static const char * const emu1616_src_texts[] = { "Silence", - "Dock Mic A", - "Dock Mic B", - "Dock ADC1 Left", - "Dock ADC1 Right", - "Dock ADC2 Left", - "Dock ADC2 Right", - "Dock SPDIF Left", - "Dock SPDIF Right", + "Mic A", + "Mic B", + "ADC1 Left", + "ADC1 Right", + "ADC2 Left", + "ADC2 Right", + "SPDIF Left", + "SPDIF Right", "ADAT 0", "ADAT 1", "ADAT 2", -- 2.40.0.152.g15d061e6df

2 1

[PATCH 0/6] ALSA: emu10k1: Various improvements related to locking
by Oswald Buddenhagen 08 May '23

08 May '23

--- This patch series needs to be applied on top of the patch titled "ALSA: emu10k1: use more existing defines instead of open-coded numbers". Oswald Buddenhagen (6): ALSA: emu10k1: remove pointless locks from timer code ALSA: emu10k1: remove pointless locks from /proc code ALSA: emu10k1: use the right lock in snd_emu10k1_shared_spdif_put() ALSA: emu10k1: fix locking in snd_emu1010_fpga_link_dst_src_write() ALSA: core: update comment on snd_card.controls_rwsem ALSA: emu10k1: remove now superfluous mixer locking include/sound/core.h | 2 +- sound/pci/emu10k1/emufx.c | 7 ------- sound/pci/emu10k1/emumixer.c | 32 ++------------------------------ sound/pci/emu10k1/emupcm.c | 2 -- sound/pci/emu10k1/emuproc.c | 6 ------ sound/pci/emu10k1/io.c | 25 +++++++++++++++++-------- sound/pci/emu10k1/timer.c | 6 ------ 7 files changed, 20 insertions(+), 60 deletions(-) -- 2.40.0.152.g15d061e6df

2 7

[PATCH v2] soundwire: bus: Don't filter slave alerts
by Charles Keepax 08 May '23

08 May '23

It makes sense to have only a single point responsible for ensuring that all currently pending IRQs are handled. The current code in sdw_handle_slave_alerts confusingly splits this process in two. This code will loop until the asserted IRQs are cleared but it will only handle IRQs that were already asserted when it was called. This means the caller must also loop (either manually, or through its IRQ mechanism) until the IRQs are all handled. It makes sense to either do all the looping in sdw_handle_slave_alerts or do no looping there and let the host controller repeatedly call it until things are handled. There are realistically two sensible host controllers, those that will generate an IRQ when the alert status changes and those that will generate an IRQ continuously whilst the alert status is high. The current code will work fine for the second of those systems but not the first with out additional looping in the host controller. Removing the code that filters out new IRQs whilst the handler is running enables both types of host controller to be supported and simplifies the code. The code will still only loop up to SDW_READ_INTR_CLEAR_RETRY times, so it shouldn't be possible for it to get completely stuck handling IRQs forever, and if you are generating IRQs faster than you can handle them you likely have bigger problems anyway. This fixes an issue on the Cadence SoundWire IP, which only generates IRQs on an alert status change, where an alert which arrives whilst another alert is being handled will never be handled and will block all future alerts from being handled. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Update commit message Thanks, Charles drivers/soundwire/bus.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 1ea6a64f8c4a5..338f4f0b5d0cc 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1588,7 +1588,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) unsigned long port; bool slave_notify; u8 sdca_cascade = 0; - u8 buf, buf2[2], _buf, _buf2[2]; + u8 buf, buf2[2]; bool parity_check; bool parity_quirk; @@ -1745,9 +1745,9 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) "SDW_SCP_INT1 recheck read failed:%d\n", ret); goto io_err; } - _buf = ret; + buf = ret; - ret = sdw_nread_no_pm(slave, SDW_SCP_INTSTAT2, 2, _buf2); + ret = sdw_nread_no_pm(slave, SDW_SCP_INTSTAT2, 2, buf2); if (ret < 0) { dev_err(&slave->dev, "SDW_SCP_INT2/3 recheck read failed:%d\n", ret); @@ -1765,12 +1765,8 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) } /* - * Make sure no interrupts are pending, but filter to limit loop - * to interrupts identified in the first status read + * Make sure no interrupts are pending */ - buf &= _buf; - buf2[0] &= _buf2[0]; - buf2[1] &= _buf2[1]; stat = buf || buf2[0] || buf2[1] || sdca_cascade; /* -- 2.30.2

3 2

[PATCH v3 0/7] soundwire: qcom: add support for SM8550 (Soundwire v2.0.0)
by Krzysztof Kozlowski 08 May '23

08 May '23

Hi, Changes since v2: 1. Use uint16 for qcom,ports-sinterval. DTS will be fixed in separate patchset. 2. Add tags. Changes since v1: 1. Patch 1: Increase maxItems to 16 for port-related properties. 2. Re-order patch 1 and 2. 3. Patch 3: Drop unneeded semicolon. 4. Patch 5: Fix lang typo in subject. Best regards, Krzysztof Krzysztof Kozlowski (7): dt-bindings: soundwire: qcom: add Qualcomm Soundwire v2.0.0 dt-bindings: soundwire: qcom: add 16-bit sample interval soundwire: qcom: allow 16-bit sample interval for ports soundwire: qcom: use consistently 'ctrl' as state variable name soundwire: qcom: prepare for handling different register layouts soundwire: qcom: add support for v2.0.0 controller soundwire: qcom: use tabs for indentation in defines .../bindings/soundwire/qcom,soundwire.yaml | 39 +- drivers/soundwire/qcom.c | 387 ++++++++++++------ 2 files changed, 287 insertions(+), 139 deletions(-) -- 2.34.1

