diff options
author | Takashi Iwai <tiwai@suse.de> | 2019-10-28 12:43:29 +0100 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2019-10-28 12:43:34 +0100 |
commit | e2e556a9549eebde9797a04729efdfc54f37e5cc (patch) | |
tree | be8e60c5c7021ebb3bda12ebe8a921fbb8883c4e /sound | |
parent | 0a671dc500553cf6f4cc3efbcb0923b5e9adccb5 (diff) | |
parent | 1a7f60b9df614bb36d14dc0c0bc898a31b2b506f (diff) | |
download | linux-stable-e2e556a9549eebde9797a04729efdfc54f37e5cc.tar.gz linux-stable-e2e556a9549eebde9797a04729efdfc54f37e5cc.tar.bz2 linux-stable-e2e556a9549eebde9797a04729efdfc54f37e5cc.zip |
Merge branch 'for-linus' into for-next
Back-merge the development process for catching up the HD-audio fix
(and apply a new one on top of that).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'sound')
34 files changed, 382 insertions, 102 deletions
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 7ac0d9f495c4..f7f0db5aa811 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -252,8 +252,7 @@ end: return err; } -static unsigned int -map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) +static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 211ca85acd8c..cfab60d88c92 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -271,6 +271,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, ret = snd_hdac_ext_bus_link_power_up(link); /* + * clear the register to invalidate all the output streams + */ + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, + ML_LOSIDV_STREAM_MASK, 0); + /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery */ diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index d3999e7b0705..7e7be8e4dcf9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -447,8 +447,6 @@ static void azx_int_disable(struct hdac_bus *bus) list_for_each_entry(azx_dev, &bus->stream_list, list) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0); - synchronize_irq(bus->irq); - /* disable SIE for all streams */ snd_hdac_chip_writeb(bus, INTCTL, 0); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca462dd39a48..2a9d87ff2e1c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1355,9 +1355,9 @@ static int azx_free(struct azx *chip) } if (bus->chip_init) { - azx_stop_chip(chip); azx_clear_irq_pending(chip); azx_stop_all_streams(chip); + azx_stop_chip(chip); } if (bus->irq >= 0) @@ -2386,6 +2386,12 @@ static const struct pci_device_id azx_ids[] = { /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Jasperlake */ + { PCI_DEVICE(0x8086, 0x38c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Tigerlake */ + { PCI_DEVICE(0x8086, 0xa0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bca5de78e9ad..795cbda32cbb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3474,6 +3474,8 @@ static int patch_nvhdmi(struct hda_codec *codec) nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + codec->link_down_at_suspend = 1; + generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin); return 0; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..80f66ba85f87 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -393,6 +393,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: alc_update_coef_idx(codec, 0x10, 1<<15, 0); break; case 0x10ec0662: @@ -408,6 +409,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0672: alc_update_coef_idx(codec, 0xd, 0, 1<<14); /* EAPD Ctrl */ break; + case 0x10ec0623: + alc_update_coef_idx(codec, 0x19, 1<<13, 0); + break; case 0x10ec0668: alc_update_coef_idx(codec, 0x7, 3<<13, 0); break; @@ -2919,6 +2923,7 @@ enum { ALC269_TYPE_ALC225, ALC269_TYPE_ALC294, ALC269_TYPE_ALC300, + ALC269_TYPE_ALC623, ALC269_TYPE_ALC700, }; @@ -2954,6 +2959,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC225: case ALC269_TYPE_ALC294: case ALC269_TYPE_ALC300: + case ALC269_TYPE_ALC623: case ALC269_TYPE_ALC700: ssids = alc269_ssids; break; @@ -5358,6 +5364,17 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); + snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); +} + static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5822,6 +5839,7 @@ enum { ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -5869,6 +5887,7 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, @@ -6558,6 +6577,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_dell_xps_13_headphone_noise2, + .chained = true, + .