diff options
Diffstat (limited to 'sound')
86 files changed, 1070 insertions, 418 deletions
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 15dc7160ba34..8cfdaee77905 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -431,33 +431,17 @@ static const struct snd_malloc_ops snd_dma_iram_ops = { */ static void *snd_dma_dev_alloc(struct snd_dma_buffer *dmab, size_t size) { - void *p; - - p = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); -#ifdef CONFIG_X86 - if (p && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wc((unsigned long)p, PAGE_ALIGN(size) >> PAGE_SHIFT); -#endif - return p; + return dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); } static void snd_dma_dev_free(struct snd_dma_buffer *dmab) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wb((unsigned long)dmab->area, - PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT); -#endif dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); } static int snd_dma_dev_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); -#endif return dma_mmap_coherent(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } @@ -471,10 +455,6 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ -#ifdef CONFIG_X86 -/* On x86, share the same ops as the standard dev ops */ -#define snd_dma_wc_ops snd_dma_dev_ops -#else /* CONFIG_X86 */ static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -497,7 +477,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .free = snd_dma_wc_free, .mmap = snd_dma_wc_mmap, }; -#endif /* CONFIG_X86 */ #ifdef CONFIG_SND_DMA_SGBUF static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 3f35972e1cf7..161a9711cd63 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -119,21 +119,18 @@ static int i915_component_master_match(struct device *dev, int subcomponent, /* check whether Intel graphics is present and reachable */ static int i915_gfx_present(struct pci_dev *hdac_pci) { - unsigned int class = PCI_BASE_CLASS_DISPLAY << 16; struct pci_dev *display_dev = NULL; - bool match = false; - do { - display_dev = pci_get_class(class, display_dev); - - if (display_dev && display_dev->vendor == PCI_VENDOR_ID_INTEL && + for_each_pci_dev(display_dev) { + if (display_dev->vendor == PCI_VENDOR_ID_INTEL && + (display_dev->class >> 16) == PCI_BASE_CLASS_DISPLAY && connectivity_check(display_dev, hdac_pci)) { pci_dev_put(display_dev); - match = true; + return true; } - } while (!match && display_dev); + } - return match; + return false; } /** diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index a8fe01764b25..ec9cbb219bc1 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -196,6 +196,12 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .ident = "UP-WHL", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + } + }, {} } }, @@ -358,6 +364,12 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .ident = "UPX-TGL", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + } + }, {} } }, diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 4063da378283..9db5ccd9aa2d 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -55,8 +55,8 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) /* find max number of channels based on format_configuration */ if (fmt_configs->fmt_count) { - dev_dbg(dev, "%s: found %d format definitions\n", - __func__, fmt_configs->fmt_count); + dev_dbg(dev, "found %d format definitions\n", + fmt_configs->fmt_count); for (i = 0; i < fmt_configs->fmt_count; i++) { struct wav_fmt_ext *fmt_ext; @@ -66,9 +66,9 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (fmt_ext->fmt.channels > max_ch) max_ch = fmt_ext->fmt.channels; } - dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch); + dev_dbg(dev, "max channels found %d\n", max_ch); } else { - dev_dbg(dev, "%s: No format information found\n", __func__); + dev_dbg(dev, "No format information found\n"); } if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) { @@ -95,17 +95,16 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) } if (dmic_geo > 0) { - dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo); + dev_dbg(dev, "Array with %d dmics\n", dmic_geo); } if (max_ch > dmic_geo) { - dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n", - __func__, max_ch, dmic_geo); + dev_dbg(dev, "max channels %d exceed dmic number %d\n", + max_ch, dmic_geo); } } } - dev_dbg(dev, "%s: dmic number %d max_ch %d\n", - __func__, dmic_geo, max_ch); + dev_dbg(dev, "dmic number %d max_ch %d\n", dmic_geo, max_ch); return dmic_geo; } diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index bd60308769ff..8634004a606b 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -74,36 +74,36 @@ static int snd_card_cs46xx_probe(struct pci_dev *pci, err = snd_cs46xx_create(card, pci, external_amp[dev], thinkpad[dev]); if (err < 0) - return err; + goto error; card->private_data = chip; chip->accept_valid = mmap_valid[dev]; err = snd_cs46xx_pcm(chip, 0); if (err < 0) - return err; + goto error; #ifdef CONFIG_SND_CS46XX_NEW_DSP err = snd_cs46xx_pcm_rear(chip, 1); if (err < 0) - return err; + goto error; err = snd_cs46xx_pcm_iec958(chip, 2); if (err < 0) - return err; + goto error; #endif err = snd_cs46xx_mixer(chip, 2); if (err < 0) - return err; + goto error; #ifdef CONFIG_SND_CS46XX_NEW_DSP if (chip->nr_ac97_codecs ==2) { err = snd_cs46xx_pcm_center_lfe(chip, 3); if (err < 0) - return err; + goto error; } #endif err = snd_cs46xx_midi(chip, 0); if (err < 0) - return err; + goto error; err = snd_cs46xx_start_dsp(chip); if (err < 0) - return err; + goto error; snd_cs46xx_gameport(chip); @@ -117,11 +117,15 @@ static int snd_card_cs46xx_probe(struct pci_dev *pci, err = snd_card_register(card); if (err < 0) - return err; + goto error; pci_set_drvdata(pci, card); dev++; return 0; + + error: + snd_card_free(card); + return err; } static struct pci_driver cs46xx_driver = { diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index cd1db943b7e0..7c6b1fe8dfcc 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -819,7 +819,7 @@ static void set_pin_targets(struct hda_codec *codec, snd_hda_set_pin_ctl_cache(codec, cfg->nid, cfg->val); } -static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth) { const char *modelname = codec->fixup_name; @@ -829,7 +829,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) if (++depth > 10) break; if (fix->chained_before) - apply_fixup(codec, fix->chain_id, action, depth + 1); + __snd_hda_apply_fixup(codec, fix->chain_id, action, depth + 1); switch (fix->type) { case HDA_FIXUP_PINS: @@ -870,6 +870,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) id = fix->chain_id; } } +EXPORT_SYMBOL_GPL(__snd_hda_apply_fixup); /** * snd_hda_apply_fixup - Apply the fixup chain with the given action @@ -879,7 +880,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) void snd_hda_apply_fixup(struct hda_codec *codec, int action) { if (codec->fixup_list) - apply_fixup(codec, codec->fixup_id, action, 0); + __snd_hda_apply_fixup(codec, codec->fixup_id, action, 0); } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1..a77165bd92a9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2525,6 +2525,9 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cf), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Meteorlake-P */ + { PCI_DEVICE(0x8086, 0x7e28), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aca592651870..682dca2057db 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -348,6 +348,7 @@ void snd_hda_apply_verbs(struct hda_codec *codec); void snd_hda_apply_pincfgs(struct hda_codec *codec, const struct hda_pintbl *cfg); void snd_hda_apply_fixup(struct hda_codec *codec, int action); +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth); void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_model_fixup *models, const struct snd_pci_quirk *quirk, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index aa360a0af284..83ae21a01bbf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -944,6 +944,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x82b4, "HP ProDesk 600 G3", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), @@ -1052,6 +1053,13 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_pick_fixup(codec, cxt5051_fixup_models, cxt5051_fixups, cxt_fixups); break; + case 0x14f15098: + codec->pin_amp_workaround = 1; + spec->gen.mixer_nid = 0x22; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + snd_hda_pick_fixup(codec, cxt5066_fixup_models, + cxt5066_fixups, cxt_fixups); + break; case 0x14f150f2: codec->power_save_node = 1; fallthrough; @@ -1072,11 +1080,11 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; - err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); + err = cx_auto_parse_beep(codec); if (err < 0) goto error; - err = cx_auto_parse_beep(codec); + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) goto error; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 31fe41795571..6c209cd26c0c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4554,6 +4554,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptorlake-P HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281d, "Meteorlake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..2f55bc43bfa9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -2632,6 +2634,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67f1, "Clevo PC70H[PRS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67f5, "Clevo PD70PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170SM", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7715, "Clevo X170KM-G", ALC1220_FIXUP_CLEVO_PB51ED), @@ -3247,6 +3250,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3279,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4915,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5031,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5182,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5282,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5397,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5499,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5794,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6397,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -6888,6 +6901,7 @@ enum { ALC298_FIXUP_LENOVO_SPK_VOLUME, ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, ALC269_FIXUP_ATIV_BOOK_8, + ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE, ALC221_FIXUP_HP_MIC_NO_PRESENCE, ALC256_FIXUP_ASUS_HEADSET_MODE, ALC256_FIXUP_ASUS_MIC, @@ -6992,6 +7006,7 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC298_FIXUP_LENOVO_C940_DUET7, ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, @@ -7010,6 +7025,23 @@ enum { ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, }; +/* A special fixup for Lenovo C940 and Yoga Duet 7; + * both have the very same PCI SSID, and we need to apply different fixups + * depending on the codec ID + */ +static void alc298_fixup_lenovo_c940_duet7(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + int id; + + if (codec->core.vendor_id == 0x10ec0298) + id = ALC298_FIXUP_LENOVO_SPK_VOLUME; /* C940 */ + else + id = ALC287_FIXUP_YOGA7_14ITL_SPEAKERS; /* Duet 7 */ + __snd_hda_apply_fixup(codec, id, action, 0); +} + static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_GPIO2] = { .type = HDA_FIXUP_FUNC, @@ -7806,6 +7838,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_NO_SHUTUP }, + [ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01813030 }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC221_FIXUP_HP_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -8709,6 +8751,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, + [ALC298_FIXUP_LENOVO_C940_DUET7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_lenovo_c940_duet7, + }, [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8851,6 +8897,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x129d, "Acer SWIFT SF313-51", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), @@ -8860,6 +8907,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05be, "Dell Latitude E6540", ALC292_FIXUP_DELL_E7X), @@ -8975,6 +9023,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2b5e, "HP 288 Pro G2 MT", ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), @@ -9010,6 +9059,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), @@ -9059,6 +9109,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -9174,6 +9229,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -9258,8 +9315,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), + SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), - SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), @@ -9315,6 +9373,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), @@ -10095,6 +10154,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -10722,6 +10782,7 @@ enum { ALC668_FIXUP_MIC_DET_COEF, ALC897_FIXUP_LENOVO_HEADSET_MIC, ALC897_FIXUP_HEADSET_MIC_PIN, + ALC897_FIXUP_HP_HSMIC_VERB, }; static const struct hda_fixup alc662_fixups[] = { @@ -11141,6 +11202,13 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MIC }, + [ALC897_FIXUP_HP_HSMIC_VERB] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -11166,7 +11234,9 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x103c, 0x8719, "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), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), @@ -11545,6 +11615,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a05304f340df..aea7fae2ca4b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -518,11 +518,11 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); + err = auto_parse_beep(codec); if (err < 0) return err; - err = auto_parse_beep(codec); + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) return err; diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 55e773f92122..93606e5afd8f 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -868,10 +868,12 @@ static void ak4613_parse_of(struct ak4613_priv *priv, /* * connected STDI + * TDM support is assuming it is probed via Audio-Graph-Card style here. + * Default is SDTIx1 if it was probed via Simple-Audio-Card for now. */ sdti_num = of_graph_get_endpoint_count(np); - if (WARN_ON((sdti_num > 3) || (sdti_num < 1))) - return; + if ((sdti_num >= SDTx_MAX) || (sdti_num < 1)) + sdti_num = 1; AK4613_CONFIG_SDTI_set(priv, sdti_num); } diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e32871b3f68a..7434aeeda292 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component, if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) return true; - val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE); - if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK)) return true; val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1); diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 920190daa4d1..dfe85dc2cd20 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index 6d3070ea9e06..198cfe54a46f 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -37,8 +37,8 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_DAC_PCM1_SRC, 0x00000008 }, { CS35L41_ASP_TX1_SRC, 0x00000018 }, { CS35L41_ASP_TX2_SRC, 0x00000019 }, - { CS35L41_ASP_TX3_SRC, 0x00000020 }, - { CS35L41_ASP_TX4_SRC, 0x00000021 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, { CS35L41_DSP1_RX1_SRC, 0x00000008 }, { CS35L41_DSP1_RX2_SRC, 0x00000009 }, { CS35L41_DSP1_RX3_SRC, 0x00000018 }, @@ -644,6 +644,8 @@ static const struct reg_sequence cs35l41_reva0_errata_patch[] = { { CS35L41_DSP1_XM_ACCEL_PL0_PRI, 0x00000000 }, { CS35L41_PWR_CTRL2, 0x00000000 }, { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, }; static const struct reg_sequence cs35l41_revb0_errata_patch[] = { @@ -655,6 +657,8 @@ static const struct reg_sequence cs35l41_revb0_errata_patch[] = { { CS35L41_DSP1_XM_ACCEL_PL0_PRI, 0x00000000 }, { CS35L41_PWR_CTRL2, 0x00000000 }, { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, }; static const struct reg_sequence cs35l41_revb2_errata_patch[] = { @@ -666,6 +670,8 @@ static const struct reg_sequence cs35l41_revb2_errata_patch[] = { { CS35L41_DSP1_XM_ACCEL_PL0_PRI, 0x00000000 }, { CS35L41_PWR_CTRL2, 0x00000000 }, { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, }; static const struct reg_sequence cs35l41_fs_errata_patch[] = { diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 3e68a07a3c8e..71ab2a5d1c55 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -333,7 +333,7 @@ static const struct snd_kcontrol_new cs35l41_aud_controls[] = { SOC_SINGLE("HW Noise Gate Enable", CS35L41_NG_CFG, 8, 63, 0), SOC_SINGLE("HW Noise Gate Delay", CS35L41_NG_CFG, 4, 7, 0), SOC_SINGLE("HW Noise Gate Threshold", CS35L41_NG_CFG, 0, 7, 0), - SOC_SINGLE("Aux Noise Gate CH1 Enable", + SOC_SINGLE("Aux Noise Gate CH1 Switch", CS35L41_MIXER_NGATE_CH1_CFG, 16, 1, 0), SOC_SINGLE("Aux Noise Gate CH1 Entry Delay", CS35L41_MIXER_NGATE_CH1_CFG, 8, 15, 0), @@ -341,15 +341,15 @@ static const struct snd_kcontrol_new cs35l41_aud_controls[] = { CS35L41_MIXER_NGATE_CH1_CFG, 0, 7, 0), SOC_SINGLE("Aux Noise Gate CH2 Entry Delay", CS35L41_MIXER_NGATE_CH2_CFG, 8, 15, 0), - SOC_SINGLE("Aux Noise Gate CH2 Enable", + SOC_SINGLE("Aux Noise Gate CH2 Switch", CS35L41_MIXER_NGATE_CH2_CFG, 16, 1, 0), SOC_SINGLE("Aux Noise Gate CH2 Threshold", CS35L41_MIXER_NGATE_CH2_CFG, 0, 7, 0), - SOC_SINGLE("SCLK Force", CS35L41_SP_FORMAT, CS35L41_SCLK_FRC_SHIFT, 1, 0), - SOC_SINGLE("LRCLK Force", CS35L41_SP_FORMAT, CS35L41_LRCLK_FRC_SHIFT, 1, 0), - SOC_SINGLE("Invert Class D", CS35L41_AMP_DIG_VOL_CTRL, + SOC_SINGLE("SCLK Force Switch", CS35L41_SP_FORMAT, CS35L41_SCLK_FRC_SHIFT, 1, 0), + SOC_SINGLE("LRCLK Force Switch", CS35L41_SP_FORMAT, CS35L41_LRCLK_FRC_SHIFT, 1, 0), + SOC_SINGLE("Invert Class D Switch", CS35L41_AMP_DIG_VOL_CTRL, CS35L41_AMP_INV_PCM_SHIFT, 1, 0), - SOC_SINGLE("Amp Gain ZC", CS35L41_AMP_GAIN_CTRL, + SOC_SINGLE("Amp Gain ZC Switch", CS35L41_AMP_GAIN_CTRL, CS35L41_AMP_GAIN_ZC_SHIFT, 1, 0), WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), WM_ADSP_FW_CONTROL("DSP1", 0), diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index aff618513c75..0e933181b5db 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -143,7 +143,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 9b182b585be4..10e696406a71 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,9 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), @@ -364,7 +366,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index dc23007336c5..510c94265b1f 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index 391fd7da331f..1c7d52bef893 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -122,6 +122,9 @@ static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + if (!!ucontrol->value.integer.value[0] == cs47l15->in1_lp_mode) + return 0; + switch (ucontrol->value.integer.value[0]) { case 0: /* Set IN1 to normal mode */ @@ -150,7 +153,7 @@ static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, break; } - return 0; + return 1; } static const struct snd_kcontrol_new cs47l15_snd_controls[] = { diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index a1b8dcdb9f7b..444026b7d54b 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum, diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 703545273900..360ca2ffd506 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f00ead97006..dd53dfd87b04 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 272041c6236a..b9f19fbd2911 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -618,7 +618,13 @@ int madera_out1_demux_put(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } EXPORT_SYMBOL_GPL(madera_out1_demux_put); @@ -893,7 +899,7 @@ static int madera_adsp_rate_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; const int adsp_num = e->shift_l; const unsigned int item = ucontrol->value.enumerated.item[0]; - int ret; + int ret = 0; if (item >= e->items) return -EINVAL; @@ -910,10 +916,10 @@ static int madera_adsp_rate_put(struct snd_kcontrol *kcontrol, "Cannot change '%s' while in use by active audio paths\n", kcontrol->id.name); ret = -EBUSY; - } else { + } else if (priv->adsp_rate_cache[adsp_num] != e->values[item]) { /* Volatile register so defer until the codec is powered up */ priv->adsp_rate_cache[adsp_num] = e->values[item]; - ret = 0; + ret = 1; } mutex_unlock(&priv->rate_lock); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index f47e956d4f55..97b64477dde6 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -862,6 +862,16 @@ static