* [GIT PULL] ALSA fixes
@ 2008-10-23 17:56 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-10-23 17:56 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, perex, linux-kernel
Hi Linus,
please pull ALSA updates for 2.6.28 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains only the following small bug fixes.
Thanks!
Takashi
==
Harvey Harrison (1):
ALSA: hda - correct bracketing in spdif test in patch_sigmatel.c
Jarkko Nikula (2):
ALSA: ASoC: OMAP: Continue fixing DSP DAI format in McBSP DAI driver
ALSA: ASoC: tlv320aic3x: Fix DSP DAI format and signal polarities matching
Johannes Berg (1):
ALSA: aoa i2sbus: don't overwrite module parameter
Mark Brown (1):
ALSA: Ensure PXA runtime data is initialised
Takashi Iwai (1):
ALSA: hda - Fix conflicting volume controls on ALC260
sound/aoa/soundbus/i2sbus/i2sbus-core.c | 6 +++---
sound/arm/pxa2xx-pcm-lib.c | 2 +-
sound/pci/hda/patch_realtek.c | 22 ++++++++++++++--------
sound/pci/hda/patch_sigmatel.c | 2 +-
sound/soc/codecs/tlv320aic3x.c | 16 ++++++++++------
sound/soc/omap/omap-mcbsp.c | 7 ++-----
6 files changed, 31 insertions(+), 24 deletions(-)
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
index e6beb92..b4590df 100644
--- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
@@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
struct i2sbus_dev *dev;
struct device_node *child = NULL, *sound = NULL;
struct resource *r;
- int i, layout = 0, rlen;
+ int i, layout = 0, rlen, ok = force;
static const char *rnames[] = { "i2sbus: %s (control)",
"i2sbus: %s (tx)",
"i2sbus: %s (rx)" };
@@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
layout = *layout_id;
snprintf(dev->sound.modalias, 32,
"sound-layout-%d", layout);
- force = 1;
+ ok = 1;
}
}
/* for the time being, until we can handle non-layout-id
@@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
* When there are two i2s busses and only one has a layout-id,
* then this depends on the order, but that isn't important
* either as the second one in that case is just a modem. */
- if (!force) {
+ if (!ok) {
kfree(dev);
return -ENODEV;
}
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 1c93eb7..75a0d74 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
goto out;
ret = -ENOMEM;
- rtd = kmalloc(sizeof(*rtd), GFP_KERNEL);
+ rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
if (!rtd)
goto out;
rtd->dma_desc_array =
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e72707c..ef4955c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4996,7 +4996,7 @@ static struct hda_verb alc260_test_init_verbs[] = {
*/
static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
- const char *pfx)
+ const char *pfx, int *vol_bits)
{
hda_nid_t nid_vol;
unsigned long vol_val, sw_val;
@@ -5018,10 +5018,14 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
} else
return 0; /* N/A */
- snprintf(name, sizeof(name), "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
- if (err < 0)
- return err;
+ if (!(*vol_bits & (1 << nid_vol))) {
+ /* first control for the volume widget */
+ snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
+ if (err < 0)
+ return err;
+ *vol_bits |= (1 << nid_vol);
+ }
snprintf(name, sizeof(name), "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
if (err < 0)
@@ -5035,6 +5039,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
{
hda_nid_t nid;
int err;
+ int vols = 0;
spec->multiout.num_dacs = 1;
spec->multiout.dac_nids = spec->private_dac_nids;
@@ -5042,21 +5047,22 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
nid = cfg->line_out_pins[0];
if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Front");
+ err = alc260_add_playback_controls(spec, nid, "Front", &vols);
if (err < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Speaker");
+ err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
if (err < 0)
return err;
}
nid = cfg->hp_pins[0];
if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Headphone");
+ err = alc260_add_playback_controls(spec, nid, "Headphone",
+ &vols);
if (err < 0)
return err;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a2ac720..788fdc6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1282,7 +1282,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
return err;
spec->multiout.share_spdif = 1;
}
- if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) {
+ if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
if (err < 0)
return err;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 05336ed..cff276e 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- /* interface format */
- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_I2S:
+ /*
+ * match both interface format and signal polarities since they
+ * are fixed
+ */
+ switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+ SND_SOC_DAIFMT_INV_MASK)) {
+ case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
break;
- case SND_SOC_DAIFMT_DSP_A:
+ case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
iface_breg |= (0x01 << 6);
break;
- case SND_SOC_DAIFMT_RIGHT_J:
+ case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x02 << 6);
break;
- case SND_SOC_DAIFMT_LEFT_J:
+ case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x03 << 6);
break;
default:
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 853b33a..8485a8a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
break;
case SND_SOC_DAIFMT_DSP_A:
regs->srgr2 |= FPER(wlen * 2 - 1);
- regs->srgr1 |= FWID(0);
+ regs->srgr1 |= FWID(wlen * 2 - 2);
break;
}
@@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
{
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
- unsigned int temp_fmt = fmt;
if (mcbsp_data->configured)
return 0;
@@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
regs->xcr2 |= XDATDLY(0);
- /* Invert bit clock and FS polarity configuration for DSP_A */
- temp_fmt ^= SND_SOC_DAIFMT_IB_IF;
break;
default:
/* Unsupported data format */
@@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
}
/* Set bit clock (CLKX/CLKR) and FS polarities */
- switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
/*
* Normal BCLK + FS.
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-11-21 16:51 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-11-21 16:51 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for 2.6.28-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains one trivial fix for sound_core.c and a few (yet again)
fixes for Dell laptops with STAC/IDT codecs.
Thanks!
Takashi
===
Hannes Eder (1):
sound/sound_core: Fix sparse warnings
Matthew Ranostay (2):
ALSA: hda: STAC_DELL_M6 EAPD
ALSA: hda: Add STAC_DELL_M4_3 quirk
Takashi Iwai (1):
ALSA: hda - Add a quirk for Dell Studio 15
---
Documentation/sound/alsa/ALSA-Configuration.txt | 1 +
sound/pci/hda/patch_sigmatel.c | 43 +++++++++++++++++++---
sound/sound_core.c | 6 ++--
3 files changed, 41 insertions(+), 9 deletions(-)
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index e0e54a2..147f176 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -1072,6 +1072,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
ref Reference board
dell-m4-1 Dell desktops
dell-m4-2 Dell desktops
+ dell-m4-3 Dell desktops
STAC92HD73*
ref Reference board
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9563b5b..2b52a40 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -84,6 +84,7 @@ enum {
STAC_92HD71BXX_REF,
STAC_DELL_M4_1,
STAC_DELL_M4_2,
+ STAC_DELL_M4_3,
STAC_HP_M4,
STAC_92HD71BXX_MODELS
};
@@ -137,6 +138,7 @@ struct sigmatel_spec {
unsigned int num_mixers;
int board_config;
+ unsigned int eapd_switch: 1;
unsigned int surr_switch: 1;
unsigned int line_switch: 1;
unsigned int mic_switch: 1;
@@ -1628,6 +1630,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
"unknown Dell", STAC_DELL_M6),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271,
"unknown Dell", STAC_DELL_M6),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x029f,
+ "Dell Studio 15", STAC_DELL_M6),
{} /* terminator */
};
@@ -1670,10 +1674,17 @@ static unsigned int dell_m4_2_pin_configs[11] = {
0x40f000f0, 0x044413b0, 0x044413b0,
};
+static unsigned int dell_m4_3_pin_configs[11] = {
+ 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
+ 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0,
+ 0x40f000f0, 0x044413b0, 0x044413b0,
+};
+
static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
[STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs,
[STAC_DELL_M4_1] = dell_m4_1_pin_configs,
[STAC_DELL_M4_2] = dell_m4_2_pin_configs,
+ [STAC_DELL_M4_3] = dell_m4_3_pin_configs,
[STAC_HP_M4] = NULL,
};
@@ -1681,6 +1692,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
[STAC_92HD71BXX_REF] = "ref",
[STAC_DELL_M4_1] = "dell-m4-1",
[STAC_DELL_M4_2] = "dell-m4-2",
+ [STAC_DELL_M4_3] = "dell-m4-3",
[STAC_HP_M4] = "hp-m4",
};
@@ -1716,6 +1728,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"unknown Dell", STAC_DELL_M4_2),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264,
"unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02aa,
+ "unknown Dell", STAC_DELL_M4_3),
{} /* terminator */
};
@@ -3901,7 +3915,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
for (i = 0; i < cfg->speaker_outs; i++)
stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
AC_PINCTL_OUT_EN);
- if (spec->eapd_mask)
+ if (spec->eapd_mask && spec->eapd_switch)
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir, spec->gpio_data &
~spec->eapd_mask);
@@ -3916,7 +3930,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
for (i = 0; i < cfg->speaker_outs; i++)
stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
AC_PINCTL_OUT_EN);
- if (spec->eapd_mask)
+ if (spec->eapd_mask && spec->eapd_switch)
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir, spec->gpio_data |
spec->eapd_mask);
@@ -4243,6 +4257,7 @@ again:
spec->num_smuxes = 0;
spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
+ spec->eapd_switch = 0;
spec->num_amps = 1;
if (!spec->init)
@@ -4274,6 +4289,7 @@ again:
default:
spec->num_dmics = STAC92HD73XX_NUM_DMICS;
spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
+ spec->eapd_switch = 1;
}
if (spec->board_config > STAC_92HD73XX_REF) {
/* GPIO0 High = Enable EAPD */
@@ -4419,7 +4435,13 @@ static int stac92hd71xx_resume(struct hda_codec *codec)
static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state)
{
+ struct sigmatel_spec *spec = codec->spec;
+
stac92hd71xx_set_power_state(codec, AC_PWRST_D3);
