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Date: Wed, 23 Mar 2011 14:49:30 +0100
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lrg@...mlogic.co.uk>,
Andrew Morton <akpm@...ux-foundation.org>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes
Linus,
please pull sound fixes for v2.6.39-rc1 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
A series of fixes for VIA HD-audio codecs and a few other small fixes
are included.
Thanks!
Takashi
===
Andres Mejia (1):
ALSA: hda - Set EAPD for Realtek ALC665
David Henningsson (1):
ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
Lydia Wang (7):
ALSA: hda - VIA: Fix stereo mixer recording no sound issue
ALSA: hda - VIA: Fix independent headphone no sound issue
ALSA: hda - VIA: Add missing support for VT1718S in A-A path
ALSA: hda - VIA: Fix invalid A-A path volume adjust issue
ALSA: hda - VIA: Fix codec type for VT1708BCE at the right timing
ALSA: hda - VIA: Correct stream names for VT1818S
ALSA: hda - VIA: Fix VT1708 can't build up Headphone control issue
Marek Belisko (1):
ASoC: mini2440: Fix uda134x codec problem.
Mark Brown (2):
ASoC: Remove bogus check for register validity in debugfs write
ASoC: Support !REGULATOR build for sgtl5000
Takashi Iwai (1):
ALSA: usb - Remove trailing spaces from USB card name strings
Xiaochen Wang (1):
sound: read i_size with i_size_read()
---
arch/arm/mach-s3c2440/mach-mini2440.c | 7 ++++
sound/pci/hda/patch_realtek.c | 27 +++++++++++----
sound/pci/hda/patch_via.c | 58 ++++++++++++++++++++++++++-------
sound/soc/codecs/sgtl5000.c | 14 ++++++++
sound/soc/codecs/uda134x.c | 3 +-
sound/soc/samsung/s3c24xx_uda134x.c | 3 +-
sound/soc/soc-core.c | 2 -
sound/sound_firmware.c | 2 +-
sound/usb/card.c | 22 ++++++++++--
9 files changed, 109 insertions(+), 29 deletions(-)
diff --git a/arch/arm/mach-s3c2440/mach-mini2440.c b/arch/arm/mach-s3c2440/mach-mini2440.c
index d80f129..dfedc9c 100644
--- a/arch/arm/mach-s3c2440/mach-mini2440.c
+++ b/arch/arm/mach-s3c2440/mach-mini2440.c
@@ -488,6 +488,11 @@ static struct i2c_board_info mini2440_i2c_devs[] __initdata = {
},
};
+static struct platform_device uda1340_codec = {
+ .name = "uda134x-codec",
+ .id = -1,
+};
+
static struct platform_device *mini2440_devices[] __initdata = {
&s3c_device_ohci,
&s3c_device_wdt,
@@ -503,7 +508,9 @@ static struct platform_device *mini2440_devices[] __initdata = {
&s3c_device_nand,
&s3c_device_sdi,
&s3c_device_iis,
+ &uda1340_codec,
&mini2440_audio,
+ &samsung_asoc_dma,
};
static void __init mini2440_map_io(void)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f1a03f2..5d582de 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1265,6 +1265,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
+ case 0x10ec0665:
case 0x10ec0862:
case 0x10ec0889:
set_eapd(codec, 0x14, 1);
@@ -4240,6 +4241,7 @@ static void alc_power_eapd(struct hda_codec *codec)
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
+ case 0x10ec0665:
case 0x10ec0862:
case 0x10ec0889:
set_eapd(codec, 0x14, 0);
@@ -16006,9 +16008,12 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec,
return err;
} else {
const char *name = pfx;
- if (!name)
+ int index = i;
+ if (!name) {
name = chname[i];
- err = __alc861_create_out_sw(codec, name, nid, i, 3);
+ index = 0;
+ }
+ err = __alc861_create_out_sw(codec, name, nid, index, 3);
if (err < 0)
return err;
}
@@ -17159,16 +17164,19 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
return err;
} else {
const char *name = pfx;
- if (!name)
+ int index = i;
+ if (!name) {
name = chname[i];
+ index = 0;
+ }
err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
- name, i,
+ name, index,
HDA_COMPOSE_AMP_VAL(nid_v, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
- name, i,
+ name, index,
HDA_COMPOSE_AMP_VAL(nid_s, 3, 2,
HDA_INPUT));
if (err < 0)
@@ -19217,12 +19225,15 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
return err;
} else {
const char *name = pfx;
- if (!name)
+ int index = i;
+ if (!name) {
name = chname[i];
- err = __alc662_add_vol_ctl(spec, name, nid, i, 3);
+ index = 0;
+ }
+ err = __alc662_add_vol_ctl(spec, name, nid, index, 3);
if (err < 0)
return err;
- err = __alc662_add_sw_ctl(spec, name, mix, i, 3);
+ err = __alc662_add_sw_ctl(spec, name, mix, index, 3);
if (err < 0)
return err;
}
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 63b0054..1371b57 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -159,6 +159,7 @@ struct via_spec {
#endif
};
+static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec);
static struct via_spec * via_new_spec(struct hda_codec *codec)
{
struct via_spec *spec;
@@ -169,6 +170,10 @@ static struct via_spec * via_new_spec(struct hda_codec *codec)
codec->spec = spec;
spec->codec = codec;
+ spec->codec_type = get_codec_type(codec);
+ /* VT1708BCE & VT1708S are almost same */
+ if (spec->codec_type == VT1708BCE)
+ spec->codec_type = VT1708S;
return spec;
}
@@ -1101,6 +1106,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ int ret;
if (!spec->mux_nids[adc_idx])
return -EINVAL;
@@ -1109,12 +1115,14 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0)
snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- /* update jack power state */
- set_jack_power_state(codec);
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ ret = snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
spec->mux_nids[adc_idx],
&spec->cur_mux[adc_idx]);
+ /* update jack power state */
+ set_jack_power_state(codec);
+
+ return ret;
}
static int via_independent_hp_info(struct snd_kcontrol *kcontrol,
@@ -1188,8 +1196,16 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
/* Get Independent Mode index of headphone pin widget */
spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
? 1 : 0;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel);
+ if (spec->codec_type == VT1718S)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
+ else
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, pinsel);
+ if (spec->codec_type == VT1812)
+ snd_hda_codec_write(codec, 0x35, 0,
+ AC_VERB_SET_CONNECT_SEL, pinsel);
if (spec->multiout.hp_nid && spec->multiout.hp_nid
!= spec->multiout.dac_nids[HDA_FRONT])
snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid,
@@ -1208,6 +1224,8 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
activate_ctl(codec, "Headphone Playback Switch",
spec->hp_independent_mode);
}
+ /* update jack power state */
+ set_jack_power_state(codec);
return 0;
}
@@ -1248,9 +1266,12 @@ static int via_hp_build(struct hda_codec *codec)
break;
}
- nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS);
- if (nums <= 1)
- return 0;
+ if (spec->codec_type != VT1708) {
+ nums = snd_hda_get_connections(codec, nid,
+ conn, HDA_MAX_CONNECTIONS);
+ if (nums <= 1)
+ return 0;
+ }
knew = via_clone_control(spec, &via_hp_mixer[0]);
if (knew == NULL)
@@ -1310,6 +1331,11 @@ static void mute_aa_path(struct hda_codec *codec, int mute)
start_idx = 2;
end_idx = 4;
break;
+ case VT1718S:
+ nid_mixer = 0x21;
+ start_idx = 1;
+ end_idx = 3;
+ break;
default:
return;
}
@@ -2185,10 +2211,6 @@ static int via_init(struct hda_codec *codec)
for (i = 0; i < spec->num_iverbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
- spec->codec_type = get_codec_type(codec);
- if (spec->codec_type == VT1708BCE)
- spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost
- same */
/* Lydia Add for EAPD enable */
if (!spec->dig_in_nid) { /* No Digital In connection */
if (spec->dig_in_pin) {
@@ -2438,7 +2460,14 @@ static int vt_auto_create_analog_input_ctls(struct hda_codec *codec,
else
type_idx = 0;
label = hda_get_autocfg_input_label(codec, cfg, i);
- err = via_new_analog_input(spec, label, type_idx, idx, cap_nid);
+ if (spec->codec_type == VT1708S ||
+ spec->codec_type == VT1702 ||
+ spec->codec_type == VT1716S)
+ err = via_new_analog_input(spec, label, type_idx,
+ idx+1, cap_nid);
+ else
+ err = via_new_analog_input(spec, label, type_idx,
+ idx, cap_nid);
if (err < 0)
return err;
snd_hda_add_imux_item(imux, label, idx, NULL);
@@ -4147,6 +4176,11 @@ static int patch_vt1708S(struct hda_codec *codec)
spec->stream_name_analog = "VT1708BCE Analog";
spec->stream_name_digital = "VT1708BCE Digital";
}
+ /* correct names for VT1818S */
+ if (codec->vendor_id == 0x11060440) {
+ spec->stream_name_analog = "VT1818S Analog";
+ spec->stream_name_digital = "VT1818S Digital";
+ }
return 0;
}
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 1f7217f..ff29380 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -772,6 +772,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+#ifdef CONFIG_REGULATOR
static int ldo_regulator_is_enabled(struct regulator_dev *dev)
{
struct ldo_regulator *ldo = rdev_get_drvdata(dev);