int max98373_sdw_probe(struct sdw_slave *slave, return max98373_init(slave, regmap); } +static int max98373_sdw_remove(struct sdw_slave *slave) +{ + struct max98373_priv *max98373 = dev_get_drvdata(&slave->dev); + + if (max98373->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + #if defined(CONFIG_OF) static const struct of_device_id max98373_of_match[] = { { .compatible = "maxim,max98373", }, @@ -893,7 +903,7 @@ static struct sdw_driver max98373_sdw_driver = { .pm = &max98373_pm, }, .probe = max98373_sdw_probe, - .remove = NULL, + .remove = max98373_sdw_remove, .ops = &max98373_slave_ops, .id_table = max98373_id, }; diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index 56eb62bb041f..34db38812807 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -342,12 +342,15 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_component *component = codec_dai->component; struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component); - unsigned int format = 0; + unsigned int format_mask, format = 0; unsigned int bclk_pol = 0; int ret, status; int reg; bool update = false; + format_mask = MAX98396_PCM_MODE_CFG_FORMAT_MASK | + MAX98396_PCM_MODE_CFG_LRCLKEDGE; + dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt); switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -395,7 +398,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) ret = regmap_read(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, ®); if (ret < 0) return -EINVAL; - if (format != (reg & MAX98396_PCM_BCLKEDGE_BSEL_MASK)) { + if (format != (reg & format_mask)) { update = true; } else { ret = regmap_read(max98396->regmap, @@ -412,8 +415,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, - MAX98396_PCM_BCLKEDGE_BSEL_MASK, - format); + format_mask, format); regmap_update_bits(max98396->regmap, MAX98396_R2042_PCM_CLK_SETUP, diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 66bbd8f4f1ad..08f6c56dc387 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -741,6 +741,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler, pll_param->pre_factor); snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); + snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | pll_param->pll_int); @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 1c11b42dd76e..72f673f278ee 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -691,6 +691,16 @@ static int rt1308_sdw_probe(struct sdw_slave *slave, return 0; } +static int rt1308_sdw_remove(struct sdw_slave *slave) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); + + if (rt1308->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt1308_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x1308, 0x2, 0, 0), {}, @@ -750,6 +760,7 @@ static struct sdw_driver rt1308_sdw_driver = { .pm = &rt1308_pm, }, .probe = rt1308_sdw_probe, + .remove = rt1308_sdw_remove, .ops = &rt1308_slave_ops, .id_table = rt1308_id, }; diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 60baa9ff1907..2d6b5f9d4d77 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -676,6 +676,16 @@ static int rt1316_sdw_probe(struct sdw_slave *slave, return rt1316_sdw_init(&slave->dev, regmap, slave); } +static int rt1316_sdw_remove(struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(&slave->dev); + + if (rt1316->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt1316_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x1316, 0x3, 0x1, 0), {}, @@ -735,6 +745,7 @@ static struct sdw_driver rt1316_sdw_driver = { .pm = &rt1316_pm, }, .probe = rt1316_sdw_probe, + .remove = rt1316_sdw_remove, .ops = &rt1316_slave_ops, .id_table = rt1316_id, }; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 69c80d80ed9d..18b3da9211e3 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000); snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000); snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); - snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) + snd_soc_component_write(component, RT5640_PWR_ANLG1, + 0x0018); + else + snd_soc_component_write(component, RT5640_PWR_ANLG1, + 0x0000); snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000); break; @@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work) static irqreturn_t rt5640_irq(int irq, void *data) { struct rt5640_priv *rt5640 = data; + int delay = 0; + + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { + cancel_delayed_work_sync(&rt5640->jack_work); + delay = 100; + } if (rt5640->jack) - queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); + queue_delayed_work(system_long_wq, &rt5640->jack_work, delay); return IRQ_HANDLED; } @@ -2580,6 +2591,12 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); + snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, + RT5640_PWR_VREF2, RT5640_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, + RT5640_PWR_FV2, RT5640_PWR_FV2); + rt5640->jack = jack; ret = request_irq(rt5640->irq, rt5640_irq, @@ -2696,16 +2713,13 @@ static int rt5640_probe(struct snd_soc_component *component) if (device_property_read_u32(component->dev, "realtek,jack-detect-source", &val) == 0) { - if (val <= RT5640_JD_SRC_GPIO4) { + if (val <= RT5640_JD_SRC_GPIO4) rt5640->jd_src = val << RT5640_JD_SFT; - } else if (val == RT5640_JD_SRC_HDA_HEADER) { + else if (val == RT5640_JD_SRC_HDA_HEADER) rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER; - snd_soc_component_update_bits(component, RT5640_DUMMY1, - 0x0300, 0x0); - } else { + else dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n", val); - } } if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted")) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 248257a2e4e0..f04e18c32489 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -719,9 +719,12 @@ static int rt5682_sdw_remove(struct sdw_slave *slave) { struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); - if (rt5682 && rt5682->hw_init) + if (rt5682->hw_init) cancel_delayed_work_sync(&rt5682->jack_detect_work); + if (rt5682->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index bda594899664..f7439e40ca8b 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -13,6 +13,7 @@ #include <linux/soundwire/sdw_type.h> #include <linux/soundwire/sdw_registers.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <linux/regmap.h> #include <sound/soc.h> #include "rt700.h" @@ -463,11 +464,14 @@ static int rt700_sdw_remove(struct sdw_slave *slave) { struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev); - if (rt700 && rt700->hw_init) { + if (rt700->hw_init) { cancel_delayed_work_sync(&rt700->jack_detect_work); cancel_delayed_work_sync(&rt700->jack_btn_check_work); } + if (rt700->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index af32295fa9b9..9bceeeb830b1 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -162,7 +162,7 @@ static void rt700_jack_detect_handler(struct work_struct *work) if (!rt700->hs_jack) return; - if (!rt700->component->card->instantiated) + if (!rt700->component->card || !rt700->component->card->instantiated) return; reg = RT700_VERB_GET_PIN_SENSE | RT700_HP_OUT; @@ -315,17 +315,27 @@ static int rt700_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + int ret; rt700->hs_jack = hs_jack; - if (!rt700->hw_init) { - dev_dbg(&rt700->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt700_jack_init(rt700); + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } @@ -1115,6 +1125,11 @@ int rt700_init(struct device *dev, struct regmap *sdw_regmap, mutex_init(&rt700->disable_irq_lock); + INIT_DELAYED_WORK(&rt700->jack_detect_work, + rt700_jack_detect_handler); + INIT_DELAYED_WORK(&rt700->jack_btn_check_work, + rt700_btn_check_handler); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1209,13 +1224,6 @@ int rt700_io_init(struct device *dev, struct sdw_slave *slave) /* Finish Initial Settings, set power to D3 */ regmap_write(rt700->regmap, RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D3); - if (!rt700->first_hw_init) { - INIT_DELAYED_WORK(&rt700->jack_detect_work, - rt700_jack_detect_handler); - INIT_DELAYED_WORK(&rt700->jack_btn_check_work, - rt700_btn_check_handler); - } - /* * if set_jack callback occurred early than io_init, * we set up the jack detection function now diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index aaf5af153d3f..a085b2f530aa 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -11,6 +11,7 @@ #include <linux/mod_devicetable.h> #include <linux/soundwire/sdw_registers.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include "rt711-sdca.h" #include "rt711-sdca-sdw.h" @@ -364,11 +365,17 @@ static int rt711_sdca_sdw_remove(struct sdw_slave *slave) { struct rt711_sdca_priv *rt711 = dev_get_drvdata(&slave->dev); - if (rt711 && rt711->hw_init) { + if (rt711->hw_init) { cancel_delayed_work_sync(&rt711->jack_detect_work); cancel_delayed_work_sync(&rt711->jack_btn_check_work); } + if (rt711->first_hw_init) + pm_runtime_disable(&slave->dev); + + mutex_destroy(&rt711->calibrate_mutex); + mutex_destroy(&rt711->disable_irq_lock); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 57629c18db38..5ad53bbc8528 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -34,7 +34,7 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to set private value: %06x <= %04x ret=%d\n", addr, value, ret); @@ -50,7 +50,7 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to get private value: %06x => %04x ret=%d\n", addr, *value, ret); @@ -294,7 +294,7 @@ static void rt711_sdca_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return; - if (!rt711->component->card->instantiated) + if (!rt711->component->card || !rt711->component->card->instantiated) return; /* SDW_SCP_SDCA_INT_SDCA_0 is used for jack detection */ @@ -487,16 +487,27 @@ static int rt711_sdca_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711->hs_jack = hs_jack; - if (!rt711->hw_init) { - dev_dbg(&rt711->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt711_sdca_jack_init(rt711); + + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } @@ -1190,14 +1201,6 @@ static int rt711_sdca_probe(struct snd_soc_component *component) return 0; } -static void rt711_sdca_remove(struct snd_soc_component *component) -{ - struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - - regcache_cache_only(rt711->regmap, true); - regcache_cache_only(rt711->mbq_regmap, true); -} - static const struct snd_soc_component_driver soc_sdca_dev_rt711 = { .probe = rt711_sdca_probe, .controls = rt711_sdca_snd_controls, @@ -1207,7 +1210,6 @@ static const struct snd_soc_component_driver soc_sdca_dev_rt711 = { .dapm_routes = rt711_sdca_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_sdca_audio_map), .set_jack = rt711_sdca_set_jack_detect, - .remove = rt711_sdca_remove, .endianness = 1, }; @@ -1412,8 +1414,12 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, rt711->regmap = regmap; rt711->mbq_regmap = mbq_regmap; + mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); + INIT_DELAYED_WORK(&rt711->jack_detect_work, rt711_sdca_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_sdca_btn_check_handler); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1545,14 +1551,6 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, RT711_PUSH_BTN_INT_CTL0, 0x20, 0x00); - if (!rt711->first_hw_init) { - INIT_DELAYED_WORK(&rt711->jack_detect_work, - rt711_sdca_jack_detect_handler); - INIT_DELAYED_WORK(&rt711->jack_btn_check_work, - rt711_sdca_btn_check_handler); - mutex_init(&rt711->calibrate_mutex); - } - /* calibration */ ret = rt711_sdca_calibration(rt711); if (ret < 0) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index bda2cc9439c9..4fe68bcf2a7c 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -13,6 +13,7 @@ #include <linux/soundwire/sdw_type.h> #include <linux/soundwire/sdw_registers.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <linux/regmap.h> #include <sound/soc.h> #include "rt711.h" @@ -464,12 +465,18 @@ static int rt711_sdw_remove(struct sdw_slave *slave) { struct rt711_priv *rt711 = dev_get_drvdata(&slave->dev); - if (rt711 && rt711->hw_init) { + if (rt711->hw_init) { cancel_delayed_work_sync(&rt711->jack_detect_work); cancel_delayed_work_sync(&rt711->jack_btn_check_work); cancel_work_sync(&rt711->calibration_work); } + if (rt711->first_hw_init) + pm_runtime_disable(&slave->dev); + + mutex_destroy(&rt711->calibrate_mutex); + mutex_destroy(&rt711->disable_irq_lock); + return 0; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 9838fb4d5b9c..9df800abfc2d 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -242,7 +242,7 @@ static void rt711_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return; - if (!rt711->component->card->instantiated) + if (!rt711->component->card || !rt711->component->card->instantiated) return; if (pm_runtime_status_suspended(rt711->slave->dev.parent)) { @@ -457,17 +457,27 @@ static int rt711_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711->hs_jack = hs_jack; - if (!rt711->hw_init) { - dev_dbg(&rt711->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt711_jack_init(rt711); + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } @@ -932,13 +942,6 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } -static void rt711_remove(struct snd_soc_component *component) -{ - struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); - - regcache_cache_only(rt711->regmap, true); -} - static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -949,7 +952,6 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, - .remove = rt711_remove, .endianness = 1, }; @@ -1204,8 +1206,13 @@ int rt711_init(struct device *dev, struct regmap *sdw_regmap, rt711->sdw_regmap = sdw_regmap; rt711->regmap = regmap; + mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); + INIT_DELAYED_WORK(&rt711->jack_detect_work, rt711_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_btn_check_handler); + INIT_WORK(&rt711->calibration_work, rt711_calibration_work); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1313,15 +1320,8 @@ int rt711_io_init(struct device *dev, struct sdw_slave *slave) if (rt711->first_hw_init) rt711_calibration(rt711); - else { - INIT_DELAYED_WORK(&rt711->jack_detect_work, - rt711_jack_detect_handler); - INIT_DELAYED_WORK(&rt711->jack_btn_check_work, - rt711_btn_check_handler); - mutex_init(&rt711->calibrate_mutex); - INIT_WORK(&rt711->calibration_work, rt711_calibration_work); + else schedule_work(&rt711->calibration_work); - } /* * if set_jack callback occurred early than io_init, diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 0ecd2948f7aa..13e731d16675 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -13,6 +13,7 @@ #include <linux/soundwire/sdw_type.h> #include <linux/soundwire/sdw_registers.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <linux/regmap.h> #include <sound/soc.h> #include "rt715-sdca.h" @@ -193,6 +194,16 @@ static int rt715_sdca_sdw_probe(struct sdw_slave *slave, return rt715_sdca_init(&slave->dev, mbq_regmap, regmap, slave); } +static int rt715_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt715_sdca_priv *rt715 = dev_get_drvdata(&slave->dev); + + if (rt715->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt715_sdca_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x3, 0x1, 0), SDW_SLAVE_ENTRY_EXT(0x025d, 0x714, 0x3, 0x1, 0), @@ -267,6 +278,7 @@ static struct sdw_driver rt715_sdw_driver = { .pm = &rt715_pm, }, .probe = rt715_sdca_sdw_probe, + .remove = rt715_sdca_sdw_remove, .ops = &rt715_sdca_slave_ops, .id_table = rt715_sdca_id, }; diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index a7b21b03c08b..b047bf87a100 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -14,6 +14,7 @@ #include <linux/soundwire/sdw_type.h> #include <linux/soundwire/sdw_registers.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <linux/of.h> #include <linux/regmap.h> #include <sound/soc.h> @@ -514,6 +515,16 @@ static int rt715_sdw_probe(struct sdw_slave *slave, return 0; } +static int rt715_sdw_remove(struct sdw_slave *slave) +{ + struct rt715_priv *rt715 = dev_get_drvdata(&slave->dev); + + if (rt715->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt715_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x714, 0x2, 0, 0), SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x2, 0, 0), @@ -575,6 +586,7 @@ static struct sdw_driver rt715_sdw_driver = { .pm = &rt715_pm, }, .probe = rt715_sdw_probe, + .remove = rt715_sdw_remove, .ops = &rt715_slave_ops, .id_table = rt715_id, }; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2aa48aef6a97..3363d1696ad7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1795,6 +1795,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT); + clk_disable_unprepare(sgtl5000->mclk); regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies); regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies); @@ -1802,6 +1805,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) return 0; } +static void sgtl5000_i2c_shutdown(struct i2c_client *client) +{ + sgtl5000_i2c_remove(client); +} + static const struct i2c_device_id sgtl5000_id[] = { {"sgtl5000", 0}, {}, @@ -1822,6 +1830,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { }, .probe_new = sgtl5000_i2c_probe, .remove = sgtl5000_i2c_remove, + .shutdown = sgtl5000_i2c_shutdown, .id_table = sgtl5000_id, }; diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 56ec5863f250..3a808c762299 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -80,6 +80,7 @@ /* * SGTL5000_CHIP_DIG_POWER */ +#define SGTL5000_DIG_POWER_DEFAULT 0x0000 #define SGTL5000_ADC_EN 0x0040 #define SGTL5000_DAC_EN 0x0020 #define SGTL5000_DAP_POWERUP 0x0010 diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index d395feffb30b..4cb788f3e5f7 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764) gpiod_set_value_cansleep(tas2764->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2764->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2764->component, TAS2764_SW_RST, TAS2764_RST); + usleep_range(1000, 2000); } static int tas2764_set_bias_level(struct snd_soc_component *component, @@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component) struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); int ret; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, TAS2764_PWR_CTRL_MASK, @@ -131,7 +135,8 @@ static const char * const tas2764_ASI1_src[] = { }; static SOC_ENUM_SINGLE_DECL( - tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, 4, tas2764_ASI1_src); + tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, TAS2764_TDM_CFG2_SCFG_SHIFT, + tas2764_ASI1_src); static const struct snd_kcontrol_new tas2764_asi1_mux = SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum); @@ -329,20 +334,22 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); - u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; - int iface; + u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0; int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; + fallthrough; case SND_SOC_DAIFMT_NB_NF: asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING; break; + case SND_SOC_DAIFMT_IB_IF: + asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; + fallthrough; case SND_SOC_DAIFMT_IB_NF: asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING; break; - default: - dev_err(tas2764->dev, "ASI format Inverse is not found\n"); - return -EINVAL; } ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1, @@ -353,13 +360,13 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; + fallthrough; case SND_SOC_DAIFMT_DSP_A: - iface = TAS2764_TDM_CFG2_SCFG_I2S; tdm_rx_start_slot = 1; break; case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_LEFT_J: - iface = TAS2764_TDM_CFG2_SCFG_LEFT_J; tdm_rx_start_slot = 0; break; default: @@ -368,14 +375,15 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1, - TAS2764_TDM_CFG1_MASK, - (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT)); + ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG0, + TAS2764_TDM_CFG0_FRAME_START, + asi_cfg_0); if (ret < 0) return ret; - ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG2, - TAS2764_TDM_CFG2_SCFG_MASK, iface); + ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1, + TAS2764_TDM_CFG1_MASK, + (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT)); if (ret < 0) return ret; @@ -501,8 +509,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component) tas2764->component = component; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2764_reset(tas2764); @@ -526,12 +536,12 @@ static int tas2764_codec_probe(struct snd_soc_component *component) } static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0); -static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10000, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1); static const struct snd_kcontrol_new tas2764_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0, TAS2764_DVC_MAX, 1, tas2764_playback_volume), - SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 0, 0x14, 0, + SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0, tas2764_digital_tlv), }; @@ -556,7 +566,7 @@ static const struct reg_default tas2764_reg_defaults[] = { { TAS2764_SW_RST, 0x00 }, { TAS2764_PWR_CTRL, 0x1a }, { TAS2764_DVC, 0x00 }, - { TAS2764_CHNL_0, 0x00 }, + { TAS2764_CHNL_0, 0x28 }, { TAS2764_TDM_CFG0, 0x09 }, { TAS2764_TDM_CFG1, 0x02 }, { TAS2764_TDM_CFG2, 0x0a }, diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h index 67d6fd903c42..f015f22a083b 100644 --- a/sound/soc/codecs/tas2764.h +++ b/sound/soc/codecs/tas2764.h @@ -47,6 +47,7 @@ #define TAS2764_TDM_CFG0_MASK GENMASK(3, 1) #define TAS2764_TDM_CFG0_44_1_48KHZ BIT(3) #define TAS2764_TDM_CFG0_88_2_96KHZ (BIT(3) | BIT(1)) +#define TAS2764_TDM_CFG0_FRAME_START BIT(0) /* TDM Configuration Reg1 */ #define TAS2764_TDM_CFG1 TAS2764_REG(0X0, 0x09) @@ -66,10 +67,7 @@ #define TAS2764_TDM_CFG2_RXS_16BITS 0x0 #define TAS2764_TDM_CFG2_RXS_24BITS BIT(0) #define TAS2764_TDM_CFG2_RXS_32BITS BIT(1) -#define TAS2764_TDM_CFG2_SCFG_MASK GENMASK(5, 4) -#define TAS2764_TDM_CFG2_SCFG_I2S 0x0 -#define TAS2764_TDM_CFG2_SCFG_LEFT_J BIT(4) -#define TAS2764_TDM_CFG2_SCFG_RIGHT_J BIT(5) +#define TAS2764_TDM_CFG2_SCFG_SHIFT 4 /* TDM Configuration Reg3 */ #define TAS2764_TDM_CFG3 TAS2764_REG(0X0, 0x0c) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index b55f0b836932..0b729658fde8 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -33,7 +33,6 @@ struct adcx140_priv { bool micbias_vg; unsigned int dai_fmt; - unsigned int tdm_delay; unsigned int slot_width; }; @@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - unsigned int lsb; - /* TDM based on DSP mode requires slots to be adjacent */ - lsb = __ffs(tx_mask); - if ((lsb + 1) != __fls(tx_mask)) { - dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + /* + * The chip itself supports arbitrary masks, but the driver currently + * only supports adjacent slots beginning at the first slot. + */ + if (tx_mask != GENMASK(__fls(tx_mask), 0)) { + dev_err(component->dev, "Only lower adjacent slots are supported\n"); return -EINVAL; } @@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - adcx140->tdm_delay = lsb; adcx140->slot_width = slot_width; return 0; diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..3cb7a3eab8c7 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -342,7 +342,7 @@ struct wcd9335_codec { struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY]; unsigned int rx_port_value[WCD9335_RX_MAX]; - unsigned int tx_port_value; + unsigned int tx_port_value[WCD9335_TX_MAX]; int hph_l_gain; int hph_r_gain; u32 rx_bias_count; @@ -1287,11 +1287,17 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, struct snd_soc_dapm_update *update = NULL; u32 port_id = w->shift; + if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0]) + return 0; + wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; + /* Remove channel from any list it's in before adding it to a new one */ + list_del_init(&wcd->rx_chs[port_id].list); + switch (wcd->rx_port_value[port_id]) { case 0: - list_del_init(&wcd->rx_chs[port_id].list); + /* Channel already removed from lists. Nothing to do here */ break; case 1: list_add_tail(&wcd->rx_chs[port_id].list, @@ -1328,8 +1334,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev); + struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc); + struct soc_mixer_control *mixer = + (struct soc_mixer_control *)kc->private_value; + int dai_id = widget->shift; + int port_id = mixer->shift; - ucontrol->value.integer.value[0] = wcd->tx_port_value; + ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id; return 0; } @@ -1352,12 +1363,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc, case AIF2_CAP: case AIF3_CAP: /* only add to the list if value not set */ - if (enable && !(wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value |= BIT(port_id); + if (enable && wcd->tx_port_value[port_id] != dai_id) { + wcd->tx_port_value[port_id] = dai_id; list_add_tail(&wcd->tx_chs[port_id].list, &wcd->dai[dai_id].slim_ch_list); - } else if (!enable && (wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value &= ~BIT(port_id); + } else if (!enable && wcd->tx_port_value[port_id] == dai_id) { + wcd->tx_port_value[port_id] = -1; list_del_init(&wcd->tx_chs[port_id].list); } break; diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index c1b61b997f69..781ae569be29 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -2519,6 +2519,9 @@ static int wcd938x_tx_mode_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int path = e->shift_l; + if (wcd938x->tx_mode[path] == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->tx_mode[path] = ucontrol->value.enumerated.item[0]; return 1; @@ -2541,6 +2544,9 @@ static int wcd938x_rx_hph_mode_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->hph_mode == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->hph_mode = ucontrol->value.enumerated.item[0]; return 1; @@ -2632,6 +2638,9 @@ static int wcd938x_ldoh_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->ldoh == ucontrol->value.integer.value[0]) + return 0; + wcd938x->ldoh = ucontrol->value.integer.value[0]; return 1; @@ -2654,6 +2663,9 @@ static int wcd938x_bcs_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->bcs_dis == ucontrol->value.integer.value[0]) + return 0; + wcd938x->bcs_dis = ucontrol->value.integer.value[0]; return 1; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index da2f8998df87..b034df47a5ef 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + int ret = 0; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + if (arizona->dac_comp_coeff != dac_comp_coeff) { + arizona->dac_comp_coeff = dac_comp_coeff; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, @@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret = 0; + + if (ucontrol->value.integer.value[0] > mc->max) + return -EINVAL; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) { + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static const char * const wm5102_osr_text[] = { diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4973ba1ed779..4ab7a672f8de 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -413,6 +413,7 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, unsigned int rnew = (!!ucontrol->value.integer.value[1]) << mc->rshift; unsigned int lold, rold; unsigned int lena, rena; + bool change = false; int ret; snd_soc_dapm_mutex_lock(dapm); @@ -440,8 +441,8 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, goto err; } - ret = regmap_update_bits(arizona->regmap, ARIZONA_DRE_ENABLE, - mask, lnew | rnew); + ret = regmap_update_bits_check(arizona->regmap, ARIZONA_DRE_ENABLE, + mask, lnew | rnew, &change); if (ret) { dev_err(arizona->dev, "Failed to set DRE: %d\n", ret); goto err; @@ -454,6 +455,9 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, if (!rnew && rold) wm5110_clear_pga_volume(arizona, mc->rshift); + if (change) + ret = 1; + err: snd_soc_dapm_mutex_unlock(dapm); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 34cd5a2a997c..5cca89364280 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3868,6 +3868,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 00b59fc9b1fe..ab5481187c71 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int mode_reg, mode_index; unsigned int mux, inmode, src_val, mode_val; + int change, ret; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) @@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + change = snd_soc_component_update_bits(component, e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); + if (ret < 0) { + dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static const char * const wm8998_inmux_texts[] = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7973a75cac05..a7784ac15dde 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -333,7 +333,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; @@ -997,7 +997,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); } - return 0; + return 1; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..e765da9a19e7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1293,6 +1293,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx8mm-sai", .data = &fsl_sai_imx8mm_data }, { .compatible = "fsl,imx8mp-sai", .data = &fsl_sai_imx8mp_data }, { .compatible = "fsl,imx8ulp-sai", .data = &fsl_sai_imx8ulp_data }, + { .compatible = "fsl,imx8mn-sai", .data = &fsl_sai_imx8mp_data }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 77ac4051b827..d34b29a49268 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -90,12 +90,12 @@ links indicates connection part of CPU side (= A). ports@0 { (X) (A) mcpu: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; (y) port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; -(y) port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; +(y) port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; }; ports@1 { (X) port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; -(y) port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; -(y) port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; +(y) port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; +(y) port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; }; }; }; diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 0d11cc8aab0b..6a06fe387d13 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -128,10 +128,10 @@ struct avs_tplg_token_parser { static int avs_parse_uuid_token(struct snd_soc_component *comp, void *elem, void *object, u32 offset) { - struct snd_soc_tplg_vendor_value_elem *tuple = elem; + struct snd_soc_tplg_vendor_uuid_elem *tuple = elem; guid_t *val = (guid_t *)((u8 *)object + offset); - guid_copy((guid_t *)val, (const guid_t *)&tuple->value); + guid_copy((guid_t *)val, (const guid_t *)&tuple->uuid); return 0; } diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 00384c6fbcaa..330c0ace1638 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -421,8 +421,17 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) priv->spkvdd_en_gpio = gpiod_get(codec_dev, "wlf,spkvdd-ena", GPIOD_OUT_LOW); put_device(codec_dev); - if (IS_ERR(priv->spkvdd_en_gpio)) - return dev_err_probe(dev, PTR_ERR(priv->spkvdd_en_gpio), "getting spkvdd-GPIO\n"); + if (IS_ERR(priv->spkvdd_en_gpio)) { + ret = PTR_ERR(priv->spkvdd_en_gpio); + /* + * The spkvdd gpio-lookup is registered by: drivers/mfd/arizona-spi.c, + * so -ENOENT means that arizona-spi hasn't probed yet. + */ + if (ret == -ENOENT) + ret = -EPROBE_DEFER; + + return dev_err_probe(dev, ret, "getting spkvdd-GPIO\n"); + } /* override platform name, if required */ byt_wm5102_card.dev = dev; diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..f4192df962d6 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -54,22 +54,29 @@ static struct snd_soc_dai_link_component cs35l41_components[] = { }, }; +/* + * Mapping between ACPI instance id and speaker position. + * + * Four speakers: + * 0: Tweeter left, 1: Woofer left + * 2: Tweeter right, 3: Woofer right + */ static struct snd_soc_codec_conf cs35l41_codec_conf[] = { { .