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authorLinus Torvalds <torvalds@linux-foundation.org>2013-12-05 09:54:35 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2013-12-05 09:54:35 -0800
commit09759d1e13e06c2f6346b1b3285fbde2c1e794c6 (patch)
tree9ef0062747b7ef7e0cfdf54463013642a506affc
parentda1965d353c7266fce4fe6c658f2096c88a8bebc (diff)
parent0756f09c4946fe2d9ce2ebcb6f2e3c58830d22a3 (diff)
downloadlwn-09759d1e13e06c2f6346b1b3285fbde2c1e794c6.tar.gz
lwn-09759d1e13e06c2f6346b1b3285fbde2c1e794c6.zip
Merge tag 'sound-3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "A usual pattern of half ASoC and half HD-audio fixes, although HD-audio fixups have more volumes, in addition to a couple of trivial fixes. Nothing to worry much is found here. For ASoC side: a few fixes for PCM rate constraints calculations, regmap byte-order fix, the rest driver specific fixes (atmel, fsl, omap, kirkwood, wm codecs). For HD-audio: Dell headset and mono out fix, ELD update in polling mode, ALC283 Chromebook fixes, a few fixes for old AD codecs and MBA2, one regression fix" * tag 'sound-3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits) ALSA: hda - Fix silent output on MacBook Air 2,1 ALSA: hda - Fix missing ELD info when using jackpoll_ms parameter ALSA: hda/realtek - remove hp_automute_hook from alc283_fixup_chromebook ASoC: wm8731: fix dsp mode configuration ALSA: hda/realtek - Independent of model for HP ALSA: hda - Fix headset mic input after muted internal mic (Dell/Realtek) ALSA: hda - Use always amps for auto-mute on AD1986A codec ALSA: hda/analog - Handle inverted EAPD properly in vmaster hook ALSA: hda - Another fixup for ASUS laptop with ALC660 codec ALSA: atmel: Fix possible array overflow ALSA: hda - Fix complete_all() timing in deferred probes ALSA: hda - Fix bad EAPD setup for HP machines with AD1984A ASoC: core: fix devres parameter in devm_snd_soc_register_card() ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare ASoC: fsl: set correct platform drvdata in pcm030_fabric_probe() ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation ALSA: hda - Fix silent output on ASUS W7J laptop ASoC: core: Use consistent byte ordering in snd_soc_bytes_get ALSA: dice: fix array limits in dice_proc_read() ...
-rw-r--r--include/sound/soc-dapm.h3
-rw-r--r--sound/atmel/abdac.c3
-rw-r--r--sound/firewire/dice.c4
-rw-r--r--sound/pci/hda/hda_intel.c9
-rw-r--r--sound/pci/hda/patch_analog.c6
-rw-r--r--sound/pci/hda/patch_hdmi.c27
-rw-r--r--sound/pci/hda/patch_realtek.c92
-rw-r--r--sound/soc/atmel/sam9x5_wm8731.c2
-rw-r--r--sound/soc/codecs/wm5110.c25
-rw-r--r--sound/soc/codecs/wm8731.c4
-rw-r--r--sound/soc/codecs/wm8990.c2
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c3
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c22
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/sh/Kconfig1
-rw-r--r--sound/soc/soc-core.c4
-rw-r--r--sound/soc/soc-devres.c4
-rw-r--r--sound/soc/soc-pcm.c18
18 files changed, 146 insertions, 87 deletions
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 2037c45adfe6..56ebdfca6273 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -104,7 +104,8 @@ struct device;
SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
.kcontrol_news = wcontrols, .num_kcontrols = 1}
#define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \
-{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, \
+{ .id = snd_soc_dapm_mux, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
.kcontrol_news = wcontrols, .num_kcontrols = 1}
#define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \
{ .id = snd_soc_dapm_virt_mux, .name = wname, \
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 872d59e35ee2..721d8fd45685 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -357,7 +357,8 @@ static int set_sample_rates(struct atmel_abdac *dac)
if (new_rate < 0)
break;
/* make sure we are below the ABDAC clock */
- if (new_rate <= clk_get_rate(dac->pclk)) {
+ if (index < MAX_NUM_RATES &&
+ new_rate <= clk_get_rate(dac->pclk)) {
dac->rates[index] = new_rate / 256;
index++;
}
diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c
index 57bcd31fcc12..c0aa64941cee 100644
--- a/sound/firewire/dice.c
+++ b/sound/firewire/dice.c
@@ -1019,7 +1019,7 @@ static void dice_proc_read(struct snd_info_entry *entry,