+ if (spec->eapd_mask)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data &
+ ~spec->eapd_mask);
return 0;
};
@@ -4562,14 +4584,21 @@ again:
switch (spec->board_config) {
case STAC_HP_M4:
- spec->num_dmics = 0;
- spec->num_smuxes = 0;
- spec->num_dmuxes = 0;
-
/* enable internal microphone */
stac92xx_set_config_reg(codec, 0x0e, 0x01813040);
stac92xx_auto_set_pinctl(codec, 0x0e,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_80);
+ /* fallthru */
+ case STAC_DELL_M4_2:
+ spec->num_dmics = 0;
+ spec->num_smuxes = 0;
+ spec->num_dmuxes = 0;
+ break;
+ case STAC_DELL_M4_1:
+ case STAC_DELL_M4_3:
+ spec->num_dmics = 1;
+ spec->num_smuxes = 0;
+ spec->num_dmuxes = 0;
break;
default:
spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
@@ -4806,6 +4835,7 @@ static int patch_stac927x(struct hda_codec *codec)
spec->num_pwrs = 0;
spec->aloopback_mask = 0x40;
spec->aloopback_shift = 0;
+ spec->eapd_switch = 1;
err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
if (!err) {
@@ -4886,6 +4916,7 @@ static int patch_stac9205(struct hda_codec *codec)
spec->aloopback_mask = 0x40;
spec->aloopback_shift = 0;
+ spec->eapd_switch = 1;
spec->multiout.dac_nids = spec->dac_nids;
switch (spec->board_config){
diff --git a/sound/sound_core.c b/sound/sound_core.c
index a75b289..10ba421 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -457,7 +457,7 @@ EXPORT_SYMBOL(unregister_sound_mixer);
void unregister_sound_midi(int unit)
{
- return sound_remove_unit(&chains[2], unit);
+ sound_remove_unit(&chains[2], unit);
}
EXPORT_SYMBOL(unregister_sound_midi);
@@ -474,7 +474,7 @@ EXPORT_SYMBOL(unregister_sound_midi);
void unregister_sound_dsp(int unit)
{
- return sound_remove_unit(&chains[3], unit);
+ sound_remove_unit(&chains[3], unit);
}
@@ -507,7 +507,7 @@ static struct sound_unit *__look_for_unit(int chain, int unit)
return NULL;
}
-int soundcore_open(struct inode *inode, struct file *file)
+static int soundcore_open(struct inode *inode, struct file *file)
{
int chain;
int unit = iminor(inode);
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-11-18 15:16 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-11-18 15:16 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA fixes for 2.6.28-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains several STAC/IDT HD-audio codec fixes (for recent Dell
and HP laptops), and one trivial error-path fix in pcxhr driver.
Thanks!
Takashi
===
Julia Lawall (1):
ALSA: sound/pci/pcxhr/pcxhr.c: introduce missing kfree and pci_disable_device
Matthew Ranostay (1):
ALSA: hda: STAC_VREF_EVENT value change
Takashi Iwai (6):
ALSA: hda - Add digital beep playback switch for STAC/IDT codecs
ALSA: hda - Missing NULL check in hda_beep.c
ALSA: hda - Check model type instead of SSID in patch_92hd71bxx()
ALSA: hda - Fix GPIO initialization in patch_stac92hd71bxx()
ALSA: hda - Add quirks for HP Pavilion DV models
ALSA: hda - Fix resume of GPIO unsol event for STAC/IDT
---
sound/pci/hda/hda_beep.c | 8 +++
sound/pci/hda/hda_beep.h | 1 +
sound/pci/hda/patch_sigmatel.c | 99 ++++++++++++++++++++++++++++++++-------
sound/pci/pcxhr/pcxhr.c | 5 ++-
4 files changed, 94 insertions(+), 19 deletions(-)
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 9b77b3e..3ecd7e7 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -37,6 +37,9 @@ static void snd_hda_generate_beep(struct work_struct *work)
container_of(work, struct hda_beep, beep_work);
struct hda_codec *codec = beep->codec;
+ if (!beep->enabled)
+ return;
+
/* generate tone */
snd_hda_codec_write_cache(codec, beep->nid, 0,
AC_VERB_SET_BEEP_CONTROL, beep->tone);
@@ -85,6 +88,10 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
snprintf(beep->phys, sizeof(beep->phys),
"card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
input_dev = input_allocate_device();
+ if (!input_dev) {
+ kfree(beep);
+ return -ENOMEM;
+ }
/* setup digital beep device */
input_dev->name = "HDA Digital PCBeep";
@@ -115,6 +122,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
beep->nid = nid;
beep->dev = input_dev;
beep->codec = codec;
+ beep->enabled = 1;
codec->beep = beep;
INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index de4036e..b9679f0 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -31,6 +31,7 @@ struct hda_beep {
char phys[32];
int tone;
int nid;
+ int enabled;
struct work_struct beep_work; /* scheduled task for beep event */
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4300a67..9563b5b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -36,9 +36,11 @@
#include "hda_beep.h"
#define NUM_CONTROL_ALLOC 32
+
+#define STAC_VREF_EVENT 0x00
+#define STAC_INSERT_EVENT 0x10
#define STAC_PWR_EVENT 0x20
#define STAC_HP_EVENT 0x30
-#define STAC_VREF_EVENT 0x40
enum {
STAC_REF,
@@ -1686,6 +1688,10 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
/* SigmaTel reference board */
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
"DFI LanParty", STAC_92HD71BXX_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
+ "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, 0x361a,
"unknown HP", STAC_HP_M4),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
@@ -2587,8 +2593,10 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
};
/* add dynamic controls */
-static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type,
- int idx, const char *name, unsigned long val)
+static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
+ struct snd_kcontrol_new *ktemp,
+ int idx, const char *name,
+ unsigned long val)
{
struct snd_kcontrol_new *knew;
@@ -2607,20 +2615,29 @@ static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type,
}
knew = &spec->kctl_alloc[spec->num_kctl_used];
- *knew = stac92xx_control_templates[type];
+ *knew = *ktemp;
knew->index = idx;
knew->name = kstrdup(name, GFP_KERNEL);
- if (! knew->name)
+ if (!knew->name)
return -ENOMEM;
knew->private_value = val;
spec->num_kctl_used++;
return 0;
}
+static inline int stac92xx_add_control_idx(struct sigmatel_spec *spec,
+ int type, int idx, const char *name,
+ unsigned long val)
+{
+ return stac92xx_add_control_temp(spec,
+ &stac92xx_control_templates[type],
+ idx, name, val);
+}
+
/* add dynamic controls */
-static int stac92xx_add_control(struct sigmatel_spec *spec, int type,
- const char *name, unsigned long val)
+static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type,
+ const char *name, unsigned long val)
{
return stac92xx_add_control_idx(spec, type, 0, name, val);
}
@@ -3062,6 +3079,43 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
return 0;
}
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+#define stac92xx_dig_beep_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_dig_beep_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = codec->beep->enabled;
+ return 0;
+}
+
+static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int enabled = !!ucontrol->value.integer.value[0];
+ if (codec->beep->enabled != enabled) {
+ codec->beep->enabled = enabled;
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = stac92xx_dig_beep_switch_info,
+ .get = stac92xx_dig_beep_switch_get,
+ .put = stac92xx_dig_beep_switch_put,
+};
+
+static int stac92xx_beep_switch_ctl(struct hda_codec *codec)
+{
+ return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl,
+ 0, "PC Beep Playback Switch", 0);
+}
+#endif
+
static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -3368,6 +3422,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
#ifdef CONFIG_SND_HDA_INPUT_BEEP
if (spec->digbeep_nid > 0) {
hda_nid_t nid = spec->digbeep_nid;
+ unsigned int caps;
err = stac92xx_auto_create_beep_ctls(codec, nid);
if (err < 0)
@@ -3375,6 +3430,14 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
err = snd_hda_attach_beep_device(codec, nid);
if (err < 0)
return err;
+ /* if no beep switch is available, make its own one */
+ caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+ if (codec->beep &&
+ !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) {
+ err = stac92xx_beep_switch_ctl(codec);
+ if (err < 0)
+ return err;
+ }
}
#endif
@@ -4419,6 +4482,13 @@ again:
stac92xx_set_config_regs(codec);
}
+ if (spec->board_config > STAC_92HD71BXX_REF) {
+ /* GPIO0 = EAPD */
+ spec->gpio_mask = 0x01;
+ spec->gpio_dir = 0x01;
+ spec->gpio_data = 0x01;
+ }
+
switch (codec->vendor_id) {
case 0x111d76b6: /* 4 Port without Analog Mixer */
case 0x111d76b7:
@@ -4429,10 +4499,10 @@ again:
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
break;
case 0x111d7608: /* 5 Port with Analog Mixer */
- switch (codec->subsystem_id) {
- case 0x103c361a:
+ switch (spec->board_config) {
+ case STAC_HP_M4:
/* Enable VREF power saving on GPIO1 detect */
- snd_hda_codec_write(codec, codec->afg, 0,
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02);
snd_hda_codec_write_cache(codec, codec->afg, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
@@ -4478,13 +4548,6 @@ again:
spec->aloopback_mask = 0x50;
spec->aloopback_shift = 0;
- if (spec->board_config > STAC_92HD71BXX_REF) {
- /* GPIO0 = EAPD */
- spec->gpio_mask = 0x01;
- spec->gpio_dir = 0x01;
- spec->gpio_data = 0x01;
- }
-
spec->powerdown_adcs = 1;
spec->digbeep_nid = 0x26;
spec->mux_nids = stac92hd71bxx_mux_nids;
@@ -4832,7 +4895,7 @@ static int patch_stac9205(struct hda_codec *codec)
stac92xx_set_config_reg(codec, 0x20, 0x1c410030);
/* Enable unsol response for GPIO4/Dock HP connection */
- snd_hda_codec_write(codec, codec->afg, 0,
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10);
snd_hda_codec_write_cache(codec, codec->afg, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 0e06c6c..73de6e9 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1229,8 +1229,11 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
return -ENOMEM;
}
- if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST))
+ if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) {
+ kfree(mgr);
+ pci_disable_device(pci);
return -ENODEV;
+ }
card_name = pcxhr_board_params[pci_id->driver_data].board_name;
mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips;
mgr->capture_chips = pcxhr_board_params[pci_id->driver_data].capture_chips;
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-11-12 15:56 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-11-12 15:56 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for 2.6.28-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains fixes for bugs with STAC/IDT HD-audio codecs, typically
hitting Dell laptops.
Thanks!
Takashi
===
Takashi Iwai (3):
ALSA: hda - Add missing analog-mux mixer creation for STAC9200
ALSA: hda - Fix input pin initialization for STAC/IDT codecs
ALSA: hda - Fix IDT/STAC multiple HP detection
---
sound/pci/hda/patch_sigmatel.c | 85 +++++++++++++++++++++++++++++----------
1 files changed, 63 insertions(+), 22 deletions(-)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e608591..4300a67 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -212,7 +212,7 @@ struct sigmatel_spec {
/* i/o switches */
unsigned int io_switch[2];
unsigned int clfe_swap;
- unsigned int hp_switch;
+ unsigned int hp_switch; /* NID of HP as line-out */
unsigned int aloopback;
struct hda_pcm pcm_rec[2]; /* PCM information */
@@ -2443,7 +2443,7 @@ static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct sigmatel_spec *spec = codec->spec;
- ucontrol->value.integer.value[0] = spec->hp_switch;
+ ucontrol->value.integer.value[0] = !!spec->hp_switch;
return 0;
}
@@ -2452,8 +2452,9 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct sigmatel_spec *spec = codec->spec;
-
- spec->hp_switch = ucontrol->value.integer.value[0];
+ int nid = kcontrol->private_value;
+
+ spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0;
/* check to be sure that the ports are upto date with
* switch changes
@@ -2862,7 +2863,8 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
if (cfg->hp_outs > 1) {
err = stac92xx_add_control(spec,
STAC_CTL_WIDGET_HP_SWITCH,
- "Headphone as Line Out Switch", 0);
+ "Headphone as Line Out Switch",
+ cfg->hp_pins[cfg->hp_outs - 1]);
if (err < 0)
return err;
}
@@ -3530,6 +3532,12 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
if ((err = stac9200_auto_create_lfe_ctls(codec, &spec->autocfg)) < 0)
return err;
+ if (spec->num_muxes > 0) {
+ err = stac92xx_auto_create_mux_input_ctls(codec);
+ if (err < 0)
+ return err;
+ }
+
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = 0x05;
if (spec->autocfg.dig_in_pin)
@@ -3647,14 +3655,18 @@ static int stac92xx_init(struct hda_codec *codec)
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = cfg->input_pins[i];
if (nid) {
- unsigned int pinctl = snd_hda_codec_read(codec, nid,
- 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- /* if PINCTL already set then skip */
- if (pinctl & AC_PINCAP_IN)
- continue;
- pinctl = AC_PINCTL_IN_EN;
- if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC)
- pinctl |= stac92xx_get_vref(codec, nid);
+ unsigned int pinctl;
+ if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) {
+ /* for mic pins, force to initialize */
+ pinctl = stac92xx_get_vref(codec, nid);
+ } else {
+ pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ /* if PINCTL already set then skip */
+ if (pinctl & AC_PINCTL_IN_EN)
+ continue;
+ }
+ pinctl |= AC_PINCTL_IN_EN;
stac92xx_auto_set_pinctl(codec, nid, pinctl);
}
}
@@ -3776,11 +3788,30 @@ static int get_hp_pin_presence(struct hda_codec *codec, hda_nid_t nid)
return 0;
}
+/* return non-zero if the hp-pin of the given array index isn't
+ * a jack-detection target
+ */
+static int no_hp_sensing(struct sigmatel_spec *spec, int i)
+{
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+
+ /* ignore sensing of shared line and mic jacks */
+ if (spec->line_switch &&
+ cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_LINE])
+ return 1;
+ if (spec->mic_switch &&
+ cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_MIC])
+ return 1;
+ /* ignore if the pin is set as line-out */
+ if (cfg->hp_pins[i] == spec->hp_switch)
+ return 1;
+ return 0;
+}
+
static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
{
struct sigmatel_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
- int nid = cfg->hp_pins[cfg->hp_outs - 1];
int i, presence;
presence = 0;
@@ -3791,15 +3822,16 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
for (i = 0; i < cfg->hp_outs; i++) {
if (presence)
break;
- if (spec->hp_switch && cfg->hp_pins[i] == nid)
- break;
+ if (no_hp_sensing(spec, i))
+ continue;
presence = get_hp_pin_presence(codec, cfg->hp_pins[i]);
}
if (presence) {
- /* disable lineouts, enable hp */
+ /* disable lineouts */
if (spec->hp_switch)
- stac92xx_reset_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+ stac92xx_reset_pinctl(codec, spec->hp_switch,
+ AC_PINCTL_OUT_EN);
for (i = 0; i < cfg->line_outs; i++)
stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
@@ -3811,9 +3843,10 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
spec->gpio_dir, spec->gpio_data &
~spec->eapd_mask);
} else {
- /* enable lineouts, disable hp */
+ /* enable lineouts */
if (spec->hp_switch)
- stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+ stac92xx_set_pinctl(codec, spec->hp_switch,
+ AC_PINCTL_OUT_EN);
for (i = 0; i < cfg->line_outs; i++)
stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
@@ -3825,8 +3858,16 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
spec->gpio_dir, spec->gpio_data |
spec->eapd_mask);
}
- if (!spec->hp_switch && cfg->hp_outs > 1 && presence)
- stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+ /* toggle hp outs */
+ for (i = 0; i < cfg->hp_outs; i++) {
+ unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
+ if (no_hp_sensing(spec, i))
+ continue;
+ if (presence)
+ stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
+ else
+ stac92xx_reset_pinctl(codec, cfg->hp_pins[i], val);
+ }
}
static void stac92xx_pin_sense(struct hda_codec *codec, int idx)
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-11-10 17:03 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-11-10 17:03 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for 2.6.28-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following trivial fixes.
Thanks!
Takashi
===
Andrew Morton (1):
alsa: fix snd_BUG_on() and friends
Michel Marti (1):
ALSA: hda - Add another HP model (6730s) for AD1884A
Takashi Iwai (2):
ALSA: hda - Limit the number of GPIOs show in proc
ALSA: hda - Add a quirk for MEDION MD96630
Tim Blechmann (2):
ALSA: HDSP: check for io box before uploading firmware
ALSA: hdsp: check for iobox and upload firmware during ioctl
Travis Place (1):
ALSA: hda - Make the HP EliteBook 8530p use AD1884A model laptop
Ville Syrjala (1):
ALSA: gusextreme: Fix build errors
---
include/sound/core.h | 10 +++++++---
sound/isa/Kconfig | 2 +-
sound/pci/hda/hda_proc.c | 2 ++
sound/pci/hda/patch_analog.c | 2 ++
sound/pci/hda/patch_realtek.c | 1 +
sound/pci/rme9652/hdsp.c | 27 ++++++++++++++++++++++-----
6 files changed, 35 insertions(+), 9 deletions(-)
diff --git a/include/sound/core.h b/include/sound/core.h
index 35424a9..1508c4e 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -385,9 +385,13 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
#else /* !CONFIG_SND_DEBUG */
-#define snd_printd(fmt, args...) /* nothing */
-#define snd_BUG() /* nothing */
-#define snd_BUG_ON(cond) ({/*(void)(cond);*/ 0;}) /* always false */
+#define snd_printd(fmt, args...) do { } while (0)
+#define snd_BUG() do { } while (0)
+static inline int __snd_bug_on(void)
+{
+ return 0;
+}
+#define snd_BUG_ON(cond) __snd_bug_on() /* always false */
#endif /* CONFIG_SND_DEBUG */
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 660beb4..ce0aa04 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -211,7 +211,7 @@ config SND_GUSCLASSIC
config SND_GUSEXTREME
tristate "Gravis UltraSound Extreme"
- select SND_HWDEP
+ select SND_OPL3_LIB
select SND_MPU401_UART
select SND_PCM
help
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 743d779..c39af98 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -483,6 +483,8 @@ static void print_gpio(struct snd_info_buffer *buffer,
(gpio & AC_GPIO_UNSOLICITED) ? 1 : 0,
(gpio & AC_GPIO_WAKE) ? 1 : 0);
max = gpio & AC_GPIO_IO_COUNT;
+ if (!max || max > 8)
+ return;
enable = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_MASK, 0);
direction = snd_hda_codec_read(codec, nid, 0,
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d3fd432..686c774 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3861,6 +3861,8 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
+ SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
+ SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a4666c9..a378c01 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8469,6 +8469,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+ SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index d723543..736246f 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -4548,11 +4548,20 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
{
struct hdsp *hdsp = (struct hdsp *)hw->private_data;
void __user *argp = (void __user *)arg;
+ int err;
switch (cmd) {
case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: {
struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg;
+ err = hdsp_check_for_iobox(hdsp);
+ if (err < 0)
+ return err;
+
+ err = hdsp_check_for_firmware(hdsp, 1);
+ if (err < 0)
+ return err;
+
if (!(hdsp->state & HDSP_FirmwareLoaded)) {
snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n");
return -EINVAL;
@@ -4572,10 +4581,14 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
unsigned long flags;
int i;
- if (!(hdsp->state & HDSP_FirmwareLoaded)) {
- snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n");
- return -EINVAL;
- }
+ err = hdsp_check_for_iobox(hdsp);
+ if (err < 0)
+ return err;
+
+ err = hdsp_check_for_firmware(hdsp, 1);
+ if (err < 0)
+ return err;
+
spin_lock_irqsave(&hdsp->lock, flags);
info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp);
info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp);
@@ -5045,6 +5058,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
/* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
ssleep(2);
+ err = hdsp_check_for_iobox(hdsp);
+ if (err < 0)
+ return err;
+
if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
#ifdef HDSP_FW_LOADER
if ((err = hdsp_request_fw_loader(hdsp)) < 0)
@@ -5057,7 +5074,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
/* init is complete, we return */
return 0;
#endif
- /* no iobox connected, we defer initialization */
+ /* we defer initialization */
snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n");
if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
return err;
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-11-03 15:42 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-11-03 15:42 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Hi Linus,
please pull ALSA updates for 2.6.28-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
including dev->bus_id removals, a few HD-audio fixes and other trivial
fixes as below.
Thanks!
Takashi
===
Kay Sievers (1):
sound: struct device - replace bus_id with dev_name(), dev_set_name()
Matthew Ranostay (1):
ALSA: hda: make a STAC_DELL_EQ option
Takashi Iwai (6):
ALSA: hda - Disable broken mic auto-muting in Realtek codes
ALSA: hda - Add digital-mic for ALC269 auto-probe mode
ALSA: rawmidi - Add open check in rawmidi callbacks
ALSA: remove direct access of dev->bus_id in sound/isa/*
ALSA: hda - Add a quirk for another Acer Aspire (1025:0090)
ALSA: emu10k1 - Add more invert_shared_spdif flag to Audigy models
Zoltan Devai (1):
ALSA: Fix PIT lockup on some chipsets when using the PC-Speaker
---
sound/aoa/soundbus/core.c | 2 +-
sound/core/rawmidi.c | 8 ++++++++
sound/drivers/ml403-ac97cr.c | 4 ++--
sound/drivers/pcsp/pcsp_input.c | 4 ++--
sound/isa/ad1848/ad1848.c | 6 +++---
sound/isa/adlib.c | 12 ++++++------
sound/isa/cs423x/cs4231.c | 8 ++++----
sound/isa/cs423x/cs4236.c | 8 ++++----
sound/isa/es1688/es1688.c | 9 +++------
sound/isa/gus/gusclassic.c | 13 +++++--------
sound/isa/gus/gusextreme.c | 19 +++++++------------
sound/isa/sb/sb8.c | 4 ++--
sound/pci/emu10k1/emu10k1_main.c | 3 +++
sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++++++++++++--
sound/pci/hda/patch_sigmatel.c | 16 ++++++++++------
sound/soc/soc-core.c | 4 ++--
16 files changed, 89 insertions(+), 60 deletions(-)
diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c
index f84f3e5..fa8ab28 100644
--- a/sound/aoa/soundbus/core.c
+++ b/sound/aoa/soundbus/core.c
@@ -176,7 +176,7 @@ int soundbus_add_one(struct soundbus_dev *dev)
return -EINVAL;
}
- snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount);
+ dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount);
dev->ofdev.dev.bus = &soundbus_bus_type;
return of_device_register(&dev->ofdev);
}
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index c4995c9..39672f6 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -148,6 +148,8 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream)
static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up)
{
+ if (!substream->opened)
+ return;
if (up) {
tasklet_hi_schedule(&substream->runtime->tasklet);
} else {
@@ -158,6 +160,8 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs
static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
{
+ if (!substream->opened)
+ return;
substream->ops->trigger(substream, up);
if (!up && substream->runtime->event)
tasklet_kill(&substream->runtime->tasklet);
@@ -857,6 +861,8 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
int result = 0, count1;
struct snd_rawmidi_runtime *runtime = substream->runtime;
+ if (!substream->opened)
+ return -EBADFD;
if (runtime->buffer == NULL) {
snd_printd("snd_rawmidi_receive: input is not active!!!\n");
return -EINVAL;
@@ -1126,6 +1132,8 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count)
int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
unsigned char *buffer, int count)
{
+ if (!substream->opened)
+ return -EBADFD;
count = snd_rawmidi_transmit_peek(substream, buffer, count);
if (count < 0)
return count;
diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
index ecdbeb6..7783843 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
/* get irq */
irq = platform_get_irq(pfdev, 0);
if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
- pfdev->dev.bus_id, (void *)ml403_ac97cr)) {
+ dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
"unable to grab IRQ %d\n",
irq);
@@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
ml403_ac97cr->irq);
irq = platform_get_irq(pfdev, 1);
if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
- pfdev->dev.bus_id, (void *)ml403_ac97cr)) {
+ dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
"unable to grab IRQ %d\n",
irq);
diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c
index cd9b83e..0444cde 100644
--- a/sound/drivers/pcsp/pcsp_input.c
+++ b/sound/drivers/pcsp/pcsp_input.c
@@ -24,13 +24,13 @@ static void pcspkr_do_sound(unsigned int count)
spin_lock_irqsave(&i8253_lock, flags);
if (count) {
- /* enable counter 2 */
- outb_p(inb_p(0x61) | 3, 0x61);
/* set command for counter 2, 2 byte write */
outb_p(0xB6, 0x43);
/* select desired HZ */
outb_p(count & 0xff, 0x42);
outb((count >> 8) & 0xff, 0x42);
+ /* enable counter 2 */
+ outb_p(inb_p(0x61) | 3, 0x61);
} else {
/* disable counter 2 */
outb(inb_p(0x61) & 0xFC, 0x61);
diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c
index b68d20e..223a6c0 100644
--- a/sound/isa/ad1848/ad1848.c
+++ b/sound/isa/ad1848/ad1848.c
@@ -70,15 +70,15 @@ static int __devinit snd_ad1848_match(struct device *dev, unsigned int n)
return 0;
if (port[n] == SNDRV_AUTO_PORT) {
- snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
+ dev_err(dev, "please specify port\n");
return 0;
}
if (irq[n] == SNDRV_AUTO_IRQ) {
- snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id);
+ dev_err(dev, "please specify irq\n");
return 0;
}
if (dma1[n] == SNDRV_AUTO_DMA) {
- snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id);
+ dev_err(dev, "please specify dma1\n");
return 0;
}
return 1;
diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c
index efa8c80..374b717 100644
--- a/sound/isa/adlib.c
+++ b/sound/isa/adlib.c
@@ -36,7 +36,7 @@ static int __devinit snd_adlib_match(struct device *dev, unsigned int n)
return 0;
if (port[n] == SNDRV_AUTO_PORT) {
- snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
+ dev_err(dev, "please specify port\n");
return 0;
}
return 1;
@@ -55,13 +55,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
if (!card) {
- snd_printk(KERN_ERR "%s: could not create card\n", dev->bus_id);
+ dev_err(dev, "could not create card\n");
return -EINVAL;
}
card->private_data = request_region(port[n], 4, CRD_NAME);
if (!card->private_data) {
- snd_printk(KERN_ERR "%s: could not grab ports\n", dev->bus_id);
+ dev_err(dev, "could not grab ports\n");
error = -EBUSY;
goto out;
}
@@ -73,13 +73,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
error = snd_opl3_create(card, port[n], port[n] + 2, OPL3_HW_AUTO, 1, &opl3);
if (error < 0) {
- snd_printk(KERN_ERR "%s: could not create OPL\n", dev->bus_id);
+ dev_err(dev, "could not create OPL\n");
goto out;
}
error = snd_opl3_hwdep_new(opl3, 0, 0, NULL);
if (error < 0) {
- snd_printk(KERN_ERR "%s: could not create FM\n", dev->bus_id);
+ dev_err(dev, "could not create FM\n");
goto out;
}
@@ -87,7 +87,7 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
error = snd_card_register(card);
if (error < 0) {
- snd_printk(KERN_ERR "%s: could not register card\n", dev->bus_id);
+ dev_err(dev, "could not register card\n");
goto out;
}
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
index ddd2891..f019d44 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -74,15 +74,15 @@ static int __devinit snd_cs4231_match(struct device *dev, unsigned int n)
return 0;
if (port[n] == SNDRV_AUTO_PORT) {
- snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id);
+ dev_err(dev, "please specify port\n");
return 0;
}
if (irq[n] == SNDRV_AUTO_IRQ) {
- snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id);
+ dev_err(dev, "please specify irq\n");
return 0;
}
if (dma1[n] == SNDRV_AUTO_DMA) {
- snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id);
+ dev_err(dev, "please specify dma1\n");
return 0;
}
return 1;
@@ -133,7 +133,7 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n)
mpu_port[n], 0, mpu_irq[n],
mpu_irq[n] >= 0 ? IRQF_DISABLED : 0,
NULL) < 0)
- printk(KERN_WARNING "%s: MPU401 not detected\n", dev->bus_id);
+ dev_warn(dev, "MPU401 not detected\n");
}
snd_card_set_dev(card, dev);
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 91f9c15..019c940 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -488,19 +488,19 @@ static int __devinit snd_cs423x_isa_match(struct device *pdev,
return 0;
if (port[dev] == SNDRV_AUTO_PORT) {
- snd_printk(KERN_ERR "%s: please specify port\n", pdev->bus_id);
+ dev_err(pdev, "please specify port\n");
return 0;
}
if (cport[dev] == SNDRV_AUTO_PORT) {
- snd_printk(KERN_ERR "%s: please specify cport\n", pdev->bus_id);
+ dev_err(pdev, "please specify cport\n");
return 0;
}
if (irq[dev] == SNDRV_AUTO_IRQ) {
- snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id);
+ dev_err(pdev, "please specify irq\n");
return 0;
}
if (dma1[dev] == SNDRV_AUTO_DMA) {
- snd_printk(KERN_ERR "%s: please specify dma1\n", pdev->bus_id);
+ dev_err(pdev, "please specify dma1\n");
return 0;
}
return 1;
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index f88639e..b463771 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -88,16 +88,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card,
if (irq[n] == SNDRV_AUTO_IRQ) {
irq[n] = snd_legacy_find_free_irq(possible_irqs);
if (irq[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free IRQ\n",
- dev->bus_id);
+ dev_err(dev, "unable to find a free IRQ\n");
return -EBUSY;
}
}
if (dma8[n] == SNDRV_AUTO_DMA) {
dma8[n] = snd_legacy_find_free_dma(possible_dmas);
if (dma8[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free DMA\n",
- dev->bus_id);
+ dev_err(dev, "unable to find a free DMA\n");
return -EBUSY;
}
}
@@ -147,8 +145,7 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n)
if (snd_opl3_create(card, chip->port, chip->port + 2,
OPL3_HW_OPL3, 0, &opl3) < 0)
- printk(KERN_WARNING "%s: opl3 not detected at 0x%lx\n",
- dev->bus_id, chip->port);
+ dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port);
else {
error = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
if (error < 0)
diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c
index 8f914b3..426532a 100644
--- a/sound/isa/gus/gusclassic.c
+++ b/sound/isa/gus/gusclassic.c
@@ -90,24 +90,21 @@ static int __devinit snd_gusclassic_create(struct snd_card *card,
if (irq[n] == SNDRV_AUTO_IRQ) {
irq[n] = snd_legacy_find_free_irq(possible_irqs);
if (irq[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free IRQ\n",
- dev->bus_id);
+ dev_err(dev, "unable to find a free IRQ\n");
return -EBUSY;
}
}
if (dma1[n] == SNDRV_AUTO_DMA) {
dma1[n] = snd_legacy_find_free_dma(possible_dmas);
if (dma1[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free DMA1\n",
- dev->bus_id);
+ dev_err(dev, "unable to find a free DMA1\n");
return -EBUSY;
}
}
if (dma2[n] == SNDRV_AUTO_DMA) {
dma2[n] = snd_legacy_find_free_dma(possible_dmas);
if (dma2[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free DMA2\n",
- dev->bus_id);
+ dev_err(dev, "unable to find a free DMA2\n");
return -EBUSY;
}
}
@@ -174,8 +171,8 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n)
error = -ENODEV;
if (gus->max_flag || gus->ess_flag) {
- snd_printk(KERN_ERR "%s: GUS Classic or ACE soundcard was "
- "not detected at 0x%lx\n", dev->bus_id, gus->gf1.port);
+ dev_err(dev, "GUS Classic or ACE soundcard was "
+ "not detected at 0x%lx\n", gus->gf1.port);
goto out;
}
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index da13185..7ad4c3b 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -106,16 +106,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card,
if (irq[n] == SNDRV_AUTO_IRQ) {
irq[n] = snd_legacy_find_free_irq(possible_irqs);
if (irq[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free IRQ "
- "for ES1688\n", dev->bus_id);
+ dev_err(dev, "unable to find a free IRQ for ES1688\n");
return -EBUSY;
}
}
if (dma8[n] == SNDRV_AUTO_DMA) {
dma8[n] = snd_legacy_find_free_dma(possible_dmas);
if (dma8[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free DMA "
- "for ES1688\n", dev->bus_id);
+ dev_err(dev, "unable to find a free DMA for ES1688\n");
return -EBUSY;
}
}
@@ -143,16 +141,14 @@ static int __devinit snd_gusextreme_gus_card_create(struct snd_card *card,
if (gf1_irq[n] == SNDRV_AUTO_IRQ) {
gf1_irq[n] = snd_legacy_find_free_irq(possible_irqs);
if (gf1_irq[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free IRQ "
- "for GF1\n", dev->bus_id);
+ dev_err(dev, "unable to find a free IRQ for GF1\n");
return -EBUSY;
}
}
if (dma1[n] == SNDRV_AUTO_DMA) {
dma1[n] = snd_legacy_find_free_dma(possible_dmas);
if (dma1[n] < 0) {
- snd_printk(KERN_ERR "%s: unable to find a free DMA "
- "for GF1\n", dev->bus_id);
+ dev_err(dev, "unable to find a free DMA for GF1\n");
return -EBUSY;
}
}
@@ -278,8 +274,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
error = -ENODEV;
if (!gus->ess_flag) {
- snd_printk(KERN_ERR "%s: GUS Extreme soundcard was not "
- "detected at 0x%lx\n", dev->bus_id, gus->gf1.port);
+ dev_err(dev, "GUS Extreme soundcard was not "
+ "detected at 0x%lx\n", gus->gf1.port);
goto out;
}
gus->codec_flag = 1;
@@ -310,8 +306,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
if (snd_opl3_create(card, es1688->port, es1688->port + 2,
OPL3_HW_OPL3, 0, &opl3) < 0)
- printk(KERN_ERR "%s: opl3 not detected at 0x%lx\n",
- dev->bus_id, es1688->port);
+ dev_warn(dev, "opl3 not detected at 0x%lx\n", es1688->port);
else {
error = snd_opl3_hwdep_new(opl3, 0, 2, NULL);
if (error < 0)
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index 336a342..667eccc 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -85,11 +85,11 @@ static int __devinit snd_sb8_match(struct device *pdev, unsigned int dev)
if (!enable[dev])
return 0;
if (irq[dev] == SNDRV_AUTO_IRQ) {
- snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id);
+ dev_err(pdev, "please specify irq\n");
return 0;
}
if (dma8[dev] == SNDRV_AUTO_DMA) {
- snd_printk(KERN_ERR "%s: please specify dma8\n", pdev->bus_id);
+ dev_err(pdev, "please specify dma8\n");
return 0;
}
return 1;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 2f283ea..de5ee8f 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1464,6 +1464,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0151_chip = 1,
.spk71 = 1,
.spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102,
.driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]",
@@ -1473,6 +1474,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0151_chip = 1,
.spk71 = 1,
.spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102,
.driver = "Audigy2", .name = "Audigy 2 ZS [2001]",
@@ -1482,6 +1484,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0151_chip = 1,
.spk71 = 1,
.spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
.ac97_chip = 1} ,
/* Audigy 2 */
/* Tested by James@superbug.co.uk 3rd July 2005 */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4eceab9..a4666c9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -829,6 +829,7 @@ static void alc_sku_automute(struct hda_codec *codec)
spec->jack_present ? 0 : PIN_OUT);
}
+#if 0 /* it's broken in some acses -- temporarily disabled */
static void alc_mic_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -849,6 +850,9 @@ static void alc_mic_automute(struct hda_codec *codec)
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
+#else
+#define alc_mic_automute(codec) /* NOP */
+#endif /* disabled */
/* unsolicited event for HP jack sensing */
static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -1058,12 +1062,14 @@ do_sku:
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC880_HP_EVENT);
+#if 0 /* it's broken in some acses -- temporarily disabled */
if (spec->autocfg.input_pins[AUTO_PIN_MIC] &&
spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC])
snd_hda_codec_write(codec,
spec->autocfg.input_pins[AUTO_PIN_MIC], 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC880_MIC_EVENT);
+#endif /* disabled */
spec->unsol_event = alc_sku_unsol_event;
}
@@ -8408,6 +8414,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
@@ -12238,8 +12245,26 @@ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
return 0;
}
-#define alc269_auto_create_analog_input_ctls \
- alc880_auto_create_analog_input_ctls
+static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int err;
+
+ err = alc880_auto_create_analog_input_ctls(spec, cfg);
+ if (err < 0)
+ return err;
+ /* digital-mic input pin is excluded in alc880_auto_create..()
+ * because it's under 0x18
+ */
+ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
+ cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
+ struct hda_input_mux *imux = &spec->private_imux;
+ imux->items[imux->num_items].label = "Int Mic";
+ imux->items[imux->num_items].index = 0x05;
+ imux->num_items++;
+ }
+ return 0;
+}
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc269_loopbacks alc880_loopbacks
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index df9b0bc..e608591 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -69,6 +69,7 @@ enum {
enum {
STAC_92HD73XX_REF,
STAC_DELL_M6,
+ STAC_DELL_EQ,
STAC_92HD73XX_MODELS
};
@@ -773,9 +774,7 @@ static struct hda_verb dell_eq_core_init[] = {
};
static struct hda_verb dell_m6_core_init[] = {
- /* set master volume to max value without distortion
- * and direct control */
- { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* setup audio connections */
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -1600,11 +1599,13 @@ static unsigned int dell_m6_pin_configs[13] = {
static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
[STAC_92HD73XX_REF] = ref92hd73xx_pin_configs,
[STAC_DELL_M6] = dell_m6_pin_configs,
+ [STAC_DELL_EQ] = dell_m6_pin_configs,
};
static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
[STAC_92HD73XX_REF] = "ref",
[STAC_DELL_M6] = "dell-m6",
+ [STAC_DELL_EQ] = "dell-eq",
};
static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
@@ -4131,12 +4132,17 @@ again:
sizeof(stac92hd73xx_dmux));
switch (spec->board_config) {
- case STAC_DELL_M6:
+ case STAC_DELL_EQ:
spec->init = dell_eq_core_init;
+ /* fallthru */
+ case STAC_DELL_M6:
spec->num_smuxes = 0;
spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
spec->num_amps = 1;
+
+ if (!spec->init)
+ spec->init = dell_m6_core_init;
switch (codec->subsystem_id) {
case 0x1028025e: /* Analog Mics */
case 0x1028025f:
@@ -4146,8 +4152,6 @@ again:
break;
case 0x10280271: /* Digital Mics */
case 0x10280272:
- spec->init = dell_m6_core_init;
- /* fall-through */
case 0x10280254:
case 0x10280255:
stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a3adbf0..16c7453 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -95,8 +95,8 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
codec->ac97->dev.parent = NULL;
codec->ac97->dev.release = soc_ac97_device_release;
- snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
- codec->card->number, 0, codec->name);
+ dev_set_name(&codec->ac97->dev, "%d-%d:%s",
+ codec->card->number, 0, codec->name);
err = device_register(&codec->ac97->dev);
if (err < 0) {
snd_printk(KERN_ERR "Can't register ac97 bus\n");
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-10-29 18:17 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-10-29 18:17 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Hi Linus,
please pull ALSA updates for 2.6.28-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the fixes below. One trivial quirk addition for old
intel8x0, and others are for HD-audio fixes as usual, in addition
to a sanity check for breakage checks in the control codes.
Thanks!
Takashi
===
Bastien Nocera (1):
ALSA: intel8x0 - add Dell Optiplex GX620 (AD1981B) to AC97 clock whitelist
Mark Brown (2):
ALSA: hda: Add HDA vendor ID for Wolfson Microelectronics
ALSA: Warn when control names are truncated
Takashi Iwai (3):
ALSA: hda - Add another HP model for AD1884A
ALSA: hda - Fix SPDIF mute on IDT/STAC codecs
ALSA: hda - Add reboot notifier
---
sound/core/control.c | 7 ++++++-
sound/pci/hda/hda_codec.c | 1 +
sound/pci/hda/hda_intel.c | 29 +++++++++++++++++++++++++++++
sound/pci/hda/patch_analog.c | 1 +
sound/pci/hda/patch_sigmatel.c | 4 +---
sound/pci/intel8x0.c | 1 +
6 files changed, 39 insertions(+), 4 deletions(-)
diff --git a/sound/core/control.c b/sound/core/control.c
index 6d71f9a..b0bf426 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -225,8 +225,13 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol,
kctl.id.iface = ncontrol->iface;
kctl.id.device = ncontrol->device;
kctl.id.subdevice = ncontrol->subdevice;
- if (ncontrol->name)
+ if (ncontrol->name) {
strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name));
+ if (strcmp(ncontrol->name, kctl.id.name) != 0)
+ snd_printk(KERN_WARNING
+ "Control name '%s' truncated to '%s'\n",
+ ncontrol->name, kctl.id.name);
+ }
kctl.id.index = ncontrol->index;
kctl.count = ncontrol->count ? ncontrol->count : 1;
access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 6447754..ba1ab73 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -64,6 +64,7 @@ static struct hda_vendor_id hda_vendor_ids[] = {
{ 0x14f1, "Conexant" },
{ 0x17e8, "Chrontel" },
{ 0x1854, "LG" },
+ { 0x1aec, "Wolfson Microelectronics" },
{ 0x434d, "C-Media" },
{ 0x8384, "SigmaTel" },
{} /* terminator */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f080f8c..35722ec 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -45,6 +45,7 @@
#include <linux/slab.h>
#include <linux/pci.h>
#include <linux/mutex.h>
+#include <linux/reboot.h>
#include <sound/core.h>
#include <sound/initval.h>
#include "hda_codec.h"
@@ -397,6 +398,9 @@ struct azx {
/* for pending irqs */
struct work_struct irq_pending_work;
+
+ /* reboot notifier (for mysterious hangup problem at power-down) */
+ struct notifier_block reboot_notifier;
};
/* driver types */
@@ -1979,12 +1983,36 @@ static int azx_resume(struct pci_dev *pci)
/*
+ * reboot notifier for hang-up problem at power-down
+ */
+static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf)
+{
+ struct azx *chip = container_of(nb, struct azx, reboot_notifier);
+ azx_stop_chip(chip);
+ return NOTIFY_OK;
+}
+
+static void azx_notifier_register(struct azx *chip)
+{
+ chip->reboot_notifier.notifier_call = azx_halt;
+ register_reboot_notifier(&chip->reboot_notifier);
+}
+
+static void azx_notifier_unregister(struct azx *chip)
+{
+ if (chip->reboot_notifier.notifier_call)
+ unregister_reboot_notifier(&chip->reboot_notifier);
+}
+
+/*
* destructor
*/
static int azx_free(struct azx *chip)
{
int i;
+ azx_notifier_unregister(chip);
+
if (chip->initialized) {
azx_clear_irq_pending(chip);
for (i = 0; i < chip->num_streams; i++)
@@ -2348,6 +2376,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
pci_set_drvdata(pci, card);
chip->running = 1;
power_down_all_codecs(chip);
+ azx_notifier_register(chip);
dev++;
return err;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2b00c4a..d3fd432 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3860,6 +3860,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
+ SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 788fdc6..df9b0bc 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -566,10 +566,8 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
nid = codec->slave_dig_outs[smux_idx - 1];
if (spec->cur_smux[smux_idx] == smux->num_items - 1)
val = AMP_OUT_MUTE;
- if (smux_idx == 0)
- nid = spec->multiout.dig_out_nid;
else
- nid = codec->slave_dig_outs[smux_idx - 1];
+ val = AMP_OUT_UNMUTE;
/* un/mute SPDIF out */
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE, val);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index c88d1ea..19d3391 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2702,6 +2702,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = {
SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000),
SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100),
SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000),
+ SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),
SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000),
{ } /* terminator */
};
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-10-27 16:21 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-10-27 16:21 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for 2.6.28-rc1 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following fixes.
Thanks!
Takashi
==
Alan Cox (1):
sound: use a common working email address
Arjan van de Ven (1):
pci: use pci_ioremap_bar() in sound/
Cliff Cai (1):
ALSA: ASoC: Blackfin: update SPORT0 port selector (v2)
Takashi Iwai (1):
ALSA: hda - Restore default pin configs for realtek codecs
---
sound/oss/kahlua.c | 2 +-
sound/pci/ad1889.c | 2 +-
sound/pci/atiixp.c | 2 +-
sound/pci/atiixp_modem.c | 2 +-
sound/pci/au88x0/au88x0.c | 3 +-
sound/pci/bt87x.c | 3 +-
sound/pci/cs4281.c | 4 +-
sound/pci/cs5530.c | 4 +-
sound/pci/hda/hda_intel.c | 2 +-
sound/pci/hda/patch_realtek.c | 77 ++++++++++++++++++++++++++++++++++++++++
sound/pci/mixart/mixart.c | 3 +-
sound/soc/blackfin/bf5xx-i2s.c | 34 +++++++++++------
sound/sound_core.c | 2 +-
13 files changed, 112 insertions(+), 28 deletions(-)
diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c
index eb9bc36..c180598 100644
--- a/sound/oss/kahlua.c
+++ b/sound/oss/kahlua.c
@@ -1,7 +1,7 @@
/*
* Initialisation code for Cyrix/NatSemi VSA1 softaudio
*
- * (C) Copyright 2003 Red Hat Inc <alan@redhat.com>
+ * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
*
* XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
* The older version (VSA1) provides fairly good soundblaster emulation
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 92f3a97..a7f38e6 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card,
goto free_and_ret;
chip->bar = pci_resource_start(pci, 0);
- chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0));
+ chip->iobase = pci_ioremap_bar(pci, 0);
if (chip->iobase == NULL) {
printk(KERN_ERR PFX "unable to reserve region.\n");
err = -EBUSY;
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 085a52b..226fe82 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
return err;
}
chip->addr = pci_resource_start(pci, 0);
- chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0));
+ chip->remap_addr = pci_ioremap_bar(pci, 0);
if (chip->remap_addr == NULL) {
snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
snd_atiixp_free(chip);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 2f10630..0e6e5cc 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
return err;
}
chip->addr = pci_resource_start(pci, 0);
- chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0));
+ chip->remap_addr = pci_ioremap_bar(pci, 0);
if (chip->remap_addr == NULL) {
snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
snd_atiixp_free(chip);
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 68368e4..a36d4d1 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0)
goto regions_out;
- chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
- pci_resource_len(pci, 0));
+ chip->mmio = pci_ioremap_bar(pci, 0);
if (!chip->mmio) {
printk(KERN_ERR "MMIO area remap failed.\n");
err = -ENOMEM;
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 3aa8d97..1aa1c04 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card,
pci_disable_device(pci);
return err;
}
- chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
- pci_resource_len(pci, 0));
+ chip->mmio = pci_ioremap_bar(pci, 0);
if (!chip->mmio) {
snd_printk(KERN_ERR "cannot remap io memory\n");
err = -ENOMEM;
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index ef9308f..192e784 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card,
chip->ba0_addr = pci_resource_start(pci, 0);
chip->ba1_addr = pci_resource_start(pci, 1);
- chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0));
- chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1));
+ chip->ba0 = pci_ioremap_bar(pci, 0);
+ chip->ba1 = pci_ioremap_bar(pci, 1);
if (!chip->ba0 || !chip->ba1) {
snd_cs4281_free(chip);
return -ENOMEM;
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index 7ff8b68..6dea5b5 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -2,7 +2,7 @@
* cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio
*
* (C) Copyright 2007 Ash Willis <ashwillis@programmer.net>
- * (C) Copyright 2003 Red Hat Inc <alan@redhat.com>
+ * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
*
* This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did
* mess with it a bit. The chip seems to have to have trouble with full duplex
@@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card,
}
chip->pci_base = pci_resource_start(pci, 0);
- mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0));
+ mem = pci_ioremap_bar(pci, 0);
if (mem == NULL) {
kfree(chip);
pci_disable_device(pci);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9f316c1..f080f8c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2158,7 +2158,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
chip->addr = pci_resource_start(pci, 0);
- chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0));
+ chip->remap_addr = pci_ioremap_bar(pci, 0);
if (chip->remap_addr == NULL) {
snd_printk(KERN_ERR SFX "ioremap error\n");
err = -ENXIO;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ef4955c..4eceab9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -307,6 +307,13 @@ struct alc_spec {
/* for PLL fix */
hda_nid_t pll_nid;
unsigned int pll_coef_idx, pll_coef_bit;
+
+#ifdef SND_HDA_NEEDS_RESUME
+#define ALC_MAX_PINS 16
+ unsigned int num_pins;
+ hda_nid_t pin_nids[ALC_MAX_PINS];
+ unsigned int pin_cfgs[ALC_MAX_PINS];
+#endif
};
/*
@@ -2778,6 +2785,64 @@ static void alc_free(struct hda_codec *codec)
codec->spec = NULL; /* to be sure */
}
+#ifdef SND_HDA_NEEDS_RESUME
+static void store_pin_configs(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid, end_nid;
+
+ end_nid = codec->start_nid + codec->num_nodes;
+ for (nid = codec->start_nid; nid < end_nid; nid++) {
+ unsigned int wid_caps = get_wcaps(codec, nid);
+ unsigned int wid_type =
+ (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wid_type != AC_WID_PIN)
+ continue;
+ if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids))
+ break;
+ spec->pin_nids[spec->num_pins] = nid;
+ spec->pin_cfgs[spec->num_pins] =
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ spec->num_pins++;
+ }
+}
+
+static void resume_pin_configs(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_pins; i++) {
+ hda_nid_t pin_nid = spec->pin_nids[i];
+ unsigned int pin_config = spec->pin_cfgs[i];
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+ pin_config & 0x000000ff);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+ (pin_config & 0x0000ff00) >> 8);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+ (pin_config & 0x00ff0000) >> 16);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+ pin_config >> 24);
+ }
+}
+
+static int alc_resume(struct hda_codec *codec)
+{
+ resume_pin_configs(codec);
+ codec->patch_ops.init(codec);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ return 0;
+}
+#else
+#define store_pin_configs(codec)
+#endif
+
/*
*/
static struct hda_codec_ops alc_patch_ops = {
@@ -2786,6 +2851,9 @@ static struct hda_codec_ops alc_patch_ops = {
.init = alc_init,
.free = alc_free,
.unsol_event = alc_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+ .resume = alc_resume,
+#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
.check_power_status = alc_check_power_status,
#endif
@@ -3832,6 +3900,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
+ store_pin_configs(codec);
return 1;
}
@@ -5250,6 +5319,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
}
spec->num_mixers++;
+ store_pin_configs(codec);
return 1;
}
@@ -10313,6 +10383,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ store_pin_configs(codec);
return 1;
}
@@ -11447,6 +11518,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ store_pin_configs(codec);
return 1;
}
@@ -12230,6 +12302,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc269_capture_mixer;
spec->num_mixers++;
+ store_pin_configs(codec);
return 1;
}
@@ -13316,6 +13389,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc861_capture_mixer;
spec->num_mixers++;
+ store_pin_configs(codec);
return 1;
}
@@ -14427,6 +14501,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ store_pin_configs(codec);
return 1;
}
@@ -16258,6 +16333,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc662_capture_mixer;
spec->num_mixers++;
+
+ store_pin_configs(codec);
return 1;
}
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 2d0dce6..ae7601f 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci,
}
for (i = 0; i < 2; i++) {
mgr->mem[i].phys = pci_resource_start(pci, i);
- mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys,
- pci_resource_len(pci, i));
+ mgr->mem[i].virt = pci_ioremap_bar(pci, i);
if (!mgr->mem[i].virt) {
printk(KERN_ERR "unable to remap resource 0x%lx\n",
mgr->mem[i].phys);
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 827587f..e020c16 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = {
}
};
-static u16 sport_req[][7] = {
- { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
- P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
- { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
- P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
-};
+/*
+ * Setting the TFS pin selector for SPORT 0 based on whether the selected
+ * port id F or G. If the port is F then no conflict should exist for the
+ * TFS. When Port G is selected and EMAC then there is a conflict between
+ * the PHY interrupt line and TFS. Current settings prevent the conflict
+ * by ignoring the TFS pin when Port G is selected. This allows both
+ * ssm2602 using Port G and EMAC concurrently.
+ */
+#ifdef CONFIG_BF527_SPORT0_PORTF
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
+#else
+#define LOCAL_SPORT0_TFS (0)
+#endif
+
+static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+ P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0},
+ {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI,
+ P_SPORT1_RSCLK, P_SPORT1_TFS, 0} };
static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
@@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
ret = -EINVAL;
break;
default:
+ printk(KERN_ERR "%s: Unknown DAI format type\n", __func__);
ret = -EINVAL;
break;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFS:
- ret = -EINVAL;
- break;
- case SND_SOC_DAIFMT_CBM_CFS:
- ret = -EINVAL;
- break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBM_CFS:
case SND_SOC_DAIFMT_CBS_CFM:
ret = -EINVAL;
break;
default:
+ printk(KERN_ERR "%s: Unknown DAI master type\n", __func__);
ret = -EINVAL;
break;
}
diff --git a/sound/sound_core.c b/sound/sound_core.c
index faef87a..a75b289 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -57,7 +57,7 @@ module_exit(cleanup_soundcore);
/*
* OSS sound core handling. Breaks out sound functions to submodules
*
- * Author: Alan Cox <alan.cox@linux.org>
+ * Author: Alan Cox <alan@lxorguk.ukuu.org.uk>
*
* Fixes:
*
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-09-30 16:01 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-09-30 16:01 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for 2.6.27-rc8 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
These are hopefully last fixes for 2.6.27: one is a patch I missed
in the previous pull request for ASoC cs4270 codec fix, and another
is a trivial fix for a Dell laptop.
Thanks!
Takashi
==
Jean Delvare (1):
ALSA: ASoC: Fix cs4270 error path
Takashi Iwai (1):
ALSA: hda - Fix model for Dell Inspiron 1525
sound/pci/hda/patch_sigmatel.c | 2 +-
sound/soc/codecs/cs4270.c | 25 ++++++++++++++++++-------
2 files changed, 19 insertions(+), 8 deletions(-)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index ad994fc..f3da621 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1683,8 +1683,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
/* Dell 3 stack systems with verb table in BIOS */
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_BIOS),
- SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell ", STAC_DELL_BIOS),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_3ST),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS),
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index d68650d..0bbd945 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -681,7 +681,7 @@ static int cs4270_probe(struct platform_device *pdev)
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "cs4270: failed to create PCMs\n");
- return ret;
+ goto error_free_codec;
}
#ifdef USE_I2C
@@ -690,8 +690,7 @@ static int cs4270_probe(struct platform_device *pdev)
ret = i2c_add_driver(&cs4270_i2c_driver);
if (ret) {
printk(KERN_ERR "cs4270: failed to attach driver");
- snd_soc_free_pcms(socdev);
- return ret;
+ goto error_free_pcms;
}
/* Did we find a CS4270 on the I2C bus? */
@@ -713,10 +712,23 @@ static int cs4270_probe(struct platform_device *pdev)
ret = snd_soc_register_card(socdev);
if (ret < 0) {
printk(KERN_ERR "cs4270: failed to register card\n");
- snd_soc_free_pcms(socdev);
- return ret;
+ goto error_del_driver;
}
+ return 0;
+
+error_del_driver:
+#ifdef USE_I2C
+ i2c_del_driver(&cs4270_i2c_driver);
+
+error_free_pcms:
+#endif
+ snd_soc_free_pcms(socdev);
+
+error_free_codec:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+
return ret;
}
@@ -727,8 +739,7 @@ static int cs4270_remove(struct platform_device *pdev)
snd_soc_free_pcms(socdev);
#ifdef USE_I2C
- if (socdev->codec->control_data)
- i2c_del_driver(&cs4270_i2c_driver);
+ i2c_del_driver(&cs4270_i2c_driver);
#endif
kfree(socdev->codec);
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-09-26 16:13 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-09-26 16:13 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Hi Linus,
please pull another ALSA updates for 2.6.27-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
These are fixes for possible mutex/semaphore deadlocks found via
recent lockdep. The deadlock possibility is quite low and I guess
it never happened in the real use yet, though.
Thanks!
Takashi
==
Takashi Iwai (2):
ALSA: fix locking in snd_pcm_open*() and snd_rawmidi_open*()
ALSA: remove unneeded power_mutex lock in snd_pcm_drop
sound/core/pcm.c | 4 ++--
sound/core/pcm_native.c | 13 +++----------
sound/core/rawmidi.c | 4 ++--
3 files changed, 7 insertions(+), 14 deletions(-)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 9dd9bc7..ece25c7 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -781,7 +781,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
return -ENODEV;
card = pcm->card;
- down_read(&card->controls_rwsem);
+ read_lock(&card->ctl_files_rwlock);
list_for_each_entry(kctl, &card->ctl_files, list) {
if (kctl->pid == current->pid) {
prefer_subdevice = kctl->prefer_pcm_subdevice;
@@ -789,7 +789,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
break;
}
}
- up_read(&card->controls_rwsem);
+ read_unlock(&card->ctl_files_rwlock);
switch (stream) {
case SNDRV_PCM_STREAM_PLAYBACK:
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index c49b9d9..c487025 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1546,16 +1546,10 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream)
card = substream->pcm->card;
if (runtime->status->state == SNDRV_PCM_STATE_OPEN ||
- runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED)
+ runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED ||
+ runtime->status->state == SNDRV_PCM_STATE_SUSPENDED)
return -EBADFD;
- snd_power_lock(card);
- if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
- result = snd_power_wait(card, SNDRV_CTL_POWER_D0);
- if (result < 0)
- goto _unlock;
- }
-
snd_pcm_stream_lock_irq(substream);
/* resume pause */
if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
@@ -1564,8 +1558,7 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream)
snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
/* runtime->control->appl_ptr = runtime->status->hw_ptr; */
snd_pcm_stream_unlock_irq(substream);
- _unlock:
- snd_power_unlock(card);
+
return result;
}
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index f7ea728..b917a9f 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -418,7 +418,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
mutex_lock(&rmidi->open_mutex);
while (1) {
subdevice = -1;
- down_read(&card->controls_rwsem);
+ read_lock(&card->ctl_files_rwlock);
list_for_each_entry(kctl, &card->ctl_files, list) {
if (kctl->pid == current->pid) {
subdevice = kctl->prefer_rawmidi_subdevice;
@@ -426,7 +426,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
break;
}
}
- up_read(&card->controls_rwsem);
+ read_unlock(&card->ctl_files_rwlock);
err = snd_rawmidi_kernel_open(rmidi->card, rmidi->device,
subdevice, fflags, rawmidi_file);
if (err >= 0)
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
2008-09-22 15:57 [GIT PULL] ALSA fix Takashi Iwai
@ 2008-09-23 16:59 ` Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-09-23 16:59 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for 2.6.27-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains a fix of MAINTAINERS in addition to the previous pending
pull request. Both are obviously safe to apply.
Thanks!
Takashi
==
Haavard Skinnemoen (1):
ALSA: ASoC: Fix at32-pcm build breakage with PM enabled
Liam Girdwood (1):
ALSA: ASoC: maintainers - update email address for Liam Girdwood
MAINTAINERS | 3 ++-
sound/soc/at32/at32-pcm.c | 5 +++--
2 files changed, 5 insertions(+), 3 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index cad81a2..42ebbfd 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -3833,11 +3833,12 @@ S: Maintained
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
P: Liam Girdwood
-M: liam.girdwood@wolfsonmicro.com
+M: lrg@slimlogic.co.uk
P: Mark Brown
M: broonie@opensource.wolfsonmicro.com
T: git opensource.wolfsonmicro.com/linux-2.6-asoc
L: alsa-devel@alsa-project.org (subscribers-only)
+W: http://alsa-project.org/main/index.php/ASoC
S: Supported
SPI SUBSYSTEM
diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c
index 435f1da..c83584f 100644
--- a/sound/soc/at32/at32-pcm.c
+++ b/sound/soc/at32/at32-pcm.c
@@ -434,7 +434,8 @@ static int at32_pcm_suspend(struct platform_device *pdev,
params = prtd->params;
/* Disable the PDC and save the PDC registers */
- ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable);
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
@@ -464,7 +465,7 @@ static int at32_pcm_resume(struct platform_device *pdev,
ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
- ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable);
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable);
return 0;
}
#else /* CONFIG_PM */
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [GIT PULL] ALSA fixes
@ 2008-06-23 15:08 Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2008-06-23 15:08 UTC (permalink / raw)
To: Linus Torvalds; +Cc: linux-kernel
Linus,
please pull ALSA updates from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
which contains the following two trivial fixes.
Thanks!
Takashi
===
Takashi Iwai (2):
ALSA: aw2 - Fix Oops at initialization
ALSA: sb - Fix wrong assertions
sound/isa/sb/sb_mixer.c | 4 ++--
sound/pci/aw2/aw2-alsa.c | 4 ++--
2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 91d1422..73d4572 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -925,7 +925,7 @@ static unsigned char als4000_saved_regs[] = {
static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
{
unsigned char *val = chip->saved_regs;
- snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return);
+ snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return);
for (; num_regs; num_regs--)
*val++ = snd_sbmixer_read(chip, *regs++);
}
@@ -933,7 +933,7 @@ static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
static void restore_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
{
unsigned char *val = chip->saved_regs;
- snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return);
+ snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return);
for (; num_regs; num_regs--)
snd_sbmixer_write(chip, *regs++, *val++);
}
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 56f87cd..3f00ddf 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -316,6 +316,8 @@ static int __devinit snd_aw2_create(struct snd_card *card,
return -ENOMEM;
}
+ /* (2) initialization of the chip hardware */
+ snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
IRQF_SHARED, "Audiowerk2", chip)) {
@@ -329,8 +331,6 @@ static int __devinit snd_aw2_create(struct snd_card *card,
}
chip->irq = pci->irq;
- /* (2) initialization of the chip hardware */
- snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
free_irq(chip->irq, (void *)chip);
^ permalink raw reply related [flat|nested] 12+ messages in thread
end of thread, other threads:[~2008-11-21 16:51 UTC | newest]
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