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authorLinus Torvalds <torvalds@linux-foundation.org>2013-11-29 09:36:42 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2013-11-29 09:36:42 -0800
commitb8495995dd8ad425ec1b78f7182586d5a004d8ec (patch)
tree251e5032e6c1b8bbb24af2a8b6a4dd0eabad8eb2
parentb01537bfbc832a09162e7189f63251a8785e2112 (diff)
parenteb9ca3ab2194ad9a6c52da0e8bf1b3f1ff9cd6f4 (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.c15
-rw-r--r--sound/pci/hda/hda_codec.h1
-rw-r--r--sound/pci/hda/hda_generic.c79
-rw-r--r--sound/pci/hda/hda_intel.c3
-rw-r--r--sound/pci/hda/patch_conexant.c23
-rw-r--r--sound/pci/hda/patch_realtek.c38
-rw-r--r--sound/pci/hda/patch_sigmatel.c3
-rw-r--r--sound/usb/endpoint.c16
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;