From 2fd3f170e5ba04acc60ee94b2c76ee166727c734 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 27 Nov 2013 14:47:26 +0800 Subject: [PATCH 1/6] 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 c205bb1747fd..1f2717f817a0 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), {} From a93455e1c301ce2a4cae11682359fd2067a8fd30 Mon Sep 17 00:00:00 2001 From: Thomas Pugliese Date: Tue, 26 Nov 2013 13:58:15 -0600 Subject: [PATCH 2/6] 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 b9ba0fcc45df..83aabea259d7 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; From ced4cefc75fdb8be95eaee325ad0f6b2fc0a484b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Nov 2013 08:33:45 +0100 Subject: [PATCH 3/6] 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 3067ed4fe3b2..3bd89ab0d9af 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, From 16c0cefe8951b2c4b824fd06011ac1b359b1ab3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Nov 2013 08:44:26 +0100 Subject: [PATCH 4/6] 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 3bd89ab0d9af..faefff1d3e81 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); From 2ded3e5b61d61d0bc90bebb8004db6184c7db6eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Nov 2013 11:05:28 +0100 Subject: [PATCH 5/6] 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 faefff1d3e81..c4671d00babd 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; From 2cede30379f3fed3f4c83010cf08be6afcc811b9 Mon Sep 17 00:00:00 2001 From: Oleksij Rempel Date: Wed, 27 Nov 2013 17:12:03 +0100 Subject: [PATCH 6/6] 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 c4ad9d198068..3a9a1b0c1dbe 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),