diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/firewire/amdtp.c | 15 | ||||
-rw-r--r-- | sound/firewire/amdtp.h | 1 | ||||
-rw-r--r-- | sound/pci/hda/Kconfig | 3 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 4 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 1 | ||||
-rw-r--r-- | sound/pci/hda/hda_generic.c | 79 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 3 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 23 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 108 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/ab8500-codec.c | 66 | ||||
-rw-r--r-- | sound/soc/codecs/arizona.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm5110.c | 43 | ||||
-rw-r--r-- | sound/soc/davinci/davinci-pcm.c | 2 | ||||
-rw-r--r-- | sound/soc/sh/rcar/core.c | 13 | ||||
-rw-r--r-- | sound/soc/sh/rcar/scu.c | 2 | ||||
-rw-r--r-- | sound/usb/endpoint.c | 16 |
17 files changed, 299 insertions, 87 deletions
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index d3226892ad6..9048777228e 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -434,17 +434,14 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle) return; index = s->packet_index; + /* this module generate empty packet for 'no data' */ syt = calculate_syt(s, cycle); - if (!(s->flags & CIP_BLOCKING)) { + if (!(s->flags & CIP_BLOCKING)) data_blocks = calculate_data_blocks(s); - } else { - if (syt != 0xffff) { - data_blocks = s->syt_interval; - } else { - data_blocks = 0; - syt = 0xffffff; - } - } + else if (syt != 0xffff) + data_blocks = s->syt_interval; + else + data_blocks = 0; buffer = s->buffer.packets[index].buffer; buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 839ebf812d7..2746ecd291a 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -4,6 +4,7 @@ #include <linux/err.h> #include <linux/interrupt.h> #include <linux/mutex.h> +#include <sound/asound.h> #include "packets-buffer.h" /** diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 8de66ccd727..4cdd9ded456 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -209,8 +209,9 @@ config SND_HDA_CODEC_CA0132 config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" - depends on SND_HDA_CODEC_CA0132 && FW_LOADER + depends on SND_HDA_CODEC_CA0132 select SND_HDA_DSP_LOADER + select FW_LOADER help Say Y here to enable the DSP for Creative CA0132 for extended features like equalizer or echo cancellation. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index afb90f48867..69178c4f411 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4000,6 +4000,10 @@ static void hda_call_codec_resume(struct hda_codec *codec) * in the resume / power-save sequence */ hda_keep_power_on(codec); + if (codec->pm_down_notified) { + codec->pm_down_notified = 0; + hda_call_pm_notify(codec->bus, true); + } hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); hda_exec_init_verbs(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 77db69480c1..7aa9870040c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -698,7 +698,6 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ - unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */ int primary_dig_out_type; /* primary digital out PCM type */ }; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3067ed4fe3b..c4671d00bab 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2506,12 +2506,8 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins, for (i = 0; i < num_pins; i++) { hda_nid_t pin = pins[i]; - if (pin == spec->hp_mic_pin) { - int ret = create_hp_mic_jack_mode(codec, pin); - if (ret < 0) - return ret; + if (pin == spec->hp_mic_pin) continue; - } if (get_out_jack_num_items(codec, pin) > 1) { struct snd_kcontrol_new *knew; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -2764,7 +2760,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol, val &= ~(AC_PINCTL_VREFEN | PIN_HP); val |= get_vref_idx(vref_caps, idx) | PIN_IN; } else - val = snd_hda_get_default_vref(codec, nid); + val = snd_hda_get_default_vref(codec, nid) | PIN_IN; } snd_hda_set_pin_ctl_cache(codec, nid, val); call_hp_automute(codec, NULL); @@ -2784,9 +2780,6 @@ static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin) struct hda_gen_spec *spec = codec->spec; struct snd_kcontrol_new *knew; - if (get_out_jack_num_items(codec, pin) <= 1 && - get_in_jack_num_items(codec, pin) <= 1) - return 0; /* no need */ knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode", &hp_mic_jack_mode_enum); if (!knew) @@ -2815,6 +2808,42 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx) return 0; } +/* return true if either a volume or a mute amp is found for the given + * aamix path; the amp has to be either in the mixer node or its direct leaf + */ +static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, + hda_nid_t pin, unsigned int *mix_val, + unsigned int *mute_val) +{ + int idx, num_conns; + const hda_nid_t *list; + hda_nid_t nid; + + idx = snd_hda_get_conn_index(codec, mix_nid, pin, true); + if (idx < 0) + return false; + + *mix_val = *mute_val = 0; + if (nid_has_volume(codec, mix_nid, HDA_INPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (nid_has_mute(codec, mix_nid, HDA_INPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (*mix_val && *mute_val) + return true; + + /* check leaf node */ + num_conns = snd_hda_get_conn_list(codec, mix_nid, &list); + if (num_conns < idx) + return false; + nid = list[idx]; + if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + + return *mix_val || *mute_val; +} + /* create input playback/capture controls for the given pin */ static int new_analog_input(struct hda_codec *codec, int input_idx, hda_nid_t pin, const char *ctlname, int ctlidx, @@ -2822,12 +2851,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, { struct hda_gen_spec *spec = codec->spec; struct nid_path *path; - unsigned int val; + unsigned int mix_val, mute_val; int err, idx; - if (!nid_has_volume(codec, mix_nid, HDA_INPUT) && - !nid_has_mute(codec, mix_nid, HDA_INPUT)) - return 0; /* no need for analog loopback */ + if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val)) + return 0; path = snd_hda_add_new_path(codec, pin, mix_nid, 0); if (!path) @@ -2836,20 +2864,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path); idx = path->idx[path->depth - 1]; - if (nid_has_volume(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val); + if (mix_val) { + err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val); if (err < 0) return err; - path->ctls[NID_PATH_VOL_CTL] = val; + path->ctls[NID_PATH_VOL_CTL] = mix_val; } - if (nid_has_mute(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val); + if (mute_val) { + err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val); if (err < 0) return err; - path->ctls[NID_PATH_MUTE_CTL] = val; + path->ctls[NID_PATH_MUTE_CTL] = mute_val; } path->active = true; @@ -4383,6 +4409,17 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* create "Headphone Mic Jack Mode" if no input selection is + * available (or user specifies add_jack_modes hint) + */ + if (spec->hp_mic_pin && + (spec->auto_mic || spec->input_mux.num_items == 1 || + spec->add_jack_modes)) { + err = create_hp_mic_jack_mode(codec, spec->hp_mic_pin); + if (err < 0) + return err; + } + if (spec->add_jack_modes) { if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = create_out_jack_modes(codec, cfg->line_outs, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7a09404579a..c6d230193da 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2994,8 +2994,7 @@ static int azx_runtime_suspend(struct device *dev) STATESTS_INT_MASK); azx_stop_chip(chip); - if (!chip->bus->avoid_link_reset) - azx_enter_link_reset(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c205bb1747f..1f2717f817a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3244,9 +3244,29 @@ enum { #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> +#include <acpi/acpi.h> static int (*led_set_func)(int, bool); +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + static void update_tpacpi_mute_led(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -3279,6 +3299,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3494,6 +3516,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 04d1e6be600..c770bdba653 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1512,6 +1512,7 @@ enum { ALC260_FIXUP_KN1, ALC260_FIXUP_FSC_S7020, ALC260_FIXUP_FSC_S7020_JWSE, + ALC260_FIXUP_VAIO_PINS, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -1652,6 +1653,24 @@ static const struct hda_fixup alc260_fixups[] = { .chained = true, .chain_id = ALC260_FIXUP_FSC_S7020, }, + [ALC260_FIXUP_VAIO_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* Pin configs are missing completely on some VAIOs */ + { 0x0f, 0x01211020 }, + { 0x10, 0x0001003f }, + { 0x11, 0x411111f0 }, + { 0x12, 0x01a15930 }, + { 0x13, 0x411111f0 }, + { 0x14, 0x411111f0 }, + { 0x15, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x411111f0 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { } + } + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -1660,6 +1679,8 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), + SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_FIXUP_VAIO_PINS), + SND_PCI_QUIRK(0x104d, 0x81e2, "Sony VAIO TX", ALC260_FIXUP_HP_PIN_0F), SND_PCI_QUIRK(0x10cf, 0x1326, "FSC LifeBook S7020", ALC260_FIXUP_FSC_S7020), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1), @@ -1761,6 +1782,8 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, + ALC887_FIXUP_ASUS_BASS, + ALC887_FIXUP_BASS_CHMAP, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1894,6 +1917,9 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, } } +static void alc_fixup_bass_chmap(struct hda_codec *codec, + const struct hda_fixup *fix, int action); + static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = HDA_FIXUP_PINS, @@ -2084,6 +2110,19 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc882_fixup_no_primary_hp, }, + [ALC887_FIXUP_ASUS_BASS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x16, 