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE + }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -6912,6 +6937,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7001,17 +7035,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), @@ -7108,6 +7142,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), @@ -7186,6 +7221,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + 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, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -7987,9 +8024,13 @@ 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 0x10ec0623: + spec->codec_variant = ALC269_TYPE_ALC623; + break; case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ @@ -9187,6 +9228,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), @@ -9204,6 +9246,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269), HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269), HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269), HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662), HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880), HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882), diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index e609abcf3220..eb709d528259 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c, max98373->i_slot = value & 0xF; else max98373->i_slot = 1; - - max98373->reset_gpio = of_get_named_gpio(dev->of_node, + if (dev->of_node) { + max98373->reset_gpio = of_get_named_gpio(dev->of_node, "maxim,reset-gpio", 0); - if (!gpio_is_valid(max98373->reset_gpio)) { - dev_err(dev, "Looking up %s property in node %s failed %d\n", - "maxim,reset-gpio", dev->of_node->full_name, - max98373->reset_gpio); + if (!gpio_is_valid(max98373->reset_gpio)) { + dev_err(dev, "Looking up %s property in node %s failed %d\n", + "maxim,reset-gpio", dev->of_node->full_name, + max98373->reset_gpio); + } else { + dev_dbg(dev, "maxim,reset-gpio=%d", + max98373->reset_gpio); + } } else { - dev_dbg(dev, "maxim,reset-gpio=%d", - max98373->reset_gpio); + /* this makes reset_gpio as invalid */ + max98373->reset_gpio = -1; } if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 9fa5d44fdc79..58b2468fb2a7 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -243,6 +243,10 @@ static const char *const rx_mix1_text[] = { "ZERO", "IIR1", "IIR2", "RX1", "RX2", "RX3" }; +static const char * const rx_mix2_text[] = { + "ZERO", "IIR1", "IIR2" +}; + static const char *const dec_mux_text[] = { "ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2" }; @@ -270,6 +274,16 @@ static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text), }; +/* RX1 MIX2 */ +static const struct soc_enum rx_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX1_B3_CTL, + 0, 3, rx_mix2_text); + +/* RX2 MIX2 */ +static const struct soc_enum rx2_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B3_CTL, + 0, 3, rx_mix2_text); + /* DEC */ static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE( LPASS_CDC_CONN_TX_B1_CTL, 0, 6, dec_mux_text); @@ -309,6 +323,10 @@ static const struct snd_kcontrol_new rx3_mix1_inp2_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP2 Mux", rx3_mix1_inp_enum[1]); static const struct snd_kcontrol_new rx3_mix1_inp3_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP3 Mux", rx3_mix1_inp_enum[2]); +static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX1 MIX2 INP1 Mux", rx_mix2_inp1_chain_enum); +static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum); /* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */ static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0); @@ -740,6 +758,10 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = { &rx3_mix1_inp2_mux), SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0, &rx3_mix1_inp3_mux), + SND_SOC_DAPM_MUX("RX1 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx1_mix2_inp1_mux), + SND_SOC_DAPM_MUX("RX2 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx2_mix2_inp1_mux), SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux), SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux), diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 762595de956c..c506c9305043 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1770,6 +1770,9 @@ static int rt5651_detect_headset(struct snd_soc_component *component) static bool rt5651_support_button_press(struct rt5651_priv *rt5651) { + if (!rt5651->hp_jack) + return false; + /* Button press support only works with internal jack-detection */ return (rt5651->hp_jack->status & SND_JACK_MICROPHONE) && rt5651->gpiod_hp_det == NULL; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 1ef470700ed5..c50b75ce82e0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -995,6 +995,16 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + rt5682->hs_jack = hs_jack; + + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + return 0; + } + switch (rt5682->pdata.jd_src) { case RT5682_JD1: snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, @@ -1032,8 +1042,6 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, break; } - rt5682->hs_jack = hs_jack; - return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c3d06e8bc54f..d5fb7f5dd551 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -533,13 +533,10 @@ static SOC_ENUM_SINGLE_DECL(dac_osr, static SOC_ENUM_SINGLE_DECL(adc_osr, WM8994_OVERSAMPLING, 1, osr_text); -static const struct snd_kcontrol_new wm8994_snd_controls[] = { +static const struct snd_kcontrol_new wm8994_common_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1_ADC1_RIGHT_VOLUME, 1, 119, 0, digital_tlv), -SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME, - WM8994_AIF1_ADC2_RIGHT_VOLUME, - 1, 119, 0, digital_tlv), SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2_ADC_RIGHT_VOLUME, 1, 119, 0, digital_tlv), @@ -556,8 +553,6 @@ SOC_ENUM("AIF2DACR Source", aif2dacr_src), SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), -SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, - WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv), @@ -565,17 +560,12 @@ SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv), SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv), SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0), -SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0), SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0), WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2), WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1), WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0), -WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2), -WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1), -WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0), - WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2), WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1), WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0), @@ -594,9 +584,6 @@ SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0), SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf), SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0), -SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf), -SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0), - SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf), SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0), @@ -637,6 +624,24 @@ SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF2_DAC_FILTERS_2, 8, 1, 0), }; +/* Controls not available on WM1811 */ +static const struct snd_kcontrol_new wm8994_snd_controls[] = { +SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1_ADC2_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), + +SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0), + +WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2), +WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1), +WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0), + +SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf), +SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0), +}; + static const struct snd_kcontrol_new wm8994_eq_controls[] = { SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0, eq_tlv), @@ -4258,13 +4263,15 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_handle_pdata(wm8994); wm_hubs_add_analogue_controls(component); - snd_soc_add_component_controls(component, wm8994_snd_controls, - ARRAY_SIZE(wm8994_snd_controls)); + snd_soc_add_component_controls(component, wm8994_common_snd_controls, + ARRAY_SIZE(wm8994_common_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets, ARRAY_SIZE(wm8994_dapm_widgets)); switch (control->type) { case WM8994: + snd_soc_add_component_controls(component, wm8994_snd_controls, + ARRAY_SIZE(wm8994_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); if (control->revision < 4) { @@ -4284,8 +4291,10 @@ static int wm8994_component_probe(struct snd_soc_component *component) } break; case WM8958: + snd_soc_add_component_controls(component, wm8994_snd_controls, + ARRAY_SIZE(wm8994_snd_controls)); snd_soc_add_component_controls(component, wm8958_snd_controls, - ARRAY_SIZE(wm8958_snd_controls)); + ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); if (control->revision < 1) { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ae28d9907c30..9b8bb7bbe945 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1259,8 +1259,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) } if (in) { - if (in & WMFW_CTL_FLAG_READABLE) - out |= rd; + out |= rd; if (in & WMFW_CTL_FLAG_WRITEABLE) out |= wr; if (in & WMFW_CTL_FLAG_VOLATILE) @@ -3697,11 +3696,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) u32 xmalg, addr, magic; int i, ret; + alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); + if (!alg_region) { + adsp_err(dsp, "No algorithm region found\n"); + return -EINVAL; + } + buf = wm_adsp_buffer_alloc(dsp); if (!buf) return -ENOMEM; - alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); xmalg = dsp->ops->sys_config_size / sizeof(__be32); addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index