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "WL", + .name_prefix = "TL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WR", + .name_prefix = "WL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TL", + .name_prefix = "TR", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "TR", + .name_prefix = "WR", }, }; @@ -101,6 +108,21 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) return ret; } +/* + * Channel map: + * + * TL/WL: ASPRX1 on slot 0, ASPRX2 on slot 1 (default) + * TR/WR: ASPRX1 on slot 1, ASPRX2 on slot 0 + */ +static const struct { + unsigned int rx[2]; +} cs35l41_channel_map[] = { + {.rx = {0, 1}}, /* TL */ + {.rx = {0, 1}}, /* WL */ + {.rx = {1, 0}}, /* TR */ + {.rx = {1, 0}}, /* WR */ +}; + static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -134,6 +156,16 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* setup channel map */ + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(cs35l41_channel_map[i].rx), + (unsigned int *)cs35l41_channel_map[i].rx); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } } return 0; diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d67a2c87a1d..4a90a0a5d831 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -69,11 +69,10 @@ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | static int is_legacy_cpu; -static struct snd_soc_jack sof_hdmi[3]; - struct sof_hdmi_pcm { struct list_head head; struct snd_soc_dai *codec_dai; + struct snd_soc_jack hdmi_jack; int device; }; @@ -434,7 +433,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) char jack_name[NAME_SIZE]; struct sof_hdmi_pcm *pcm; int err; - int i = 0; /* HDMI is not supported by SOF on Baytrail/CherryTrail */ if (is_legacy_cpu || !ctx->idisp_codec) @@ -455,17 +453,15 @@ static int sof_card_late_probe(struct snd_soc_card *card) snprintf(jack_name, sizeof(jack_name), "HDMI/DP, pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &sof_hdmi[i]); + SND_JACK_AVOUT, &pcm->hdmi_jack); if (err) return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &sof_hdmi[i]); + &pcm->hdmi_jack); if (err < 0) return err; - - i++; } if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1f00679b4240..ad826ad82d51 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1398,6 +1398,33 @@ static struct snd_soc_card card_sof_sdw = { .late_probe = sof_sdw_card_late_probe, }; +static void mc_dailink_exit_loop(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int ret; + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (!codec_info_list[i].exit) + continue; + /* + * We don't need to call .exit function if there is no matched + * dai link found. + */ + for_each_card_prelinks(card, j, link) { + if (!strcmp(link->codecs[0].dai_name, + codec_info_list[i].dai_name)) { + ret = codec_info_list[i].exit(card, link); + if (ret) + dev_warn(card->dev, + "codec exit failed %d\n", + ret); + break; + } + } + } +} + static int mc_probe(struct platform_device *pdev) { struct snd_soc_card *card = &card_sof_sdw; @@ -1462,6 +1489,7 @@ static int mc_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(card->dev, "snd_soc_register_card failed %d\n", ret); + mc_dailink_exit_loop(card); return ret; } @@ -1473,29 +1501,8 @@ static int mc_probe(struct platform_device *pdev) static int mc_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct snd_soc_dai_link *link; - int ret; - int i, j; - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { - if (!codec_info_list[i].exit) - continue; - /* - * We don't need to call .exit function if there is no matched - * dai link found. - */ - for_each_card_prelinks(card, j, link) { - if (!strcmp(link->codecs[0].dai_name, - codec_info_list[i].dai_name)) { - ret = codec_info_list[i].exit(card, link); - if (ret) - dev_warn(&pdev->dev, - "codec exit failed %d\n", - ret); - break; - } - } - } + mc_dailink_exit_loop(card); return 0; } diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 2439a574ac2f..deb7b820325e 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -99,7 +99,6 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, struct nhlt_fmt_cfg *fmt_cfg; struct wav_fmt_ext *wav_fmt; unsigned long rate; - bool present = false; int rate_index = 0; u16 channels, bps; u8 clk_src; @@ -112,9 +111,12 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, if (fmt->fmt_count == 0) return; + fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; for (i = 0; i < fmt->fmt_count; i++) { - fmt_cfg = &fmt->fmt_config[i]; - wav_fmt = &fmt_cfg->fmt_ext; + struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg; + bool present = false; + + wav_fmt = &saved_fmt_cfg->fmt_ext; channels = wav_fmt->fmt.channels; bps = wav_fmt->fmt.bits_per_sample; @@ -132,12 +134,18 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, * derive the rate. */ for (j = i; j < fmt->fmt_count; j++) { - fmt_cfg = &fmt->fmt_config[j]; - wav_fmt = &fmt_cfg->fmt_ext; + struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg; + + wav_fmt = &tmp_fmt_cfg->fmt_ext; if ((fs == wav_fmt->fmt.samples_per_sec) && - (bps == wav_fmt->fmt.bits_per_sample)) + (bps == wav_fmt->fmt.bits_per_sample)) { channels = max_t(u16, channels, wav_fmt->fmt.channels); + saved_fmt_cfg = tmp_fmt_cfg; + } + /* Move to the next nhlt_fmt_cfg */ + tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps + + tmp_fmt_cfg->config.size); } rate = channels * bps * fs; @@ -153,8 +161,11 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, /* Fill rate and parent for sclk/sclkfs */ if (!present) { + struct nhlt_fmt_cfg *first_fmt_cfg; + + first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; i2s_config_ext = (struct skl_i2s_config_blob_ext *) - fmt->fmt_config[0].config.caps; + first_fmt_cfg->config.caps; /* MCLK Divider Source Select */ if (is_legacy_blob(i2s_config_ext->hdr.sig)) { @@ -168,6 +179,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, parent = skl_get_parent_clk(clk_src); + /* Move to the next nhlt_fmt_cfg */ + fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps + + fmt_cfg->config.size); /* * Do not copy the config data if there is no parent * clock available for this clock source select @@ -176,9 +190,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, continue; sclk[id].rate_cfg[rate_index].rate = rate; - sclk[id].rate_cfg[rate_index].config = fmt_cfg; + sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg; sclkfs[id].rate_cfg[rate_index].rate = rate; - sclkfs[id].rate_cfg[rate_index].config = fmt_cfg; + sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg; sclk[id].parent_name = parent->name; sclkfs[id].parent_name = parent->name; @@ -192,13 +206,13 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk, { struct skl_i2s_config_blob_ext *i2s_config_ext; struct skl_i2s_config_blob_legacy *i2s_config; - struct nhlt_specific_cfg *fmt_cfg; + struct nhlt_fmt_cfg *fmt_cfg; struct skl_clk_parent_src *parent; u32 clkdiv, div_ratio; u8 clk_src; - fmt_cfg = &fmt->fmt_config[0].config; - i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->caps; + fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; + i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps; /* MCLK Divider Source Select and divider */ if (is_legacy_blob(i2s_config_ext->hdr.sig)) { @@ -227,7 +241,7 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk, return; mclk[id].rate_cfg[0].rate = parent->rate/div_ratio; - mclk[id].rate_cfg[0].config = &fmt->fmt_config[0]; + mclk[id].rate_cfg[0].config = fmt_cfg; mclk[id].parent_name = parent->name; } diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f03a7ae49d50..b41ab7a321ae 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -898,7 +898,7 @@ static int lpass_platform_cdc_dma_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; unsigned long size, offset; - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); size = vma->vm_end - vma->vm_start; offset = vma->vm_pgoff << PAGE_SHIFT; return io_remap_pfn_range(vma, vma->vm_start, diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 19c4a90ec1ea..ee59ef36b85a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -147,6 +147,12 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, cfg.num_channels = runtime->channels; cfg.bit_width = prtd->bits_per_sample; + if (prtd->state) { + /* clear the previous setup if any */ + q6apm_graph_stop(prtd->graph); + q6apm_unmap_memory_regions(prtd->graph, substream->stream); + } + prtd->pcm_count = snd_pcm_lib_period_bytes(substream); prtd->pos = 0; /* rate and channels are sent to audio driver */ diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index f424d7aa389a..794019286c70 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL); if (id < 0) { dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id); + kfree(graph->graph); kfree(graph); mutex_unlock(&apm->lock); return