if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0)
return;
- quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx));
+ quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4);
for (stream = 0; stream < tx_rx_header.number; ++stream) {
if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 +
stream * tx_rx_header.size,
@@ -1045,7 +1045,7 @@ static void dice_proc_read(struct snd_info_entry *entry,
if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0)
return;
- quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx));
+ quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4);
for (stream = 0; stream < tx_rx_header.number; ++stream) {
if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 +
stream * tx_rx_header.size,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c6d230193da6..27aa14007cbd 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -3876,7 +3876,8 @@ static int azx_probe(struct pci_dev *pci,
}
dev++;
- complete_all(&chip->probe_wait);
+ if (chip->disabled)
+ complete_all(&chip->probe_wait);
return 0;
out_free:
@@ -3953,10 +3954,10 @@ static int azx_probe_continue(struct azx *chip)
if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo)
pm_runtime_put_noidle(&pci->dev);
- return 0;
-
out_free:
- chip->init_failed = 1;
+ if (err < 0)
+ chip->init_failed = 1;
+ complete_all(&chip->probe_wait);
return err;
}
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1a83559f4cbd..cac015be3325 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled)
if (!spec->eapd_nid)
return;
+ if (codec->inv_eapd)
+ enabled = !enabled;
snd_hda_codec_update_cache(codec, spec->eapd_nid, 0,
AC_VERB_SET_EAPD_BTLENABLE,
enabled ? 0x02 : 0x00);
@@ -359,6 +361,9 @@ static int patch_ad1986a(struct hda_codec *codec)
*/
spec->gen.multiout.no_share_stream = 1;
+ /* AD1986A can't manage the dynamic pin on/off smoothly */
+ spec->gen.auto_mute_via_amp = 1;
+
snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
ad1986a_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -962,6 +967,7 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
switch (action) {
case HDA_FIXUP_ACT_PRE_PROBE:
spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+ spec->gen.own_eapd_ctl = 1;
snd_hda_sequence_write_cache(codec, gpio_init_verbs);
break;
case HDA_FIXUP_ACT_PROBE:
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 08407bed093e..c4a66ef6cf6f 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
-static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack)
{
struct hdmi_spec *spec = codec->spec;
+ int pin_idx = pin_nid_to_pin_index(spec, jack->nid);
+ if (pin_idx < 0)
+ return;
+
+ if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
+ snd_hda_jack_report_sync(codec);
+}
+
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+{
int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
- int pin_nid;
- int pin_idx;
struct hda_jack_tbl *jack;
int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
if (!jack)
return;
- pin_nid = jack->nid;
jack->jack_dirty = 1;
_snd_printd(SND_PR_VERBOSE,
"HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
- codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
+ codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
- pin_idx = pin_nid_to_pin_index(spec, pin_nid);
- if (pin_idx < 0)
- return;
-
- if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
- snd_hda_jack_report_sync(codec);
+ jack_callback(codec, jack);
}
static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
@@ -2095,7 +2097,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
hda_nid_t pin_nid = per_pin->pin_nid;
hdmi_init_pin(codec, pin_nid);
- snd_hda_jack_detect_enable(codec, pin_nid, pin_nid);
+ snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid,
+ codec->jackpoll_interval > 0 ? jack_callback : NULL);
}
return 0;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c770bdba6531..c5ea483d7559 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1780,6 +1780,7 @@ enum {
ALC889_FIXUP_DAC_ROUTE,
ALC889_FIXUP_MBP_VREF,
ALC889_FIXUP_IMAC91_VREF,
+ ALC889_FIXUP_MBA21_VREF,
ALC882_FIXUP_INV_DMIC,
ALC882_FIXUP_NO_PRIMARY_HP,
ALC887_FIXUP_ASUS_BASS,
@@ -1884,17 +1885,13 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec,
}
}
-/* Set VREF on speaker pins on imac91 */
-static void alc889_fixup_imac91_vref(struct hda_codec *codec,
