* [GIT PULL] sound fixes
@ 2009-05-26 17:41 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-05-26 17:41 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.30-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains a few fixes for HD-audio as below, including the one
you mentioned in the previous post.
Thanks!
Takashi
===
Ozan Çağlayan (1):
ALSA: hda - Add forced codec-slots for ASUS W5Fm
Takashi Iwai (2):
ALSA: hda - Add 5stack-no-fp model for STAC927x
ALSA: hda - Add missing check of pin vref 50 and others in Realtek codecs
---
Documentation/sound/alsa/HD-Audio-Models.txt | 1 +
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_realtek.c | 6 ++++++
sound/pci/hda/patch_sigmatel.c | 10 ++++++++++
4 files changed, 18 insertions(+), 0 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 8eec05b..322869f 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -334,6 +334,7 @@ STAC9227/9228/9229/927x
ref-no-jd Reference board without HP/Mic jack detection
3stack D965 3stack
5stack D965 5stack + SPDIF
+ 5stack-no-fp D965 5stack without front panel
dell-3stack Dell Dimension E520
dell-bios Fixes with Dell BIOS setup
auto BIOS setup (default)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 21e99cf..3128e1a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2141,6 +2141,7 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
/* forced codec slots */
+ SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103),
SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bcbb736..0fd258e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -776,6 +776,12 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
if (pincap & AC_PINCAP_VREF_80)
val = PIN_VREF80;
+ else if (pincap & AC_PINCAP_VREF_50)
+ val = PIN_VREF50;
+ else if (pincap & AC_PINCAP_VREF_100)
+ val = PIN_VREF100;
+ else if (pincap & AC_PINCAP_VREF_GRD)
+ val = PIN_VREFGRD;
}
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 03b3646..d2fd8ef 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -150,6 +150,7 @@ enum {
STAC_D965_REF,
STAC_D965_3ST,
STAC_D965_5ST,
+ STAC_D965_5ST_NO_FP,
STAC_DELL_3ST,
STAC_DELL_BIOS,
STAC_927X_MODELS
@@ -2154,6 +2155,13 @@ static unsigned int d965_5st_pin_configs[14] = {
0x40000100, 0x40000100
};
+static unsigned int d965_5st_no_fp_pin_configs[14] = {
+ 0x40000100, 0x40000100, 0x0181304e, 0x01014010,
+ 0x01a19040, 0x01011012, 0x01016011, 0x40000100,
+ 0x40000100, 0x40000100, 0x40000100, 0x01442070,
+ 0x40000100, 0x40000100
+};
+
static unsigned int dell_3st_pin_configs[14] = {
0x02211230, 0x02a11220, 0x01a19040, 0x01114210,
0x01111212, 0x01116211, 0x01813050, 0x01112214,
@@ -2166,6 +2174,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
[STAC_D965_REF] = ref927x_pin_configs,
[STAC_D965_3ST] = d965_3st_pin_configs,
[STAC_D965_5ST] = d965_5st_pin_configs,
+ [STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs,
[STAC_DELL_3ST] = dell_3st_pin_configs,
[STAC_DELL_BIOS] = NULL,
};
@@ -2176,6 +2185,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
[STAC_D965_REF] = "ref",
[STAC_D965_3ST] = "3stack",
[STAC_D965_5ST] = "5stack",
+ [STAC_D965_5ST_NO_FP] = "5stack-no-fp",
[STAC_DELL_3ST] = "dell-3stack",
[STAC_DELL_BIOS] = "dell-bios",
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2012-02-03 13:40 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2012-02-03 13:40 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
Linus,
please pull sound fixes for v3.3-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git sound-fixes
Most of commits are either regression fixes for varioud HD-audio
codecs or small ASoC fixes. Also a trivial build fix is included.
Thanks!
Takashi
===
Axel Lin (1):
ALSA: Add #ifdef CONFIG_PCI guard for snd_pci_quirk_* functions
David Henningsson (4):
ALSA: HDA: Remove quirk for Asus N53Jq
ALSA: HDA: Fix jack creation for codecs with front and rear Line In
ALSA: HDA: Remove quirk for Toshiba Qosmio G50
ALSA: HDA: Fix duplicated output to more than one codec
Denis 'GNUtoo' Carikli (1):
ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine
Dylan Reid (1):
ALSA: hda - Fix calling cs_automic twice for Cirrus codecs.
Mark Brown (6):
ASoC: wm5100: Make sure we switch to button reporting mode
ASoC: wm5100: Fix microphone configuration
ASoC: wm8996: Mark register cache as dirty when regulators are disabled
ASoC: wm8962: Mark register cache as dirty when regulators are disabled
ASoC: wm5100: Mark register cache as dirty when regulators are disabled
ASoC: wm_hubs: Enable line out VMID buffer for single ended line outputs
Takashi Iwai (5):
ALSA: hda - Apply 0x0f-VREF fix to all ASUS laptops with ALC861/660
ALSA: hda - Check power-state before changing in patch_via.c
ALSA: hda - Fix the logic to detect VIA analog low-current mode
ALSA: hda - Allow analog low-current mode when dynamic power-control is on
ALSA: hda - Disable dynamic-power control for VIA as default
UK KIM (1):
ASoC: wm_hubs: fix wrong bits for LINEOUT2 N/P mixer
---
include/sound/core.h | 2 +
sound/pci/hda/hda_codec.c | 2 +-
sound/pci/hda/hda_jack.c | 24 ++-
sound/pci/hda/patch_cirrus.c | 6 +-
sound/pci/hda/patch_realtek.c | 57 +++++---
sound/pci/hda/patch_via.c | 284 ++++++++++++++++--------------------
sound/soc/codecs/wm5100.c | 12 ++-
sound/soc/codecs/wm8962.c | 2 +-
sound/soc/codecs/wm8996.c | 2 +-
sound/soc/codecs/wm_hubs.c | 10 +-
sound/soc/samsung/neo1973_wm8753.c | 65 +--------
11 files changed, 208 insertions(+), 258 deletions(-)
^ permalink raw reply [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2012-01-27 14:01 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2012-01-27 14:01 UTC (permalink / raw)
To: Linus Torvalds; +Cc: linux-kernel
Linus,
please pull a few another sound fixes for v3.3-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus
The topmost commit is b3a81520bd37a28f77cb0f7002086fb14061824d
This contains small regression-fixes and a fix for and old hardware.
Thanks!
Takashi
===
Gustavo Maciel Dias Vieira (1):
ALSA: hda: set mute led polarity for laptops with buggy BIOS based on SSID
Jesper Juhl (1):
ALSA: Fix memory leak on error in snd_compr_set_params()
Raymond Yau (1):
ALSA: ymfpci - Don't create invalid PCM & mixers when AC97 doesn't support
Takashi Iwai (2):
ALSA: hda - Fix silent output on ASUS A6Rp
ALSA: hda - Fix silent output on Haier W18 laptop
---
sound/core/compress_offload.c | 13 ++++++++-----
sound/pci/hda/patch_realtek.c | 11 +++++++++++
sound/pci/hda/patch_sigmatel.c | 9 ++++++++-
sound/pci/ymfpci/ymfpci.c | 21 +++++++++++++--------
sound/pci/ymfpci/ymfpci_main.c | 21 ++++++++++++++-------
5 files changed, 54 insertions(+), 21 deletions(-)
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index dac3633..a68aed7 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;
- if (copy_from_user(params, (void __user *)arg, sizeof(*params)))
- return -EFAULT;
+ if (copy_from_user(params, (void __user *)arg, sizeof(*params))) {
+ retval = -EFAULT;
+ goto out;
+ }
retval = snd_compr_allocate_buffer(stream, params);
if (retval) {
- kfree(params);
- return -ENOMEM;
+ retval = -ENOMEM;
+ goto out;
}
retval = stream->ops->set_params(stream, params);
if (retval)
goto out;
stream->runtime->state = SNDRV_PCM_STATE_SETUP;
- } else
+ } else {
return -EPERM;
+ }
out:
kfree(params);
return retval;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c95c8bd..0db1dc4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5586,6 +5586,7 @@ static const struct hda_amp_list alc861_loopbacks[] = {
/* Pin config fixes */
enum {
PINFIX_FSC_AMILO_PI1505,
+ PINFIX_ASUS_A6RP,
};
static const struct alc_fixup alc861_fixups[] = {
@@ -5597,9 +5598,19 @@ static const struct alc_fixup alc861_fixups[] = {
{ }
}
},
+ [PINFIX_ASUS_A6RP] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* node 0x0f VREF seems controlling the master output */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
+ { }
+ },
+ },
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", PINFIX_ASUS_A6RP),
+ SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP),
SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 336cfcd..948f0be 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4870,7 +4870,14 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
/* BIOS bug: unfilled OEM string */
if (strstr(dev->name, "HP_Mute_LED_P_G")) {
set_hp_led_gpio(codec);
- spec->gpio_led_polarity = 1;
+ switch (codec->subsystem_id) {
+ case 0x103c148a:
+ spec->gpio_led_polarity = 0;
+ break;
+ default:
+ spec->gpio_led_polarity = 1;
+ break;
+ }
return 1;
}
}
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index e57b89e8..94ab728 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -286,17 +286,22 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm_4ch(chip, 2, NULL)) < 0) {
+ err = snd_ymfpci_mixer(chip, rear_switch[dev]);
+ if (err < 0) {
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm2(chip, 3, NULL)) < 0) {
- snd_card_free(card);
- return err;
- }
- if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) {
- snd_card_free(card);
- return err;
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ err = snd_ymfpci_pcm_4ch(chip, 2, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = snd_ymfpci_pcm2(chip, 3, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
}
if ((err = snd_ymfpci_timer(chip, 0)) < 0) {
snd_card_free(card);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 03ee4e3..12a9a2b 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -1614,6 +1614,14 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return change;
}
+static struct snd_kcontrol_new snd_ymfpci_dup4ch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "4ch Duplication",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_ymfpci_info_dup4ch,
+ .get = snd_ymfpci_get_dup4ch,
+ .put = snd_ymfpci_put_dup4ch,
+};
static struct snd_kcontrol_new snd_ymfpci_controls[] __devinitdata = {
{
@@ -1642,13 +1650,6 @@ YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4),
-{
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "4ch Duplication",
- .info = snd_ymfpci_info_dup4ch,
- .get = snd_ymfpci_get_dup4ch,
- .put = snd_ymfpci_put_dup4ch,
-},
};
@@ -1838,6 +1839,12 @@ int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch)
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_controls[idx], chip))) < 0)
return err;
}
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ kctl = snd_ctl_new1(&snd_ymfpci_dup4ch, chip);
+ err = snd_ctl_add(chip->card, kctl);
+ if (err < 0)
+ return err;
+ }
/* add S/PDIF control */
if (snd_BUG_ON(!chip->pcm_spdif))
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-12-16 17:30 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-12-16 17:30 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
Linus,
please pull sound fixes for v3.2-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus
The topmost commit is 844ec3142af3b8fd7b9aa95886015eabed80983e
All small and trivial fixes as expected for rc6.
Thanks!
Takashi
===
Axel Lin (5):
ASoC: Include linux/io.h for jz4740 codec
ASoC: Fix build dependency for SND_SOC_JZ4740_CODEC
ASoC: Fix a typo in jive_wm8750
ASoC: Include linux/module.h for smdk2443_wm9710
ASoC: Fix hx4700 error handling to free gpios if snd_soc_register_card fails
David Henningsson (2):
ALSA: HDA: Set position fix to LPIB for an Atom/Poulsbo based device
ALSA: HDA: Use LPIB Position fix for Intel SCH Poulsbo
Gustavo Maciel Dias Vieira (1):
ALSA: hda: fix mute led polarity for HP laptops with buggy BIOS
Jonathan Neuschäfer (1):
ASoC: WM8958: correctly show firmware magic on mismatch
Lothar Waßmann (2):
ASoC: mxs: Add missing MODULE_LICENSE("GPL")
ASoC: mxs: Add appropriate MODULE_ALIAS()
Mark Brown (1):
ASoC: Fix WM8996 24.576MHz clock operation
---
sound/pci/hda/hda_intel.c | 4 +++-
sound/pci/hda/patch_sigmatel.c | 6 ++++++
sound/soc/codecs/Kconfig | 2 +-
sound/soc/codecs/jz4740.c | 1 +
sound/soc/codecs/wm8958-dsp2.c | 2 ++
sound/soc/codecs/wm8996.c | 1 +
sound/soc/mxs/mxs-pcm.c | 3 +++
sound/soc/mxs/mxs-sgtl5000.c | 1 +
sound/soc/pxa/hx4700.c | 5 +++--
sound/soc/samsung/jive_wm8750.c | 3 +--
sound/soc/samsung/smdk2443_wm9710.c | 1 +
11 files changed, 23 insertions(+), 6 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7d98240..c2f79e6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2507,6 +2507,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
@@ -2970,7 +2971,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
- AZX_DCAPS_BUFSIZE},
+ AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */
+ /* ICH */
{ PCI_DEVICE(0x8086, 0x2668),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
AZX_DCAPS_BUFSIZE }, /* ICH6 */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index eeb25d52..616678f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4929,6 +4929,12 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity)
set_hp_led_gpio(codec);
return 1;
}
+ /* BIOS bug: unfilled OEM string */
+ if (strstr(dev->name, "HP_Mute_LED_P_G")) {
+ set_hp_led_gpio(codec);
+ spec->gpio_led_polarity = 1;
+ return 1;
+ }
}
/*
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4584514..fa787d4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -33,7 +33,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CX20442
select SND_SOC_DA7210 if I2C
select SND_SOC_DFBMCS320
- select SND_SOC_JZ4740_CODEC if SOC_JZ4740
+ select SND_SOC_JZ4740_CODEC
select SND_SOC_LM4857 if I2C
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98095 if I2C
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index e373f8f..3e1f4e1 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -15,6 +15,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
+#include <linux/io.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 0293763..5a14d5c 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -60,6 +60,8 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
}
if (memcmp(fw->data, "WMFW", 4) != 0) {
+ memcpy(&data32, fw->data, sizeof(data32));
+ data32 = be32_to_cpu(data32);
dev_err(codec->dev, "%s: firmware has bad file magic %08x\n",
name, data32);
goto err;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 645c980..a33b04d 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1968,6 +1968,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
break;
case 24576000:
ratediv = WM8996_SYSCLK_DIV;
+ wm8996->sysclk /= 2;
case 12288000:
snd_soc_update_bits(codec, WM8996_AIF_RATE,
WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE);
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index dea5aa4..f39d7dd 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -357,3 +357,6 @@ static void __exit snd_mxs_pcm_exit(void)
platform_driver_unregister(&mxs_pcm_driver);
}
module_exit(snd_mxs_pcm_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mxs-pcm-audio");
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 7fbeaec..1c57f66 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -171,3 +171,4 @@ module_exit(mxs_sgtl5000_exit);
MODULE_AUTHOR("Freescale Semiconductor, Inc.");
MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mxs-sgtl5000");
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 65c1248..c664e33 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -209,9 +209,10 @@ static int __devinit hx4700_audio_probe(struct platform_device *pdev)
snd_soc_card_hx4700.dev = &pdev->dev;
ret = snd_soc_register_card(&snd_soc_card_hx4700);
if (ret)
- return ret;
+ gpio_free_array(hx4700_audio_gpios,
+ ARRAY_SIZE(hx4700_audio_gpios));
- return 0;
+ return ret;
}
static int __devexit hx4700_audio_remove(struct platform_device *pdev)
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 1826acf..8e523fd 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -101,7 +101,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
/* These endpoints are not being used. */
snd_soc_dapm_nc_pin(dapm, "LINPUT2");
@@ -131,7 +130,7 @@ static struct snd_soc_card snd_soc_machine_jive = {
.dai_link = &jive_dai,
.num_links = 1,
- .dapm_widgtets = wm8750_dapm_widgets,
+ .dapm_widgets = wm8750_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c
index 3a0dbfc..8bd1dc5 100644
--- a/sound/soc/samsung/smdk2443_wm9710.c
+++ b/sound/soc/samsung/smdk2443_wm9710.c
@@ -12,6 +12,7 @@
*
*/
+#include <linux/module.h>
#include <sound/soc.h>
static struct snd_soc_card smdk2443;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-12-09 11:33 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-12-09 11:33 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
Linus,
please pull sound fixes for v3.2-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus
The topmost commit is 0a34b42b6245ccc710be6759cded5b7fe41ea1f9
A few HD-audio regression fixes and a fix for old SIS7019 controller
are included here, in addition to the other trivial fixes in ASoC.