a437567b8cee..4f6e58c3954a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -308,6 +308,9 @@ static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; @@ -318,10 +321,6 @@ static const struct snd_soc_dapm_route sof_map[] = { /* other jacks */ { "IN1P", NULL, "Headset Mic" }, - - /* digital mics */ - {"DMic", NULL, "SoC DMIC"}, - }; static const struct snd_soc_dapm_route speaker_map[] = { @@ -329,6 +328,11 @@ static const struct snd_soc_dapm_route speaker_map[] = { { "Spk", NULL, "Speaker" }, }; +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -342,6 +346,28 @@ static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { .name = "sof_rt5682", @@ -445,6 +471,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].name = "dmic01"; links[id].cpus = &cpus[id]; links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; if (dmic_be_num > 1) { /* set up 2 BE links at most */ links[id + 1].name = "dmic16k"; @@ -576,6 +603,15 @@ static int sof_audio_probe(struct platform_device *pdev) /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret); + return ret; + } + ret = clk_prepare_enable(ctx->mclk); if (ret < 0) { dev_err(&pdev->dev, @@ -621,8 +657,24 @@ static int sof_audio_probe(struct platform_device *pdev) &sof_audio_card_rt5682); } +static int sof_rt5682_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_component *component = NULL; + + for_each_card_components(card, component) { + if (!strcmp(component->name, rt5682_component[0].name)) { + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + static struct platform_driver sof_audio = { .probe = sof_audio_probe, + .remove = sof_rt5682_remove, .driver = { .name = "sof_rt5682", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af2d5a6124c8..61c984f10d8e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -677,7 +677,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - return ret; + goto err_suspend; } return 0; diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index c213913eb984..fd8c6642fb0d 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -5,6 +5,7 @@ // Author: Claude <claude@insginal.co.kr> #include <linux/module.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/clk.h> @@ -74,6 +75,17 @@ static struct snd_soc_card arndale_rt5631 = { .num_links = ARRAY_SIZE(arndale_rt5631_dai), }; +static void arndale_put_of_nodes(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + int i; + + for_each_card_prelinks(card, i, dai_link) { + of_node_put(dai_link->cpus->of_node); + of_node_put(dai_link->codecs->of_node); + } +} + static int arndale_audio_probe(struct platform_device *pdev) { int n, ret; @@ -103,18 +115,31 @@ static int arndale_audio_probe(struct platform_device *pdev) if (!arndale_rt5631_dai[0].codecs->of_node) { dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto err_put_of_nodes; } } ret = devm_snd_soc_register_card(card->dev, card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + goto err_put_of_nodes; + } + return 0; - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); - +err_put_of_nodes: + arndale_put_of_nodes(card); return ret; } +static int arndale_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + arndale_put_of_nodes(card); + return 0; +} + static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { { .compatible = "samsung,arndale-rt5631", }, { .compatible = "samsung,arndale-alc5631", }, @@ -129,6 +154,7 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), }, .probe = arndale_audio_probe, + .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bda5b958d0dc..e9596c2096cd 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -761,6 +761,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set format */ + rdai->bit_clk_inv = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..b600d3eaaf5c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1070,7 +1070,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - snd_soc_dai_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; @@ -1097,7 +1097,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; @@ -1146,6 +1146,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; + char *name; /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1171,9 +1172,15 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "<-" : "->", be->dai_link->name); #ifdef CONFIG_DEBUG_FS - dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name, - fe->debugfs_dpcm_root); - debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state); + name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name, + stream ? "capture" : "playback"); + if (name) { + dpcm->debugfs_state = debugfs_create_dir(name, + fe->debugfs_dpcm_root); + debugfs_create_u32("state", 0644, dpcm->debugfs_state, + &dpcm->state); + kfree(name); + } #endif