diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2013-11-29 09:36:42 -0800 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2013-11-29 09:36:42 -0800 |
commit | b8495995dd8ad425ec1b78f7182586d5a004d8ec (patch) | |
tree | 251e5032e6c1b8bbb24af2a8b6a4dd0eabad8eb2 | |
parent | b01537bfbc832a09162e7189f63251a8785e2112 (diff) | |
parent | eb9ca3ab2194ad9a6c52da0e8bf1b3f1ff9cd6f4 (diff) |
Merge tag 'sound-3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Quite a few HD-Audio fixes, a WUSB audio fix and a fix for FireWire
audio. The HD-audio part contains a couple of fixes for the generic
parser, and these are the only intrusive fixes. The rest are mostly
device-specific fixes"
* tag 'sound-3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Add LFE chmap to ASUS ET2700
ALSA: hda - Initialize missing bass speaker pin for ASUS AIO ET2700
ALSA: hda - limit mic boost on Asus UX31[A,E]
ALSA: hda - Check leaf nodes to find aamix amps
ALSA: hda - Fix hp-mic mode without VREF bits
ALSA: hda - Create Headhpone Mic Jack Mode when really needed
ALSA: usb: use multiple packets per urb for Wireless USB inbound audio
ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec
ALSA: hda - Drop bus->avoid_link_reset flag
ALSA: hda/realtek - Set pcbeep amp for ALC668
ALSA: hda/realtek - Add support of ALC231 codec
ALSA: firewire-lib: fix wrong value for FDF field as an empty packet
-rw-r--r-- | sound/firewire/amdtp.c | 15 | ||||
-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 | 38 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 3 | ||||
-rw-r--r-- | sound/usb/endpoint.c | 16 |
8 files changed, 138 insertions, 40 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/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 5e42059f10a..c770bdba653 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1782,6 +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, @@ -1915,6 +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, @@ -2105,6 +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[] = { @@ -2138,6 +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), @@ -3798,6 +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, @@ -4075,6 +4095,12 @@ static const struct hda_fixup alc269_fixups[] = { .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, .v.func = alc269_fixup_limit_int_mic_boost, @@ -4189,8 +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), @@ -4715,7 +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) { @@ -4923,7 +4949,7 @@ 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 }, @@ -4936,7 +4962,7 @@ static const struct hda_fixup alc662_fixups[] = { }, [ALC662_FIXUP_BASS_1A_CHMAP] = { .type = HDA_FIXUP_FUNC, - .v.func = alc662_fixup_bass_chmap, + .v.func = alc_fixup_bass_chmap, .chained = true, .chain_id = ALC662_FIXUP_BASS_1A, }, @@ -5118,6 +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: @@ -5175,6 +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/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; |