diff options
Diffstat (limited to 'sound')
35 files changed, 525 insertions, 150 deletions
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6913d113bb4e..5f0e7765b8bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1365,8 +1365,21 @@ static void azx_free(struct azx *chip) if (use_vga_switcheroo(hda)) { if (chip->disabled && hda->probe_continued) snd_hda_unlock_devices(&chip->bus); - if (hda->vga_switcheroo_registered) + if (hda->vga_switcheroo_registered) { vga_switcheroo_unregister_client(chip->pci); + + /* Some GPUs don't have sound, and azx_first_init fails, + * leaving the device probed but non-functional. As long + * as it's probed, the PCI subsystem keeps its runtime + * PM status as active. Force it to suspended (as we + * actually stop the chip) to allow GPU to suspend via + * vga_switcheroo, and print a warning. + */ + dev_warn(&pci->dev, "GPU sound probed, but not operational: please add a quirk to driver_denylist\n"); + pm_runtime_disable(&pci->dev); + pm_runtime_set_suspended(&pci->dev); + pm_runtime_enable(&pci->dev); + } } if (bus->chip_init) { @@ -2212,6 +2225,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), /* KONTRON SinglePC may cause a stall at runtime resume */ SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0), + /* Dell ALC3271 */ + SND_PCI_QUIRK(0x1028, 0x0962, "Dell ALC3271", 0), {} }; #endif /* CONFIG_PM */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 520144df0174..59f6d70689df 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1098,6 +1098,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8231, "HP ProBook 450 G4", CXT_FIXUP_MUTE_LED_GPIO), 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), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e4ab772f2331..e1de24c9f626 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -576,6 +576,9 @@ static void alc_shutup_pins(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + if (spec->no_shutup_pins) + return; + switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: @@ -591,8 +594,7 @@ static void alc_shutup_pins(struct hda_codec *codec) alc_headset_mic_no_shutup(codec); break; default: - if (!spec->no_shutup_pins) - snd_hda_shutup_pins(codec); + snd_hda_shutup_pins(codec); break; } } @@ -3770,6 +3772,7 @@ static void alc225_init(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); msleep(75); + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */ } } @@ -3824,6 +3827,79 @@ static void alc225_shutup(struct hda_codec *codec) } } +static void alc222_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = alc_get_hp_pin(spec); + bool hp1_pin_sense, hp2_pin_sense; + + if (!hp_pin) + return; + + msleep(30); + + hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); + hp2_pin_sense = snd_hda_jack_detect(codec, 0x14); + + if (hp1_pin_sense || hp2_pin_sense) { + msleep(2); + + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + msleep(75); + + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + + msleep(75); + } +} + +static void alc222_shutup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = alc_get_hp_pin(spec); + bool hp1_pin_sense, hp2_pin_sense; + + if (!hp_pin) + hp_pin = 0x21; + + hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); + hp2_pin_sense = snd_hda_jack_detect(codec, 0x14); + + if (hp1_pin_sense || hp2_pin_sense) { + msleep(2); + + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + msleep(75); + + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + msleep(75); + } + alc_auto_setup_eapd(codec, false); + alc_shutup_pins(codec); +} + static void alc_default_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4683,6 +4759,21 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, } } +static void alc295_fixup_hp_mute_led_coefbit11(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef.idx = 0xb; + spec->mute_led_coef.mask = 3 << 3; + spec->mute_led_coef.on = 1 << 3; + spec->mute_led_coef.off = 1 << 4; + snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set); + } +} + static void alc285_fixup_hp_mute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6922,6 +7013,7 @@ enum { ALC290_FIXUP_MONO_SPEAKERS_HSJACK, ALC290_FIXUP_SUBWOOFER, ALC290_FIXUP_SUBWOOFER_HSJACK, + ALC295_FIXUP_HP_MUTE_LED_COEFBIT11, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, @@ -8468,6 +8560,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC283_FIXUP_INT_MIC, }, + [ALC295_FIXUP_HP_MUTE_LED_COEFBIT11] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_hp_mute_led_coefbit11, + }, [ALC298_FIXUP_SAMSUNG_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc298_fixup_samsung_amp, @@ -9182,6 +9278,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11), SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), @@ -9228,6 +9325,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x881e, "HP Laptop 15s-du3xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), @@ -9517,6 +9615,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1849, 0x0269, "Positivo Master C6400", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), @@ -10372,8 +10471,11 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC300; spec->gen.mixer_nid = 0; /* no loopback on ALC300 */ break; + case 0x10ec0222: case 0x10ec0623: spec->codec_variant = ALC269_TYPE_ALC623; + spec->shutup = alc222_shutup; + spec->init_hook = alc222_init; break; case 0x10ec0700: case 0x10ec0701: diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3e2f79466317..a64172136389 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1779,6 +1779,7 @@ config SND_SOC_WM8993 config SND_SOC_WM8994 tristate + depends on MFD_WM8994 config SND_SOC_WM8995 tristate diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e32871b3f68a..be207350b712 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -967,7 +967,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3L_ENA_SHIFT: case ARIZONA_OUT3R_ENA_SHIFT: priv->out_up_pending++; - priv->out_up_delay += 17; + priv->out_up_delay += 17000; break; case ARIZONA_OUT4L_ENA_SHIFT: case ARIZONA_OUT4R_ENA_SHIFT: @@ -977,7 +977,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case WM8997: break; default: - priv->out_up_delay += 10; + priv->out_up_delay += 10000; break; } break; @@ -999,7 +999,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, if (!priv->out_up_pending && priv->out_up_delay) { dev_dbg(component->dev, "Power up delay: %d\n", priv->out_up_delay); - msleep(priv->out_up_delay); + fsleep(priv->out_up_delay); priv->out_up_delay = 0; } break; @@ -1017,7 +1017,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3L_ENA_SHIFT: case ARIZONA_OUT3R_ENA_SHIFT: priv->out_down_pending++; - priv->out_down_delay++; + priv->out_down_delay += 1000; break; case ARIZONA_OUT4L_ENA_SHIFT: case ARIZONA_OUT4R_ENA_SHIFT: @@ -1028,10 +1028,10 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, break; case WM8998: case WM1814: - priv->out_down_delay += 5; + priv->out_down_delay += 5000; break; default: - priv->out_down_delay++; + priv->out_down_delay += 1000; break; } break; @@ -1053,7 +1053,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, if (!priv->out_down_pending && priv->out_down_delay) { dev_dbg(component->dev, "Power down delay: %d\n", priv->out_down_delay); - msleep(priv->out_down_delay); + fsleep(priv->out_down_delay); priv->out_down_delay = 0; } break; diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index ca3b1c00fa78..fe4763805df9 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -234,7 +234,6 @@ static const struct snd_kcontrol_new es8328_right_line_controls = /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0), SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0), @@ -244,7 +243,6 @@ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0), - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0), }; @@ -337,10 +335,10 @@ static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = { SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER, ES8328_DACPOWER_LDAC_OFF, 1), - SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MIXER("Left Mixer", ES8328_DACCONTROL17, 7, 0, &es8328_left_mixer_controls[0], ARRAY_SIZE(es8328_left_mixer_controls)), - SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MIXER("Right Mixer", ES8328_DACCONTROL20, 7, 0, &es8328_right_mixer_controls[0], ARRAY_SIZE(es8328_right_mixer_controls)), @@ -419,19 +417,14 @@ static const struct snd_soc_dapm_route es8328_dapm_routes[] = { { "Right Line Mux", "PGA", "Right PGA Mux" }, { "Right Line Mux", "Differential", "Differential Mux" }, - { "Left Out 1", NULL, "Left DAC" }, - { "Right Out 1", NULL, "Right DAC" }, - { "Left Out 2", NULL, "Left DAC" }, - { "Right Out 2", NULL, "Right DAC" }, - - { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", NULL, "Left DAC" }, { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, { "Left Mixer", "Right Playback Switch", "Right DAC" }, { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, { "Right Mixer", "Left Playback Switch", "Left DAC" }, { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, - { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", NULL, "Right DAC" }, { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, { "DAC DIG", NULL, "DAC STM" }, diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 5dbe5de81dc3..0f8f08363b53 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -62,6 +62,10 @@ #define CDC_WSA_TX_SPKR_PROT_CLK_DISABLE 0 #define CDC_WSA_TX_SPKR_PROT_PCM_RATE_MASK GENMASK(3, 0) #define CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K 0 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_16K 1 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_24K 2 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_32K 3 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_48K 4 #define CDC_WSA_TX0_SPKR_PROT_PATH_CFG0 (0x0248) #define CDC_WSA_TX1_SPKR_PROT_PATH_CTL (0x0264) #define CDC_WSA_TX1_SPKR_PROT_PATH_CFG0 (0x0268) @@ -344,6 +348,7 @@ struct wsa_macro { int ear_spkr_gain; int spkr_gain_offset; int spkr_mode; + u32 pcm_rate_vi; int is_softclip_on[WSA_MACRO_SOFTCLIP_MAX]; int softclip_clk_users[WSA_MACRO_SOFTCLIP_MAX]; struct regmap *regmap; @@ -967,6 +972,7 @@ static int wsa_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); int ret; switch (substream->stream) { @@ -979,6 +985,11 @@ static int wsa_macro_hw_params(struct snd_pcm_substream *substream, return ret; } break; + case SNDRV_PCM_STREAM_CAPTURE: + if (dai->id == WSA_MACRO_AIF_VI) + wsa->pcm_rate_vi = params_rate(params); + + break; default: break; } @@ -1135,35 +1146,11 @@ static void wsa_macro_mclk_enable(struct wsa_macro *wsa, bool mclk_enable) } } -static int wsa_macro_mclk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static void wsa_macro_enable_disable_vi_sense(struct snd_soc_component *component, bool enable, + u32 tx_reg0, u32 tx_reg1, u32 val) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); - - wsa_macro_mclk_enable(wsa, event == SND_SOC_DAPM_PRE_PMU); - return 0; -} - -static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) -{ - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); - u32 tx_reg0, tx_reg1; - - if (test_bit(WSA_MACRO_TX0, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { - tx_reg0 = CDC_WSA_TX0_SPKR_PROT_PATH_CTL; - tx_reg1 = CDC_WSA_TX1_SPKR_PROT_PATH_CTL; - } else if (test_bit(WSA_MACRO_TX1, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { - tx_reg0 = CDC_WSA_TX2_SPKR_PROT_PATH_CTL; - tx_reg1 = CDC_WSA_TX3_SPKR_PROT_PATH_CTL; - } - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* Enable V&I sensing */ + if (enable) { + /* Enable V&I sensing */ snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_RESET_MASK, CDC_WSA_TX_SPKR_PROT_RESET); @@ -1172,10 +1159,10 @@ static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, CDC_WSA_TX_SPKR_PROT_RESET); snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_PCM_RATE_MASK, - CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K); + val); snd_soc_component_update_bits(component, tx_reg1, CDC_WSA_TX_SPKR_PROT_PCM_RATE_MASK, - CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K); + val); snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_CLK_EN_MASK, CDC_WSA_TX_SPKR_PROT_CLK_ENABLE); @@ -1188,9 +1175,7 @@ static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, tx_reg1, CDC_WSA_TX_SPKR_PROT_RESET_MASK, CDC_WSA_TX_SPKR_PROT_NO_RESET); - break; - case SND_SOC_DAPM_POST_PMD: - /* Disable V&I sensing */ + } else { snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_RESET_MASK, CDC_WSA_TX_SPKR_PROT_RESET); @@ -1203,6 +1188,72 @@ static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, tx_reg1, CDC_WSA_TX_SPKR_PROT_CLK_EN_MASK, CDC_WSA_TX_SPKR_PROT_CLK_DISABLE); + } +} + +static void wsa_macro_enable_disable_vi_feedback(struct snd_soc_component *component, + bool enable, u32 rate) +{ + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); + + if (test_bit(WSA_MACRO_TX0, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) + wsa_macro_enable_disable_vi_sense(component, enable, + CDC_WSA_TX0_SPKR_PROT_PATH_CTL, + CDC_WSA_TX1_SPKR_PROT_PATH_CTL, rate); + + if (test_bit(WSA_MACRO_TX1, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) + wsa_macro_enable_disable_vi_sense(component, enable, + CDC_WSA_TX2_SPKR_PROT_PATH_CTL, + CDC_WSA_TX3_SPKR_PROT_PATH_CTL, rate); +} + +static int wsa_macro_mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); + + wsa_macro_mclk_enable(wsa, event == SND_SOC_DAPM_PRE_PMU); + return 0; +} + +static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); + u32 rate_val; + + switch (wsa->pcm_rate_vi) { + case 8000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K; + break; + case 16000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_16K; + break; + case 24000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_24K; + break; + case 32000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_32K; + break; + case 48000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_48K; + break; + default: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K; + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Enable V&I sensing */ + wsa_macro_enable_disable_vi_feedback(component, true, rate_val); + break; + case SND_SOC_DAPM_POST_PMD: + /* Disable V&I sensing */ + wsa_macro_enable_disable_vi_feedback(component, false, rate_val); break; } diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index fd4fa1d5d2d1..5775898fc6f9 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -2322,10 +2322,10 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case CS42L92: case CS47L92: case CS47L93: - out_up_delay = 6; + out_up_delay = 6000; break; default: - out_up_delay = 17; + out_up_delay = 17000; break; } @@ -2356,7 +2356,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case MADERA_OUT3R_ENA_SHIFT: priv->out_up_pending--; if (!priv->out_up_pending) { - msleep(priv->out_up_delay); + fsleep(priv->out_up_delay); priv->out_up_delay = 0; } break; @@ -2375,7 +2375,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case MADERA_OUT3L_ENA_SHIFT: case MADERA_OUT3R_ENA_SHIFT: priv->out_down_pending++; - priv->out_down_delay++; + priv->out_down_delay += 1000; break; default: break; @@ -2392,7 +2392,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case MADERA_OUT3R_ENA_SHIFT: priv->out_down_pending--; if (!priv->out_down_pending) { - msleep(priv->out_down_delay); + fsleep(priv->out_down_delay); priv->out_down_delay = 0; } break; diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 1951bae95b31..273bf4027a6e 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -315,7 +315,7 @@ 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_0 = 0, asi_cfg_1 = 0; + u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0, asi_cfg_4 = 0; int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -324,12 +324,14 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fallthrough; case SND_SOC_DAIFMT_NB_NF: asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING; + asi_cfg_4 = TAS2764_TDM_CFG4_TX_FALLING; 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; + asi_cfg_4 = TAS2764_TDM_CFG4_TX_RISING; break; } @@ -339,6 +341,12 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (ret < 0) return ret; + ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG4, + TAS2764_TDM_CFG4_TX_MASK, + asi_cfg_4); + if (ret < 0) + return ret; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START; diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h index f015f22a083b..337bc611bee9 100644 --- a/sound/soc/codecs/tas2764.h +++ b/sound/soc/codecs/tas2764.h @@ -25,7 +25,7 @@ /* Power Control */ #define TAS2764_PWR_CTRL TAS2764_REG(0X0, 0x02) -#define TAS2764_PWR_CTRL_MASK GENMASK(1, 0) +#define TAS2764_PWR_CTRL_MASK GENMASK(2, 0) #define TAS2764_PWR_CTRL_ACTIVE 0x0 #define TAS2764_PWR_CTRL_MUTE BIT(0) #define TAS2764_PWR_CTRL_SHUTDOWN BIT(1) @@ -75,6 +75,12 @@ #define TAS2764_TDM_CFG3_RXS_SHIFT 0x4 #define TAS2764_TDM_CFG3_MASK GENMASK(3, 0) +/* TDM Configuration Reg4 */ +#define TAS2764_TDM_CFG4 TAS2764_REG(0X0, 0x0d) +#define TAS2764_TDM_CFG4_TX_MASK BIT(0) +#define TAS2764_TDM_CFG4_TX_RISING 0x0 +#define TAS2764_TDM_CFG4_TX_FALLING BIT(0) + /* TDM Configuration Reg5 */ #define TAS2764_TDM_CFG5 TAS2764_REG(0X0, 0x0e) #define TAS2764_TDM_CFG5_VSNS_MASK BIT(6) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index ec0df3b1ef61..4e71dc1cf588 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -508,7 +508,7 @@ static int tas2770_codec_probe(struct snd_soc_component *component) } static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0); -static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -10050, 50, 0); static const struct snd_kcontrol_new tas2770_snd_controls[] = { SOC_SINGLE_TLV("Speaker Playback Volume", TAS2770_PLAY_CFG_REG2, diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 94ffd2ba29ae..765ac2a3e963 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -2281,7 +2281,7 @@ static irqreturn_t wcd934x_slim_irq_handler(int irq, void *data) { struct wcd934x_codec *wcd = data; unsigned long status = 0; - int i, j, port_id; + unsigned int i, j, port_id; unsigned int val, int_val = 0; irqreturn_t ret = IRQ_NONE; bool tx; diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 28b4656c4e14..b2f87af1bfc8 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -952,7 +952,7 @@ static int wm0010_spi_probe(struct spi_device *spi) if (ret) { dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n", irq, ret); - return ret; + goto free_irq; } if (spi->max_speed_hz) @@ -964,9 +964,18 @@ static int wm0010_spi_probe(struct spi_device *spi) &soc_component_dev_wm0010, wm0010_dai, ARRAY_SIZE(wm0010_dai)); if (ret < 0) - return ret; + goto disable_irq_wake; return 0; + +disable_irq_wake: + irq_set_irq_wake(wm0010->irq, 0); + +free_irq: + if (wm0010->irq) + free_irq(wm0010->irq, wm0010); + + return ret; } static int wm0010_spi_remove(struct spi_device *spi) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 7c6e01720d65..bc3dfb53ba95 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -302,7 +302,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w) } else { wseq = wm5110_no_dre_left_enable; nregs = ARRAY_SIZE(wm5110_no_dre_left_enable); - priv->out_up_delay += 10; + priv->out_up_delay += 10000; } break; case ARIZONA_OUT1R_ENA_SHIFT: @@ -312,7 +312,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w) } else { wseq = wm5110_no_dre_right_enable; nregs = ARRAY_SIZE(wm5110_no_dre_right_enable); - priv->out_up_delay += 10; + priv->out_up_delay += 10000; } break; default: @@ -338,7 +338,7 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w) snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, ARIZONA_WS_TRG1, 0); - priv->out_down_delay += 27; + priv->out_down_delay += 27000; } break; case ARIZONA_OUT1R_ENA_SHIFT: @@ -350,7 +350,7 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w) snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, ARIZONA_WS_TRG2, 0); - priv->out_down_delay += 27; + priv->out_down_delay += 27000; } break; default: diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index a87278c8ed3c..6c4141cfe971 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -492,11 +492,17 @@ static int fsl_audmix_probe(struct platform_device *pdev) goto err_disable_pm; } - priv->pdev = platform_device_register_data(dev, "imx-audmix", 0, NULL, 0); - if (IS_ERR(priv->pdev)) { - ret = PTR_ERR(priv->pdev); - dev_err(dev, "failed to register platform: %d\n", ret); - goto err_disable_pm; + /* + * If dais property exist, then register the imx-audmix card driver. + * otherwise, it should be linked by audio graph card. + */ + if (of_find_property(pdev->dev.of_node, "dais", NULL)) { + priv->pdev = platform_device_register_data(dev, "imx-audmix", 0, NULL, 0); + if (IS_ERR(priv->pdev)) { + ret = PTR_ERR(priv->pdev); + dev_err(dev, "failed to register platform: %d\n", ret); + goto err_disable_pm; + } } return 0; diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 67434524076b..0f3b5c121797 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -764,6 +764,8 @@ static int imx_card_probe(struct platform_device *pdev) data->dapm_routes[i].sink = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%d %s", i + 1, "Playback"); + if (!data->dapm_routes[i].sink) + return -ENOMEM; data->dapm_routes[i].source = "CPU-Playback"; } } @@ -781,6 +783,8 @@ static int imx_card_probe(struct platform_device *pdev) data->dapm_routes[i].source = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%d %s", i + 1, "Capture"); + if (!data->dapm_routes[i].source) + return -ENOMEM; data->dapm_routes[i].sink = "CPU-Capture"; } } diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8706fef8ccce..721b9971fd74 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1102,7 +1102,22 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, - { /* Vexia Edu Atla 10 tablet */ + { + /* Vexia Edu Atla 10 tablet 5V version */ + .matches = { + /* Having all 3 of these not set is somewhat unique */ + DMI_MATCH(DMI_SYS_VENDOR, "To be filled by O.E.M."), + DMI_MATCH(DMI_PRODUCT_NAME, "To be filled by O.E.M."), + DMI_MATCH(DMI_BOARD_NAME, "To be filled by O.E.M."), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "05/14/2015"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, + { /* Vexia Edu Atla 10 tablet 9V version */ .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index b74b67720ef4..ad2c9788e321 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -902,9 +902,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, if (ret < 0) { dev_err(dev, "q6asm_open_write failed\n"); - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; - return ret; + goto open_err; } } @@ -913,7 +911,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, prtd->session_id, dir); if (ret) { dev_err(dev, "Stream reg failed ret:%d\n", ret); - return ret; + goto q6_err; } ret = __q6asm_dai_compr_set_codec_params(component, stream, @@ -921,7 +919,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, prtd->stream_id); if (ret) { dev_err(dev, "codec param setup failed ret:%d\n", ret); - return ret; + goto q6_err; } ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, @@ -930,12 +928,21 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, if (ret < 0) { dev_err(dev, "Buffer Mapping failed ret:%d\n", ret); - return -ENOMEM; + ret = -ENOMEM; + goto q6_err; } prtd->state = Q6ASM_STREAM_RUNNING; return 0; + +q6_err: + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); + +open_err: + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return ret; } static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component, diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a2221ebb1b6a..c04c38d58804 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -214,8 +214,9 @@ config SND_SOC_SAMSUNG_TM2_WM5110 config SND_SOC_SAMSUNG_ARIES_WM8994 tristate "SoC I2S Audio support for WM8994 on Aries" - depends on SND_SOC_SAMSUNG && MFD_WM8994 && IIO && EXTCON + depends on SND_SOC_SAMSUNG && I2C && IIO && EXTCON select SND_SOC_BT_SCO + select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -227,8 +228,9 @@ config SND_SOC_SAMSUNG_ARIES_WM8994 config SND_SOC_SAMSUNG_MIDAS_WM1811 tristate "SoC I2S Audio support for Midas boards" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S + select MFD_WM8994 select SND_SOC_WM8994 help Say Y if you want to add support for SoC audio on the Midas boards. diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index af8ef2a27d34..65022ba5c587 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1694,20 +1694,6 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) return 1; } -int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) -{ - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct device *dev = rsnd_priv_to_dev(priv); - - if (!runtime) { - dev_warn(dev, "Can't update kctrl when idle\n"); - return 0; - } - - return 1; -} - struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index f8ef6836ef84..690f4932357c 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -742,7 +742,6 @@ struct rsnd_kctrl_cfg_s { #define rsnd_kctrl_vals(x) ((x).val) /* = (x).cfg.val[0] */ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); -int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index f832165e46bc..9893839666d7 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -530,6 +530,22 @@ static irqreturn_t rsnd_src_interrupt(int irq, void *data) return IRQ_HANDLED; } +static int rsnd_src_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + if (!runtime) { + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_warn(dev, "\"SRC Out Rate\" can use during running\n"); + + return 0; + } + + return 1; +} + static int rsnd_src_probe_(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -593,7 +609,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", - rsnd_kctrl_accept_runtime, + rsnd_src_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 2e33a1fa0a6f..b6a7011b4e3b 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -243,8 +243,7 @@ static void rz_ssi_stream_quit(struct rz_ssi_priv *ssi, static int rz_ssi_clk_setup(struct rz_ssi_priv *ssi, unsigned int rate, unsigned int channels) { - static s8 ckdv[16] = { 1, 2, 4, 8, 16, 32, 64, 128, - 6, 12, 24, 48, 96, -1, -1, -1 }; + static u8 ckdv[] = { 1, 2, 4, 8, 16, 32, 64, 128, 6, 12, 24, 48, 96 }; unsigned int channel_bits = 32; /* System Word Length */ unsigned long bclk_rate = rate * channels * channel_bits; unsigned int div; @@ -488,6 +487,8 @@ static int rz_ssi_pio_send(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) sample_space = strm->fifo_sample_size; ssifsr = rz_ssi_reg_readl(ssi, SSIFSR); sample_space -= (ssifsr >> SSIFSR_TDC_SHIFT) & SSIFSR_TDC_MASK; + if (sample_space < 0) + return -EINVAL; /* Only add full frames at a time */ while (frames_left && (sample_space >= runtime->channels)) { diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 57caa91a4376..d8d0a26a554d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && ((int)val + min) > mc->platform_max) + if (mc->platform_max && val > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && ((int)val2 + min) > mc->platform_max) + if (mc->platform_max && val2 > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; @@ -485,17 +485,16 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; + uinfo->value.integer.max = max; return 0; } diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6744318de612..0449e7a2669f 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -258,6 +258,7 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev) } EXPORT_SYMBOL_NS(hda_codec_i915_exit, SND_SOC_SOF_HDA_AUDIO_CODEC_I915); +MODULE_SOFTDEP("pre: snd-hda-codec-hdmi"); #endif MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index dd8d13f3fd12..e885af039b92 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -175,6 +175,7 @@ struct sun4i_spdif_quirks { unsigned int reg_dac_txdata; bool has_reset; unsigned int val_fctl_ftx; + unsigned int mclk_multiplier; }; struct sun4i_spdif_dev { @@ -311,6 +312,7 @@ static int sun4i_spdif_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + mclk *= host->quirks->mclk_multiplier; ret = clk_set_rate(host->spdif_clk, mclk); if (ret < 0) { @@ -345,6 +347,7 @@ static int sun4i_spdif_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + mclk_div *= host->quirks->mclk_multiplier; reg_val = 0; reg_val |= SUN4I_SPDIF_TXCFG_ASS; @@ -427,24 +430,28 @@ static struct snd_soc_dai_driver sun4i_spdif_dai = { static const struct sun4i_spdif_quirks sun4i_a10_spdif_quirks = { .reg_dac_txdata = SUN4I_SPDIF_TXFIFO, .val_fctl_ftx = SUN4I_SPDIF_FCTL_FTX, + .mclk_multiplier = 1, }; static const struct sun4i_spdif_quirks sun6i_a31_spdif_quirks = { .reg_dac_txdata = SUN4I_SPDIF_TXFIFO, .val_fctl_ftx = SUN4I_SPDIF_FCTL_FTX, .has_reset = true, + .mclk_multiplier = 1, }; static const struct sun4i_spdif_quirks sun8i_h3_spdif_quirks = { .reg_dac_txdata = SUN8I_SPDIF_TXFIFO, .val_fctl_ftx = SUN4I_SPDIF_FCTL_FTX, .has_reset = true, + .mclk_multiplier = 4, }; static const struct sun4i_spdif_quirks sun50i_h6_spdif_quirks = { .reg_dac_txdata = SUN8I_SPDIF_TXFIFO, .val_fctl_ftx = SUN50I_H6_SPDIF_FCTL_FTX, .has_reset = true, + .mclk_multiplier = 1, }; static const struct of_device_id sun4i_spdif_of_match[] = { diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 149f4e2ce999..7f2734318452 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -182,6 +182,8 @@ static int j721e_configure_refclk(struct j721e_priv *priv, clk_id = J721E_CLK_PARENT_48000; else if (!(rate % 11025) && priv->pll_rates[J721E_CLK_PARENT_44100]) clk_id = J721E_CLK_PARENT_44100; + else if (!(rate % 11025) && priv->pll_rates[J721E_CLK_PARENT_48000]) + clk_id = J721E_CLK_PARENT_48000; else return ret; diff --git a/sound/usb/format.c b/sound/usb/format.c index 3b3a5ea6fcbf..f33d25a4e4cc 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -263,7 +263,8 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof } /* Jabra Evolve 65 headset */ - if (chip->usb_id == USB_ID(0x0b0e, 0x030b)) { + if (chip->usb_id == USB_ID(0x0b0e, 0x030b) || + chip->usb_id == USB_ID(0x0b0e, 0x030c)) { /* only 48kHz for playback while keeping 16kHz for capture */ if (fp->nr_rates != 1) return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 9a361b202a09..c6586da43a04 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -489,16 +489,84 @@ static void ch345_broken_sysex_input(struct snd_usb_midi_in_endpoint *ep, /* * CME protocol: like the standard protocol, but SysEx commands are sent as a - * single USB packet preceded by a 0x0F byte. + * single USB packet preceded by a 0x0F byte, as are system realtime + * messages and MIDI Active Sensing. + * Also, multiple messages can be sent in the same packet. */ static void snd_usbmidi_cme_input(struct snd_usb_midi_in_endpoint *ep, uint8_t *buffer, int buffer_length) { - if (buffer_length < 2 || (buffer[0] & 0x0f) != 0x0f) - snd_usbmidi_standard_input(ep, buffer, buffer_length); - else - snd_usbmidi_input_data(ep, buffer[0] >> 4, - &buffer[1], buffer_length - 1); + int remaining = buffer_length; + + /* + * CME send sysex, song position pointer, system realtime + * and active sensing using CIN 0x0f, which in the standard + * is only intended for single byte unparsed data. + * So we need to interpret these here before sending them on. + * By default, we assume single byte data, which is true + * for system realtime (midi clock, start, stop and continue) + * and active sensing, and handle the other (known) cases + * separately. + * In contrast to the standard, CME does not split sysex + * into multiple 4-byte packets, but lumps everything together + * into one. In addition, CME can string multiple messages + * together in the same packet; pressing the Record button + * on an UF6 sends a sysex message directly followed + * by a song position pointer in the same packet. + * For it to have any reasonable meaning, a sysex message + * needs to be at least 3 bytes in length (0xf0, id, 0xf7), + * corresponding to a packet size of 4 bytes, and the ones sent + * by CME devices are 6 or 7 bytes, making the packet fragments + * 7 or 8 bytes long (six or seven bytes plus preceding CN+CIN byte). + * For the other types, the packet size is always 4 bytes, + * as per the standard, with the data size being 3 for SPP + * and 1 for the others. + * Thus all packet fragments are at least 4 bytes long, so we can + * skip anything that is shorter; this also conveniantly skips + * packets with size 0, which CME devices continuously send when + * they have nothing better to do. + * Another quirk is that sometimes multiple messages are sent + * in the same packet. This has been observed for midi clock + * and active sensing i.e. 0x0f 0xf8 0x00 0x00 0x0f 0xfe 0x00 0x00, + * but also multiple note ons/offs, and control change together + * with MIDI clock. Similarly, some sysex messages are followed by + * the song position pointer in the same packet, and occasionally + * additionally by a midi clock or active sensing. + * We handle this by looping over all data and parsing it along the way. + */ + while (remaining >= 4) { + int source_length = 4; /* default */ + + if ((buffer[0] & 0x0f) == 0x0f) { + int data_length = 1; /* default */ + + if (buffer[1] == 0xf0) { + /* Sysex: Find EOX and send on whole message. */ + /* To kick off the search, skip the first + * two bytes (CN+CIN and SYSEX (0xf0). + */ + uint8_t *tmp_buf = buffer + 2; + int tmp_length = remaining - 2; + + while (tmp_length > 1 && *tmp_buf != 0xf7) { + tmp_buf++; + tmp_length--; + } + data_length = tmp_buf - buffer; + source_length = data_length + 1; + } else if (buffer[1] == 0xf2) { + /* Three byte song position pointer */ + data_length = 3; + } + snd_usbmidi_input_data(ep, buffer[0] >> 4, + &buffer[1], data_length); + } else { + /* normal channel events */ + snd_usbmidi_standard_input(ep, buffer, source_length); + } + buffer += source_length; + remaining -= source_length; + } } /* @@ -1145,7 +1213,7 @@ static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { struct usbmidi_out_port *port = substream->runtime->private_data; - cancel_work_sync(&port->ep->work); + flush_work(&port->ep->work); return substream_open(substream, 0, 0); } diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e45f3d3e11b4..5eccc8af839f 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3553,6 +3553,52 @@ static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer, } } +/* + * Some Plantronics headsets have control names that don't meet ALSA naming + * standards. This function fixes nonstandard source names. By the time + * this function is called the control name should look like one of these: + * "source names Playback Volume" + * "source names Playback Switch" + * "source names Capture Volume" + * "source names Capture Switch" + * If any of the trigger words are found in the name then the name will + * be changed to: + * "Headset Playback Volume" + * "Headset Playback Switch" + * "Headset Capture Volume" + * "Headset Capture Switch" + * depending on the current suffix. + */ +static void snd_fix_plt_name(struct snd_usb_audio *chip, + struct snd_ctl_elem_id *id) +{ + /* no variant of "Sidetone" should be added to this list */ + static const char * const trigger[] = { + "Earphone", "Microphone", "Receive", "Transmit" + }; + static const char * const suffix[] = { + " Playback Volume", " Playback Switch", + " Capture Volume", " Capture Switch" + }; + int i; + + for (i = 0; i < ARRAY_SIZE(trigger); i++) + if (strstr(id->name, trigger[i])) + goto triggered; + usb_audio_dbg(chip, "no change in %s\n", id->name); + return; + +triggered: + for (i = 0; i < ARRAY_SIZE(suffix); i++) + if (strstr(id->name, suffix[i])) { + usb_audio_dbg(chip, "fixing kctl name %s\n", id->name); + snprintf(id->name, sizeof(id->name), "Headset%s", + suffix[i]); + return; + } + usb_audio_dbg(chip, "something wrong in kctl name %s\n", id->name); +} + void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl) @@ -3570,5 +3616,10 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, cval->min_mute = 1; break; } + + /* ALSA-ify some Plantronics headset control names */ + if (USB_ID_VENDOR(mixer->chip->usb_id) == 0x047f && + (cval->control == UAC_FU_MUTE || cval->control == UAC_FU_VOLUME)) + snd_fix_plt_name(mixer->chip, &kctl->id); } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8661399e60d5..10430c76475b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1502,6 +1502,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */ case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; @@ -1836,6 +1837,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0c45, 0x6340, /* Sonix HD USB Camera */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0d8c, 0x0014, /* USB Audio Device */ + QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0ecb, 0x205c, /* JBL Quantum610 Wireless */ QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ @@ -1932,6 +1935,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x2d95, 0x8021, /* VIVO USB-C-XE710 HEADSET */ QUIRK_FLAG_CTL_MSG_DELAY_1M), + DEVICE_FLG(0x2fc6, 0xf0b7, /* iBasso DC07 Pro */ + QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index c3292afa883e..b8f0c0298f14 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -151,6 +151,12 @@ static int snd_usx2y_card_used[SNDRV_CARDS]; static void snd_usx2y_card_private_free(struct snd_card *card); static void usx2y_unlinkseq(struct snd_usx2y_async_seq *s); +#ifdef USX2Y_NRPACKS_VARIABLE +int nrpacks = USX2Y_NRPACKS; /* number of packets per urb */ +module_param(nrpacks, int, 0444); +MODULE_PARM_DESC(nrpacks, "Number of packets per URB."); +#endif + /* * pipe 4 is used for switching the lamps, setting samplerate, volumes .... */ @@ -433,6 +439,11 @@ static int snd_usx2y_probe(struct usb_interface *intf, struct snd_card *card; int err; +#ifdef USX2Y_NRPACKS_VARIABLE + if (nrpacks < 0 || nrpacks > USX2Y_NRPACKS_MAX) + return -EINVAL; +#endif + if (le16_to_cpu(device->descriptor.idVendor) != 0x1604 || (le16_to_cpu(device->descriptor.idProduct) != USB_ID_US122 && le16_to_cpu(device->descriptor.idProduct) != USB_ID_US224 && diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 8d82f5cc2fe1..0538c457921e 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -7,6 +7,32 @@ #define NRURBS 2 +/* Default value used for nr of packs per urb. + * 1 to 4 have been tested ok on uhci. + * To use 3 on ohci, you'd need a patch: + * look for "0000425-linux-2.6.9-rc4-mm1_ohci-hcd.patch.gz" on + * "https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=0000425" + * + * 1, 2 and 4 work out of the box on ohci, if I recall correctly. + * Bigger is safer operation, smaller gives lower latencies. + */ +#define USX2Y_NRPACKS 4 + +#define USX2Y_NRPACKS_MAX 1024 + +/* If your system works ok with this module's parameter + * nrpacks set to 1, you might as well comment + * this define out, and thereby produce smaller, faster code. + * You'd also set USX2Y_NRPACKS to 1 then. + */ +#define USX2Y_NRPACKS_VARIABLE 1 + +#ifdef USX2Y_NRPACKS_VARIABLE +extern int nrpacks; +#define nr_of_packs() nrpacks +#else +#define nr_of_packs() USX2Y_NRPACKS +#endif #define URBS_ASYNC_SEQ 10 #define URB_DATA_LEN_ASYNC_SEQ 32 diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index c39cc6851e2d..a6ed4f0230b7 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -28,33 +28,6 @@ #include "usx2y.h" #include "usbusx2y.h" -/* Default value used for nr of packs per urb. - * 1 to 4 have been tested ok on uhci. - * To use 3 on ohci, you'd need a patch: - * look for "0000425-linux-2.6.9-rc4-mm1_ohci-hcd.patch.gz" on - * "https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=0000425" - * - * 1, 2 and 4 work out of the box on ohci, if I recall correctly. - * Bigger is safer operation, smaller gives lower latencies. - */ -#define USX2Y_NRPACKS 4 - -/* If your system works ok with this module's parameter - * nrpacks set to 1, you might as well comment - * this define out, and thereby produce smaller, faster code. - * You'd also set USX2Y_NRPACKS to 1 then. - */ -#define USX2Y_NRPACKS_VARIABLE 1 - -#ifdef USX2Y_NRPACKS_VARIABLE -static int nrpacks = USX2Y_NRPACKS; /* number of packets per urb */ -#define nr_of_packs() nrpacks -module_param(nrpacks, int, 0444); -MODULE_PARM_DESC(nrpacks, "Number of packets per URB."); -#else -#define nr_of_packs() USX2Y_NRPACKS -#endif - static int usx2y_urb_capt_retire(struct snd_usx2y_substream *subs) { struct urb *urb = subs->completed_urb; diff --git a/sound/virtio/virtio_pcm.c b/sound/virtio/virtio_pcm.c index c10d91fff2fb..1ddec1f4f05d 100644 --- a/sound/virtio/virtio_pcm.c +++ b/sound/virtio/virtio_pcm.c @@ -337,6 +337,21 @@ int virtsnd_pcm_parse_cfg(struct virtio_snd *snd) if (!snd->substreams) return -ENOMEM; + /* + * Initialize critical substream fields early in case we hit an + * error path and end up trying to clean up uninitialized structures + * elsewhere. + */ + for (i = 0; i < snd->nsubstreams; ++i) { + struct virtio_pcm_substream *vss = &snd->substreams[i]; + + vss->snd = snd; + vss->sid = i; + INIT_WORK(&vss->elapsed_period, virtsnd_pcm_period_elapsed); + init_waitqueue_head(&vss->msg_empty); + spin_lock_init(&vss->lock); + } + info = kcalloc(snd->nsubstreams, sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; @@ -350,12 +365,6 @@ int virtsnd_pcm_parse_cfg(struct virtio_snd *snd) struct virtio_pcm_substream *vss = &snd->substreams[i]; struct virtio_pcm *vpcm; - vss->snd = snd; - vss->sid = i; - INIT_WORK(&vss->elapsed_period, virtsnd_pcm_period_elapsed); - init_waitqueue_head(&vss->msg_empty); - spin_lock_init(&vss->lock); - rc = virtsnd_pcm_build_hw(vss, &info[i]); if (rc) goto on_exit; |