Thanks!
Takashi
===
Axel Lin (4):
ASoC: kirkwood: Make SND_KIRKWOOD_SOC_OPENRD and SND_KIRKWOOD_SOC_T5325 depend on I2C
ASoC: uda1380: Return proper error in uda1380_modinit failure path
ASoC: Fix dependency for SND_SOC_RAUMFELD and SND_PXA2XX_SOC_HX4700
ASoC: Make SND_SOC_MX27VIS_AIC32X4 depend on I2C
David Dillow (1):
ALSA: sis7019 - give slow codecs more time to reset
Mark Brown (3):
ASoC: Mark WM8994 ADC muxes as virtual
ASoC: Remove references to corgi and spitz from machine driver document
ASoC: Provide a more complete DMA driver stub
Takashi Iwai (6):
ALSA: hda/realtek - Fix Oops in alc_mux_select()
ALSA: hda - Fix GPIO LED setup for IDT 92HD75 codecs
ALSA: hda - Fix remaining VREF mute-LED NID check in post-3.1 changes
ALSA: hda/realtek - Don't create extra controls with channel suffix
ALSA: hda/realtek - Create "Bass Speaker" for two speaker pins
ALSA: hda/realtek - Fix lost speaker volume controls
---
Documentation/sound/alsa/soc/machine.txt | 6 +--
sound/pci/hda/patch_realtek.c | 65 ++++++++++++++++++++++-------
sound/pci/hda/patch_sigmatel.c | 47 +++++++++++----------
sound/pci/sis7019.c | 62 ++++++++++++++++++++++++-----
sound/soc/codecs/uda1380.c | 4 +-
sound/soc/codecs/wm8994.c | 12 +++---
sound/soc/imx/Kconfig | 2 +-
sound/soc/kirkwood/Kconfig | 3 +-
sound/soc/pxa/Kconfig | 3 +-
sound/soc/soc-utils.c | 31 ++++++++++++++-
10 files changed, 171 insertions(+), 64 deletions(-)
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index 3e2ec9c..d50c14d 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -50,8 +50,7 @@ Machine DAI Configuration
The machine DAI configuration glues all the codec and CPU DAIs together. It can
also be used to set up the DAI system clock and for any machine related DAI
initialisation e.g. the machine audio map can be connected to the codec audio
-map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c
-for examples.
+map, unconnected codec pins can be set as such.
struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
@@ -83,8 +82,7 @@ Machine Power Map
The machine driver can optionally extend the codec power map and to become an
audio power map of the audio subsystem. This allows for automatic power up/down
of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
-sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for
-details.
+sockets in the machine init function.
Machine Controls
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cbde019..1d07e8f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -297,6 +297,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
imux = &spec->input_mux[mux_idx];
if (!imux->num_items && mux_idx > 0)
imux = &spec->input_mux[0];
+ if (!imux->num_items)
+ return 0;
if (idx >= imux->num_items)
idx = imux->num_items - 1;
@@ -2629,6 +2631,8 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
case AUTO_PIN_SPEAKER_OUT:
if (cfg->line_outs == 1)
return "Speaker";
+ if (cfg->line_outs == 2)
+ return ch ? "Bass Speaker" : "Speaker";
break;
case AUTO_PIN_HP_OUT:
/* for multi-io case, only the primary out */
@@ -2902,7 +2906,7 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
if (!nid)
continue;
if (found_in_nid_list(nid, spec->multiout.dac_nids,
- spec->multiout.num_dacs))
+ ARRAY_SIZE(spec->private_dac_nids)))
continue;
if (found_in_nid_list(nid, spec->multiout.hp_out_nid,
ARRAY_SIZE(spec->multiout.hp_out_nid)))
@@ -2923,6 +2927,7 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin)
return 0;
}
+/* return 0 if no possible DAC is found, 1 if one or more found */
static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
const hda_nid_t *pins, hda_nid_t *dacs)
{
@@ -2940,7 +2945,7 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
if (!dacs[i])
dacs[i] = alc_auto_look_for_dac(codec, pins[i]);
}
- return 0;
+ return 1;
}
static int alc_auto_fill_multi_ios(struct hda_codec *codec,
@@ -2950,7 +2955,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec,
static int alc_auto_fill_dac_nids(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- const struct auto_pin_cfg *cfg = &spec->autocfg;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
bool redone = false;
int i;
@@ -2961,6 +2966,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
spec->multiout.extra_out_nid[0] = 0;
memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
spec->multiout.dac_nids = spec->private_dac_nids;
+ spec->multi_ios = 0;
/* fill hard-wired DACs first */
if (!redone) {
@@ -2994,10 +3000,12 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
for (i = 0; i < cfg->line_outs; i++) {
if (spec->private_dac_nids[i])
spec->multiout.num_dacs++;
- else
+ else {
memmove(spec->private_dac_nids + i,
spec->private_dac_nids + i + 1,
sizeof(hda_nid_t) * (cfg->line_outs - i - 1));
+ spec->private_dac_nids[cfg->line_outs - 1] = 0;
+ }
}
if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
@@ -3019,9 +3027,28 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
if (cfg->line_out_type != AUTO_PIN_HP_OUT)
alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins,
spec->multiout.hp_out_nid);
- if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT)
- alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins,
- spec->multiout.extra_out_nid);
+ if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+ int err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs,
+ cfg->speaker_pins,
+ spec->multiout.extra_out_nid);
+ /* if no speaker volume is assigned, try again as the primary
+ * output
+ */
+ if (!err && cfg->speaker_outs > 0 &&
+ cfg->line_out_type == AUTO_PIN_HP_OUT) {
+ cfg->hp_outs = cfg->line_outs;
+ memcpy(cfg->hp_pins, cfg->line_out_pins,
+ sizeof(cfg->hp_pins));
+ cfg->line_outs = cfg->speaker_outs;
+ memcpy(cfg->line_out_pins, cfg->speaker_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->speaker_outs = 0;
+ memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
+ cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
+ redone = false;
+ goto again;
+ }
+ }
return 0;
}
@@ -3171,7 +3198,8 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
}
static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
- hda_nid_t dac, const char *pfx)
+ hda_nid_t dac, const char *pfx,
+ int cidx)
{
struct alc_spec *spec = codec->spec;
hda_nid_t sw, vol;
@@ -3187,15 +3215,15 @@ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
if (is_ctl_used(spec->sw_ctls, val))
return 0; /* already created */
mark_ctl_usage(spec->sw_ctls, val);
- return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val);
+ return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, cidx, val);
}
sw = alc_look_for_out_mute_nid(codec, pin, dac);
vol = alc_look_for_out_vol_nid(codec, pin, dac);
- err = alc_auto_add_stereo_vol(codec, pfx, 0, vol);
+ err = alc_auto_add_stereo_vol(codec, pfx, cidx, vol);
if (err < 0)
return err;
- err = alc_auto_add_stereo_sw(codec, pfx, 0, sw);
+ err = alc_auto_add_stereo_sw(codec, pfx, cidx, sw);
if (err < 0)
return err;
return 0;
@@ -3236,16 +3264,21 @@ static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins,
hda_nid_t dac = *dacs;
if (!dac)
dac = spec->multiout.dac_nids[0];
- return alc_auto_create_extra_out(codec, *pins, dac, pfx);
+ return alc_auto_create_extra_out(codec, *pins, dac, pfx, 0);
}
if (dacs[num_pins - 1]) {
/* OK, we have a multi-output system with individual volumes */
for (i = 0; i < num_pins; i++) {
- snprintf(name, sizeof(name), "%s %s",
- pfx, channel_name[i]);
- err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
- name);
+ if (num_pins >= 3) {
+ snprintf(name, sizeof(name), "%s %s",
+ pfx, channel_name[i]);
+ err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
+ name, 0);
+ } else {
+ err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
+ pfx, i);
+ }
if (err < 0)
return err;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d8d2f9d..eeb25d52 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -215,6 +215,7 @@ struct sigmatel_spec {
unsigned int gpio_mute;
unsigned int gpio_led;
unsigned int gpio_led_polarity;
+ unsigned int vref_mute_led_nid; /* pin NID for mute-LED vref control */
unsigned int vref_led;
/* stream */
@@ -4318,12 +4319,10 @@ static void stac_store_hints(struct hda_codec *codec)
spec->eapd_switch = val;
get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity);
if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
- if (spec->gpio_led <= 8) {
- spec->gpio_mask |= spec->gpio_led;
- spec->gpio_dir |= spec->gpio_led;
- if (spec->gpio_led_polarity)
- spec->gpio_data |= spec->gpio_led;
- }
+ spec->gpio_mask |= spec->gpio_led;
+ spec->gpio_dir |= spec->gpio_led;
+ if (spec->gpio_led_polarity)
+ spec->gpio_data |= spec->gpio_led;
}
}
@@ -4443,7 +4442,7 @@ static int stac92xx_init(struct hda_codec *codec)
/* power on when no jack detection is available */
/* or when the VREF is used for controlling LED */
if (!spec->hp_detect ||
- (spec->gpio_led > 8 && spec->gpio_led == nid)) {
+ spec->vref_mute_led_nid == nid) {
stac_toggle_power_map(codec, nid, 1);
continue;
}
@@ -4915,8 +4914,14 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity)
if (sscanf(dev->name, "HP_Mute_LED_%d_%x",
&spec->gpio_led_polarity,
&spec->gpio_led) == 2) {
- if (spec->gpio_led < 4)
+ unsigned int max_gpio;
+ max_gpio = snd_hda_param_read(codec, codec->afg,
+ AC_PAR_GPIO_CAP);
+ max_gpio &= AC_GPIO_IO_COUNT;
+ if (spec->gpio_led < max_gpio)
spec->gpio_led = 1 << spec->gpio_led;
+ else
+ spec->vref_mute_led_nid = spec->gpio_led;
return 1;
}
if (sscanf(dev->name, "HP_Mute_LED_%d",
@@ -5045,15 +5050,12 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
struct sigmatel_spec *spec = codec->spec;
/* sync mute LED */
- if (spec->gpio_led) {
- if (spec->gpio_led <= 8) {
- stac_gpio_set(codec, spec->gpio_mask,
- spec->gpio_dir, spec->gpio_data);
- } else {
- stac_vrefout_set(codec,
- spec->gpio_led, spec->vref_led);
- }
- }
+ if (spec->vref_mute_led_nid)
+ stac_vrefout_set(codec, spec->vref_mute_led_nid,
+ spec->vref_led);
+ else if (spec->gpio_led)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data);
return 0;
}
@@ -5064,7 +5066,7 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
struct sigmatel_spec *spec = codec->spec;
if (power_state == AC_PWRST_D3) {
- if (spec->gpio_led > 8) {
+ if (spec->vref_mute_led_nid) {
/* with vref-out pin used for mute led control
* codec AFG is prevented from D3 state
*/
@@ -5117,7 +5119,7 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
}
}
/*polarity defines *not* muted state level*/
- if (spec->gpio_led <= 8) {
+ if (!spec->vref_mute_led_nid) {
if (muted)
spec->gpio_data &= ~spec->gpio_led; /* orange */
else
@@ -5135,7 +5137,8 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
muted_lvl = spec->gpio_led_polarity ?
AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
spec->vref_led = muted ? muted_lvl : notmtd_lvl;
- stac_vrefout_set(codec, spec->gpio_led, spec->vref_led);
+ stac_vrefout_set(codec, spec->vref_mute_led_nid,
+ spec->vref_led);
}
return 0;
}
@@ -5649,7 +5652,7 @@ again:
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
- if (spec->gpio_led <= 8) {
+ if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
spec->gpio_dir |= spec->gpio_led;
spec->gpio_data |= spec->gpio_led;
@@ -5962,7 +5965,7 @@ again:
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
- if (spec->gpio_led <= 8) {
+ if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
spec->gpio_dir |= spec->gpio_led;
spec->gpio_data |= spec->gpio_led;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index a391e62..28dfafb 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -41,6 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}");
static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
static int enable = 1;
+static int codecs = 1;
module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator.");
@@ -48,6 +49,8 @@ module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator.");
module_param(enable, bool, 0444);
MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator.");
+module_param(codecs, int, 0444);
+MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)");
static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = {
{ PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) },
@@ -140,6 +143,9 @@ struct sis7019 {
dma_addr_t silence_dma_addr;
};
+/* These values are also used by the module param 'codecs' to indicate
+ * which codecs should be present.
+ */
#define SIS_PRIMARY_CODEC_PRESENT 0x0001
#define SIS_SECONDARY_CODEC_PRESENT 0x0002
#define SIS_TERTIARY_CODEC_PRESENT 0x0004
@@ -1078,6 +1084,7 @@ static int sis_chip_init(struct sis7019 *sis)
{
unsigned long io = sis->ioport;
void __iomem *ioaddr = sis->ioaddr;
+ unsigned long timeout;
u16 status;
int count;
int i;
@@ -1104,21 +1111,45 @@ static int sis_chip_init(struct sis7019 *sis)
while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)
udelay(1);
+ /* Command complete, we can let go of the semaphore now.
+ */
+ outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
+ if (!count)
+ return -EIO;
+
/* Now that we've finished the reset, find out what's attached.
+ * There are some codec/board combinations that take an extremely
+ * long time to come up. 350+ ms has been observed in the field,
+ * so we'll give them up to 500ms.
*/
- status = inl(io + SIS_AC97_STATUS);
- if (status & SIS_AC97_STATUS_CODEC_READY)
- sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
- if (status & SIS_AC97_STATUS_CODEC2_READY)
- sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
- if (status & SIS_AC97_STATUS_CODEC3_READY)
- sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
+ sis->codecs_present = 0;
+ timeout = msecs_to_jiffies(500) + jiffies;
+ while (time_before_eq(jiffies, timeout)) {
+ status = inl(io + SIS_AC97_STATUS);
+ if (status & SIS_AC97_STATUS_CODEC_READY)
+ sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
+ if (status & SIS_AC97_STATUS_CODEC2_READY)
+ sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
+ if (status & SIS_AC97_STATUS_CODEC3_READY)
+ sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
+
+ if (sis->codecs_present == codecs)
+ break;
- /* All done, let go of the semaphore, and check for errors
+ msleep(1);
+ }
+
+ /* All done, check for errors.
*/
- outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
- if (!sis->codecs_present || !count)
+ if (!sis->codecs_present) {
+ printk(KERN_ERR "sis7019: could not find any codecs\n");
return -EIO;
+ }
+
+ if (sis->codecs_present != codecs) {
+ printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n",
+ sis->codecs_present, codecs);
+ }
/* Let the hardware know that the audio driver is alive,
* and enable PCM slots on the AC-link for L/R playback (3 & 4) and
@@ -1390,6 +1421,17 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci,
if (!enable)
goto error_out;
+ /* The user can specify which codecs should be present so that we
+ * can wait for them to show up if they are slow to recover from
+ * the AC97 cold reset. We default to a single codec, the primary.
+ *
+ * We assume that SIS_PRIMARY_*_PRESENT matches bits 0-2.