@@ -901,6 +902,19 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec)
return 0;
}
+#else
+static int ldo_regulator_register(struct snd_soc_codec *codec,
+ struct regulator_init_data *init_data,
+ int voltage)
+{
+ return -EINVAL;
+}
+
+static int ldo_regulator_remove(struct snd_soc_codec *codec)
+{
+ return 0;
+}
+#endif
/*
* set dac bias
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index e76847a..48ffd40 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -486,7 +486,8 @@ static struct snd_soc_dai_driver uda134x_dai = {
static int uda134x_soc_probe(struct snd_soc_codec *codec)
{
struct uda134x_priv *uda134x;
- struct uda134x_platform_data *pd = dev_get_drvdata(codec->card->dev);
+ struct uda134x_platform_data *pd = codec->card->dev->platform_data;
+
int ret;
printk(KERN_INFO "UDA134X SoC Audio Codec\n");
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index 3cb7007..dc9d551 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -219,7 +219,7 @@ static struct snd_soc_ops s3c24xx_uda134x_ops = {
static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
.name = "UDA134X",
.stream_name = "UDA134X",
- .codec_name = "uda134x-hifi",
+ .codec_name = "uda134x-codec",
.codec_dai_name = "uda134x-hifi",
.cpu_dai_name = "s3c24xx-iis",
.ops = &s3c24xx_uda134x_ops,
@@ -314,6 +314,7 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev)
platform_set_drvdata(s3c24xx_uda134x_snd_device,
&snd_soc_s3c24xx_uda134x);
+ platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x));
ret = platform_device_add(s3c24xx_uda134x_snd_device);
if (ret) {
printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 17efacd..4dda589 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -259,8 +259,6 @@ static ssize_t codec_reg_write_file(struct file *file,
while (*start == ' ')
start++;
reg = simple_strtoul(start, &start, 16);
- if ((reg >= codec->driver->reg_cache_size) || (reg % step))
- return -EINVAL;
while (*start == ' ')
start++;
if (strict_strtoul(start, 16, &value))
diff --git a/sound/sound_firmware.c b/sound/sound_firmware.c
index 340a0bc..7e96249 100644
--- a/sound/sound_firmware.c
+++ b/sound/sound_firmware.c
@@ -19,7 +19,7 @@ static int do_mod_firmware_load(const char *fn, char **fp)
printk(KERN_INFO "Unable to load '%s'.\n", fn);
return 0;
}
- l = filp->f_path.dentry->d_inode->i_size;
+ l = i_size_read(filp->f_path.dentry->d_inode);
if (l <= 0 || l > 131072)
{
printk(KERN_INFO "Invalid firmware '%s'\n", fn);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 40722f8..a90662a 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -41,6 +41,7 @@
#include <linux/list.h>
#include <linux/slab.h>
#include <linux/string.h>
+#include <linux/ctype.h>
#include <linux/usb.h>
#include <linux/moduleparam.h>
#include <linux/mutex.h>
@@ -283,6 +284,15 @@ static int snd_usb_audio_dev_free(struct snd_device *device)
return snd_usb_audio_free(chip);
}
+static void remove_trailing_spaces(char *str)
+{
+ char *p;
+
+ if (!*str)
+ return;
+ for (p = str + strlen(str) - 1; p >= str && isspace(*p); p--)
+ *p = 0;
+}
/*
* create a chip instance and set its names.
@@ -351,7 +361,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
snd_component_add(card, component);
/* retrieve the device string as shortname */
- if (quirk && quirk->product_name) {
+ if (quirk && quirk->product_name && *quirk->product_name) {
strlcpy(card->shortname, quirk->product_name, sizeof(card->shortname));
} else {
if (!dev->descriptor.iProduct ||
@@ -363,9 +373,10 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
USB_ID_PRODUCT(chip->usb_id));
}
}
+ remove_trailing_spaces(card->shortname);
/* retrieve the vendor and device strings as longname */
- if (quirk && quirk->vendor_name) {
+ if (quirk && quirk->vendor_name && *quirk->vendor_name) {
len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
} else {
if (dev->descriptor.iManufacturer)
@@ -375,8 +386,11 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
len = 0;
/* we don't really care if there isn't any vendor string */
}
- if (len > 0)
- strlcat(card->longname, " ", sizeof(card->longname));
+ if (len > 0) {
+ remove_trailing_spaces(card->longname);
+ if (*card->longname)
+ strlcat(card->longname, " ", sizeof(card->longname));
+ }
strlcat(card->longname, card->shortname, sizeof(card->longname));
--
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