ERR_PTR(id); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 869c76506b66..a8e842e02cdc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -62,6 +62,8 @@ struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); +static unsigned int soc_dapm_read(struct snd_soc_dapm_context *dapm, int reg); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 1, @@ -442,6 +444,9 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); + } else if (e->reg != SND_SOC_NOPM) { + data->value = soc_dapm_read(widget->dapm, e->reg) & + (e->mask << e->shift_l); } break; default: diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e693070f51fe..d867f449d82d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -526,7 +526,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -547,7 +547,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 000ea906670c..e24eea725acb 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -181,12 +181,20 @@ int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask) * Power Management. */ -static int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; unsigned int cpa; u32 adspcs; int ret; + /* restrict core_mask to host managed cores mask */ + core_mask &= chip->host_managed_cores_mask; + /* return if core_mask is not valid */ + if (!core_mask) + return 0; + /* update bits */ snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS, HDA_DSP_ADSPCS_SPA_MASK(core_mask), diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 64290125d7cd..145d483bd129 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -95,9 +95,9 @@ out_put: } /* - * first boot sequence has some extra steps. core 0 waits for power - * status on core 1, so power up core 1 also momentarily, keep it in - * reset/stall and then turn it off + * first boot sequence has some extra steps. + * power on all host managed cores and only unstall/run the boot core to boot the + * DSP then turn off all non boot cores (if any) is powered on. */ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) { @@ -110,7 +110,7 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) int ret; /* step 1: power up corex */ - ret = hda_dsp_enable_core(sdev, chip->host_managed_cores_mask); + ret = hda_dsp_core_power_up(sdev, chip->host_managed_cores_mask); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, "error: dsp core 0/1 power up failed\n"); @@ -127,7 +127,7 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) snd_sof_dsp_write(sdev, HDA_DSP_BAR, chip->ipc_req, ipc_hdr); /* step 3: unset core 0 reset state & unstall/run core 0 */ - ret = hda_dsp_core_run(sdev, BIT(0)); + ret = hda_dsp_core_run(sdev, chip->init_core_mask); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, @@ -389,7 +389,8 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) struct snd_dma_buffer dmab; int ret, ret1, i; - if (hda->imrboot_supported && !sdev->first_boot) { + if (sdev->system_suspend_target < SOF_SUSPEND_S4 && + hda->imrboot_supported && !sdev->first_boot) { dev_dbg(sdev->dev, "IMR restore supported, booting from IMR directly\n"); hda->boot_iteration = 0; ret = hda_dsp_boot_imr(sdev); diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index dc1f743730c0..6888e0a4665d 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -192,79 +192,7 @@ snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, goto found; } - switch (sof_hda_position_quirk) { - case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: - /* - * This legacy code, inherited from the Skylake driver, - * mixes DPIB registers and DPIB DDR updates and - * does not seem to follow any known hardware recommendations. - * It's not clear e.g. why there is a different flow - * for capture and playback, the only information that matters is - * what traffic class is used, and on all SOF-enabled platforms - * only VC0 is supported so the work-around was likely not necessary - * and quite possibly wrong. - */ - - /* DPIB/posbuf position mode: - * For Playback, Use DPIB register from HDA space which - * reflects the actual data transferred. - * For Capture, Use the position buffer for pointer, as DPIB - * is not accurate enough, its update may be completed - * earlier than the data written to DDR. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - } else { - /* - * For capture stream, we need more workaround to fix the - * position incorrect issue: - * - * 1. Wait at least 20us before reading position buffer after - * the interrupt generated(IOC), to make sure position update - * happens on frame boundary i.e. 20.833uSec for 48KHz. - * 2. Perform a dummy Read to DPIB register to flush DMA - * position value. - * 3. Read the DMA Position from posbuf. Now the readback - * value should be >= period boundary. - */ - usleep_range(20, 21); - snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - pos = snd_hdac_stream_get_pos_posbuf(hstream); - } - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: - /* - * In case VC1 traffic is disabled this is the recommended option - */ - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: - /* - * This is the recommended option when VC1 is enabled. - * While this isn't needed for SOF platforms it's added for - * consistency and debug. - */ - pos = snd_hdac_stream_get_pos_posbuf(hstream); - break; - default: - dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", - sof_hda_position_quirk); - pos = 0; - break; - } - - if (pos >= hstream->bufsize) - pos = 0; - + pos = hda_dsp_stream_get_position(hstream, substream->stream, true); found: pos = bytes_to_frames(substream->runtime, pos); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index daeb64c495e4..d95ae17e81cc 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -707,12 +707,13 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) } static void -hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) +hda_dsp_compr_bytes_transferred(struct hdac_stream *hstream, int direction) { + u64 buffer_size = hstream->bufsize; u64 prev_pos, pos, num_bytes; div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos); - pos = snd_hdac_stream_get_pos_posbuf(hstream); + pos = hda_dsp_stream_get_position(hstream, direction, false); if (pos < prev_pos) num_bytes = (buffer_size - prev_pos) + pos; @@ -748,8 +749,7 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); } else if (s->cstream) { - hda_dsp_set_bytes_transferred(s, - s->cstream->runtime->buffer_size); + hda_dsp_compr_bytes_transferred(s, s->cstream->direction); snd_compr_fragment_elapsed(s->cstream); } } @@ -1009,3 +1009,89 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) devm_kfree(sdev->dev, hda_stream); } } + +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep) +{ + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + struct sof_intel_hda_stream *hda_stream = hstream_to_sof_hda_stream(hext_stream); + struct snd_sof_dev *sdev = hda_stream->sdev; + snd_pcm_uframes_t pos; + + switch (sof_hda_position_quirk) { + case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: + /* + * This legacy code, inherited from the Skylake driver, + * mixes DPIB registers and DPIB DDR updates and + * does not seem to follow any known hardware recommendations. + * It's not clear e.g. why there is a different flow + * for capture and playback, the only information that matters is + * what traffic class is used, and on all SOF-enabled platforms + * only VC0 is supported so the work-around was likely not necessary + * and quite possibly wrong. + */ + + /* DPIB/posbuf position mode: + * For Playback, Use DPIB register from HDA space which + * reflects the actual data transferred. + * For Capture, Use the position buffer for pointer, as DPIB + * is not accurate enough, its update may be completed + * earlier than the data written to DDR. + */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + } else { + /* + * For capture stream, we need more workaround to fix the + * position incorrect issue: + * + * 1. Wait at least 20us before reading position buffer after + * the interrupt generated(IOC), to make sure position update + * happens on frame boundary i.e. 20.833uSec for 48KHz. + * 2. Perform a dummy Read to DPIB register to flush DMA + * position value. + * 3. Read the DMA Position from posbuf. Now the readback + * value should be >= period boundary. + */ + if (can_sleep) + usleep_range(20, 21); + + snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + } + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: + /* + * In case VC1 traffic is disabled this is the recommended option + */ + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: + /* + * This is the recommended option when VC1 is enabled. + * While this isn't needed for SOF platforms it's added for + * consistency and debug. + */ + pos = snd_hdac_stream_get_pos_posbuf(hstream); + break; + default: + dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", + sof_hda_position_quirk); + pos = 0; + break; + } + + if (pos >= hstream->bufsize) + pos = 0; + + return pos; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3e0f7b0c586a..06476ffe96d7 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -497,6 +497,7 @@ struct sof_intel_hda_stream { */ int hda_dsp_probe(struct snd_sof_dev *sdev); int hda_dsp_remove(struct snd_sof_dev *sdev); +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_enable_core(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev, @@ -564,6 +565,9 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep); + struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag); diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 043554d7cb4a..10740c55294d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1577,24 +1577,23 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ struct sof_ipc_ctrl_data *cdata; int ret; - scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); - if (!scontrol->ipc_control_data) - return -ENOMEM; - - if (scontrol->max_size < sizeof(*cdata) || - scontrol->max_size < sizeof(struct sof_abi_hdr)) { - ret = -EINVAL; - goto err; + if (scontrol->max_size < (sizeof(*cdata) + sizeof(struct sof_abi_hdr))) { + dev_err(sdev->dev, "%s: insufficient size for a bytes control: %zu.