0x99130130}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC887_FIXUP_BASS_CHMAP, + }, + [ALC887_FIXUP_BASS_CHMAP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_bass_chmap, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2117,6 +2156,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), + SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), @@ -3393,7 +3433,7 @@ static void alc_update_headset_mode_hook(struct hda_codec *codec, static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct alc_spec *spec = codec->spec; - spec->current_headset_type = ALC_HEADSET_MODE_UNKNOWN; + spec->current_headset_type = ALC_HEADSET_TYPE_UNKNOWN; snd_hda_gen_hp_automute(codec, jack); } @@ -3652,9 +3692,29 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> +#include <acpi/acpi.h> static int (*led_set_func)(int, bool); +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + static void update_tpacpi_mute_led(void *private_data, int enabled) { if (led_set_func) @@ -3680,6 +3740,8 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3755,6 +3817,7 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -3923,6 +3986,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -4027,6 +4092,14 @@ static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_LIMIT_INT_MIC_BOOST] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, + [ALC269VB_FIXUP_ASUS_ZENBOOK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269VB_FIXUP_DMIC, }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, @@ -4070,8 +4143,6 @@ static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_thinkpad_acpi, - .chained = true, - .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, [ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -4128,6 +4199,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -4143,8 +4215,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), - SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), @@ -4173,7 +4245,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), @@ -4181,6 +4253,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 @@ -4668,7 +4741,7 @@ static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = { }; /* override the 2.1 chmap */ -static void alc662_fixup_bass_chmap(struct hda_codec *codec, +static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action) { if (action == HDA_FIXUP_ACT_BUILD) { @@ -4698,6 +4771,8 @@ enum { ALC668_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_HEADSET_MODE, ALC662_FIXUP_BASS_CHMAP, + ALC662_FIXUP_BASS_1A, + ALC662_FIXUP_BASS_1A_CHMAP, }; static const struct hda_fixup alc662_fixups[] = { @@ -4874,10 +4949,23 @@ static const struct hda_fixup alc662_fixups[] = { }, [ALC662_FIXUP_BASS_CHMAP] = { .type = HDA_FIXUP_FUNC, - .v.func = alc662_fixup_bass_chmap, + .v.func = alc_fixup_bass_chmap, .chained = true, .chain_id = ALC662_FIXUP_ASUS_MODE4 }, + [ALC662_FIXUP_BASS_1A] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x80106111}, /* bass speaker */ + {} + }, + }, + [ALC662_FIXUP_BASS_1A_CHMAP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_bass_chmap, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_1A, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -4890,8 +4978,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), @@ -5054,6 +5144,7 @@ static int patch_alc662(struct hda_codec *codec) case 0x10ec0272: case 0x10ec0663: case 0x10ec0665: + case 0x10ec0668: set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); break; case 0x10ec0273: @@ -5111,6 +5202,7 @@ static int patch_alc680(struct hda_codec *codec) */ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, + { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 }, { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d2cc0041d9d..088a5afbd1b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2094,7 +2094,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ - codec->bus->avoid_link_reset = 1; + /* resetting controller clears GPIO, so we need to keep on */ + codec->bus->power_keep_link_on = 1; } } diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 21ae8d4fdbf..1ad92cbf0b2 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -126,8 +126,6 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { - struct regmap *regmap; - /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -168,34 +166,48 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static int ab8500_codec_read_reg(void *context, unsigned int reg, - unsigned int *value) +static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, + unsigned int reg) { - struct device *dev = context; int status; + unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(dev, AB8500_AUDIO, - reg, &value8); - *value = (unsigned int)value8; + status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, + reg, &value8); + if (status < 0) { + dev_err(codec->dev, + "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + } else { + dev_dbg(codec->dev, + "%s: Read 0x%02x from register 