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-rw-r--r--sound/core/seq/oss/seq_oss_event.c14
-rw-r--r--sound/core/seq/seq_timer.c8
-rw-r--r--sound/core/vmaster.c5
-rw-r--r--sound/oss/sequencer.c6
-rw-r--r--sound/pci/asihpi/asihpi.c3
-rw-r--r--sound/pci/hda/hda_codec.c35
-rw-r--r--sound/pci/hda/patch_ca0132.c36
-rw-r--r--sound/pci/hda/patch_cirrus.c4
-rw-r--r--sound/pci/hda/patch_realtek.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c29
-rw-r--r--sound/pci/ice1712/ice1712.c2
-rw-r--r--sound/soc/codecs/wm5102.c15
-rw-r--r--sound/soc/codecs/wm5110.c16
-rw-r--r--sound/soc/codecs/wm8350.c4
-rw-r--r--sound/soc/codecs/wm8960.c8
-rw-r--r--sound/soc/tegra/tegra20_i2s.h2
-rw-r--r--sound/soc/tegra/tegra30_i2s.h2
-rw-r--r--sound/usb/card.c15
18 files changed, 153 insertions, 53 deletions
diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c
index 066f5f3e3f4c..c3908862bc8b 100644
--- a/sound/core/seq/oss/seq_oss_event.c
+++ b/sound/core/seq/oss/seq_oss_event.c
@@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev
static int
note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
{
- struct seq_oss_synthinfo *info = &dp->synths[dev];
+ struct seq_oss_synthinfo *info;
+
+ if (!snd_seq_oss_synth_is_valid(dp, dev))
+ return -ENXIO;
+
+ info = &dp->synths[dev];
switch (info->arg.event_passing) {
case SNDRV_SEQ_OSS_PROCESS_EVENTS:
if (! info->ch || ch < 0 || ch >= info->nr_voices) {
@@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st
static int
note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
{
- struct seq_oss_synthinfo *info = &dp->synths[dev];
+ struct seq_oss_synthinfo *info;
+
+ if (!snd_seq_oss_synth_is_valid(dp, dev))
+ return -ENXIO;
+
+ info = &dp->synths[dev];
switch (info->arg.event_passing) {
case SNDRV_SEQ_OSS_PROCESS_EVENTS:
if (! info->ch || ch < 0 || ch >= info->nr_voices) {
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 160b1bd0cd62..24d44b2f61ac 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -290,10 +290,10 @@ int snd_seq_timer_open(struct snd_seq_queue *q)
tid.device = SNDRV_TIMER_GLOBAL_SYSTEM;
err = snd_timer_open(&t, str, &tid, q->queue);
}
- if (err < 0) {
- snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
- return err;
- }
+ }
+ if (err < 0) {
+ snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
+ return err;
}
t->callback = snd_seq_timer_interrupt;
t->callback_data = q;
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 857586135d18..0097f3619faa 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -213,7 +213,10 @@ static int slave_put(struct snd_kcontrol *kcontrol,
}
if (!changed)
return 0;
- return slave_put_val(slave, ucontrol);
+ err = slave_put_val(slave, ucontrol);
+ if (err < 0)
+ return err;
+ return 1;
}
static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index 30bcfe470f83..4ff60a6427d9 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -545,6 +545,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
case MIDI_PGM_CHANGE:
if (seq_mode == SEQ_2)
{
+ if (chn > 15)
+ break;
+
synth_devs[dev]->chn_info[chn].pgm_num = p1;
if ((int) dev >= num_synths)
synth_devs[dev]->set_instr(dev, chn, p1);
@@ -596,6 +599,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
case MIDI_PITCH_BEND:
if (seq_mode == SEQ_2)
{
+ if (chn > 15)
+ break;
+
