From 82755abfe872c9f2b9a108641f98d08435762dc4 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 21 Nov 2013 14:21:06 +0900 Subject: ALSA: firewire-lib: fix wrong value for FDF field as an empty packet This commit fix out of specification about the value of FDF field in out packet with 'no data'. This affects blocking mode. According to IEC 61883-6, there is two way to generate AMDTP packets include no data in blocking mode. Way 1. an empty packet defined in IEC 61883-1 - Size of packet is 2 quadlets. - The value of FDF is sfc. - The packet includes only CIP headers Way 2. a special non-empty packet defined in IEC 61883-6 - Size of packet is following to blocking mode - The value of FDF is 0xff. This value is 'NO-DATA'. This means 'The receiver' must ignore all the data in a CIP with this FDF code'. - The packet includes dummy data. But current implementation is a combination of them. - Size of packet is 2 (way 1) - FDF = 0xff (way 2) This causes BeBoB chipset cannot sound. This patch applies Way 1. Signed-off-by: Takashi Sakamoto Cc: Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 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) | -- cgit v1.2.3-18-g5258 From ba4c4d0a9021ab034554d532a98133d668b87599 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 26 Nov 2013 15:17:50 +0800 Subject: ALSA: hda/realtek - Add support of ALC231 codec It's compatible with ALC269. Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e42059f10a..60973fd6eb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5175,6 +5175,7 @@ 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 }, -- cgit v1.2.3-18-g5258 From 9ad54547cf6f4410eba83bb95dfd2a0966718d6d Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 26 Nov 2013 15:41:40 +0800 Subject: ALSA: hda/realtek - Set pcbeep amp for ALC668 Set the missing pcbeep default amp for ALC668. Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 60973fd6eb8..c4ad9d19806 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5118,6 +5118,7 @@ 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: -- cgit v1.2.3-18-g5258 From 873ce8ad502cce3ba9295890d52afcce385d4107 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Nov 2013 11:58:40 +0100 Subject: ALSA: hda - Drop bus->avoid_link_reset flag Use bus->power_keep_link_on instead. The controller shouldn't go to D3 when the link isn't reset, so essentially avoiding the link reset means avoiding the runtime PM. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_intel.c | 3 +-- sound/pci/hda/patch_sigmatel.c | 3 ++- 3 files changed, 3 insertions(+), 4 deletions(-) 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_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_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; } } -- cgit v1.2.3-18-g5258 From 2fd3f170e5ba04acc60ee94b2c76ee166727c734 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 27 Nov 2013 14:47:26 +0800 Subject: ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec Most Thinkpad Edge series laptops use conexant codec, so far although the codecs have different minor Vendor Id and minor Subsystem Id, they all belong to the cxt5066 family, this change can make the mute/mic-mute LEDs support more generic among cxt_5066 family. This design refers to the similar solution for the realtek codec ALC269 family in the patch_realtek.c. Cc: Alex Hung Signed-off-by: Hui Wang Acked-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) 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 +#include 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), {} -- cgit v1.2.3-18-g5258 From a93455e1c301ce2a4cae11682359fd2067a8fd30 Mon Sep 17 00:00:00 2001 From: Thomas Pugliese Date: Tue, 26 Nov 2013 13:58:15 -0600 Subject: ALSA: usb: use multiple packets per urb for Wireless USB inbound audio For Wireless USB audio devices, use multiple isoc packets per URB for inbound endpoints with a datainterval < 5. This allows the WUSB host controller to take advantage of bursting to service endpoints whose logical polling interval is less than the 4ms minimum polling interval limit in WUSB. Signed-off-by: Thomas Pugliese Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) 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; -- cgit v1.2.3-18-g5258 From ced4cefc75fdb8be95eaee325ad0f6b2fc0a484b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Nov 2013 08:33:45 +0100 Subject: ALSA: hda - Create Headhpone Mic Jack Mode when really needed When a headphone jack is configurable as input, the generic parser tries to make it retaskable as Headphone Mic. The switching can be done smoothly if Capture Source control exists (i.e. there is another input source). Or when user explicitly enables the creation of jack mode controls, "Headhpone Mic Jack Mode" will be created accordingly. However, if the headphone mic is the only input source, we have to create "Headphone Mic Jack Mode" control because there is no capture source selection. Otherwise, the generic parser assumes that the input is constantly enabled, thus the headphone is permanently set as input. This situation happens on the old MacBook Airs where no input is supported properly, for example. This patch fixes the problem: now "Headphone Mic Jack Mode" is created when such an input selection isn't possible. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3067ed4fe3b..3bd89ab0d9a 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]; @@ -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) @@ -4383,6 +4376,17 @@ 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, -- cgit v1.2.3-18-g5258 From 16c0cefe8951b2c4b824fd06011ac1b359b1ab3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Nov 2013 08:44:26 +0100 Subject: ALSA: hda - Fix hp-mic mode without VREF bits When the hp mic pin has no VREF bits, the driver forgot to set PIN_IN bit. Spotted during debugging old MacBook Airs. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3bd89ab0d9a..faefff1d3e8 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2760,7 +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); -- cgit v1.2.3-18-g5258 From 2ded3e5b61d61d0bc90bebb8004db6184c7db6eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Nov 2013 11:05:28 +0100 Subject: ALSA: hda - Check leaf nodes to find aamix amps The current generic parser assumes blindly that the volume and mute amps are found in the aamix node itself. But on some codecs, typically Analog Devices ones, the aamix amps are separately implemented in each leaf node of the aamix node, and the current driver can't establish the correct amp controls. This is a regression compared with the previous static quirks. This patch extends the search for the amps to the leaf nodes for allowing the aamix controls again on such codecs. In this implementation, I didn't code to loop through the whole paths, since usually one depth should suffice, and we can't search too deeply, as it may result in the conflicting control assignments. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 57 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 45 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index faefff1d3e8..c4671d00bab 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2808,6 +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, @@ -2815,12 +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) @@ -2829,20 +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; -- cgit v1.2.3-18-g5258 From 2cede30379f3fed3f4c83010cf08be6afcc811b9 Mon Sep 17 00:00:00 2001 From: Oleksij Rempel Date: Wed, 27 Nov 2013 17:12:03 +0100 Subject: ALSA: hda - limit mic boost on Asus UX31[A,E] This both devices need limit for internal dmic. [cosmetic change; renamed fixup name by tiwai] Signed-off-by: Oleksij Rempel Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4ad9d19806..3a9a1b0c1db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3798,6 +3798,7 @@ 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 +4076,12 @@ 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 +4196,8 @@ 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), -- cgit v1.2.3-18-g5258 From 1f0bbf03cb829162ec8e6d03c98aaaed88c6f534 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Nov 2013 15:21:21 +0100 Subject: ALSA: hda - Initialize missing bass speaker pin for ASUS AIO ET2700 Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700 AiO desktop. The channel map will be added in the next patch, so that this can be backported easily to stable kernels. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a9a1b0c1db..fd835c52618 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1782,6 +1782,7 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, + ALC887_FIXUP_ASUS_BASS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2105,6 +2106,13 @@ 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 */ + {} + }, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2138,6 +2146,7 @@ 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), -- cgit v1.2.3-18-g5258 From eb9ca3ab2194ad9a6c52da0e8bf1b3f1ff9cd6f4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Nov 2013 15:24:34 +0100 Subject: ALSA: hda - Add LFE chmap to ASUS ET2700 As the previous commit 1f0bbf03cb82 added the pin config for the bass speaker, this patch adds the corresponding LFE-only channel map on ASUS ET2700. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fd835c52618..c770bdba653 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1783,6 +1783,7 @@ enum { 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, @@ -1916,6 +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, @@ -2112,6 +2116,12 @@ static const struct hda_fixup alc882_fixups[] = { {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, }, }; @@ -4731,7 +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) { @@ -4939,7 +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 }, @@ -4952,7 +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, }, -- cgit v1.2.3-18-g5258