0x%02x:0x%02x\n", + __func__, value8, (u8)AB8500_AUDIO, (u8)reg); + value = (unsigned int)value8; + } - return status; + return value; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(void *context, unsigned int reg, - unsigned int value) +static int ab8500_codec_write_reg(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) { - struct device *dev = context; + int status; - return abx500_set_register_interruptible(dev, AB8500_AUDIO, - reg, value); -} + status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, + reg, value); + if (status < 0) + dev_err(codec->dev, + "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + else + dev_dbg(codec->dev, + "%s: Wrote 0x%02x into register %02x:%02x\n", + __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); -static const struct regmap_config ab8500_codec_regmap = { - .reg_read = ab8500_codec_read_reg, - .reg_write = ab8500_codec_write_reg, -}; + return status; +} /* * Controls - DAPM @@ -2473,13 +2485,9 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); - codec->control_data = drvdata->regmap; - if (np) { if (!pdata) pdata = devm_kzalloc(dev, @@ -2557,6 +2565,9 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, + .read = ab8500_codec_read_reg, + .write = ab8500_codec_write_reg, + .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2581,15 +2592,6 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); - drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, - &ab8500_codec_regmap); - if (IS_ERR(drvdata->regmap)) { - status = PTR_ERR(drvdata->regmap); - dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", - __func__, status); - return status; - } - dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6f05b17d196..fea991031be 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1529,6 +1529,8 @@ static void arizona_enable_fll(struct arizona_fll *fll, try_wait_for_completion(&fll->ok); regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, @@ -1546,6 +1548,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) struct arizona *arizona = fll->arizona; bool change; + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); regmap_update_bits_check(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index f2d1094424b..c3c7396a618 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -37,6 +37,47 @@ struct wm5110_priv { struct arizona_fll fll[2]; }; +static const struct reg_default wm5110_sysclk_revd_patch[] = { + { 0x3093, 0x1001 }, + { 0x30E3, 0x1301 }, + { 0x3133, 0x1201 }, + { 0x3183, 0x1501 }, + { 0x31D3, 0x1401 }, +}; + +static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 3: + patch = wm5110_sysclk_revd_patch; + patch_size = ARRAY_SIZE(wm5110_sysclk_revd_patch); + break; + default: + return 0; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + + default: + break; + } + + return 0; +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -400,7 +441,7 @@ static const struct snd_kcontrol_new wm5110_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, NULL, 0), + 0, wm5110_sysclk_ev, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index fa64cd85204..fb5d107f560 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -238,7 +238,7 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) print_buf_info(prtd->ram_channel, "i ram_channel"); pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); - if (unlikely(ch_status != DMA_COMPLETE)) + if (unlikely(ch_status != EDMA_DMA_COMPLETE)) return; if (snd_pcm_running(substream)) { diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 78c35b44fc0..b3653d37f75 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -200,9 +200,8 @@ static void rsnd_dma_do_work(struct work_struct *work) return; } + dma_async_issue_pending(dma->chan); } - - dma_async_issue_pending(dma->chan); } int rsnd_dma_available(struct rsnd_dma *dma) @@ -288,15 +287,13 @@ int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); - - if (!mod) { - dev_err(dev, "NULL mod\n"); + if (!mod) return -EIO; - } if (!list_empty(&mod->list)) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + dev_err(dev, "%s%d is not empty\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index f4453e33a84..fa8fa15860b 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -68,7 +68,7 @@ static int rsnd_scu_set_route(struct rsnd_priv *priv, return 0; id = rsnd_mod_id(mod); - if (id < 0 || id > ARRAY_SIZE(routes)) + if (id < 0 || id >= ARRAY_SIZE(routes)) return -EIO; /* diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b9ba0fcc45d..83aabea259d 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -636,8 +636,22 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, if (usb_pipein(ep->pipe) || snd_usb_endpoint_implicit_feedback_sink(ep)) { + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } /* make capture URBs <= 1 ms and smaller than a period */ - urb_packs = min(max_packs_per_urb, packs_per_ms); + urb_packs = min(max_packs_per_urb, urb_packs); while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) urb_packs >>= 1; ep->nurbs = MAX_URBS; |