- const struct hda_fixup *fix, int action)
+static void alc889_fixup_mac_pins(struct hda_codec *codec,
+ const hda_nid_t *nids, int num_nids)
{
struct alc_spec *spec = codec->spec;
- static hda_nid_t nids[2] = { 0x18, 0x1a };
int i;
- if (action != HDA_FIXUP_ACT_INIT)
- return;
- for (i = 0; i < ARRAY_SIZE(nids); i++) {
+ for (i = 0; i < num_nids; i++) {
unsigned int val;
val = snd_hda_codec_get_pin_target(codec, nids[i]);
val |= AC_PINCTL_VREF_50;
@@ -1903,6 +1900,26 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec,
spec->gen.keep_vref_in_automute = 1;
}
+/* Set VREF on speaker pins on imac91 */
+static void alc889_fixup_imac91_vref(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static hda_nid_t nids[2] = { 0x18, 0x1a };
+
+ if (action == HDA_FIXUP_ACT_INIT)
+ alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
+}
+
+/* Set VREF on speaker pins on mba21 */
+static void alc889_fixup_mba21_vref(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static hda_nid_t nids[2] = { 0x18, 0x19 };
+
+ if (action == HDA_FIXUP_ACT_INIT)
+ alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
+}
+
/* Don't take HP output as primary
* Strangely, the speaker output doesn't work on Vaio Z and some Vaio
* all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05
@@ -2102,6 +2119,12 @@ static const struct hda_fixup alc882_fixups[] = {
.chained = true,
.chain_id = ALC882_FIXUP_GPIO1,
},
+ [ALC889_FIXUP_MBA21_VREF] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc889_fixup_mba21_vref,
+ .chained = true,
+ .chain_id = ALC889_FIXUP_MBP_VREF,
+ },
[ALC882_FIXUP_INV_DMIC] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_inv_dmic_0x12,
@@ -2172,7 +2195,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF),
- SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF),
+ SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF),
SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO),
@@ -3287,6 +3310,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
alc_write_coef_idx(codec, 0x18, 0x7388);
break;
case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x11, 0x0001);
alc_write_coef_idx(codec, 0x15, 0x0d60);
alc_write_coef_idx(codec, 0xc3, 0x0000);
break;
@@ -3315,6 +3339,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
alc_write_coef_idx(codec, 0x18, 0x7388);
break;
case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x11, 0x0001);
alc_write_coef_idx(codec, 0x15, 0x0d50);
alc_write_coef_idx(codec, 0xc3, 0x0000);
break;
@@ -3600,11 +3625,6 @@ static void alc283_hp_automute_hook(struct hda_codec *codec,
vref);
}
-static void alc283_chromebook_caps(struct hda_codec *codec)
-{
- snd_hda_override_wcaps(codec, 0x03, 0);
-}
-
static void alc283_fixup_chromebook(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -3613,9 +3633,26 @@ static void alc283_fixup_chromebook(struct hda_codec *codec,
switch (action) {
case HDA_FIXUP_ACT_PRE_PROBE:
- alc283_chromebook_caps(codec);
+ snd_hda_override_wcaps(codec, 0x03, 0);
/* Disable AA-loopback as it causes white noise */
spec->gen.mixer_nid = 0;
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ /* Enable Line1 input control by verb */
+ val = alc_read_coef_idx(codec, 0x1a);
+ alc_write_coef_idx(codec, 0x1a, val | (1 << 4));
+ break;
+ }
+}
+
+static void alc283_fixup_sense_combo_jack(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ int val;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
spec->gen.hp_automute_hook = alc283_hp_automute_hook;
break;
case HDA_FIXUP_ACT_INIT:
@@ -3623,9 +3660,6 @@ static void alc283_fixup_chromebook(struct hda_codec *codec,
/* Set to manual mode */
val = alc_read_coef_idx(codec, 0x06);
alc_write_coef_idx(codec, 0x06, val & ~0x000c);
- /* Enable Line1 input control by verb */
- val = alc_read_coef_idx(codec, 0x1a);
- alc_write_coef_idx(codec, 0x1a, val | (1 << 4));
break;
}
}
@@ -3821,6 +3855,7 @@ enum {
ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED,
ALC269VB_FIXUP_ORDISSIMO_EVE2,
ALC283_FIXUP_CHROME_BOOK,
+ ALC283_FIXUP_SENSE_COMBO_JACK,
ALC282_FIXUP_ASUS_TX300,
ALC283_FIXUP_INT_MIC,
ALC290_FIXUP_MONO_SPEAKERS,
@@ -4120,6 +4155,12 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc283_fixup_chromebook,
},
+ [ALC283_FIXUP_SENSE_COMBO_JACK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc283_fixup_sense_combo_jack,