synth_devs[dev]->chn_info[chn].bender_value = w14;
if ((int) dev < num_synths)
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 3536b076b529..0aabfedeecba 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -2549,7 +2549,7 @@ static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi,
static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
{
- struct snd_card *card = asihpi->card;
+ struct snd_card *card;
unsigned int idx = 0;
unsigned int subindex = 0;
int err;
@@ -2557,6 +2557,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
if (snd_BUG_ON(!asihpi))
return -EINVAL;
+ card = asihpi->card;
strcpy(card->mixername, "Asihpi Mixer");
err =
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 04b57383e8cb..a9ebcf9e3710 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -494,7 +494,7 @@ static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid)
int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid)
{
- return get_num_conns(codec, nid) & AC_CLIST_LENGTH;
+ return snd_hda_get_raw_connections(codec, nid, NULL, 0);
}
/**
@@ -517,9 +517,6 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t prev_nid;
int null_count = 0;
- if (snd_BUG_ON(!conn_list || max_conns <= 0))
- return -EINVAL;
-
parm = get_num_conns(codec, nid);
if (!parm)
return 0;
@@ -545,7 +542,8 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
AC_VERB_GET_CONNECT_LIST, 0);
if (parm == -1 && codec->bus->rirb_error)
return -EIO;
- conn_list[0] = parm & mask;
+ if (conn_list)
+ conn_list[0] = parm & mask;
return 1;
}
@@ -580,14 +578,20 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
continue;
}
for (n = prev_nid + 1; n <= val; n++) {
+ if (conn_list) {
+ if (conns >= max_conns)
+ return -ENOSPC;
+ conn_list[conns] = n;
+ }
+ conns++;
+ }
+ } else {
+ if (conn_list) {
if (conns >= max_conns)
return -ENOSPC;
- conn_list[conns++] = n;
+ conn_list[conns] = val;
}
- } else {
- if (conns >= max_conns)
- return -ENOSPC;
- conn_list[conns++] = val;
+ conns++;
}
prev_nid = val;
}
@@ -3334,6 +3338,8 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec,
return -EBUSY;
}
spdif = snd_array_new(&codec->spdif_out);
+ if (!spdif)
+ return -ENOMEM;
for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
kctl = snd_ctl_new1(dig_mix, codec);
if (!kctl)
@@ -3431,11 +3437,16 @@ static struct snd_kcontrol_new spdif_share_sw = {
int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
struct hda_multi_out *mout)
{
+ struct snd_kcontrol *kctl;
+
if (!mout->dig_out_nid)
return 0;
+
+ kctl = snd_ctl_new1(&spdif_share_sw, mout);
+ if (!kctl)
+ return -ENOMEM;
/* ATTENTION: here mout is passed as private_data, instead of codec */
- return snd_hda_ctl_add(codec, mout->dig_out_nid,
- snd_ctl_new1(&spdif_share_sw, mout));
+ return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl);
}
EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index db02c1e96b08..0792b5725f9c 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2298,6 +2298,11 @@ static int dspxfr_one_seg(struct hda_codec *codec,
hda_frame_size_words = ((sample_rate_div == 0) ? 0 :
(num_chans * sample_rate_mul / sample_rate_div));
+ if (hda_frame_size_words == 0) {
+ snd_printdd(KERN_ERR "frmsz zero\n");
+ return -EINVAL;
+ }
+
buffer_size_words = min(buffer_size_words,
(unsigned int)(UC_RANGE(chip_addx, 1) ?