3 10

[PATCH v2 1/5] ASoC: tas2781: Add Header file for tas2781 driver
by Shenghao Ding 08 May '23

08 May '23

Create Header file for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v2: - fixed issue | Reported-by: kernel test robot <lkp(a)intel.com> | Link: https://lore.kernel.org/oe-kbuild-all/202305022033.LiI7Ojm4-lkp@intel.com/ Changes to be committed: new file: include/sound/tas2781-dsp.h new file: include/sound/tas2781-tlv.h new file: include/sound/tas2781.h --- include/sound/tas2781-dsp.h | 189 ++++++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 22 +++++ include/sound/tas2781.h | 176 +++++++++++++++++++++++++++++++++ 3 files changed, 387 insertions(+) create mode 100644 include/sound/tas2781-dsp.h create mode 100644 include/sound/tas2781-tlv.h create mode 100644 include/sound/tas2781.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h new file mode 100644 index 000000000000..091c3ffade17 --- /dev/null +++ b/include/sound/tas2781-dsp.h @@ -0,0 +1,189 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define RCA_CONFIGID_BYPASS_ALL 0 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + int nr_retry; + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int nr_cmds; + unsigned int blk_size; + unsigned int nr_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int nr_blk; + struct tasdev_blk *dev_blks; +}; + +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short nr_programs; + struct tasdevice_prog *programs; + unsigned short nr_configurations; + struct tasdevice_config *configs; + unsigned short nr_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +void tasdevice_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_config_info_remove(void *context); +void tasdevice_dsp_remove(void *context); +int tasdevice_dsp_parser(void *context); +int tasdevice_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); +int tasdevice_prmg_load(void *context, int prm_no); +int tasdevice_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_tuning_switch(void *context, int state); +int tas2781_load_calibration(void *context, char *file_name, + enum channel i); + +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h new file mode 100644 index 000000000000..f4310dce655a --- /dev/null +++ b/include/sound/tas2781-tlv.h @@ -0,0 +1,22 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_TLV_H__ +#define __TAS2781_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +#endif /* __TAS2781_LIB_H__ */ diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h new file mode 100644 index 000000000000..c98955bda07e --- /dev/null +++ b/include/sound/tas2781.h @@ -0,0 +1,176 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include <linux/kernel.h> +#include "tas2781-dsp.h" + +/* version number */ +#define TAS2781_DRV_VER 1 +#define SMARTAMP_MODULE_NAME "tas2781" +#define TAS2781_GLOBAL_ADDR 0x40 +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) +#define TASDEVICE_MAX_CHANNELS 8 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) (reg % (256 * 128)) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + +/*Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x02) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* Enable Global addresses */ +#define TAS2781_MISC_CFG2 TASDEVICE_REG(0x0, 0X0, 0x07) +#define TAS2781_GLOBAL_ADDR_MASK BIT(1) +#define TAS2781_GLOBAL_ADDR_ENABLE BIT(1) + +/*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +struct tasdevice { + struct tasdevice_fw *cali_data_fmw; + unsigned int dev_addr; + unsigned int err_code; + unsigned char cur_book; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + int ref_cnt; + unsigned int dev_addr; + unsigned char cur_book; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct calidata { + unsigned char *data; + unsigned long total_sz; +}; + +struct tasdevice_priv { + struct tasdevice tasdevice[max_chn]; + struct tasdevice_irqinfo irq_info; + struct global_addr glb_addr; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct regmap *regmap; + struct mutex codec_lock; + struct calidata cali_data; + struct gpio_desc *reset; + struct device *dev; + struct tm tm; + + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char cal_binaryname[max_chn][64]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned char ndev; + unsigned int magic_num; + unsigned int chip_id; + unsigned int sysclk; + const char *acpi_subsystem_id; + int index; + int cur_prog; + int cur_conf; + int fw_state; + void *client; + void *codec; + bool isacpi; + bool playback_started; + void (*set_global_mode)(struct tasdevice_priv *tas_priv); + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); +}; + +void tas2781_reset(struct tasdevice_priv *tas_dev); +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)); +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); +int tasdevice_init(struct tasdevice_priv *tas_priv); +void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int *value); +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int value); +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value); +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); + +#endif /* __TAS2781_H__ */ -- 2.34.1

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[PATCH] ALSA: hda/realtek: Add quirk for ASUS GU603ZM
by frshuov＠163.com 08 May '23

08 May '23

From: Larry Chi <frshuov(a)163.com> ASUS ROG Zephyrus M16 2022 GU603ZM (1043:1c62) added SND_PCI_QUIRK for speaker and headset-mic working properly Signed-off-by: Larry Chi <frshuov(a)163.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b..374a9755d19a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9522,6 +9522,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603ZM", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), -- 2.40.1

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[PATCH] ALSA: hda/realtek: Add a quirk for HP EliteDesk 805
by Ai Chao 08 May '23

08 May '23

Add a quirk for HP EliteDesk 805 to fixup ALC3867 headset MIC no sound. Signed-off-by: Ai Chao <aichao(a)kylinos.cn> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 068ce0db9562..cbb3c3f6d502 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11662,6 +11662,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x103c, 0x870c, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB), + SND_PCI_QUIRK(0x103c, 0x872b, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x103c, 0x877e, "HP 288 Pro G6", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2), -- 2.25.1

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[PATCH v2 0/1] ALSA: hda/realtek: Add quirk for 2nd ASUS GU603
by Luke D. Jones 08 May '23

08 May '23

Add quirk for GU603 with alt codec ID. Changelog: - V2 - Move line to correct place in order Luke D. Jones (1): ALSA: hda/realtek: Add quirk for 2nd ASUS GU603 sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) -- 2.40.0

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