\n", + __func__, scontrol->max_size); + return -EINVAL; } - /* init the get/put bytes data */ if (scontrol->priv_size > scontrol->max_size - sizeof(*cdata)) { - dev_err(sdev->dev, "err: bytes data size %zu exceeds max %zu.\n", + dev_err(sdev->dev, + "%s: bytes data size %zu exceeds max %zu.\n", __func__, scontrol->priv_size, scontrol->max_size - sizeof(*cdata)); - ret = -EINVAL; - goto err; + return -EINVAL; } + scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); + if (!scontrol->ipc_control_data) + return -ENOMEM; + scontrol->size = sizeof(struct sof_ipc_ctrl_data) + scontrol->priv_size; cdata = scontrol->ipc_control_data; diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 3333a0634e29..e006532caf2f 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -392,7 +392,7 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) PLATFORM_DEVID_NONE, pdev, sizeof(*pdev)); if (IS_ERR(priv->ipc_dev)) { - ret = IS_ERR(priv->ipc_dev); + ret = PTR_ERR(priv->ipc_dev); dev_err(sdev->dev, "failed to create mtk-adsp-ipc device\n"); goto err_adsp_off; } diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 18eb327a57f0..df740be645e8 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -23,6 +23,9 @@ static u32 snd_sof_dsp_power_target(struct snd_sof_dev *sdev) u32 target_dsp_state; switch (sdev->system_suspend_target) { + case SOF_SUSPEND_S5: + case SOF_SUSPEND_S4: + /* DSP should be in D3 if the system is suspending to S3+ */ case SOF_SUSPEND_S3: /* DSP should be in D3 if the system is suspending to S3 */ target_dsp_state = SOF_DSP_PM_D3; @@ -335,8 +338,24 @@ int snd_sof_prepare(struct device *dev) return 0; #if defined(CONFIG_ACPI) - if (acpi_target_system_state() == ACPI_STATE_S0) + switch (acpi_target_system_state()) { + case ACPI_STATE_S0: sdev->system_suspend_target = SOF_SUSPEND_S0IX; + break; + case ACPI_STATE_S1: + case ACPI_STATE_S2: + case ACPI_STATE_S3: + sdev->system_suspend_target = SOF_SUSPEND_S3; + break; + case ACPI_STATE_S4: + sdev->system_suspend_target = SOF_SUSPEND_S4; + break; + case ACPI_STATE_S5: + sdev->system_suspend_target = SOF_SUSPEND_S5; + break; + default: + break; + } #endif return 0; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 8d740635a4bb..28976098a89e 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -318,7 +318,7 @@ sink_prepare: p->walking = false; if (ret < 0) { /* unprepare the source widget */ - if (!widget_ops[widget->id].ipc_unprepare && swidget->prepared) { + if (widget_ops[widget->id].ipc_unprepare && swidget->prepared) { widget_ops[widget->id].ipc_unprepare(swidget); swidget->prepared = false; } diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index 03490a4d4ae7..6bdfa527b7f7 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -150,7 +150,7 @@ static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *bu { struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; - size_t size; + ssize_t size; int ret; if (*ppos) @@ -158,8 +158,10 @@ static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *bu size = simple_write_to_buffer(priv->tx_buffer, priv->max_msg_size, ppos, buffer, count); + if (size < 0) + return size; if (size != count) - return size > 0 ? -EFAULT : size; + return -EFAULT; memset(priv->rx_buffer, 0, priv->max_msg_size); @@ -179,7 +181,7 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; struct sof_ipc4_msg *ipc4_msg = priv->tx_buffer; - size_t size; + ssize_t size; int ret; if (*ppos) @@ -192,18 +194,20 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, size = simple_write_to_buffer(&ipc4_msg->header_u64, sizeof(ipc4_msg->header_u64), ppos, buffer, count); + if (size < 0) + return size; if (size != sizeof(ipc4_msg->header_u64)) - return size > 0 ? -EFAULT : size; + return -EFAULT; count -= size; - if (!count) { - /* Copy the payload */ - size = simple_write_to_buffer(ipc4_msg->data_ptr, - priv->max_msg_size, ppos, buffer, - count); - if (size != count) - return size > 0 ? -EFAULT : size; - } + /* Copy the payload */ + size = simple_write_to_buffer(ipc4_msg->data_ptr, + priv->max_msg_size, ppos, buffer, + count); + if (size < 0) + return size; + if (size != count) + return -EFAULT; ipc4_msg->data_size = count; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70..f0f3d72c0da7 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -85,6 +85,8 @@ enum sof_system_suspend_state { SOF_SUSPEND_NONE = 0, SOF_SUSPEND_S0IX, SOF_SUSPEND_S3, + SOF_SUSPEND_S4, + SOF_SUSPEND_S5, }; enum sof_dfsentry_type { diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..7e8179cae92e 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 4479d74f0a45..9933b33c80ca 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index b7b6f3834ed5..6eb7d93b358d 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -637,10 +637,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b470404a5376..e692ae04436a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -291,6 +291,9 @@ int snd_usb_audioformat_set_sync_ep(struct snd_usb_audio *chip, bool is_playback; int err; + if (fmt->sync_ep) + return 0; /* already set up */ + alts = snd_usb_get_host_interface(chip, fmt->iface, fmt->altsetting); if (!alts) return 0; @@ -304,7 +307,7 @@ int snd_usb_audioformat_set_sync_ep(struct snd_usb_audio *chip, * Generic sync EP handling */ - if (altsd->bNumEndpoints < 2) + if (fmt->ep_idx > 0 || altsd->bNumEndpoints < 2) return 0; is_playback = !(get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 78eb41b621d6..f93201a830b5 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2658,7 +2658,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), .nr_rates = 2, .rate_table = (unsigned int[]) { 44100, 48000 - } + }, + .sync_ep = 0x82, + .sync_iface = 0, + .sync_altsetting = 1, + .sync_ep_idx = 1, + .implicit_fb = 1, } }, { @@ -3798,6 +3803,54 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_FLAG_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + USB_AUDIO_DEVICE(0x534d, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + +/* * MacroSilicon MS2109 based HDMI capture cards * * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. @@ -4114,6 +4167,206 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Fiero SC-01 (firmware v1.0.0 @ 48 kHz) + */ + USB_DEVICE(0x2b53, 0x0023), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Fiero", + .product_name = "SC-01", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .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 }, + .clock = 0x29 + } + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 }, + .clock = 0x29 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ + /* + * Fiero SC-01 (firmware v1.0.0 @ 96 kHz) + */ + USB_DEVICE(0x2b53, 0x0024), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Fiero", + .product_name = "SC-01", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_96000, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 96000 }, + .clock = 0x29 + } + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_96000, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 96000 }, + .clock = 0x29 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ + /* + * Fiero SC-01 (firmware v1.1.0) + */ + USB_DEVICE(0x2b53, 0x0031), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Fiero", + .product_name = "SC-01", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .rate_min = 48000, + .rate_max = 96000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { 48000, 96000 }, + .clock = 0x29 + } + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .rate_min = 48000, + .rate_max = 96000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { 48000, 96000 }, + .clock = 0x29 + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e8468f9b007d..968d90caeefa 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1478,6 +1478,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; @@ -1842,6 +1843,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1397, 0x0508, /* Behringer UMC204HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x1397, 0x0509, /* Behringer UMC404HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x13e5, 0x0001, /* Serato Phono */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x154e, 0x1002, /* Denon DCD-1500RE */ @@ -1904,10 +1909,18 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x534d, 0x0021, /* MacroSilicon MS2100/MS2106 */ + QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x2b53, 0x0023, /* Fiero SC-01 (firmware v1.0.0 @ 48 kHz) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x2b53, 0x0024, /* Fiero SC-01 (firmware v1.0.0 @ 96 kHz) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 0d828e35b401..ab95fb34a635 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -33,6 +33,8 @@ #include <drm/intel_lpe_audio.h> #include "intel_hdmi_audio.h" +#define INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS 5000 + #define for_each_pipe(card_ctx, pipe) \ for ((pipe) = 0; (pipe) < (card_ctx)->num_pipes; (pipe)++) #define for_each_port(card_ctx, port) \ @@ -1066,7 +1068,9 @@ static int had_pcm_open(struct snd_pcm_substream *substream) intelhaddata = snd_pcm_substream_chip(substream); runtime = substream->runtime; - pm_runtime_get_sync(intelhaddata->dev); + retval = pm_runtime_resume_and_get(intelhaddata->dev); + if (retval < 0) + return retval; /* set the runtime hw parameter with local snd_pcm_hardware struct */ runtime->hw = had_pcm_hardware; @@ -1534,8 +1538,12 @@ static void had_audio_wq(struct work_struct *work) container_of(work, struct snd_intelhad, hdmi_audio_wq); struct intel_hdmi_lpe_audio_pdata *pdata = ctx->dev->platform_data; struct intel_hdmi_lpe_audio_port_pdata *ppdata = &pdata->port[ctx->port]; + int ret; + + ret = pm_runtime_resume_and_get(ctx->dev); + if (ret < 0) + return; - pm_runtime_get_sync(ctx->dev); mutex_lock(&ctx->mutex); if (ppdata->pipe < 0) { dev_dbg(ctx->dev, "%s: Event: HAD_NOTIFY_HOT_UNPLUG : port = %d\n", @@ -1802,8 +1810,11 @@ static int __hdmi_lpe_audio_probe(struct platform_device *pdev) pdata->notify_audio_lpe = notify_audio_lpe; spin_unlock_irq(&pdata->lpe_audio_slock); + pm_runtime_set_autosuspend_delay(&pdev->dev, INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_idle(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { |