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-rw-r--r--sound/core/oss/pcm_oss.c2
-rw-r--r--sound/pci/Kconfig4
-rw-r--r--sound/pci/hda/hda_codec.c9
-rw-r--r--sound/pci/hda/hda_codec.h1
-rw-r--r--sound/pci/hda/hda_proc.c3
-rw-r--r--sound/pci/hda/patch_conexant.c14
-rw-r--r--sound/pci/hda/patch_realtek.c8
-rw-r--r--sound/pci/hda/patch_sigmatel.c58
-rw-r--r--sound/pci/intel8x0.c2
-rw-r--r--sound/pci/oxygen/virtuoso.c3
-rw-r--r--sound/soc/atmel/atmel-pcm.c2
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c2
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.h2
-rw-r--r--sound/soc/codecs/wm8350.c2
-rw-r--r--sound/soc/codecs/wm8753.c9
-rw-r--r--sound/soc/codecs/wm8990.c3
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c18
-rw-r--r--sound/soc/omap/omap-mcbsp.c4
-rw-r--r--sound/soc/omap/omap-pcm.c5
-rw-r--r--sound/usb/usbaudio.c1
20 files changed, 98 insertions, 54 deletions
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index e17836680f4..0a1798eafb0 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1767,7 +1767,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
AFMT_S8 | AFMT_U16_LE |
AFMT_U16_BE |
AFMT_S32_LE | AFMT_S32_BE |
- AFMT_S24_LE | AFMT_S24_LE |
+ AFMT_S24_LE | AFMT_S24_BE |
AFMT_S24_PACKED;
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 6e3a1848447..82b9bddcdcd 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -744,8 +744,8 @@ config SND_VIRTUOSO
select SND_OXYGEN_LIB
help
Say Y here to include support for sound cards based on the
- Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and
- HDAV1.3 (Deluxe).
+ Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
+ Support for the HDAV1.3 (Deluxe) is very experimental.
To compile this driver as a module, choose M here: the module
will be called snd-virtuoso.
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b7bba7dc7cf..0b708134d12 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -487,7 +487,6 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
{
struct hda_bus *bus;
int err;
- char qname[8];
static struct snd_device_ops dev_ops = {
.dev_register = snd_hda_bus_dev_register,
.dev_free = snd_hda_bus_dev_free,
@@ -517,10 +516,12 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
mutex_init(&bus->cmd_mutex);
INIT_LIST_HEAD(&bus->codec_list);
- snprintf(qname, sizeof(qname), "hda%d", card->number);
- bus->workq = create_workqueue(qname);
+ snprintf(bus->workq_name, sizeof(bus->workq_name),
+ "hd-audio%d", card->number);
+ bus->workq = create_singlethread_workqueue(bus->workq_name);
if (!bus->workq) {
- snd_printk(KERN_ERR "cannot create workqueue %s\n", qname);
+ snd_printk(KERN_ERR "cannot create workqueue %s\n",
+ bus->workq_name);
kfree(bus);
return -ENOMEM;
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 5810ef58840..09a332ada0c 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -614,6 +614,7 @@ struct hda_bus {
/* unsolicited event queue */
struct hda_bus_unsolicited *unsol;
+ char workq_name[16];
struct workqueue_struct *workq; /* common workqueue for codecs */
/* assigned PCMs */
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 7ca66d65414..144b85276d5 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -399,7 +399,8 @@ static void print_conn_list(struct snd_info_buffer *buffer,
{
int c, curr = -1;
- if (conn_len > 1 && wid_type != AC_WID_AUD_MIX)
+ if (conn_len > 1 && wid_type != AC_WID_AUD_MIX &&
+ wid_type != AC_WID_VOL_KNB)
curr = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_CONNECT_SEL, 0);
snd_iprintf(buffer, " Connection: %d\n", conn_len);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 75de40aaab0..0177ef8f4c9 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -347,6 +347,7 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol,
&spec->cur_mux[adc_idx]);
}
+#ifdef CONFIG_SND_JACK
static int conexant_add_jack(struct hda_codec *codec,
hda_nid_t nid, int type)
{
@@ -394,7 +395,6 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid)
static int conexant_init_jacks(struct hda_codec *codec)
{
-#ifdef CONFIG_SND_JACK
struct conexant_spec *spec = codec->spec;
int i;
@@ -422,10 +422,19 @@ static int conexant_init_jacks(struct hda_codec *codec)
++hv;
}
}
-#endif
return 0;
}
+#else
+static inline void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid)
+{
+}
+
+static inline int conexant_init_jacks(struct hda_codec *codec)
+{
+ return 0;
+}
+#endif
static int conexant_init(struct hda_codec *codec)
{
@@ -1566,6 +1575,7 @@ static struct snd_pci_quirk cxt5047_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP),
SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP),
SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP),
+ SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP),
SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 82dd0843197..ae5c8a0d147 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1037,6 +1037,7 @@ do_sku:
case 0x10ec0267:
case 0x10ec0268:
case 0x10ec0269:
+ case 0x10ec0272:
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
@@ -1065,6 +1066,7 @@ do_sku:
case 0x10ec0882:
case 0x10ec0883:
case 0x10ec0885:
+ case 0x10ec0887:
case 0x10ec0889:
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 7);
@@ -7012,12 +7014,14 @@ static int patch_alc882(struct hda_codec *codec)
break;
case 0x106b1000: /* iMac 24 */
case 0x106b2800: /* AppleTV */
+ case 0x106b3e00: /* iMac 24 Aluminium */
board_config = ALC885_IMAC24;
break;