+ .chained = true,
+ .chain_id = ALC283_FIXUP_CHROME_BOOK,
+ },
[ALC282_FIXUP_ASUS_TX300] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc282_fixup_asus_tx300,
@@ -4202,6 +4243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS),
SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
@@ -4210,7 +4252,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED),
- SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -4318,6 +4359,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
{.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
{.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"},
+ {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-chrome"},
+ {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
{}
};
@@ -4493,6 +4536,7 @@ enum {
ALC861_FIXUP_AMP_VREF_0F,
ALC861_FIXUP_NO_JACK_DETECT,
ALC861_FIXUP_ASUS_A6RP,
+ ALC660_FIXUP_ASUS_W7J,
};
/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */
@@ -4542,10 +4586,22 @@ static const struct hda_fixup alc861_fixups[] = {
.v.func = alc861_fixup_asus_amp_vref_0f,
.chained = true,
.chain_id = ALC861_FIXUP_NO_JACK_DETECT,
+ },
+ [ALC660_FIXUP_ASUS_W7J] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* ASUS W7J needs a magic pin setup on unused NID 0x10
+ * for enabling outputs
+ */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ { }
+ },
}
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J),
+ SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J),
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP),
SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F),
SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT),
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 992ae38d5a15..1b372283bd01 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
goto out;
}
+ snd_soc_card_set_drvdata(card, priv);
+
card->dev = &pdev->dev;
card->owner = THIS_MODULE;
card->dai_link = dai;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c3c7396a6181..99b359e19d35 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
-SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUT1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
- ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUT3_OSR_SHIFT, 1, 0),
-SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
- ARIZONA_OUT4_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
- ARIZONA_OUT5_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
- ARIZONA_OUT6_OSR_SHIFT, 1, 0),
-
SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUTPUT_PATH_CONFIG_1R,
- ARIZONA_OUT1L_PGA_VOL_SHIFT,
- 0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
- ARIZONA_OUTPUT_PATH_CONFIG_2R,
- ARIZONA_OUT2L_PGA_VOL_SHIFT,
- 0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUTPUT_PATH_CONFIG_3R,
- ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
-
SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 456bb8c6d759..bc7472c968e3 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
+ iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0013;
+ iface |= 0x0003;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 253c88bb7a4c..4f05fb88bddf 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
+
+ codec->cache_sync = 1;
break;
}
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index eb4373840bb6..3665f612819d 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op)
return -ENOMEM;
card->dev = &op->dev;
- platform_set_drvdata(op, pdata);
pdata->card = card;
@@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
if (ret)
dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
+ platform_set_drvdata(op, pdata);
+
return ret;
}
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index d34d91743e3f..0b18f654b413 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -33,6 +33,10 @@
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
+#define KIRKWOOD_SPDIF_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
@@ -244,15 +248,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
ctl);
}
- if (dai->id == 0)
- ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */
- else
- ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */
-
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
/* configure */
ctl = priv->ctl_play;
+ if (dai->id == 0)
+ ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */
+ else
+ ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */
+
value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
@@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = {
.channels_max = 2,
.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
SNDRV_PCM_RATE_96000,
- .formats = KIRKWOOD_I2S_FORMATS,
+ .formats = KIRKWOOD_SPDIF_FORMATS,
},
.capture = {
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
SNDRV_PCM_RATE_96000,
- .formats = KIRKWOOD_I2S_FORMATS,
+ .formats = KIRKWOOD_SPDIF_FORMATS,
},
.ops = &kirkwood_i2s_dai_ops,
},
@@ -493,7 +497,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
.rates = SNDRV_PCM_RATE_8000_192000 |
SNDRV_PCM_RATE_CONTINUOUS |
SNDRV_PCM_RATE_KNOT,
- .formats = KIRKWOOD_I2S_FORMATS,
+ .formats = KIRKWOOD_SPDIF_FORMATS,
},
.capture = {
.channels_min = 1,
@@ -501,7 +505,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
.rates = SNDRV_PCM_RATE_8000_192000 |
SNDRV_PCM_RATE_CONTINUOUS |
SNDRV_PCM_RATE_KNOT,
- .formats = KIRKWOOD_I2S_FORMATS,
+ .formats = KIRKWOOD_SPDIF_FORMATS,
},
.ops = &kirkwood_i2s_dai_ops,
},
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 6d216cb6c19b..3fde9e402710 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
n810_ext_control(&codec->dapm);
- return clk_enable(sys_clkout2);
+ return clk_prepare_enable(sys_clkout2);
}
static void n810_shutdown(struct snd_pcm_substream *substream)
{
- clk_disable(sys_clkout2);
+ clk_disable_unprepare(sys_clkout2);
}
static int n810_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 14011d90d70a..ff60e11ecb56 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU
config SND_SOC_RCAR
tristate "R-Car series SRU/SCU/SSIU/SSI support"
select SND_SIMPLE_CARD
+ select REGMAP
help
This option enables R-Car SUR/SCU/SSIU/SSI sound support
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e53d87e881d..a66783e13a9c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
break;
case 2:
((u16 *)(&ucontrol->value.bytes.data))[0]
- &= ~params->mask;
+ &= cpu_to_be16(~params->mask);
break;
case 4:
((u32 *)(&ucontrol->value.bytes.data))[0]
- &= ~params->mask;
+ &= cpu_to_be32(~params->mask);
break;
default:
return -EINVAL;
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index b1d732255c02..3449c1e909ae 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res)
*/
int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
{
- struct device **ptr;
+ struct snd_soc_card **ptr;
int ret;
ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL);
@@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
ret = snd_soc_register_card(card);
if (ret == 0) {
- *ptr = dev;
+ *ptr = card;
devres_add(dev, ptr);
} else {
devres_free(ptr);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..11a90cd027fa 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
}
}
-static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
+static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
struct snd_soc_pcm_stream *codec_stream,
struct snd_soc_pcm_stream *cpu_stream)
{
- hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min);
- hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max);
+ struct snd_pcm_hardware *hw = &runtime->hw;
+
hw->channels_min = max(codec_stream->channels_min,
cpu_stream->channels_min);
hw->channels_max = min(codec_stream->channels_max,
@@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
if (cpu_stream->rates
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
hw->rates |= codec_stream->rates;
+
+ snd_pcm_limit_hw_rates(runtime);
+
+ hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
+ hw->rate_min = max(hw->rate_min, codec_stream->rate_min);
+ hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
+ hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max);
}
/*
@@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
/* Check that the codec and cpu DAIs are compatible */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback,
+ soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback,
&cpu_dai_drv->playback);
} else {
- soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture,
+ soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture,
&cpu_dai_drv->capture);
}
ret = -EINVAL;
- snd_pcm_limit_hw_rates(runtime);
if (!runtime->hw.rates) {
printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
codec_dai->name, cpu_dai->name);