+ */
+ codecs &= SIS_PRIMARY_CODEC_PRESENT | SIS_SECONDARY_CODEC_PRESENT |
+ SIS_TERTIARY_CODEC_PRESENT;
+ if (!codecs)
+ codecs = SIS_PRIMARY_CODEC_PRESENT;
+
rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card);
if (rc < 0)
goto error_out;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index c5ca8cf..0441893 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = {
static int __init uda1380_modinit(void)
{
- int ret;
+ int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&uda1380_i2c_driver);
if (ret != 0)
pr_err("Failed to register UDA1380 I2C driver: %d\n", ret);
#endif
- return 0;
+ return ret;
}
module_init(uda1380_modinit);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 6c29885..d0c545b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1325,15 +1325,15 @@ SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
};
static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
-SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
- adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
- adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
};
static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
};
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index b133bfc..7383917 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1
config SND_SOC_MX27VIS_AIC32X4
tristate "SoC audio support for Visstrim M10 boards"
- depends on MACH_IMX27_VISSTRIM_M10
+ depends on MACH_IMX27_VISSTRIM_M10 && I2C
select SND_SOC_TLV320AIC32X4
select SND_MXC_SOC_MX2
help
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 8f49e16..c62d715 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S
config SND_KIRKWOOD_SOC_OPENRD
tristate "SoC Audio support for Kirkwood Openrd Client"
depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE)
+ depends on I2C
select SND_KIRKWOOD_SOC_I2S
select SND_SOC_CS42L51
help
@@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD
config SND_KIRKWOOD_SOC_T5325
tristate "SoC Audio support for HP t5325"
- depends on SND_KIRKWOOD_SOC && MACH_T5325
+ depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C
select SND_KIRKWOOD_SOC_I2S
select SND_SOC_ALC5623
help
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ffd2242..a0f7d3c 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE
config SND_SOC_RAUMFELD
tristate "SoC Audio support Raumfeld audio adapter"
depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR)
+ depends on I2C && SPI_MASTER
select SND_PXA_SOC_SSP
select SND_SOC_CS4270
select SND_SOC_AK4104
@@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD
config SND_PXA2XX_SOC_HX4700
tristate "SoC Audio support for HP iPAQ hx4700"
- depends on SND_PXA2XX_SOC && MACH_H4700
+ depends on SND_PXA2XX_SOC && MACH_H4700 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_AK4641
help
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 0c12b98..4220bb0 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -58,7 +58,36 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
}
EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
-static struct snd_soc_platform_driver dummy_platform;
+static const struct snd_pcm_hardware dummy_dma_hardware = {
+ .formats = 0xffffffff,
+ .channels_min = 1,
+ .channels_max = UINT_MAX,
+
+ /* Random values to keep userspace happy when checking constraints */
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .buffer_bytes_max = 128*1024,
+ .period_bytes_min = PAGE_SIZE,
+ .period_bytes_max = PAGE_SIZE*2,
+ .periods_min = 2,
+ .periods_max = 128,
+};
+
+static int dummy_dma_open(struct snd_pcm_substream *substream)
+{
+ snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
+
+ return 0;
+}
+
+static struct snd_pcm_ops dummy_dma_ops = {
+ .open = dummy_dma_open,
+ .ioctl = snd_pcm_lib_ioctl,
+};
+
+static struct snd_soc_platform_driver dummy_platform = {
+ .ops = &dummy_dma_ops,
+};
static __devinit int snd_soc_dummy_probe(struct platform_device *pdev)
{
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-12-02 9:55 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-12-02 9:55 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
Linus,
please pull sound fixes for v3.2-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus
The topmost commit is cf54d47c13c2b171f946289de445102c676d4258
This contains a few trivial/small ASoC fixes, sigma-firmware fixes,
a drop of an obsolete playpaq support, and a couple of HD-audio
regression fixes.
Thanks!
Takashi
===
Axel Lin (3):
ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET
ASoC: max9877: Update register if either val or val2 is changed
ASoC: cs42l51: Fix off-by-one for reg_cache_size
Boojin Kim (1):
ASoC: SAMSUNG: Fix build error
Charles Chin (1):
ALSA: hda - Fix S3/S4 problem on machines with VREF-pin mute-LED
Daniel Mack (1):
ASoC: cs4720: use snd_soc_cache_sync()
Eric Miao (1):
ASoC: skip resume of soc-audio devices without codecs
John F Leach (1):
ALSA: usb-audio - Support for Roland GAIA SH-01 Synthesizer
Lars-Peter Clausen (3):
firmware: Sigma: Prevent out of bounds memory access
firmware: Sigma: Skip header during CRC generation
firmware: Sigma: Fix endianess issues
Marc Vertes (1):
ALSA: hda_intel - revert a quirk that affect VIA chipsets
Mark Brown (3):
ASoC: Correct name of Speyside Main Speaker widget
ASoC: Error out if we can't generate a LRCLK at all for WM8994
ASoC: Supply dcs_codes for newer WM1811 revisions
Paul Bolle (1):
ASoC: drop support for PlayPaq with WM8510
Takashi Iwai (1):
ALSA: hda - Avoid touching mute-VREF pin for IDT codecs
Timur Tabi (1):
ASoC: mpc8610: tell the CS4270 codec that it's the master
---
drivers/firmware/sigma.c | 81 +++++--
include/linux/sigma.h | 13 +-
sound/pci/hda/hda_intel.c | 1 -
sound/pci/hda/patch_sigmatel.c | 22 +--
sound/soc/atmel/Kconfig | 21 +--
sound/soc/atmel/Makefile | 4 -
sound/soc/atmel/playpaq_wm8510.c | 473 --------------------------------------
sound/soc/codecs/ad1836.h | 2 +-
sound/soc/codecs/cs4270.c | 10 +-
sound/soc/codecs/cs42l51.c | 2 +-
sound/soc/codecs/max9877.c | 10 +-
sound/soc/codecs/wm8994.c | 7 +
sound/soc/fsl/mpc8610_hpcd.c | 24 ++-
sound/soc/samsung/smdk_wm8994.c | 1 +
sound/soc/samsung/speyside.c | 2 +-
sound/soc/soc-core.c | 6 +
sound/usb/quirks-table.h | 31 +++
17 files changed, 136 insertions(+), 574 deletions(-)
delete mode 100644 sound/soc/atmel/playpaq_wm8510.c
diff --git a/drivers/firmware/sigma.c b/drivers/firmware/sigma.c
index f10fc52..1eedb6f 100644
--- a/drivers/firmware/sigma.c
+++ b/drivers/firmware/sigma.c
@@ -14,13 +14,34 @@
#include <linux/module.h>
#include <linux/sigma.h>
-/* Return: 0==OK, <0==error, =1 ==no more actions */
+static size_t sigma_action_size(struct sigma_action *sa)
+{
+ size_t payload = 0;
+
+ switch (sa->instr) {
+ case SIGMA_ACTION_WRITEXBYTES:
+ case SIGMA_ACTION_WRITESINGLE:
+ case SIGMA_ACTION_WRITESAFELOAD:
+ payload = sigma_action_len(sa);
+ break;
+ default:
+ break;
+ }
+
+ payload = ALIGN(payload, 2);
+
+ return payload + sizeof(struct sigma_action);
+}
+
+/*
+ * Returns a negative error value in case of an error, 0 if processing of
+ * the firmware should be stopped after this action, 1 otherwise.
+ */
static int
-process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw)
+process_sigma_action(struct i2c_client *client, struct sigma_action *sa)
{
- struct sigma_action *sa = (void *)(ssfw->fw->data + ssfw->pos);
size_t len = sigma_action_len(sa);
- int ret = 0;
+ int ret;
pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__,
sa->instr, sa->addr, len);
@@ -29,44 +50,50 @@ process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw)
case SIGMA_ACTION_WRITEXBYTES:
case SIGMA_ACTION_WRITESINGLE:
case SIGMA_ACTION_WRITESAFELOAD:
- if (ssfw->fw->size < ssfw->pos + len)
- return -EINVAL;
ret = i2c_master_send(client, (void *)&sa->addr, len);
if (ret < 0)
return -EINVAL;
break;
-
case SIGMA_ACTION_DELAY:
- ret = 0;
udelay(len);
len = 0;
break;
-
case SIGMA_ACTION_END:
- return 1;
-
+ return 0;
default:
return -EINVAL;
}
- /* when arrive here ret=0 or sent data */
- ssfw->pos += sigma_action_size(sa, len);
- return ssfw->pos == ssfw->fw->size;
+ return 1;
}
static int
process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw)
{
- pr_debug("%s: processing %p\n", __func__, ssfw);
+ struct sigma_action *sa;
+ size_t size;
+ int ret;
+
+ while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) {
+ sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos);
+
+ size = sigma_action_size(sa);
+ ssfw->pos += size;
+ if (ssfw->pos > ssfw->fw->size || size == 0)
+ break;
+
+ ret = process_sigma_action(client, sa);
- while (1) {
- int ret = process_sigma_action(client, ssfw);
pr_debug("%s: action returned %i\n", __func__, ret);
- if (ret == 1)
- return 0;
- else if (ret)
+
+ if (ret <= 0)
return ret;
}
+
+ if (ssfw->pos != ssfw->fw->size)
+ return -EINVAL;
+
+ return 0;
}
int process_sigma_firmware(struct i2c_client *client, const char *name)
@@ -89,16 +116,24 @@ int process_sigma_firmware(struct i2c_client *client, const char *name)
/* then verify the header */
ret = -EINVAL;
- if (fw->size < sizeof(*ssfw_head))
+
+ /*
+ * Reject too small or unreasonable large files. The upper limit has been
+ * chosen a bit arbitrarily, but it should be enough for all practical
+ * purposes and having the limit makes it easier to avoid integer
+ * overflows later in the loading process.
+ */
+ if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000)
goto done;
ssfw_head = (void *)fw->data;
if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic)))
goto done;
- crc = crc32(0, fw->data, fw->size);
+ crc = crc32(0, fw->data + sizeof(*ssfw_head),
+ fw->size - sizeof(*ssfw_head));
pr_debug("%s: crc=%x\n", __func__, crc);
- if (crc != ssfw_head->crc)
+ if (crc != le32_to_cpu(ssfw_head->crc))
goto done;
ssfw.pos = sizeof(*ssfw_head);
diff --git a/include/linux/sigma.h b/include/linux/sigma.h
index e2accb3..d0de882 100644
--- a/include/linux/sigma.h
+++ b/include/linux/sigma.h
@@ -24,7 +24,7 @@ struct sigma_firmware {
struct sigma_firmware_header {
unsigned char magic[7];
u8 version;
- u32 crc;
+ __le32 crc;
};
enum {
@@ -40,19 +40,14 @@ enum {
struct sigma_action {
u8 instr;
u8 len_hi;
- u16 len;
- u16 addr;
+ __le16 len;
+ __be16 addr;
unsigned char payload[];
};
static inline u32 sigma_action_len(struct sigma_action *sa)
{
- return (sa->len_hi << 16) | sa->len;
-}
-
-static inline size_t sigma_action_size(struct sigma_action *sa, u32 payload_len)
-{
- return sizeof(*sa) + payload_len + (payload_len % 2);
+ return (sa->len_hi << 16) | le16_to_cpu(sa->len);
}
extern int process_sigma_firmware(struct i2c_client *client, const char *name);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 096507d..7d98240 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f365865..d8d2f9d 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4441,7 +4441,9 @@ static int stac92xx_init(struct hda_codec *codec)
int pinctl, def_conf;
/* power on when no jack detection is available */
- if (!spec->hp_detect) {
+ /* or when the VREF is used for controlling LED */
+ if (!spec->hp_detect ||
+ (spec->gpio_led > 8 && spec->gpio_led == nid)) {
stac_toggle_power_map(codec, nid, 1);
continue;
}
@@ -5055,20 +5057,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
return 0;
}
-static int stac92xx_post_suspend(struct hda_codec *codec)
-{
- struct sigmatel_spec *spec = codec->spec;
- if (spec->gpio_led > 8) {
- /* with vref-out pin used for mute led control
- * codec AFG is prevented from D3 state, but on
- * system suspend it can (and should) be used
- */
- snd_hda_codec_read(codec, codec->afg, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- }
- return 0;
-}
-
static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
@@ -5668,8 +5656,6 @@ again:
} else {
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
- codec->patch_ops.post_suspend =
- stac92xx_post_suspend;
}
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
@@ -5983,8 +5969,6 @@ again:
} else {
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
- codec->patch_ops.post_suspend =
- stac92xx_post_suspend;
}
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index bee3c94..d1fcc81 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -1,6 +1,6 @@
config SND_ATMEL_SOC
tristate "SoC Audio for the Atmel System-on-Chip"
- depends on ARCH_AT91 || AVR32
+ depends on ARCH_AT91
help
Say Y or M if you want to add support for codecs attached to
the ATMEL SSC interface. You will also need
@@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731
Say Y if you want to add support for SoC audio on WM8731-based
AT91sam9g20 evaluation board.
-config SND_AT32_SOC_PLAYPAQ
- tristate "SoC Audio support for PlayPaq with WM8510"
- depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
- select SND_ATMEL_SOC_SSC
- select SND_SOC_WM8510
- help
- Say Y or M here if you want to add support for SoC audio
- on the LRS PlayPaq.
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
- bool "Run CODEC on PlayPaq in slave mode"
- depends on SND_AT32_SOC_PLAYPAQ
- default n
- help
- Say Y if you want to run with the AT32 SSC generating the BCLK
- and FRAME signals on the PlayPaq. Unless you want to play
- with the AT32 as the SSC master, you probably want to say N here,
- as this will give you better sound quality.
-
config SND_AT91_SOC_AFEB9260
tristate "SoC Audio support for AFEB9260 board"
depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index e7ea56b..a5c0bf1 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
# AT91 Machine Support
snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
deleted file mode 100644
index 73ae99a..0000000
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 444747f..dd7be0d 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -34,7 +34,7 @@
#define AD1836_ADC_CTRL2 13
#define AD1836_ADC_WORD_LEN_MASK 0x30
-#define AD1836_ADC_WORD_OFFSET 5
+#define AD1836_ADC_WORD_OFFSET 4
#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1f237e..73f46eb 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
static int cs4270_soc_resume(struct snd_soc_codec *codec)
{
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
- struct i2c_client *i2c_client = to_i2c_client(codec->dev);
int reg;
regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
@@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
ndelay(500);
/* first restore the entire register cache ... */
- for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
- u8 val = snd_soc_read(codec, reg);
-
- if (i2c_smbus_write_byte_data(i2c_client, reg, val)) {
- dev_err(codec->dev, "i2c write failed\n");
- return -EIO;
- }
- }
+ snd_soc_cache_sync(codec);
/* ... then disable the power-down bits */
reg = snd_soc_read(codec, CS4270_PWRCTL);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 8c3c820..1ee66361 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
.probe = cs42l51_probe,
- .reg_cache_size = CS42L51_NUMREGS,
+ .reg_cache_size = CS42L51_NUMREGS + 1,
.reg_word_size = sizeof(u8),
};
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 9e7e964..dcf6f2a 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
unsigned int mask = mc->max;
unsigned int val = (ucontrol->value.integer.value[0] & mask);
unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
- unsigned int change = 1;
+ unsigned int change = 0;
- if (((max9877_regs[reg] >> shift) & mask) == val)
- change = 0;
+ if (((max9877_regs[reg] >> shift) & mask) != val)
+ change = 1;
- if (((max9877_regs[reg2] >> shift) & mask) == val2)
- change = 0;
+ if (((max9877_regs[reg2] >> shift) & mask) != val2)
+ change = 1;
if (change) {
max9877_regs[reg] &= ~(mask << shift);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 9c982e4..6c29885 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT;
lrclk = bclk_rate / params_rate(params);
+ if (!lrclk) {
+ dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n",
+ bclk_rate);
+ return -EINVAL;
+ }
dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
lrclk, bclk_rate / lrclk);
@@ -3178,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
switch (wm8994->revision) {
case 0:
case 1:
+ case 2:
+ case 3:
wm8994->hubs.dcs_codes_l = -9;
wm8994->hubs.dcs_codes_r = -5;
break;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 31af405..ae49f1c 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
if (strcasecmp(sprop, "i2s-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
@@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
machine_data->clk_frequency = be32_to_cpup(iprop);
} else if (strcasecmp(sprop, "i2s-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "lj-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "lj-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "rj-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "rj-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "ac97-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "ac97-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else {
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index f75e439..ad9ac42 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -9,6 +9,7 @@
#include "../codecs/wm8994.h"
#include <sound/pcm_params.h>
+#include <linux/module.h>
/*
* Default CFG switch settings to use this driver:
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 85bf541..4b8e354 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card)
snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC");
snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC");
- snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker");
snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output");
snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a5d3685..a25fa63 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev)
struct snd_soc_card *card = dev_get_drvdata(dev);
int i, ac97_control = 0;
+ /* If the initialization of this soc device failed, there is no codec
+ * associated with it. Just bail out in this case.
+ */
+ if (list_empty(&card->codec_dev_list))
+ return 0;
+
/* AC97 devices might have other drivers hanging off them so
* need to resume immediately. Other drivers don't have that
* problem and may take a substantial amount of time to resume
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index b61945f..32d2a21 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1633,6 +1633,37 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
{
+ /* Roland GAIA SH-01 */
+ USB_DEVICE(0x0582, 0x0111),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "GAIA",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = &(const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
USB_DEVICE(0x0582, 0x0113),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
/* .vendor_name = "BOSS", */
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-05-12 19:34 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-05-12 19:34 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.39 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/asoc
This contains only the following ASoC fixes.
All small fixes, as expected.
Thanks!
Takashi
===
Lars-Peter Clausen (4):
ASoC: JZ4740: Fix i2s shutdown
ASoC: SSM2602: Properly annotate i2c probe and remove functions
ASoC: SSM2602: Fix 'Mic Boost2' control
ASoC: SSM2602: Fix reg_cache_size
Marek Belisko (1):
ASoC: UDA134x: Remove POWER_OFF_ON_STANDBY define.
Mark Brown (1):
ASoC: Don't crash on PM operations
Stephen Warren (1):
ASoC: WM8903: Fix Digital Capture Volume range
xingchao (1):
ASoC: sst_platform: add hw_free callback to fix resource leak
---
sound/soc/codecs/ssm2602.c | 10 +++++-----
sound/soc/codecs/uda134x.c | 2 --
sound/soc/codecs/wm8903.c | 2 +-
sound/soc/jz4740/jz4740-i2s.c | 2 +-
sound/soc/mid-x86/sst_platform.c | 6 ++++++
sound/soc/soc-core.c | 2 ++
6 files changed, 15 insertions(+), 9 deletions(-)
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 2727bef..b04d280 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -139,7 +139,7 @@ SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
-SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 7, 1, 0),
+SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1),
@@ -602,7 +602,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.read = ssm2602_read_reg_cache,
.write = ssm2602_write,
.set_bias_level = ssm2602_set_bias_level,
- .reg_cache_size = sizeof(ssm2602_reg),
+ .reg_cache_size = ARRAY_SIZE(ssm2602_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = ssm2602_reg,
};
@@ -614,7 +614,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
* low = 0x1a
* high = 0x1b
*/
-static int ssm2602_i2c_probe(struct i2c_client *i2c,
+static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct ssm2602_priv *ssm2602;
@@ -635,7 +635,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c,
return ret;
}
-static int ssm2602_i2c_remove(struct i2c_client *client)
+static int __devexit ssm2602_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
kfree(i2c_get_clientdata(client));
@@ -655,7 +655,7 @@ static struct i2c_driver ssm2602_i2c_driver = {
.owner = THIS_MODULE,
},
.probe = ssm2602_i2c_probe,
- .remove = ssm2602_i2c_remove,
+ .remove = __devexit_p(ssm2602_i2c_remove),
.id_table = ssm2602_i2c_id,
};
#endif
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 48ffd40..a7b8f30 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -601,9 +601,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
.reg_cache_step = 1,
.read = uda134x_read_reg_cache,
.write = uda134x_write,
-#ifdef POWER_OFF_ON_STANDBY
.set_bias_level = uda134x_set_bias_level,
-#endif
};
static int __devinit uda134x_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index f52b623..824d1c8 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -692,7 +692,7 @@ SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
- WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv),
+ WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
SOC_ENUM("ADC Companding Mode", adc_companding),
SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 419bf4f..cd22a54 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -133,7 +133,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream,
struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
uint32_t conf;
- if (!dai->active)
+ if (dai->active)
return;
conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index d567c32..6b1f9d3 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -376,6 +376,11 @@ static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
static struct snd_pcm_ops sst_platform_ops = {
.open = sst_platform_open,
.close = sst_platform_close,
@@ -384,6 +389,7 @@ static struct snd_pcm_ops sst_platform_ops = {
.trigger = sst_platform_pcm_trigger,
.pointer = sst_platform_pcm_pointer,
.hw_params = sst_platform_pcm_hw_params,
+ .hw_free = sst_platform_pcm_hw_free,
};
static void sst_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d8562ce..dd55d10 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3291,6 +3291,8 @@ int snd_soc_register_card(struct snd_soc_card *card)
if (!card->name || !card->dev)
return -EINVAL;
+ dev_set_drvdata(card->dev, card);
+
snd_soc_initialize_card_lists(card);
soc_init_card_debugfs(card);
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-05-09 11:54 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-05-09 11:54 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.39-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/asoc
This contains only small ASoC fixes as below.
Thanks!
Takashi
===
Ben Gardiner (4):
ASoC: davinci-mcasp: correct tdm_slots limit
davinci-mcasp: use bitfield definitions for PDIR
davinci-mcasp: fix _CBM_CFS hw_params
davinci-mcasp: fix _CBM_CFS pin directions
Mark Brown (2):
ASoC: Fix CODEC name in Goni
ASoC: Fix CODEC DAI names for Goni
---
sound/soc/davinci/davinci-mcasp.c | 19 ++++++++++++-------
sound/soc/samsung/goni_wm8994.c | 8 ++++----
2 files changed, 16 insertions(+), 11 deletions(-)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index a5af834..4ddc6d3 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -434,17 +434,21 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
- mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26));
+ mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+ ACLKX | AHCLKX | AFSX);
break;
case SND_SOC_DAIFMT_CBM_CFS:
/* codec is clock master and frame slave */
- mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
- mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
- mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26));
+ mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+ ACLKX | ACLKR);
+ mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+ AFSX | AFSR);
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* codec is clock and frame master */
@@ -454,7 +458,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
- mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26));
+ mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+ ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
break;
default:
@@ -644,7 +649,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
- if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+ if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
FSXMOD(dev->tdm_slots), FSXMOD(0x1FF));
else
@@ -660,7 +665,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
AHCLKRE);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
- if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+ if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG,
FSRMOD(dev->tdm_slots), FSRMOD(0x1FF));
else
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index f6b3a3c..0e80dae 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -236,18 +236,18 @@ static struct snd_soc_dai_link goni_dai[] = {
.name = "WM8994",
.stream_name = "WM8994 HiFi",
.cpu_dai_name = "samsung-i2s.0",
- .codec_dai_name = "wm8994-hifi",
+ .codec_dai_name = "wm8994-aif1",
.platform_name = "samsung-audio",
- .codec_name = "wm8994-codec.0-0x1a",
+ .codec_name = "wm8994-codec.0-001a",
.init = goni_wm8994_init,
.ops = &goni_hifi_ops,
}, {
.name = "WM8994 Voice",
.stream_name = "Voice",
.cpu_dai_name = "goni-voice-dai",
- .codec_dai_name = "wm8994-voice",
+ .codec_dai_name = "wm8994-aif2",
.platform_name = "samsung-audio",
- .codec_name = "wm8994-codec.0-0x1a",
+ .codec_name = "wm8994-codec.0-001a",
.ops = &goni_voice_ops,
},
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-05-02 16:00 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-05-02 16:00 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.39-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All small fixes as below.
Thanks!
Takashi
===
Daniel Cordero (1):
ALSA: hda: add beep quirk for Realtek 0x1043:831a
David Henningsson (1):
ALSA: HDA: Fix automute for Gateway NV79
Lydia Wang (1):
ALSA: hda - VIA: Fix notify_aa_path_ctls() invalid issue.
Raymond Yau (1):
ALSA - au88x0 - Add buffer bytes constraints
Takashi Iwai (2):
Revert "ALSA: hda - Fix pin-config of Gigabyte mobo"
ALSA: hda - Fix Realtek's chained fixup checks
Wolfgang Breyha (1):
ALSA: usb-audio - Terratec Aureon 7.1 USB ID as C-Media cm6206 quirks
---
sound/pci/au88x0/au88x0_pcm.c | 7 +++++--
sound/pci/hda/patch_realtek.c | 30 +++++++++++-------------------
sound/pci/hda/patch_via.c | 10 +++++++---
sound/usb/format.c | 4 +++-
sound/usb/quirks.c | 1 +
5 files changed, 27 insertions(+), 25 deletions(-)
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 33f0ba5..62e9591 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -44,10 +44,10 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = {
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 0x10000,
- .period_bytes_min = 0x1,
+ .period_bytes_min = 0x20,
.period_bytes_max = 0x1000,
.periods_min = 2,
- .periods_max = 32,
+ .periods_max = 1024,
};
#ifndef CHIP_AU8820
@@ -140,6 +140,9 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0)
return err;
+ snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64);
+
if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
#ifndef CHIP_AU8820
if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d3bd2c1..c82979a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1704,11 +1704,11 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
codec->chip_name, fix->type);
break;
}
- if (!fix[id].chained)
+ if (!fix->chained)
break;
if (++depth > 10)
break;
- id = fix[id].chain_id;
+ id = fix->chain_id;
}
}
@@ -5645,6 +5645,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
static struct snd_pci_quirk beep_white_list[] = {
SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
+ SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
{}
};
@@ -9863,6 +9864,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
@@ -10699,7 +10701,6 @@ enum {
PINFIX_LENOVO_Y530,
PINFIX_PB_M5210,
PINFIX_ACER_ASPIRE_7736,
- PINFIX_GIGABYTE_880GM,
};
static const struct alc_fixup alc882_fixups[] = {
@@ -10731,13 +10732,6 @@ static const struct alc_fixup alc882_fixups[] = {
.type = ALC_FIXUP_SKU,
.v.sku = ALC_FIXUP_SKU_IGNORE,
},
- [PINFIX_GIGABYTE_880GM] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x1114410 }, /* set as speaker */
- { }
- }
- },
};
static struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -10745,7 +10739,6 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
- SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM),
{}
};
@@ -18805,6 +18798,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+ ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
@@ -19478,7 +19473,7 @@ enum {
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
ALC662_FIXUP_CZC_P10T,
- ALC662_FIXUP_GIGABYTE,
+ ALC662_FIXUP_SKU_IGNORE,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -19507,20 +19502,17 @@ static const struct alc_fixup alc662_fixups[] = {
{}
}
},
- [ALC662_FIXUP_GIGABYTE] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x1114410 }, /* set as speaker */
- { }
- }
+ [ALC662_FIXUP_SKU_IGNORE] = {
+ .type = ALC_FIXUP_SKU,
+ .v.sku = ALC_FIXUP_SKU_IGNORE,
},
};
static struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
- SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 1371b57..0997031 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1292,14 +1292,18 @@ static void notify_aa_path_ctls(struct hda_codec *codec)
{
int i;
struct snd_ctl_elem_id id;
- const char *labels[] = {"Mic", "Front Mic", "Line"};
+ const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"};
+ struct snd_kcontrol *ctl;
memset(&id, 0, sizeof(id));
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
for (i = 0; i < ARRAY_SIZE(labels); i++) {
sprintf(id.name, "%s Playback Volume", labels[i]);
- snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &id);
+ ctl = snd_hda_find_mixer_ctl(codec, id.name);
+ if (ctl)
+ snd_ctl_notify(codec->bus->card,
+ SNDRV_CTL_EVENT_MASK_VALUE,
+ &ctl->id);
}
}
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5b792d2..f079b5e 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -176,9 +176,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
if (!rate)
continue;
/* C-Media CM6501 mislabels its 96 kHz altsetting */
+ /* Terratec Aureon 7.1 USB C-Media 6206, too */
if (rate == 48000 && nr_rates == 1 &&
(chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
- chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
+ chip->usb_id == USB_ID(0x0d8c, 0x0102) ||
+ chip->usb_id == USB_ID(0x0ccd, 0x00b1)) &&
fp->altsetting == 5 && fp->maxpacksize == 392)
rate = 96000;
/* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ec07e62..1b94ec3 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -533,6 +533,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
case USB_ID(0x0d8c, 0x0102):
/* C-Media CM6206 / CM106-Like Sound Device */
+ case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */
return snd_usb_cm6206_boot_quirk(dev);
case USB_ID(0x133e, 0x0815):
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-03-27 9:13 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-03-27 9:13 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
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
This contains the last-minute fixes picked up recently, including
two OSS security fixes, a couple of long-standing regression fixes,
asihpi minor updates, and a few quirks for USB-audio and HD-audio.
Thanks!
Takashi
===
Benjamin Herrenschmidt (1):
ALSA: vmalloc buffers should use normal mmap
Clemens Ladisch (1):
ALSA: usb-audio: add Cakewalk UM-1G support
Dan Rosenberg (2):
sound/oss: remove offset from load_patch callbacks
sound/oss/opl3: validate voice and channel indexes
David Henningsson (1):
ALSA: HDA: New AD1984A model for Dell Precision R5500
Eliot Blennerhassett (3):
ALSA: asihpi - Support single-rate no-SRC cards
ALSA: asihpi - Improve non-busmaster adapter operation
ALSA: asihpi - Update verbose debug print macros
Keith A. Milner (1):
ALSA: usb-audio - Support for Boss JS-8 Jam Station
Takashi Iwai (2):
ALSA: hda - Fix SPDIF out regression on ALC889
ALSA: Fix yet another race in disconnection
---
include/sound/pcm.h | 4 +-
sound/core/init.c | 4 +
sound/core/pcm_native.c | 9 ---
sound/oss/dev_table.h | 2 +-
sound/oss/midi_synth.c | 30 ++++-----
sound/oss/midi_synth.h | 2 +-
sound/oss/opl3.c | 23 +++++---
sound/oss/sequencer.c | 2 +-
sound/pci/asihpi/asihpi.c | 137 +++++++++++++++++++----------------------
sound/pci/hda/patch_analog.c | 89 ++++++++++++++++++++++++++
sound/pci/hda/patch_realtek.c | 2 +-
sound/usb/quirks-table.h | 40 ++++++++++++
12 files changed, 230 insertions(+), 114 deletions(-)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 430a9cc..e1bad11 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1031,9 +1031,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s
#define snd_pcm_lib_mmap_iomem NULL
#endif
-int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream,
- struct vm_area_struct *area);
-#define snd_pcm_lib_mmap_vmalloc snd_pcm_lib_mmap_noncached
+#define snd_pcm_lib_mmap_vmalloc NULL
static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
{
diff --git a/sound/core/init.c b/sound/core/init.c
index 3e65da2..a0080aa 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -848,6 +848,7 @@ int snd_card_file_add(struct snd_card *card, struct file *file)
return -ENOMEM;
mfile->file = file;
mfile->disconnected_f_op = NULL;
+ INIT_LIST_HEAD(&mfile->shutdown_list);
spin_lock(&card->files_lock);
if (card->shutdown) {
spin_unlock(&card->files_lock);
@@ -883,6 +884,9 @@ int snd_card_file_remove(struct snd_card *card, struct file *file)
list_for_each_entry(mfile, &card->files_list, list) {
if (mfile->file == file) {
list_del(&mfile->list);
+ spin_lock(&shutdown_lock);
+ list_del(&mfile->shutdown_list);
+ spin_unlock(&shutdown_lock);
if (mfile->disconnected_f_op)
fops_put(mfile->disconnected_f_op);
found = mfile;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index ae42b65..fe5c803 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -3201,15 +3201,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream,
EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem);
#endif /* SNDRV_PCM_INFO_MMAP */
-/* mmap callback with pgprot_noncached */
-int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream,
- struct vm_area_struct *area)
-{
- area->vm_page_prot = pgprot_noncached(area->vm_page_prot);
- return snd_pcm_default_mmap(substream, area);
-}
-EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached);
-
/*
* mmap DMA buffer
*/
diff --git a/sound/oss/dev_table.h b/sound/oss/dev_table.h
index b7617be..0199a31 100644
--- a/sound/oss/dev_table.h
+++ b/sound/oss/dev_table.h
@@ -271,7 +271,7 @@ struct synth_operations
void (*reset) (int dev);
void (*hw_control) (int dev, unsigned char *event);
int (*load_patch) (int dev, int format, const char __user *addr,
- int offs, int count, int pmgr_flag);
+ int count, int pmgr_flag);
void (*aftertouch) (int dev, int voice, int pressure);
void (*controller) (int dev, int voice, int ctrl_num, int value);
void (*panning) (int dev, int voice, int value);
diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c
index 3c09374..2292c23 100644
--- a/sound/oss/midi_synth.c
+++ b/sound/oss/midi_synth.c
@@ -476,7 +476,7 @@ EXPORT_SYMBOL(midi_synth_hw_control);
int
midi_synth_load_patch(int dev, int format, const char __user *addr,
- int offs, int count, int pmgr_flag)
+ int count, int pmgr_flag)
{
int orig_dev = synth_devs[dev]->midi_dev;
@@ -491,33 +491,29 @@ midi_synth_load_patch(int dev, int format, const char __user *addr,
if (!prefix_cmd(orig_dev, 0xf0))
return 0;
+ /* Invalid patch format */
if (format != SYSEX_PATCH)
- {
-/* printk("MIDI Error: Invalid patch format (key) 0x%x\n", format);*/
return -EINVAL;
- }
+
+ /* Patch header too short */
if (count < hdr_size)
- {
-/* printk("MIDI Error: Patch header too short\n");*/
return -EINVAL;
- }
+
count -= hdr_size;
/*
- * Copy the header from user space but ignore the first bytes which have
- * been transferred already.
+ * Copy the header from user space
*/
- if(copy_from_user(&((char *) &sysex)[offs], &(addr)[offs], hdr_size - offs))
+ if (copy_from_user(&sysex, addr, hdr_size))
return -EFAULT;
-
- if (count < sysex.len)
- {
-/* printk(KERN_WARNING "MIDI Warning: Sysex record too short (%d<%d)\n", count, (int) sysex.len);*/
+
+ /* Sysex record too short */
+ if ((unsigned)count < (unsigned)sysex.len)
sysex.len = count;
- }
- left = sysex.len;
- src_offs = 0;
+
+ left = sysex.len;
+ src_offs = 0;
for (i = 0; i < left && !signal_pending(current); i++)
{
diff --git a/sound/oss/midi_synth.h b/sound/oss/midi_synth.h
index 6bc9d00..b64ddd6 100644
--- a/sound/oss/midi_synth.h
+++ b/sound/oss/midi_synth.h
@@ -8,7 +8,7 @@ int midi_synth_open (int dev, int mode);
void midi_synth_close (int dev);
void midi_synth_hw_control (int dev, unsigned char *event);
int midi_synth_load_patch (int dev, int format, const char __user * addr,
- int offs, int count, int pmgr_flag);
+ int count, int pmgr_flag);
void midi_synth_panning (int dev, int channel, int pressure);
void midi_synth_aftertouch (int dev, int channel, int pressure);
void midi_synth_controller (int dev, int channel, int ctrl_num, int value);
diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c
index 938c48c..407cd67 100644
--- a/sound/oss/opl3.c
+++ b/sound/oss/opl3.c
@@ -820,7 +820,7 @@ static void opl3_hw_control(int dev, unsigned char *event)
}
static int opl3_load_patch(int dev, int format, const char __user *addr,
- int offs, int count, int pmgr_flag)
+ int count, int pmgr_flag)
{
struct sbi_instrument ins;
@@ -830,11 +830,7 @@ static int opl3_load_patch(int dev, int format, const char __user *addr,
return -EINVAL;
}
- /*
- * What the fuck is going on here? We leave junk in the beginning
- * of ins and then check the field pretty close to that beginning?
- */
- if(copy_from_user(&((char *) &ins)[offs], addr + offs, sizeof(ins) - offs))
+ if (copy_from_user(&ins, addr, sizeof(ins)))
return -EFAULT;
if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR)
@@ -849,6 +845,10 @@ static int opl3_load_patch(int dev, int format, const char __user *addr,
static void opl3_panning(int dev, int voice, int value)
{
+
+ if (voice < 0 || voice >= devc->nr_voice)
+ return;
+
devc->voc[voice].panning = value;
}
@@ -1066,8 +1066,15 @@ static int opl3_alloc_voice(int dev, int chn, int note, struct voice_alloc_info
static void opl3_setup_voice(int dev, int voice, int chn)
{
- struct channel_info *info =
- &synth_devs[dev]->chn_info[chn];
+ struct channel_info *info;
+
+ if (voice < 0 || voice >= devc->nr_voice)
+ return;
+
+ if (chn < 0 || chn > 15)
+ return;
+
+ info = &synth_devs[dev]->chn_info[chn];
opl3_set_instr(dev, voice, info->pgm_num);
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index 5ea1098..30bcfe4 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -241,7 +241,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun
return -ENXIO;
fmt = (*(short *) &event_rec[0]) & 0xffff;
- err = synth_devs[dev]->load_patch(dev, fmt, buf, p + 4, c, 0);
+ err = synth_devs[dev]->load_patch(dev, fmt, buf + p, c, 0);
if (err < 0)
return err;
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 0ac1f98..f53a31e 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -22,21 +22,6 @@
* for any purpose including commercial applications.
*/
-/* >0: print Hw params, timer vars. >1: print stream write/copy sizes */
-#define REALLY_VERBOSE_LOGGING 0
-
-#if REALLY_VERBOSE_LOGGING
-#define VPRINTK1 snd_printd
-#else
-#define VPRINTK1(...)
-#endif
-
-#if REALLY_VERBOSE_LOGGING > 1
-#define VPRINTK2 snd_printd
-#else
-#define VPRINTK2(...)
-#endif
-
#include "hpi_internal.h"
#include "hpimsginit.h"
#include "hpioctl.h"
@@ -57,11 +42,25 @@
#include <sound/tlv.h>
#include <sound/hwdep.h>
-
MODULE_LICENSE("GPL");
MODULE_AUTHOR("AudioScience inc. <support@audioscience.com>");
MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx");
+#if defined CONFIG_SND_DEBUG_VERBOSE
+/**
+ * snd_printddd - very verbose debug printk
+ * @format: format string
+ *
+ * Works like snd_printk() for debugging purposes.
+ * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
+ * Must set snd module debug parameter to 3 to enable at runtime.
+ */
+#define snd_printddd(format, args...) \
+ __snd_printk(3, __FILE__, __LINE__, format, ##args)
+#else
+#define snd_printddd(format, args...) do { } while (0)
+#endif
+
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
@@ -289,7 +288,6 @@ static u16 handle_error(u16 err, int line, char *filename)
#define hpi_handle_error(x) handle_error(x, __LINE__, __FILE__)
/***************************** GENERAL PCM ****************/
-#if REALLY_VERBOSE_LOGGING
static void print_hwparams(struct snd_pcm_hw_params *p)
{
snd_printd("HWPARAMS \n");
@@ -304,9 +302,6 @@ static void print_hwparams(struct snd_pcm_hw_params *p)
snd_printd("periods %d \n", params_periods(p));
snd_printd("buffer_size %d \n", params_buffer_size(p));
}
-#else
-#define print_hwparams(x)
-#endif
static snd_pcm_format_t hpi_to_alsa_formats[] = {
-1, /* INVALID */
@@ -381,13 +376,13 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi,
"No local sampleclock, err %d\n", err);
}
- for (idx = 0; idx < 100; idx++) {
- if (hpi_sample_clock_query_local_rate(
- h_control, idx, &sample_rate)) {
- if (!idx)
- snd_printk(KERN_ERR
- "Local rate query failed\n");
-
+ for (idx = -1; idx < 100; idx++) {
+ if (idx == -1) {
+ if (hpi_sample_clock_get_sample_rate(h_control,
+ &sample_rate))
+ continue;
+ } else if (hpi_sample_clock_query_local_rate(h_control,
+ idx, &sample_rate)) {
break;
}
@@ -440,8 +435,6 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi,
}
}
- /* printk(KERN_INFO "Supported rates %X %d %d\n",
- rates, rate_min, rate_max); */
pcmhw->rates = rates;
pcmhw->rate_min = rate_min;
pcmhw->rate_max = rate_max;
@@ -466,7 +459,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
if (err)
return err;
- VPRINTK1(KERN_INFO "format %d, %d chans, %d_hz\n",
+ snd_printdd("format %d, %d chans, %d_hz\n",
format, params_channels(params),
params_rate(params));
@@ -489,13 +482,12 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
err = hpi_stream_host_buffer_attach(dpcm->h_stream,
params_buffer_bytes(params), runtime->dma_addr);
if (err == 0) {
- VPRINTK1(KERN_INFO
+ snd_printdd(
"stream_host_buffer_attach succeeded %u %lu\n",
params_buffer_bytes(params),
(unsigned long)runtime->dma_addr);
} else {
- snd_printd(KERN_INFO
- "stream_host_buffer_attach error %d\n",
+ snd_printd("stream_host_buffer_attach error %d\n",
err);
return -ENOMEM;
}
@@ -504,7 +496,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
&dpcm->hpi_buffer_attached,
NULL, NULL, NULL);
- VPRINTK1(KERN_INFO "stream_host_buffer_attach status 0x%x\n",
+ snd_printdd("stream_host_buffer_attach status 0x%x\n",
dpcm->hpi_buffer_attached);
}
bytes_per_sec = params_rate(params) * params_channels(params);
@@ -517,7 +509,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
dpcm->bytes_per_sec = bytes_per_sec;
dpcm->buffer_bytes = params_buffer_bytes(params);
dpcm->period_bytes = params_period_bytes(params);
- VPRINTK1(KERN_INFO "buffer_bytes=%d, period_bytes=%d, bps=%d\n",
+ snd_printdd("buffer_bytes=%d, period_bytes=%d, bps=%d\n",
dpcm->buffer_bytes, dpcm->period_bytes, bytes_per_sec);
return 0;
@@ -573,7 +565,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
struct snd_pcm_substream *s;
u16 e;
- VPRINTK1(KERN_INFO "%c%d trigger\n",
+ snd_printdd("%c%d trigger\n",
SCHR(substream->stream), substream->number);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -597,7 +589,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
* data??
*/
unsigned int preload = ds->period_bytes * 1;
- VPRINTK2(KERN_INFO "%d preload x%x\n", s->number, preload);
+ snd_printddd("%d preload x%x\n", s->number, preload);
hpi_handle_error(hpi_outstream_write_buf(
ds->h_stream,
&runtime->dma_area[0],
@@ -607,7 +599,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
}
if (card->support_grouping) {
- VPRINTK1(KERN_INFO "\t%c%d group\n",
+ snd_printdd("\t%c%d group\n",
SCHR(s->stream),
s->number);
e = hpi_stream_group_add(
@@ -622,7 +614,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
} else
break;
}
- VPRINTK1(KERN_INFO "start\n");
+ snd_printdd("start\n");
/* start the master stream */
snd_card_asihpi_pcm_timer_start(substream);
if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ||
@@ -644,14 +636,14 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
s->runtime->status->state = SNDRV_PCM_STATE_SETUP;
if (card->support_grouping) {
- VPRINTK1(KERN_INFO "\t%c%d group\n",
+ snd_printdd("\t%c%d group\n",
SCHR(s->stream),
s->number);
snd_pcm_trigger_done(s, substream);
} else
break;
}
- VPRINTK1(KERN_INFO "stop\n");
+ snd_printdd("stop\n");
/* _prepare and _hwparams reset the stream */
hpi_handle_error(hpi_stream_stop(dpcm->h_stream));
@@ -664,12 +656,12 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- VPRINTK1(KERN_INFO "pause release\n");
+ snd_printdd("pause release\n");
hpi_handle_error(hpi_stream_start(dpcm->h_stream));
snd_card_asihpi_pcm_timer_start(substream);
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- VPRINTK1(KERN_INFO "pause\n");
+ snd_printdd("pause\n");
snd_card_asihpi_pcm_timer_stop(substream);
hpi_handle_error(hpi_stream_stop(dpcm->h_stream));
break;
@@ -741,7 +733,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
u16 state;
u32 buffer_size, bytes_avail, samples_played, on_card_bytes;
- VPRINTK1(KERN_INFO "%c%d snd_card_asihpi_timer_function\n",
+ snd_printdd("%c%d snd_card_asihpi_timer_function\n",
SCHR(substream->stream), substream->number);
/* find minimum newdata and buffer pos in group */
@@ -770,10 +762,10 @@ static void snd_card_asihpi_timer_function(unsigned long data)
if ((bytes_avail == 0) &&
(on_card_bytes < ds->pcm_buf_host_rw_ofs)) {
hpi_handle_error(hpi_stream_start(ds->h_stream));
- VPRINTK1(KERN_INFO "P%d start\n", s->number);
+ snd_printdd("P%d start\n", s->number);
}
} else if (state == HPI_STATE_DRAINED) {
- VPRINTK1(KERN_WARNING "P%d drained\n",
+ snd_printd(KERN_WARNING "P%d drained\n",
s->number);
/*snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN);
continue; */
@@ -794,13 +786,13 @@ static void snd_card_asihpi_timer_function(unsigned long data)
newdata);
}
- VPRINTK1(KERN_INFO "PB timer hw_ptr x%04lX, appl_ptr x%04lX\n",
+ snd_printdd("hw_ptr x%04lX, appl_ptr x%04lX\n",
(unsigned long)frames_to_bytes(runtime,
runtime->status->hw_ptr),
(unsigned long)frames_to_bytes(runtime,
runtime->control->appl_ptr));
- VPRINTK1(KERN_INFO "%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X,"
+ snd_printdd("%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X,"
" aux=x%04X space=x%04X\n",
loops, SCHR(s->stream), s->number,
state, ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs, (int)bytes_avail,
@@ -822,7 +814,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
next_jiffies = max(next_jiffies, 1U);
dpcm->timer.expires = jiffies + next_jiffies;
- VPRINTK1(KERN_INFO "jif %d buf pos x%04X newdata x%04X xfer x%04X\n",
+ snd_printdd("jif %d buf pos x%04X newdata x%04X xfer x%04X\n",
next_jiffies, pcm_buf_dma_ofs, newdata, xfercount);
snd_pcm_group_for_each_entry(s, substream) {
@@ -837,7 +829,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
if (xfercount && (on_card_bytes <= ds->period_bytes)) {
if (card->support_mmap) {
if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- VPRINTK2(KERN_INFO "P%d write x%04x\n",
+ snd_printddd("P%d write x%04x\n",
s->number,
ds->period_bytes);
hpi_handle_error(
@@ -848,7 +840,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
xfercount,
&ds->format));
} else {
- VPRINTK2(KERN_INFO "C%d read x%04x\n",
+ snd_printddd("C%d read x%04x\n",
s->number,
xfercount);
hpi_handle_error(
@@ -871,7 +863,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream,
unsigned int cmd, void *arg)
{
- /* snd_printd(KERN_INFO "Playback ioctl %d\n", cmd); */
+ snd_printdd(KERN_INFO "Playback ioctl %d\n", cmd);
return snd_pcm_lib_ioctl(substream, cmd, arg);
}
@@ -881,7 +873,7 @@ static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream *
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
- VPRINTK1(KERN_INFO "playback prepare %d\n", substream->number);
+ snd_printdd("playback prepare %d\n", substream->number);
hpi_handle_error(hpi_outstream_reset(dpcm->h_stream));
dpcm->pcm_buf_host_rw_ofs = 0;
@@ -898,7 +890,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream)
snd_pcm_uframes_t ptr;
ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs % dpcm->buffer_bytes);
- /* VPRINTK2(KERN_INFO "playback_pointer=x%04lx\n", (unsigned long)ptr); */
+ snd_printddd("playback_pointer=x%04lx\n", (unsigned long)ptr);
return ptr;
}
@@ -1014,12 +1006,13 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream)
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
card->update_interval_frames);
+
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
card->update_interval_frames * 2, UINT_MAX);
snd_pcm_set_sync(substream);
- VPRINTK1(KERN_INFO "playback open\n");
+ snd_printdd("playback open\n");
return 0;
}
@@ -1030,7 +1023,7 @@ static int snd_card_asihpi_playback_close(struct snd_pcm_substream *substream)
struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
hpi_handle_error(hpi_outstream_close(dpcm->h_stream));
- VPRINTK1(KERN_INFO "playback close\n");
+ snd_printdd("playback close\n");
return 0;
}
@@ -1050,13 +1043,13 @@ static int snd_card_asihpi_playback_copy(struct snd_pcm_substream *substream,
if (copy_from_user(runtime->dma_area, src, len))
return -EFAULT;
- VPRINTK2(KERN_DEBUG "playback copy%d %u bytes\n",
+ snd_printddd("playback copy%d %u bytes\n",
substream->number, len);
hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream,
runtime->dma_area, len, &dpcm->format));
- dpcm->pcm_buf_host_rw_ofs = dpcm->pcm_buf_host_rw_ofs + len;
+ dpcm->pcm_buf_host_rw_ofs += len;
return 0;
}
@@ -1066,16 +1059,11 @@ static int snd_card_asihpi_playback_silence(struct snd_pcm_substream *
snd_pcm_uframes_t pos,
snd_pcm_uframes_t count)
{
- unsigned int len;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
-
- len = frames_to_bytes(runtime, count);
- VPRINTK1(KERN_INFO "playback silence %u bytes\n", len);
-
- memset(runtime->dma_area, 0, len);
- hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream,
- runtime->dma_area, len, &dpcm->format));
+ /* Usually writes silence to DMA buffer, which should be overwritten
+ by real audio later. Our fifos cannot be overwritten, and are not
+ free-running DMAs. Silence is output on fifo underflow.
+ This callback is still required to allow the copy callback to be used.
+ */
return 0;
}
@@ -1110,7 +1098,7 @@ snd_card_asihpi_capture_pointer(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
- VPRINTK2(KERN_INFO "capture pointer %d=%d\n",
+ snd_printddd("capture pointer %d=%d\n",
substream->number, dpcm->pcm_buf_dma_ofs);
/* NOTE Unlike playback can't use actual samples_played
for the capture position, because those samples aren't yet in
@@ -1135,7 +1123,7 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream)
dpcm->pcm_buf_dma_ofs = 0;
dpcm->pcm_buf_elapsed_dma_ofs = 0;
- VPRINTK1("Capture Prepare %d\n", substream->number);
+ snd_printdd("Capture Prepare %d\n", substream->number);
return 0;
}
@@ -1198,7 +1186,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream)
if (dpcm == NULL)
return -ENOMEM;
- VPRINTK1("hpi_instream_open adapter %d stream %d\n",
+ snd_printdd("capture open adapter %d stream %d\n",
card->adapter_index, substream->number);
err = hpi_handle_error(
@@ -1268,7 +1256,7 @@ static int snd_card_asihpi_capture_copy(struct snd_pcm_substream *substream,
len = frames_to_bytes(runtime, count);
- VPRINTK2(KERN_INFO "capture copy%d %d bytes\n", substream->number, len);
+ snd_printddd("capture copy%d %d bytes\n", substream->number, len);
hpi_handle_error(hpi_instream_read_buf(dpcm->h_stream,
runtime->dma_area, len));
@@ -2887,6 +2875,9 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev,
if (err)
asihpi->update_interval_frames = 512;
+ if (!asihpi->support_mmap)
+ asihpi->update_interval_frames *= 2;
+
hpi_handle_error(hpi_instream_open(asihpi->adapter_index,
0, &h_stream));
@@ -2909,7 +2900,6 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev,
asihpi->support_mrx
);
-
err = snd_card_asihpi_pcm_new(asihpi, 0, pcm_substreams);
if (err < 0) {
snd_printk(KERN_ERR "pcm_new failed\n");
@@ -2944,6 +2934,7 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev,
sprintf(card->longname, "%s %i",
card->shortname, asihpi->adapter_index);
err = snd_card_register(card);
+
if (!err) {
hpi_card->snd_card_asihpi = card;
dev++;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 734c6ee..2942d2a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -4256,6 +4256,84 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
}
/*
+ * Precision R5500
+ * 0x12 - HP/line-out
+ * 0x13 - speaker (mono)
+ * 0x15 - mic-in
+ */
+
+static struct hda_verb ad1984a_precision_verbs[] = {
+ /* Unmute main output path */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */
+ /* Analog mixer; mute as default */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Select mic as input */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */
+ /* Configure as mic */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+ /* HP unmute */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* turn on EAPD */
+ {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ /* unsolicited event for pin-sense */
+ {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1984a_precision_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+
+/* mute internal speaker if HP is plugged */
+static void ad1984a_precision_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_jack_detect(codec, 0x12);
+ snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+
+/* unsolicited event for HP jack sensing */
+static void ad1984a_precision_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != AD1884A_HP_EVENT)
+ return;
+ ad1984a_precision_automute(codec);
+}
+
+/* initialize jack-sensing, too */
+static int ad1984a_precision_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1984a_precision_automute(codec);
+ return 0;
+}
+
+
+/*
* HP Touchsmart
* port-A (0x11) - front hp-out
* port-B (0x14) - unused
@@ -4384,6 +4462,7 @@ enum {
AD1884A_MOBILE,
AD1884A_THINKPAD,
AD1984A_TOUCHSMART,
+ AD1984A_PRECISION,
AD1884A_MODELS
};
@@ -4393,9 +4472,11 @@ static const char * const ad1884a_models[AD1884A_MODELS] = {
[AD1884A_MOBILE] = "mobile",
[AD1884A_THINKPAD] = "thinkpad",
[AD1984A_TOUCHSMART] = "touchsmart",
+ [AD1984A_PRECISION] = "precision",
};
static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION),
SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
@@ -4489,6 +4570,14 @@ static int patch_ad1884a(struct hda_codec *codec)
codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
codec->patch_ops.init = ad1984a_thinkpad_init;
break;
+ case AD1984A_PRECISION:
+ spec->mixers[0] = ad1984a_precision_mixers;
+ spec->init_verbs[spec->num_init_verbs++] =
+ ad1984a_precision_verbs;
+ spec->multiout.dig_out_nid = 0;
+ codec->patch_ops.unsol_event = ad1984a_precision_unsol_event;
+ codec->patch_ops.init = ad1984a_precision_init;
+ break;
case AD1984A_TOUCHSMART:
spec->mixers[0] = ad1984a_touchsmart_mixers;
spec->init_verbs[0] = ad1984a_touchsmart_verbs;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5d582de..0ef0035 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1290,7 +1290,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case 0x10ec0883:
case 0x10ec0885:
case 0x10ec0887:
- case 0x10ec0889:
+ /*case 0x10ec0889:*/ /* this causes an SPDIF problem */
alc889_coef_init(codec);
break;
case 0x10ec0888:
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index c0dcfca..c66d3f6 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1568,6 +1568,46 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
{
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0104),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "Roland", */
+ /* .product_name = "UM-1G", */
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* Boss JS-8 Jam Station */
+ USB_DEVICE(0x0582, 0x0109),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "BOSS", */
+ /* .product_name = "JS-8", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
/* has ID 0x0110 when not in Advanced Driver mode */
USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-03-23 13:49 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-03-23 13:49 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
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));
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-02-27 9:10 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-02-27 9:10 UTC (permalink / raw)
To: Linus Torvalds
Cc: Jaroslav Kysela, Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.38-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All small and simple changes.
Thanks!
Takashi
===
David Henningsson (3):
ALSA: HDA: Fix mic initialization in VIA auto parser
ALSA: HDA: Add a new Conexant codec 506e (20590)
ALSA: HDA: Add ideapad quirk for two Dell machines
Dmitry Eremin-Solenikov (1):
ASoC: correct pxa AC97 DAI names
Eric Bénard (1):
eukrea-tlv320: fix platform_name
Lu Guanqun (1):
ALSA: fix one memory leak in sound jack
Mark Brown (3):
ASoC: Fix WM8958 default microphone detection argument ordering
ASoC: Correct definition of WM8903_VMID_RES_5K
ASoC: Hook wm_hubs micbiases up to CLK_SYS
Takashi Iwai (1):
ALSA: usb-audio: fix oops due to cleanup race when disconnecting
Vitaliy Kulikov (1):
ALSA: hda - Add support for new IDT 92HD98 and 92HD99 codecs
---
sound/core/jack.c | 1 +
sound/pci/hda/patch_conexant.c | 5 +++++
sound/pci/hda/patch_sigmatel.c | 15 ++++++++++++---
sound/pci/hda/patch_via.c | 2 +-
sound/soc/codecs/wm8903.h | 2 +-
sound/soc/codecs/wm8994.c | 5 ++---
sound/soc/codecs/wm_hubs.c | 3 +++
sound/soc/imx/eukrea-tlv320.c | 2 +-
sound/soc/pxa/e740_wm9705.c | 4 ++--
sound/soc/pxa/e750_wm9705.c | 4 ++--
sound/soc/pxa/e800_wm9712.c | 4 ++--
sound/soc/pxa/em-x270.c | 4 ++--
sound/soc/pxa/mioa701_wm9713.c | 4 ++--
sound/soc/pxa/palm27x.c | 4 ++--
sound/soc/pxa/tosa.c | 4 ++--
sound/soc/pxa/zylonite.c | 4 ++--
sound/usb/card.c | 4 ++++
sound/usb/pcm.c | 7 +++++--
sound/usb/usbaudio.h | 1 +
19 files changed, 52 insertions(+), 27 deletions(-)
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 4902ae5..53b53e9 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
fail_input:
input_free_device(jack->input_dev);
+ kfree(jack->id);
kfree(jack);
return err;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index dd7c5c1..4d5004e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3114,6 +3114,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
@@ -3937,6 +3939,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_cxt5066 },
{ .id = 0x14f15069, .name = "CX20585",
.patch = patch_cxt5066 },
+ { .id = 0x14f1506e, .name = "CX20590",
+ .patch = patch_cxt5066 },
{ .id = 0x14f15097, .name = "CX20631",
.patch = patch_conexant_auto },
{ .id = 0x14f15098, .name = "CX20632",
@@ -3963,6 +3967,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066");
MODULE_ALIAS("snd-hda-codec-id:14f15067");
MODULE_ALIAS("snd-hda-codec-id:14f15068");
MODULE_ALIAS("snd-hda-codec-id:14f15069");
+MODULE_ALIAS("snd-hda-codec-id:14f1506e");
MODULE_ALIAS("snd-hda-codec-id:14f15097");
MODULE_ALIAS("snd-hda-codec-id:14f15098");
MODULE_ALIAS("snd-hda-codec-id:14f150a1");
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9ea48b4..bd7b123 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -586,7 +586,12 @@ static hda_nid_t stac92hd83xxx_pin_nids[10] = {
0x0f, 0x10, 0x11, 0x1f, 0x20,
};
-static hda_nid_t stac92hd88xxx_pin_nids[10] = {
+static hda_nid_t stac92hd87xxx_pin_nids[6] = {
+ 0x0a, 0x0b, 0x0c, 0x0d,
+ 0x0f, 0x11,
+};
+
+static hda_nid_t stac92hd88xxx_pin_nids[8] = {
0x0a, 0x0b, 0x0c, 0x0d,
0x0f, 0x11, 0x1f, 0x20,
};
@@ -5430,12 +5435,13 @@ again:
switch (codec->vendor_id) {
case 0x111d76d1:
case 0x111d76d9:
+ case 0x111d76e5:
spec->dmic_nids = stac92hd87b_dmic_nids;
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd87b_dmic_nids,
STAC92HD87B_NUM_DMICS);
- spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids);
- spec->pin_nids = stac92hd88xxx_pin_nids;
+ spec->num_pins = ARRAY_SIZE(stac92hd87xxx_pin_nids);
+ spec->pin_nids = stac92hd87xxx_pin_nids;
spec->mono_nid = 0;
spec->num_pwrs = 0;
break;
@@ -5443,6 +5449,7 @@ again:
case 0x111d7667:
case 0x111d7668:
case 0x111d7669:
+ case 0x111d76e3:
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd88xxx_dmic_nids,
STAC92HD88XXX_NUM_DMICS);
@@ -6387,6 +6394,8 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index a76c326..63b0054 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
hda_nid_t nid = cfg->inputs[i].pin;
if (spec->smart51_enabled && is_smart51_pins(spec, nid))
ctl = PIN_OUT;
- else if (i == AUTO_PIN_MIC)
+ else if (cfg->inputs[i].type == AUTO_PIN_MIC)
ctl = PIN_VREF50;
else
ctl = PIN_IN;
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index e8490f3..e3ec243 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -165,7 +165,7 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec,
#define WM8903_VMID_RES_50K 2
#define WM8903_VMID_RES_250K 3
-#define WM8903_VMID_RES_5K 4
+#define WM8903_VMID_RES_5K 6
/*
* R8 (0x08) - Analogue DAC 0
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index a60b5db..ebaee5c 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3000,11 +3000,10 @@ static void wm8958_default_micdet(u16 status, void *data)
report |= SND_JACK_BTN_5;
done:
- snd_soc_jack_report(wm8994->micdet[0].jack,
+ snd_soc_jack_report(wm8994->micdet[0].jack, report,
SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 |
SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 |
- SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT,
- report);
+ SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT);
}
/**
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 613df5d..5168927 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -674,6 +674,9 @@ SND_SOC_DAPM_OUTPUT("LINEOUT2N"),
};
static const struct snd_soc_dapm_route analogue_routes[] = {
+ { "MICBIAS1", NULL, "CLK_SYS" },
+ { "MICBIAS2", NULL, "CLK_SYS" },
+
{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index e20c9e1..1e9bcca 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -79,7 +79,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "imx-pcm-audio.0",
+ .platform_name = "imx-fiq-pcm-audio.0",
.codec_name = "tlv320aic23-codec.0-001a",
.cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 28333e7..dc65650 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -117,7 +117,7 @@ static struct snd_soc_dai_link e740_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
@@ -126,7 +126,7 @@ static struct snd_soc_dai_link e740_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9705-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 01bf316..51897fc 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -99,7 +99,7 @@ static struct snd_soc_dai_link e750_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
@@ -109,7 +109,7 @@ static struct snd_soc_dai_link e750_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9705-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index c6a37c6..053ed20 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -89,7 +89,7 @@ static struct snd_soc_dai_link e800_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -98,7 +98,7 @@ static struct snd_soc_dai_link e800_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index fc22e6e..b13a425 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 0d70fc8..38ca675 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -162,7 +162,7 @@ static struct snd_soc_dai_link mioa701_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9713-hifi",
.codec_name = "wm9713-codec",
.init = mioa701_wm9713_init,
@@ -172,7 +172,7 @@ static struct snd_soc_dai_link mioa701_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9713-aux",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 857db96..504e400 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -132,7 +132,7 @@ static struct snd_soc_dai_link palm27x_dai[] = {
{
.name = "AC97 HiFi",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.codec_name = "wm9712-codec",
.platform_name = "pxa-pcm-audio",
@@ -141,7 +141,7 @@ static struct snd_soc_dai_link palm27x_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9712-aux",
.codec_name = "wm9712-codec",
.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f75804e..4b6e5d6 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -219,7 +219,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -229,7 +229,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index b222a7d..25bba10 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -166,7 +166,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.stream_name = "AC97 HiFi",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_name = "wm9713-hifi",
.init = zylonite_wm9713_init,
},
@@ -175,7 +175,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.stream_name = "AC97 Aux",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_name = "wm9713-aux",
},
{
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 800f7cb..c0f8270 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -323,6 +323,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
return -ENOMEM;
}
+ mutex_init(&chip->shutdown_mutex);
chip->index = idx;
chip->dev = dev;
chip->card = card;
@@ -531,6 +532,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
chip = ptr;
card = chip->card;
mutex_lock(®ister_mutex);
+ mutex_lock(&chip->shutdown_mutex);
chip->shutdown = 1;
chip->num_interfaces--;
if (chip->num_interfaces <= 0) {
@@ -548,9 +550,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
snd_usb_mixer_disconnect(p);
}
usb_chip[chip->index] = NULL;
+ mutex_unlock(&chip->shutdown_mutex);
mutex_unlock(®ister_mutex);
snd_card_free_when_closed(card);
} else {
+ mutex_unlock(&chip->shutdown_mutex);
mutex_unlock(®ister_mutex);
}
}
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 4132522..e3f6805 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -361,6 +361,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
}
if (changed) {
+ mutex_lock(&subs->stream->chip->shutdown_mutex);
/* format changed */
snd_usb_release_substream_urbs(subs, 0);
/* influenced: period_bytes, channels, rate, format, */
@@ -368,6 +369,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
params_rate(hw_params),
snd_pcm_format_physical_width(params_format(hw_params)) *
params_channels(hw_params));
+ mutex_unlock(&subs->stream->chip->shutdown_mutex);
}
return ret;
@@ -385,8 +387,9 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream)
subs->cur_audiofmt = NULL;
subs->cur_rate = 0;
subs->period_bytes = 0;
- if (!subs->stream->chip->shutdown)
- snd_usb_release_substream_urbs(subs, 0);
+ mutex_lock(&subs->stream->chip->shutdown_mutex);
+ snd_usb_release_substream_urbs(subs, 0);
+ mutex_unlock(&subs->stream->chip->shutdown_mutex);
return snd_pcm_lib_free_vmalloc_buffer(substream);
}
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index db3eb21..6e66fff 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -36,6 +36,7 @@ struct snd_usb_audio {
struct snd_card *card;
u32 usb_id;
int shutdown;
+ struct mutex shutdown_mutex;
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
int num_interfaces;
int num_suspended_intf;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-02-20 9:17 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-02-20 9:17 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.38-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing a few driver-specific small changes / quirks and a fairly
minor security fix.
Thanks!
Takashi
===
David Henningsson (3):
ALSA: HDA: Add position_fix quirk for an Asus device
ALSA: HDA: Conexant auto: Handle multiple connections to ADC node
ALSA: HDA: Do not announce false surround in Conexant auto
Raymond Yau (1):
ALSA: au88x0 - Modify pointer callback to give accurate playback position
Takashi Iwai (1):
ALSA: caiaq - Fix possible string-buffer overflow
---
sound/pci/au88x0/au88x0_core.c | 14 +++++++--
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_conexant.c | 63 +++++++++++++++++++++++++++++++---------
sound/usb/caiaq/audio.c | 2 +-
sound/usb/caiaq/midi.c | 2 +-
5 files changed, 63 insertions(+), 19 deletions(-)
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f3..16c0bdf 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
{
stream_t *dma = &vortex->dma_adb[adbdma];
- int temp;
+ int temp, page, delta;
temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
- temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
- return temp;
+ page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+ if (dma->nr_periods >= 4)
+ delta = (page - dma->period_real) & 3;
+ else {
+ delta = (page - dma->period_real);
+ if (delta < 0)
+ delta += dma->nr_periods;
+ }
+ return (dma->period_virt + delta) * dma->period_bytes
+ + (temp & (dma->period_bytes - 1));
}
static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0baffcd..fcedad9 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fbe97d3..dd7c5c1 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3410,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
}
}
spec->multiout.dac_nids = spec->private_dac_nids;
- spec->multiout.max_channels = nums * 2;
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (cfg->hp_outs > 0)
spec->auto_mute = 1;
@@ -3729,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec)
return 0;
}
-static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
+static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
const char *dir, int cidx,
- hda_nid_t nid, int hda_dir)
+ hda_nid_t nid, int hda_dir, int amp_idx)
{
static char name[32];
static struct snd_kcontrol_new knew[] = {
@@ -3743,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
for (i = 0; i < 2; i++) {
struct snd_kcontrol *kctl;
- knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir);
+ knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+ hda_dir);
knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
knew[i].index = cidx;
snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]);
@@ -3759,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
return 0;
}
+#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \
+ cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+
#define cx_auto_add_pb_volume(codec, nid, str, idx) \
cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
@@ -3808,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
struct conexant_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
static const char *prev_label;
- int i, err, cidx;
+ int i, err, cidx, conn_len;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
+ int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
+ int adc_nid = spec->adc_nids[0];
+
+ conn_len = snd_hda_get_connections(codec, adc_nid, conn,
+ HDA_MAX_CONNECTIONS);
+ if (conn_len < 0)
+ return conn_len;
+
+ multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
+ if (!multi_adc_volume) {
+ err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
+ HDA_INPUT);
+ if (err < 0)
+ return err;
+ }
- err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0],
- HDA_INPUT);
- if (err < 0)
- return err;
prev_label = NULL;
cidx = 0;
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
const char *label;
- if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+ int j;
+ int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
+ if (!pin_amp && !multi_adc_volume)
continue;
+
label = hda_get_autocfg_input_label(codec, cfg, i);
if (label == prev_label)
cidx++;
else
cidx = 0;
prev_label = label;
- err = cx_auto_add_volume(codec, label, " Capture", cidx,
- nid, HDA_INPUT);
- if (err < 0)
- return err;
+
+ if (pin_amp) {
+ err = cx_auto_add_volume(codec, label, " Boost", cidx,
+ nid, HDA_INPUT);
+ if (err < 0)
+ return err;
+ }
+
+ if (!multi_adc_volume)
+ continue;
+ for (j = 0; j < conn_len; j++) {
+ if (conn[j] == nid) {
+ err = cx_auto_add_volume_idx(codec, label,
+ " Capture", cidx, adc_nid, HDA_INPUT, j);
+ if (err < 0)
+ return err;
+ break;
+ }
+ }
}
return 0;
}
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 68b9747..66eabaf 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
}
dev->pcm->private_data = dev;
- strcpy(dev->pcm->name, dev->product_name);
+ strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name));
memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 2f218c7..a1a4708 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device)
if (ret < 0)
return ret;
- strcpy(rmidi->name, device->product_name);
+ strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name));
rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX;
rmidi->private_data = device;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-02-13 9:12 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-02-13 9:12 UTC (permalink / raw)
To: Linus Torvalds
Cc: Jaroslav Kysela, Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.38-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All changes are small regression / trivial fixes or new quirks.
Thanks!
Takashi
===
Anisse Astier (1):
ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662
Clemens Ladisch (2):
ALSA: hrtimer: handle delayed timer interrupts
ALSA: hrtimer: remove superfluous tasklet invocation
David Henningsson (1):
ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G
Janusz Krzysztofik (1):
ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init()
Joseph Teichman (1):
ALSA: usbaudio - Enable the E-MU 0204 USB
Mark Brown (2):
ASoC: Create an AIF1ADCDAT signal widget to match AIF2
ASoC: Improve WM8994 digital power sequencing
Takashi Iwai (2):
ALSA: hda - Fix missing CA initialization for HDMI/DP
ALSA: hda - Don't handle empty patch files
Vaibhav Bedia (1):
asoc: davinci: da830/omap-l137: correct cpu_dai_name
Łukasz Wojniłowicz (1):
ALSA: hda - switch lfe with side in mixer for 4930g
---
sound/core/hrtimer.c | 7 ++++---
sound/pci/hda/hda_intel.c | 2 +-
sound/pci/hda/patch_hdmi.c | 2 ++
sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++++-
sound/soc/codecs/wm8994.c | 26 ++++++++++++++++++++++----
sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++---
sound/soc/soc-core.c | 2 +-
sound/usb/mixer.c | 4 ++--
sound/usb/quirks-table.h | 7 +++++++
sound/usb/quirks.c | 3 ++-
10 files changed, 82 insertions(+), 16 deletions(-)
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 7730575..b8b31c4 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
{
struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
struct snd_timer *t = stime->timer;
+ unsigned long oruns;
if (!atomic_read(&stime->running))
return HRTIMER_NORESTART;
- hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
- snd_timer_interrupt(stime->timer, t->sticks);
+ oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
+ snd_timer_interrupt(stime->timer, t->sticks * oruns);
if (!atomic_read(&stime->running))
return HRTIMER_NORESTART;
@@ -104,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
}
static struct snd_timer_hardware hrtimer_hw = {
- .flags = SNDRV_TIMER_HW_AUTO,
+ .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
.open = snd_hrtimer_open,
.close = snd_hrtimer_close,
.start = snd_hrtimer_start,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2e91a99..0baffcd 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2703,7 +2703,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
if (err < 0)
goto out_free;
#ifdef CONFIG_SND_HDA_PATCH_LOADER
- if (patch[dev]) {
+ if (patch[dev] && *patch[dev]) {
snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
patch[dev]);
err = snd_hda_load_patch(chip->bus, patch[dev]);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2d5b83f..a587677 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
hdmi_ai->ver = 0x01;
hdmi_ai->len = 0x0a;
hdmi_ai->CC02_CT47 = channels - 1;
+ hdmi_ai->CA = ca;
hdmi_checksum_audio_infoframe(hdmi_ai);
} else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */
struct dp_audio_infoframe *dp_ai;
@@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
dp_ai->len = 0x1b;
dp_ai->ver = 0x11 << 2;
dp_ai->CC02_CT47 = channels - 1;
+ dp_ai->CA = ca;
} else {
snd_printd("HDMI: unknown connection type at pin %d\n",
pin_nid);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2fa9ed9..3328a25 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2290,6 +2290,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -10359,7 +10382,7 @@ static struct alc_config_preset alc882_presets[] = {
.init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_4930G] = {
- .mixers = { alc888_base_mixer,
+ .mixers = { alc888_acer_aspire_4930g_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
alc888_acer_aspire_4930g_verbs },
@@ -18802,6 +18825,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
{}
};
@@ -19494,6 +19518,7 @@ static const struct alc_fixup alc662_fixups[] = {
};
static struct snd_pci_quirk alc662_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3351f77..37b8aa8 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
0, WM8994_POWER_MANAGEMENT_4, 8, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
0, WM8994_POWER_MANAGEMENT_4, 10, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
/* AIF3 output */
@@ -1578,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "Right Headphone Mux", "DAC", "DAC1R" },
};
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+ { "AIF1DACDAT", NULL, "AIF2DACDAT" },
+ { "AIF2DACDAT", NULL, "AIF1DACDAT" },
+ { "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+ { "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+};
+
static const struct snd_soc_dapm_route wm8994_intercon[] = {
{ "AIF2DACL", NULL, "AIF2DAC Mux" },
{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -3129,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
case WM8994:
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
+
+ if (wm8994->revision < 4)
+ snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+ ARRAY_SIZE(wm8994_revd_intercon));
+
break;
case WM8958:
snd_soc_dapm_add_routes(dapm, wm8958_intercon,
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index b36f0b3..fe79842 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
.ops = &evm_spdif_ops,
},
};
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai_name = "davinci-mcasp.1",
+ .codec_dai_name = "tlv320aic3x-hifi",
+ .codec_name = "tlv320aic3x-codec.1-0018",
+ .platform_name = "davinci-pcm-audio",
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
static struct snd_soc_card da830_snd_soc_card = {
.name = "DA830/OMAP-L137 EVM",
- .dai_link = &da8xx_evm_dai,
+ .dai_link = &da830_evm_dai,
.num_links = 1,
};
static struct snd_soc_card da850_snd_soc_card = {
.name = "DA850/OMAP-L138 EVM",
- .dai_link = &da8xx_evm_dai,
+ .dai_link = &da850_evm_dai,
.num_links = 1,
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c4b6061..c3f6f1e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
rtd = &card->rtd_aux[num];
name = aux_dev->name;
}
+ rtd->card = card;
/* machine controls, routes and widgets are not prefixed */
temp = codec->name_prefix;
@@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
/* register the rtd device */
rtd->codec = codec;
- rtd->card = card;
rtd->dev.parent = card->dev;
rtd->dev.release = rtd_release;
rtd->dev.init_name = name;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7df89b3..85af605 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -95,7 +95,7 @@ enum {
};
-/*E-mu 0202(0404) eXtension Unit(XU) control*/
+/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/
enum {
USB_XU_CLOCK_RATE = 0xe301,
USB_XU_CLOCK_SOURCE = 0xe302,
@@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
cval->initialized = 1;
} else {
if (type == USB_XU_CLOCK_RATE) {
- /* E-Mu USB 0404/0202/TrackerPre
+ /* E-Mu USB 0404/0202/TrackerPre/0204
* samplerate control quirk
*/
cval->min = 0;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 3599987..921a86f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -79,6 +79,13 @@
.idProduct = 0x3f0a,
.bInterfaceClass = USB_CLASS_AUDIO,
},
+{
+ /* E-Mu 0204 USB */
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x041e,
+ .idProduct = 0x3f19,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
/*
* Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index cf8bf08..e314cdb 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat
}
/*
- * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device,
+ * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device,
* not for interface.
*/
@@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
+ case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
set_format_emu_quirk(subs, fmt);
break;
}
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-02-06 12:05 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-02-06 12:05 UTC (permalink / raw)
To: Linus Torvalds
Cc: Jaroslav Kysela, Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.38-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All small fixes but for slightly intensive changes found in Conexant
HD-audio parser, which are straight-forward fixes for ASUS laptops and
SPDIF handling.
Thanks!