return 1; } diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index aa9a1fca46fa..0fd032914a31 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1582,7 +1582,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, /* map user to kernel widget ID */ template.id = get_widget_id(le32_to_cpu(w->id)); - if (template.id < 0) + if ((int)template.id < 0) return template.id; /* strings are allocated here, but used and freed by the widget */ diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index a4983f90ff5b..2b8711eda362 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -60,13 +60,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = - mixer_to_ipc(ucontrol->value.integer.value[i], + value = mixer_to_ipc(ucontrol->value.integer.value[i], scontrol->volume_table, sm->max + 1); + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -76,8 +79,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, SOF_CTRL_TYPE_VALUE_CHAN_GET, SOF_CTRL_CMD_VOLUME, true); - - return 0; + return change; } int snd_sof_switch_get(struct snd_kcontrol *kcontrol, @@ -105,11 +107,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.integer.value[i]; + value = ucontrol->value.integer.value[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -120,7 +126,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_SWITCH, true); - return 0; + return change; } int snd_sof_enum_get(struct snd_kcontrol *kcontrol, @@ -148,11 +154,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.enumerated.item[i]; + value = ucontrol->value.enumerated.item[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of enum updates */ @@ -163,7 +173,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_ENUM, true); - return 0; + return change; } int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 806dfa0e5eae..05f4aed13af9 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -273,6 +273,16 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC Say Y if you want to enable HDAudio codecs with SOF. If unsure select "N". +config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1 + bool "SOF enable DMI Link L1" + help + This option enables DMI L1 for both playback and capture + and disables known workarounds for specific HDaudio platforms. + Only use to look into power optimizations on platforms not + affected by DMI L1 issues. This option is not recommended. + Say Y if you want to enable DMI Link L1 + If unsure, select "N". + endif ## SND_SOC_SOF_HDA_COMMON config SND_SOC_SOF_HDA_LINK_BASELINE diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index e282179263e8..80e2826fb447 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -37,6 +37,7 @@ #define MBOX_SIZE 0x1000 #define MBOX_DUMP_SIZE 0x30 #define EXCEPT_OFFSET 0x800 +#define EXCEPT_MAX_HDR_SIZE 0x400 /* DSP peripherals */ #define DMAC0_OFFSET 0xFE000 @@ -228,6 +229,11 @@ static void bdw_get_registers(struct snd_sof_dev *sdev, /* note: variable AR register array is not read */ /* then get panic info */ + if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { + dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", + xoops->arch_hdr.totalsize); + return; + } offset += xoops->arch_hdr.totalsize; sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info)); @@ -451,6 +457,7 @@ static int bdw_probe(struct snd_sof_dev *sdev) /* TODO: add offsets */ sdev->mmio_bar = BDW_DSP_BAR; sdev->mailbox_bar = BDW_DSP_BAR; + sdev->dsp_oops_offset = MBOX_OFFSET; /* PCI base */ mmio = platform_get_resource(pdev, IORESOURCE_MEM, diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 5e7a6aaa627a..a1e514f71739 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -28,6 +28,7 @@ #define MBOX_OFFSET 0x144000 #define MBOX_SIZE 0x1000 #define EXCEPT_OFFSET 0x800 +#define EXCEPT_MAX_HDR_SIZE 0x400 /* DSP peripherals */ #define DMAC0_OFFSET 0x098000 @@ -126,6 +127,11 @@ static void byt_get_registers(struct snd_sof_dev *sdev, /* note: variable AR register array is not read */ /* then get panic info */ + if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { + dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", + xoops->arch_hdr.totalsize); + return; + } offset += xoops->arch_hdr.totalsize; sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info)); diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index bc41028a7a01..df1909e1d950 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -139,20 +139,16 @@ void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable) */ int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable) { -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - struct hdac_bus *bus = sof_to_bus(sdev); -#endif u32 val; /* enable/disable audio dsp clock gating */ val = enable ? PCI_CGCTL_ADSPDCGE : 0; snd_sof_pci_update_bits(sdev, PCI_CGCTL, PCI_CGCTL_ADSPDCGE, val); -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - /* enable/disable L1 support */ - val = enable ? SOF_HDA_VS_EM2_L1SEN : 0; - snd_hdac_chip_updatel(bus, VS_EM2, SOF_HDA_VS_EM2_L1SEN, val); -#endif + /* enable/disable DMI Link L1 support */ + val = enable ? HDA_VS_INTEL_EM2_L1SEN : 0; + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, val); /* enable/disable audio dsp power gating */ val = enable ? 0 : PCI_PGCTL_ADSPPGD; diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 6427f0b3a2f1..65c2af3fcaab 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -44,6 +44,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, return -ENODEV; } hstream = &dsp_stream->hstream; + hstream->substream = NULL; /* allocate DMA buffer */ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index ad8d41f22e92..2c7447188402 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -185,6 +185,17 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction) direction == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + /* + * Disable DMI Link L1 entry when capture stream is opened. + * Workaround to address a known issue with host DMA that results + * in xruns during pause/release in capture scenarios. + */ + if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (stream && direction == SNDRV_PCM_STREAM_CAPTURE) + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, 0); + return stream; } @@ -193,23 +204,43 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) { struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *s; + bool active_capture_stream = false; + bool found = false; spin_lock_irq(&bus->reg_lock); - /* find used stream */ + /* + * close stream matching the stream tag + * and check if there are any open capture streams. + */ list_for_each_entry(s, &bus->stream_list, list) { - if (s->direction == direction && - s->opened && s->stream_tag == stream_tag) { + if (!s->opened) + continue; + + if (s->direction == direction && s->stream_tag == stream_tag) { s->opened = false; - spin_unlock_irq(&bus->reg_lock); - return 0; + found = true; + } else if (s->direction == SNDRV_PCM_STREAM_CAPTURE) { + active_capture_stream = true; } } spin_unlock_irq(&bus->reg_lock); - dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag); - return -ENODEV; + /* Enable DMI L1 entry if there are no capture streams open */ + if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!active_capture_stream) + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, + HDA_VS_INTEL_EM2_L1SEN); + + if (!found) { + dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag); + return -ENODEV; + } + + return 0; } int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c72e9a09eee1..06e84679087b 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -35,6 +35,8 @@ #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) #define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8) +#define EXCEPT_MAX_HDR_SIZE 0x400 + /* * Debug */ @@ -131,6 +133,11 @@ static void hda_dsp_get_registers(struct snd_sof_dev *sdev, /* note: variable AR register array is not read */ /* then get panic info */ + if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { + dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", + xoops->arch_hdr.totalsize); + return; + } offset += xoops->arch_hdr.totalsize; sof_block_read(sdev, sdev->mmio_bar, offset, panic_info, sizeof(*panic_info)); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 5591841a1b6f..23e430d3e056 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -39,7 +39,6 @@ #define SOF_HDA_WAKESTS 0x0E #define SOF_HDA_WAKESTS_INT_MASK ((1 << 8) - 1) #define SOF_HDA_RIRBSTS 0x5d -#define SOF_HDA_VS_EM2_L1SEN BIT(13) /* SOF_HDA_GCTL register bist */ #define SOF_HDA_GCTL_RESET BIT(0) @@ -228,6 +227,10 @@ #define HDA_DSP_REG_HIPCIE (HDA_DSP_IPC_BASE + 0x0C) #define HDA_DSP_REG_HIPCCTL (HDA_DSP_IPC_BASE + 0x10) +/* Intel Vendor Specific Registers */ +#define HDA_VS_INTEL_EM2 0x1030 +#define HDA_VS_INTEL_EM2_L1SEN BIT(13) + /* HIPCI */ #define HDA_DSP_REG_HIPCI_BUSY BIT(31) #define HDA_DSP_REG_HIPCI_MSG_MASK 0x7FFFFFFF diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index d7f32745fefe..9a9a381a908d 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -546,10 +546,10 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev) msecs_to_jiffies(sdev->boot_timeout)); if (ret == 0) { dev_err(sdev->dev, "error: firmware boot failure\n"); - /* after this point FW_READY msg should be ignored */ - sdev->boot_complete = true; snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX | SOF_DBG_TEXT | SOF_DBG_PCI); + /* after this point FW_READY msg should be ignored */ + sdev->boot_complete = true; return -EIO; } diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index