65536 : 32768));
@@ -2308,8 +2313,7 @@ static int dspxfr_one_seg(struct hda_codec *codec,
chip_addx, hda_frame_size_words, num_chans,
sample_rate_mul, sample_rate_div, buffer_size_words);
- if ((buffer_addx == NULL) || (hda_frame_size_words == 0) ||
- (buffer_size_words < hda_frame_size_words)) {
+ if (buffer_size_words < hda_frame_size_words) {
snd_printdd(KERN_ERR "dspxfr_one_seg:failed\n");
return -EINVAL;
}
@@ -3235,7 +3239,7 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val)
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
- if (!dspload_is_loaded(codec))
+ if (spec->dsp_state != DSP_DOWNLOADED)
return 0;
/* if CrystalVoice if off, vipsource should be 0 */
@@ -4263,11 +4267,12 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
*/
static void ca0132_setup_defaults(struct hda_codec *codec)
{
+ struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
int num_fx;
int idx, i;
- if (!dspload_is_loaded(codec))
+ if (spec->dsp_state != DSP_DOWNLOADED)
return;
/* out, in effects + voicefx */
@@ -4347,12 +4352,16 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
return false;
dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
- dspload_image(codec, dsp_os_image, 0, 0, true, 0);
+ if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
+ pr_err("ca0132 dspload_image failed.\n");
+ goto exit_download;
+ }
+
dsp_loaded = dspload_wait_loaded(codec);
+exit_download:
release_firmware(fw_entry);
-
return dsp_loaded;
}
@@ -4363,16 +4372,13 @@ static void ca0132_download_dsp(struct hda_codec *codec)
#ifndef CONFIG_SND_HDA_CODEC_CA0132_DSP
return; /* NOP */
#endif
- spec->dsp_state = DSP_DOWNLOAD_INIT;
- if (spec->dsp_state == DSP_DOWNLOAD_INIT) {
- chipio_enable_clocks(codec);
- spec->dsp_state = DSP_DOWNLOADING;
- if (!ca0132_download_dsp_images(codec))
- spec->dsp_state = DSP_DOWNLOAD_FAILED;
- else
- spec->dsp_state = DSP_DOWNLOADED;
- }
+ chipio_enable_clocks(codec);
+ spec->dsp_state = DSP_DOWNLOADING;
+ if (!ca0132_download_dsp_images(codec))
+ spec->dsp_state = DSP_DOWNLOAD_FAILED;
+ else
+ spec->dsp_state = DSP_DOWNLOADED;
if (spec->dsp_state == DSP_DOWNLOADED)
ca0132_set_dsp_msr(codec, true);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 72ebb8a36b13..60d08f669f0c 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -506,6 +506,8 @@ static int patch_cs420x(struct hda_codec *codec)
if (!spec)
return -ENOMEM;
+ spec->gen.automute_hook = cs_automute;
+
snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
cs420x_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -893,6 +895,8 @@ static int patch_cs4210(struct hda_codec *codec)
if (!spec)
return -ENOMEM;
+ spec->gen.automute_hook = cs_automute;
+
snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl,
cs421x_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2d4237bc0d8e..563c24df4d6f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3163,6 +3163,7 @@ static int patch_alc269(struct hda_codec *codec)
case 0x10ec0290:
spec->codec_variant = ALC269_TYPE_ALC280;
break;
+ case 0x10ec0233:
case 0x10ec0282:
case 0x10ec0283:
spec->codec_variant = ALC269_TYPE_ALC282;
@@ -3862,6 +3863,7 @@ 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 = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 83d5335ac348..dafe04ae8c72 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
return 0;
}
+/* check whether a built-in speaker is included in parsed pins */
+static bool has_builtin_speaker(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t *nid_pin;
+ int nids, i;
+
+ if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) {
+ nid_pin = spec->gen.autocfg.line_out_pins;
+ nids = spec->gen.autocfg.line_outs;
+ } else {
+ nid_pin = spec->gen.autocfg.speaker_pins;
+ nids = spec->gen.autocfg.speaker_outs;
+ }
+
+ for (i = 0; i < nids; i++) {
+ unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]);
+ if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT)
+ return true;
+ }
+ return false;
+}
+
/*
* PC beep controls
*/
@@ -3890,6 +3913,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
return err;
}
+ /* Don't GPIO-mute speakers if there are no internal speakers, because
+ * the GPIO might be necessary for Headphone
+ */
+ if (spec->eapd_switch && !has_builtin_speaker(codec))
+ spec->eapd_switch = 0;
+
codec->proc_widget_hook = stac92hd7x_proc_hook;
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 2ffdc35d5ffd..806407a3973e 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2594,6 +2594,8 @@ static int snd_ice1712_create(struct snd_card *card,
snd_ice1712_proc_init(ice);
synchronize_irq(pci->irq);
+ card->private_data = ice;
+
err = pci_request_regions(pci, "ICE1712");
if (err < 0) {
kfree(ice);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index b8d461db369f..b82bbf584146 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -573,6 +573,13 @@ static const struct reg_default wm5102_sysclk_reva_patch[] = {
{ 0x025e, 0x0112 },
};
+static const struct reg_default wm5102_sysclk_revb_patch[] = {
+ { 0x3081, 0x08FE },
+ { 0x3083, 0x00ED },
+ { 0x30C1, 0x08FE },
+ { 0x30C3, 0x00ED },
+};
+
static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -587,6 +594,10 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
patch = wm5102_sysclk_reva_patch;
patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch);
break;
+ default:
+ patch = wm5102_sysclk_revb_patch;
+ patch_size = ARRAY_SIZE(wm5102_sysclk_revb_patch);
+ break;
}
switch (event) {
@@ -755,7 +766,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
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("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
@@ -767,7 +778,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
0xbf, 0, digital_tlv),
SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index cd17b477781d..cdeb301da1f6 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -213,9 +213,9 @@ 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("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("OUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+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),
@@ -226,9 +226,9 @@ SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
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("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("OUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+SOC_DOUBLE_R("HPOUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
@@ -240,10 +240,10 @@ SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L,
SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_VOL_SHIFT,
0xbf, 0, digital_tlv),
SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
@@ -260,11 +260,11 @@ 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("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+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("OUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+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),
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ec0efc1443ba..0e8b3aaf6c8d 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1301,7 +1301,7 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data)
if (device_may_wakeup(wm8350->dev))
pm_wakeup_event(wm8350->dev, 250);
- schedule_delayed_work(&priv->hpl.work, 200);
+ schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200));
return IRQ_HANDLED;
}
@@ -1318,7 +1318,7 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data)
if (device_may_wakeup(wm8350->dev))
pm_wakeup_event(wm8350->dev, 250);
- schedule_delayed_work(&priv->hpr.work, 200);
+ schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200));
return IRQ_HANDLED;
}
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 9bb927325993..a64b93425ae3 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -53,8 +53,8 @@
* using 2 wire for device control, so we cache them instead.
*/
static const struct reg_default wm8960_reg_defaults[] = {
- { 0x0, 0x0097 },
- { 0x1, 0x0097 },
+ { 0x0, 0x00a7 },
+ { 0x1, 0x00a7 },
{ 0x2, 0x0000 },
{ 0x3, 0x0000 },
{ 0x4, 0x0000 },
@@ -323,8 +323,8 @@ SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
wm8960_rin, ARRAY_SIZE(wm8960_rin)),
-SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0),
-SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER1, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0),
SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index c27069d24d77..729958713cd4 100644
--- a/sound/soc/tegra/tegra20_i2s.h
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -121,7 +121,7 @@
#define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12)
#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0
-#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff
+#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff
#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
/* Fields in TEGRA20_I2S_FIFO_SCR */
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 34dc47b9581c..a294d942b9f7 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -110,7 +110,7 @@
#define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12)
#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0
-#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff
+#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff
#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
/* Fields in TEGRA30_I2S_OFFSET */
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 803953a9bff3..2da8ad75fd96 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -244,6 +244,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
if (!assoc) {
+ /*
+ * Firmware writers cannot count to three. So to find
+ * the IAD on the NuForce UDH-100, also check the next
+ * interface.
+ */
+ struct usb_interface *iface =
+ usb_ifnum_to_if(dev, ctrlif + 1);
+ if (iface &&
+ iface->intf_assoc &&
+ iface->intf_assoc->bFunctionClass == USB_CLASS_AUDIO &&
+ iface->intf_assoc->bFunctionProtocol == UAC_VERSION_2)
+ assoc = iface->intf_assoc;
+ }
+
+ if (!assoc) {
snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n");
return -EINVAL;
}
Copyright © 1996 – 2023 Jason A. Donenfeld. All Rights Reverse Engineered.