case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
case 0x106b00a4: /* MacbookPro4,1 */
case 0x106b2c00: /* Macbook Pro rev3 */
case 0x106b3600: /* Macbook 3.1 */
+ case 0x106b3800: /* MacbookPro4,1 - latter revision */
board_config = ALC885_MBP3;
break;
default:
@@ -8478,6 +8482,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
@@ -8512,6 +8517,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
+ SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550",
+ ALC883_FUJITSU_PI2515),
SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515),
SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530",
ALC888_FUJITSU_XA3530),
@@ -8526,6 +8533,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL),
SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
{}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c39deebb588..38428e22428 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -81,6 +81,7 @@ enum {
enum {
STAC_92HD83XXX_REF,
+ STAC_92HD83XXX_PWR_REF,
STAC_92HD83XXX_MODELS
};
@@ -334,7 +335,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
};
static unsigned int stac92hd83xxx_pwr_mapping[4] = {
- 0x03, 0x0c, 0x10, 0x40,
+ 0x03, 0x0c, 0x20, 0x40,
};
static hda_nid_t stac92hd83xxx_amp_nids[1] = {
@@ -841,10 +842,6 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = {
};
static struct hda_verb stac92hd83xxx_core_init[] = {
- /* start of config #1 */
- { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3},
-
- /* start of config #2 */
{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
@@ -885,8 +882,8 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
static struct hda_verb stac925x_core_init[] = {
/* set dac0mux for dac converter */
{ 0x06, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* unmute and set max the selector */
- { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f },
+ /* mute the master volume */
+ { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{}
};
@@ -1138,6 +1135,8 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
};
static struct snd_kcontrol_new stac925x_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
STAC_INPUT_SOURCE(1),
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
@@ -1736,10 +1735,12 @@ static unsigned int ref92hd83xxx_pin_configs[14] = {
static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
+ [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
};
static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
[STAC_92HD83XXX_REF] = "ref",
+ [STAC_92HD83XXX_PWR_REF] = "mic-ref",
};
static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1799,8 +1800,12 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"HP dv5", STAC_HP_M4),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
"HP dv7", STAC_HP_M4),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
+ "HP dv4", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
"HP dv7", STAC_HP_M4),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600,
+ "HP dv5", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603,
"HP dv5", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
@@ -2536,6 +2541,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
info->name = "STAC92xx Analog";
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dac_nids[0];
info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs;
@@ -3573,13 +3580,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
err = stac92xx_auto_fill_dac_nids(codec);
if (err < 0)
return err;
+ err = stac92xx_auto_create_multi_out_ctls(codec,
+ &spec->autocfg);
+ if (err < 0)
+ return err;
}
- err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg);
-
- if (err < 0)
- return err;
-
/* setup analog beep controls */
if (spec->anabeep_nid > 0) {
err = stac92xx_auto_create_beep_ctls(codec,
@@ -4753,7 +4759,9 @@ static struct hda_input_mux stac92hd83xxx_dmux = {
static int patch_stac92hd83xxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
+ hda_nid_t conn[STAC92HD83_DAC_COUNT + 1];
int err;
+ int num_dacs;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -4772,15 +4780,16 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
spec->multiout.dac_nids = spec->dac_nids;
- spec->init = stac92hd83xxx_core_init;
- switch (codec->vendor_id) {
- case 0x111d7605:
- break;
- default:
- spec->num_pwrs--;
- spec->init++; /* switch to config #2 */
- }
+ /* set port 0xe to select the last DAC
+ */
+ num_dacs = snd_hda_get_connections(codec, 0x0e,
+ conn, STAC92HD83_DAC_COUNT + 1) - 1;
+
+ snd_hda_codec_write_cache(codec, 0xe, 0,
+ AC_VERB_SET_CONNECT_SEL, num_dacs);
+
+ spec->init = stac92hd83xxx_core_init;
spec->mixer = stac92hd83xxx_mixer;
spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids);
@@ -4806,6 +4815,15 @@ again:
return err;
}
+ switch (codec->vendor_id) {
+ case 0x111d7604:
+ case 0x111d7605:
+ if (spec->board_config == STAC_92HD83XXX_PWR_REF)
+ break;
+ spec->num_pwrs = 0;
+ break;
+ }
+
err = stac92xx_parse_auto_config(codec, 0x1d, 0);
if (!err) {
if (spec->board_config < 0) {
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 19d3391e229..e900cdc8484 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -617,7 +617,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip)
int time = 100;
if (chip->buggy_semaphore)
return 0; /* just ignore ... */
- while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY))
+ while (--time && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY))
udelay(1);
if (! time && ! chip->in_ac97_init)
snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n");
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index e9e829e83d7..18c7c91786b 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -683,7 +683,7 @@ static void xonar_hdav_uart_input(struct oxygen *chip)
if (chip->uart_input_count >= 2 &&
chip->uart_input[chip->uart_input_count - 2] == 'O' &&
chip->uart_input[chip->uart_input_count - 1] == 'K') {
- printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:");
+ printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n");
print_hex_dump_bytes("", DUMP_PREFIX_OFFSET,
chip->uart_input, chip->uart_input_count);
chip->uart_input_count = 0;
@@ -908,6 +908,7 @@ static const struct oxygen_model model_xonar_hdav = {
.dac_channels = 8,
.dac_volume_min = 0x0f,
.dac_volume_max = 0xff,
+ .misc_flags = OXYGEN_MISC_MIDI,
.function_flags = OXYGEN_FUNCTION_2WIRE,
.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 1fac5efd285..3dcdc4e3cfa 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -44,8 +44,6 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <mach/hardware.h>
-
#include "atmel-pcm.h"
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index c5d67900d66..ff0054b7650 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -10,7 +10,7 @@
* Based on at91-ssc.c by
* Frank Mandarino <fmandarino@endrelia.com>
* Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index a828746e8a2..391135f9c6c 100644
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -10,7 +10,7 @@
* Based on at91-ssc.c by
* Frank Mandarino <fmandarino@endrelia.com>
* Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index e3989d406f5..35d99750c38 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -3,7 +3,7 @@
*
* Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
*
- * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com>
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 6c21b50c937..77620ab9875 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1451,7 +1451,14 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
},
};
-struct snd_soc_dai wm8753_dai[2];
+struct snd_soc_dai wm8753_dai[] = {
+ {
+ .name = "WM8753 DAI 0",
+ },
+ {
+ .name = "WM8753 DAI 1",
+ },
+};
EXPORT_SYMBOL_GPL(wm8753_dai);
static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5b5afc14447..1cbb7b9b51c 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -2,8 +2,7 @@
* wm8990.c -- WM8990 ALSA Soc Audio driver
*
* Copyright 2008 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index bcec3f60bad..acf39a646b2 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -183,16 +183,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
};
/**
- * mpc8610_hpcd_machine: ASoC machine data
- */
-static struct snd_soc_card mpc8610_hpcd_machine = {
- .probe = mpc8610_hpcd_machine_probe,
- .remove = mpc8610_hpcd_machine_remove,
- .name = "MPC8610 HPCD",
- .num_links = 1,
-};
-
-/**
* mpc8610_hpcd_probe: OF probe function for the fabric driver
*
* This function gets called when an SSI node is found in the device tree.
@@ -455,7 +445,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */
machine_data->dai.ops = &mpc8610_hpcd_ops;
- mpc8610_hpcd_machine.dai_link = &machine_data->dai;
+ machine_data->machine.probe = mpc8610_hpcd_machine_probe;
+ machine_data->machine.remove = mpc8610_hpcd_machine_remove;
+ machine_data->machine.name = "MPC8610 HPCD";
+ machine_data->machine.num_links = 1;
+ machine_data->machine.dai_link = &machine_data->dai;
/* Allocate a new audio platform device structure */
sound_device = platform_device_alloc("soc-audio", -1);
@@ -465,7 +459,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
goto error;
}
- machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
+ machine_data->sound_devdata.card = &machine_data->machine;
machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
machine_data->machine.platform = &fsl_soc_platform;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index ec5e18a7875..05dd5abcddf 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -302,6 +302,10 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
regs->spcr1 |= RINTM(3);
regs->rcr2 |= RFIG;
regs->xcr2 |= XFIG;
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ regs->xccr = DXENDLY(1) | XDMAEN;
+ regs->rccr = RFULL_CYCLE | RDMAEN;
+ }
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index b0362dfd5b7..dd3bb293376 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -175,9 +175,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct omap_runtime_data *prtd = runtime->private_data;
+ unsigned long flags;
int ret = 0;
- spin_lock_irq(&prtd->lock);
+ spin_lock_irqsave(&prtd->lock, flags);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
@@ -195,7 +196,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
default:
ret = -EINVAL;
}
- spin_unlock_irq(&prtd->lock);
+ spin_unlock_irqrestore(&prtd->lock, flags);
return ret;
}
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c709b956322..2ab83129d9b 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2966,6 +2966,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
return -EINVAL;
}
alts = &iface->altsetting[fp->altset_idx];
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
usb_set_interface(chip->dev, fp->iface, 0);
init_usb_pitch(chip->dev, fp->iface, alts, fp);
init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max);
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-19 03:42:59 +0000