e3f6a6dc0f36..2b876d497447 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -244,7 +244,7 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_sof_pcm *spcm; - int ret; + int ret, err = 0; /* nothing to do for BE */ if (rtd->dai_link->no_pcm) @@ -254,26 +254,26 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream) if (!spcm) return -EINVAL; - if (!spcm->prepared[substream->stream]) - return 0; - dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); + if (spcm->prepared[substream->stream]) { + ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); + if (ret < 0) + err = ret; + } snd_pcm_lib_free_pages(substream); cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work); - if (ret < 0) - return ret; - ret = snd_sof_pcm_platform_hw_free(sdev, substream); - if (ret < 0) + if (ret < 0) { dev_err(sdev->dev, "error: platform hw free failed\n"); + err = ret; + } - return ret; + return err; } static int sof_pcm_prepare(struct snd_pcm_substream *substream) @@ -323,6 +323,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct sof_ipc_stream stream; struct sof_ipc_reply reply; bool reset_hw_params = false; + bool ipc_first = false; int ret; /* nothing to do for BE */ @@ -343,6 +344,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_PAUSE_PUSH: stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE; + ipc_first = true; break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE; @@ -363,6 +365,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; + ipc_first = true; reset_hw_params = true; break; default: @@ -370,12 +373,22 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return -EINVAL; } - snd_sof_pcm_platform_trigger(sdev, substream, cmd); + /* + * DMA and IPC sequence is different for start and stop. Need to send + * STOP IPC before stop DMA + */ + if (!ipc_first) + snd_sof_pcm_platform_trigger(sdev, substream, cmd); /* send IPC to the DSP */ ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream, sizeof(stream), &reply, sizeof(reply)); + /* need to STOP DMA even if STOP IPC failed */ + if (ipc_first) + snd_sof_pcm_platform_trigger(sdev, substream, cmd); + + /* free PCM if reset_hw_params is set and the STOP IPC is successful */ if (!ret && reset_hw_params) ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index fc85efbad378..0aabb3190ddc 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -920,7 +920,9 @@ static void sof_parse_word_tokens(struct snd_soc_component *scomp, for (j = 0; j < count; j++) { /* match token type */ if (!(tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_WORD || - tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT)) + tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT || + tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BYTE || + tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BOOL)) continue; /* match token id */ diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d7501f88aaa6..a4060813bc74 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -505,10 +505,20 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - (unsigned int)~SAI_XCR1_NODIV); + freq ? 0 : SAI_XCR1_NODIV); if (ret < 0) return ret; + /* Assume shutdown if requested frequency is 0Hz */ + if (!freq) { + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } + return 0; + } + /* If master clock is used, set parent clock now */ ret = stm32_sai_set_parent_clock(sai, freq); if (ret) @@ -1093,15 +1103,6 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - SAI_XCR1_NODIV); - - /* Release mclk rate only if rate was actually set */ - if (sai->mclk_rate) { - clk_rate_exclusive_put(sai->sai_mclk); - sai->mclk_rate = 0; - } - clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 33cd26763c0e..ff5ab24f3bd1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; + case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ + /* BOSS Katana amplifiers do not need quirks */ + return 0; } if (attr == USB_ENDPOINT_SYNC_ASYNC && diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fbfde996fee7..0bbe1201a6ac 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1657,6 +1657,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x23ba: /* Playback Designs */ case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ + case 0x292b: /* Gustard/Ess based devices */ case 0x2ab6: /* T+A devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ diff --git a/sound/usb/validate.c b/sound/usb/validate.c index 3c8f73a0eb12..a5e584b60dcd 100644 --- a/sound/usb/validate.c +++ b/sound/usb/validate.c @@ -75,7 +75,7 @@ static bool validate_processing_unit(const void *p, if (d->bLength < sizeof(*d)) return false; - len = d->bLength < sizeof(*d) + d->bNrInPins; + len = sizeof(*d) + d->bNrInPins; if (d->bLength < len) return false; switch (v->protocol) { |