Takashi
===
Amerigo Wang (1):
sound: silent echo'ed messages in Makefile
Andy Robinson (1):
ALSA: HDA: cxt5066 - Use asus model for Asus U50F, select correct SPDIF output
Clemens Ladisch (1):
ALSA: oxygen: fix output routing on Xonar DG
David Henningsson (3):
ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx
ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
ALSA: HDA: Fix microphone(s) on Lenovo Edge 13
Janusz Krzysztofik (2):
ASoC: Amstrad Delta: fix const related build error
ASoC: CX20442: fix NULL pointer dereference
Jarkko Nikula (1):
ASoC: Fix module refcount for auxiliary devices
Manjunathappa, Prakash (1):
ASoC: DaVinci: fix kernel panic due to uninitialized platform_data
Stephen Warren (1):
ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw()
Takashi Iwai (2):
ALSA: hda - Fix memory leaks in conexant jack arrays
ALSA: use linux/io.h to fix compile warnings
---
Documentation/sound/alsa/HD-Audio-Models.txt | 1 +
sound/drivers/mtpav.c | 3 +-
sound/oss/Makefile | 4 +-
sound/pci/hda/patch_conexant.c | 149 +++++++++++++++-----------
sound/pci/oxygen/oxygen.h | 2 +
sound/pci/oxygen/oxygen_mixer.c | 2 +
sound/pci/oxygen/xonar_dg.c | 36 ++++++
sound/pcmcia/pdaudiocf/pdaudiocf.h | 2 +-
sound/pcmcia/vx/vxp_ops.c | 2 +-
sound/soc/codecs/cq93vc.c | 2 +-
sound/soc/codecs/cx20442.c | 3 +
sound/soc/omap/ams-delta.c | 2 -
sound/soc/soc-core.c | 3 -
sound/soc/soc-dapm.c | 6 +-
14 files changed, 138 insertions(+), 79 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16ae430..0caf77e 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -296,6 +296,7 @@ Conexant 5066
=============
laptop Basic Laptop config (default)
hp-laptop HP laptops, e g G60
+ asus Asus K52JU, Lenovo G560
dell-laptop Dell laptops
dell-vostro Dell Vostro
olpc-xo-1_5 OLPC XO 1.5
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index da03597..5c426df 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -55,14 +55,13 @@
#include <linux/err.h>
#include <linux/platform_device.h>
#include <linux/ioport.h>
+#include <linux/io.h>
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/rawmidi.h>
#include <linux/delay.h>
-#include <asm/io.h>
-
/*
* globals
*/
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index 96f14dc..90ffb99 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
$(obj)/bin2hex pss_synth < $< > $@
else
$(obj)/pss_boot.h:
- ( \
+ $(Q)( \
echo 'static unsigned char * pss_synth = NULL;'; \
echo 'static int pss_synthLen = 0;'; \
) > $@
@@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
$(obj)/hex2hex -i trix_boot < $< > $@
else
$(obj)/trix_boot.h:
- ( \
+ $(Q)( \
echo 'static unsigned char * trix_boot = NULL;'; \
echo 'static int trix_boot_len = 0;'; \
) > $@
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9bb030a..fbe97d3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -85,6 +85,7 @@ struct conexant_spec {
unsigned int auto_mic;
int auto_mic_ext; /* autocfg.inputs[] index for ext mic */
unsigned int need_dac_fix;
+ hda_nid_t slave_dig_outs[2];
/* capture */
unsigned int num_adc_nids;
@@ -127,6 +128,7 @@ struct conexant_spec {
unsigned int ideapad:1;
unsigned int thinkpad:1;
unsigned int hp_laptop:1;
+ unsigned int asus:1;
unsigned int ext_mic_present;
unsigned int recording;
@@ -352,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec)
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
spec->dig_in_nid;
}
+ if (spec->slave_dig_outs[0])
+ codec->slave_dig_outs = spec->slave_dig_outs;
}
return 0;
@@ -403,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec,
struct conexant_spec *spec;
struct conexant_jack *jack;
const char *name;
- int err;
+ int i, err;
spec = codec->spec;
snd_array_init(&spec->jacks, sizeof(*jack), 32);
+
+ jack = spec->jacks.list;
+ for (i = 0; i < spec->jacks.used; i++, jack++)
+ if (jack->nid == nid)
+ return 0 ; /* already present */
+
jack = snd_array_new(&spec->jacks);
name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ;
@@ -2100,7 +2110,7 @@ static int patch_cxt5051(struct hda_codec *codec)
static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-#define CXT5066_SPDIF_OUT 0x21
+static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
/* OLPC's microphone port is DC coupled for use with external sensors,
* therefore we use a 50% mic bias in order to center the input signal with
@@ -2312,6 +2322,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
}
}
+
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_asus_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_jack_detect(codec, 0x1b);
+ snd_printdd("CXT5066: external microphone present=%d\n", present);
+ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 1 : 0);
+}
+
+
/* toggle input of built-in digital mic and mic jack appropriately */
static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
{
@@ -2387,79 +2410,55 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
cxt5066_update_speaker(codec);
}
-/* unsolicited event for jack sensing */
-static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
+/* Dispatch the right mic autoswitch function */
+static void cxt5066_automic(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
- snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
- /* ignore mic events in DC mode; we're always using the jack */
- if (!spec->dc_enable)
- cxt5066_olpc_automic(codec);
- break;
- }
-}
-/* unsolicited event for jack sensing */
-static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res)
-{
- snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
+ if (spec->dell_vostro)
cxt5066_vostro_automic(codec);
- break;
- }
-}
-
-/* unsolicited event for jack sensing */
-static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
-{
- snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
+ else if (spec->ideapad)
cxt5066_ideapad_automic(codec);
- break;
- }
+ else if (spec->thinkpad)
+ cxt5066_thinkpad_automic(codec);
+ else if (spec->hp_laptop)
+ cxt5066_hp_laptop_automic(codec);
+ else if (spec->asus)
+ cxt5066_asus_automic(codec);
}
/* unsolicited event for jack sensing */
-static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
{
- snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+ struct conexant_spec *spec = codec->spec;
+ snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
switch (res >> 26) {
case CONEXANT_HP_EVENT:
cxt5066_hp_automute(codec);
break;
case CONEXANT_MIC_EVENT:
- cxt5066_hp_laptop_automic(codec);
+ /* ignore mic events in DC mode; we're always using the jack */
+ if (!spec->dc_enable)
+ cxt5066_olpc_automic(codec);
break;
}
}
/* unsolicited event for jack sensing */
-static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
{
- snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
+ snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
switch (res >> 26) {
case CONEXANT_HP_EVENT:
cxt5066_hp_automute(codec);
break;
case CONEXANT_MIC_EVENT:
- cxt5066_thinkpad_automic(codec);
+ cxt5066_automic(codec);
break;
}
}
+
static const struct hda_input_mux cxt5066_analog_mic_boost = {
.num_items = 5,
.items = {
@@ -2633,6 +2632,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
spec->recording = 0;
}
+static void conexant_check_dig_outs(struct hda_codec *codec,
+ hda_nid_t *dig_pins,
+ int num_pins)
+{
+ struct conexant_spec *spec = codec->spec;
+ hda_nid_t *nid_loc = &spec->multiout.dig_out_nid;
+ int i;
+
+ for (i = 0; i < num_pins; i++, dig_pins++) {
+ unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins);
+ if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE)
+ continue;
+ if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1)
+ continue;
+ if (spec->slave_dig_outs[0])
+ nid_loc++;
+ else
+ nid_loc = spec->slave_dig_outs;
+ }
+}
+
static struct hda_input_mux cxt5066_capture_source = {
.num_items = 4,
.items = {
@@ -3039,20 +3059,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
/* initialize jack-sensing, too */
static int cxt5066_init(struct hda_codec *codec)
{
- struct conexant_spec *spec = codec->spec;
-
snd_printdd("CXT5066: init\n");
conexant_init(codec);
if (codec->patch_ops.unsol_event) {
cxt5066_hp_automute(codec);
- if (spec->dell_vostro)
- cxt5066_vostro_automic(codec);
- else if (spec->ideapad)
- cxt5066_ideapad_automic(codec);
- else if (spec->thinkpad)
- cxt5066_thinkpad_automic(codec);
- else if (spec->hp_laptop)
- cxt5066_hp_laptop_automic(codec);
+ cxt5066_automic(codec);
}
cxt5066_set_mic_boost(codec);
return 0;
@@ -3080,6 +3091,7 @@ enum {
CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */
CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */
CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */
+ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
CXT5066_HP_LAPTOP, /* HP Laptop */
CXT5066_MODELS
};
@@ -3091,6 +3103,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
[CXT5066_DELL_VOSTRO] = "dell-vostro",
[CXT5066_IDEAPAD] = "ideapad",
[CXT5066_THINKPAD] = "thinkpad",
+ [CXT5066_ASUS] = "asus",
[CXT5066_HP_LAPTOP] = "hp-laptop",
};
@@ -3102,7 +3115,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3111,7 +3126,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
{}
};
@@ -3133,7 +3150,8 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->multiout.max_channels = 2;
spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
spec->multiout.dac_nids = cxt5066_dac_nids;
- spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+ conexant_check_dig_outs(codec, cxt5066_digout_pin_nids,
+ ARRAY_SIZE(cxt5066_digout_pin_nids));
spec->num_adc_nids = 1;
spec->adc_nids = cxt5066_adc_nids;
spec->capsrc_nids = cxt5066_capsrc_nids;
@@ -3167,17 +3185,20 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->num_init_verbs++;
spec->dell_automute = 1;
break;
+ case CXT5066_ASUS:
case CXT5066_HP_LAPTOP:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->init_verbs[spec->num_init_verbs] =
cxt5066_init_verbs_hp_laptop;
spec->num_init_verbs++;
- spec->hp_laptop = 1;
+ spec->hp_laptop = board_config == CXT5066_HP_LAPTOP;
+ spec->asus = board_config == CXT5066_ASUS;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
/* no S/PDIF out */
- spec->multiout.dig_out_nid = 0;
+ if (board_config == CXT5066_HP_LAPTOP)
+ spec->multiout.dig_out_nid = 0;
/* input source automatically selected */
spec->input_mux = NULL;
spec->port_d_mode = 0;
@@ -3207,7 +3228,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_DELL_VOSTRO:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_vostro_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->init_verbs[0] = cxt5066_init_verbs_vostro;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
@@ -3224,7 +3245,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_IDEAPAD:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_ideapad_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->init_verbs[0] = cxt5066_init_verbs_ideapad;
@@ -3240,7 +3261,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_THINKPAD:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_thinkpad_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->init_verbs[0] = cxt5066_init_verbs_thinkpad;
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c2ae63d..f53897a 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -92,6 +92,8 @@ struct oxygen_model {
void (*update_dac_volume)(struct oxygen *chip);
void (*update_dac_mute)(struct oxygen *chip);
void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed);
+ unsigned int (*adjust_dac_routing)(struct oxygen *chip,
+ unsigned int play_routing);
void (*gpio_changed)(struct oxygen *chip);
void (*uart_input)(struct oxygen *chip);
void (*ac97_switch)(struct oxygen *chip,
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 9bff14d..26c7e8b 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip)
(1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
(2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
(3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+ if (chip->model.adjust_dac_routing)
+ reg_value = chip->model.adjust_dac_routing(chip, reg_value);
oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
OXYGEN_PLAY_DAC0_SOURCE_MASK |
OXYGEN_PLAY_DAC1_SOURCE_MASK |
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e1fa602..bc6eb58 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -24,6 +24,11 @@
*
* SPI 0 -> CS4245
*
+ * I²S 1 -> CS4245
+ * I²S 2 -> CS4361 (center/LFE)
+ * I²S 3 -> CS4361 (surround)
+ * I²S 4 -> CS4361 (front)
+ *
* GPIO 3 <- ?
* GPIO 4 <- headphone detect
* GPIO 5 -> route input jack to line-in (0) or mic-in (1)
@@ -36,6 +41,7 @@
* input 1 <- aux
* input 2 <- front mic
* input 4 <- line/mic
+ * DAC out -> headphones
* aux out -> front panel headphones
*/
@@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip,
cs4245_write_cached(chip, CS4245_ADC_CTRL, value);
}
+static inline unsigned int shift_bits(unsigned int value,
+ unsigned int shift_from,
+ unsigned int shift_to,
+ unsigned int mask)
+{
+ if (shift_from < shift_to)
+ return (value << (shift_to - shift_from)) & mask;
+ else
+ return (value >> (shift_from - shift_to)) & mask;
+}
+
+static unsigned int adjust_dg_dac_routing(struct oxygen *chip,
+ unsigned int play_routing)
+{
+ return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC1_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC2_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC0_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC3_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
static int output_switch_info(struct snd_kcontrol *ctl,
struct snd_ctl_elem_info *info)
{
@@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = {
.resume = dg_resume,
.set_dac_params = set_cs4245_dac_params,
.set_adc_params = set_cs4245_adc_params,
+ .adjust_dac_routing = adjust_dg_dac_routing,
.dump_registers = dump_cs4245_registers,
.model_data_size = sizeof(struct dg),
.device_config = PLAYBACK_0_TO_I2S |
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index bd26e09..6ce9ad7 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -22,7 +22,7 @@
#define __PDAUDIOCF_H
#include <sound/pcm.h>
-#include <asm/io.h>
+#include <linux/io.h>
#include <linux/interrupt.h>
#include <pcmcia/cistpl.h>
#include <pcmcia/ds.h>
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 989e04a..fe33e12 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -23,8 +23,8 @@
#include <linux/delay.h>
#include <linux/device.h>
#include <linux/firmware.h>
+#include <linux/io.h>
#include <sound/core.h>
-#include <asm/io.h>
#include "vxpocket.h"
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd0..347a567 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
static int cq93vc_probe(struct snd_soc_codec *codec)
{
- struct davinci_vc *davinci_vc = codec->dev->platform_data;
+ struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
davinci_vc->cq93vc.codec = codec;
codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e86..bb4bf65 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
return 0;
}
+static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+
static struct snd_soc_codec_driver cx20442_codec_dev = {
.probe = cx20442_codec_probe,
.remove = cx20442_codec_remove,
+ .reg_cache_default = &cx20442_reg,
.reg_cache_size = 1,
.reg_word_size = sizeof(u8),
.read = cx20442_read_reg_cache,
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdc..3167be6 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
/* Set up digital mute if not provided by the codec */
if (!codec_dai->driver->ops) {
codec_dai->driver->ops = &ams_delta_dai_ops;
- } else if (!codec_dai->driver->ops->digital_mute) {
- codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
} else {
ams_delta_ops.startup = ams_delta_startup;
ams_delta_ops.shutdown = ams_delta_shutdown;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bac7291..c4b6061 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
goto out;
found:
- if (!try_module_get(codec->dev->driver->owner))
- return -ENODEV;
-
ret = soc_probe_codec(card, codec);
if (ret < 0)
return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730a..8194f15 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- unsigned int val, val_mask;
+ unsigned int val;
int connect, change;
struct snd_soc_dapm_update update;
@@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
if (invert)
val = max - val;
- val_mask = mask << shift;
+ mask = mask << shift;
val = val << shift;
mutex_lock(&widget->codec->mutex);
widget->value = val;
- change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+ change = snd_soc_test_bits(widget->codec, reg, mask, val);
if (change) {
if (val)
/* new connection */
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-01-28 9:15 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-01-28 9:15 UTC (permalink / raw)
To: Linus Torvalds
Cc: Jaroslav Kysela, Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.38-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All small fixes.
Thanks!
Takashi
===
Andreas Mohr (1):
ALSA: azt3328 - fix broken AZF_FMT_XLATE macro
David Henningsson (2):
ALSA: HDA: Fix dmesg output of HDMI supported bits
ALSA: HDA: Fix automute on Thinkpad L412/L512
Dimitris Papastamos (1):
ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()
Dmitry Eremin-Solenikov (1):
ASoC: correct link specifications for corgi, poodle and spitz
Hans-Christian Egtvedt (1):
ALSA: fix invalid hardware.h include in ac97c for AVR32 architecture
Jesper Juhl (1):
ALSA: Xonar, CS43xx: Don't overrun static array
Lars-Peter Clausen (2):
ASoC: Fix codec device id format used by some dai_links
ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s
Mark Brown (1):
ASoC: Handle low measured DC offsets for wm_hubs devices
Qiao Zhou (1):
ASoC: WM8994: fix wrong value in tristate function
Rajashekhara, Sudhakar (1):
ASoC: da8xx/omap-l1xx: match codec_name with i2c ids
---
sound/atmel/ac97c.c | 5 ++-
sound/pci/azt3328.c | 38 ++++++++++--------------
sound/pci/hda/hda_eld.c | 2 +-
sound/pci/hda/patch_realtek.c | 6 ++-
sound/pci/oxygen/xonar_cs43xx.c | 2 +-
sound/soc/atmel/snd-soc-afeb9260.c | 2 +-
sound/soc/blackfin/bf5xx-ssm2602.c | 2 +-
sound/soc/codecs/wm8994.c | 2 +-
sound/soc/codecs/wm8995.c | 2 +-
sound/soc/codecs/wm_hubs.c | 15 ++++-----
sound/soc/davinci/davinci-evm.c | 2 +-
sound/soc/pxa/corgi.c | 4 +-
sound/soc/pxa/poodle.c | 2 +-
sound/soc/pxa/spitz.c | 4 +-
sound/soc/samsung/neo1973_gta02_wm8753.c | 6 ++--
sound/soc/samsung/neo1973_wm8753.c | 6 ++--
sound/soc/samsung/s3c24xx_simtec_hermes.c | 4 +-
sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 +-
sound/soc/samsung/s3c24xx_uda134x.c | 2 +-
19 files changed, 54 insertions(+), 56 deletions(-)
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 10c3a87..b310702 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -33,9 +33,12 @@
#include <linux/dw_dmac.h>
#include <mach/cpu.h>
-#include <mach/hardware.h>
#include <mach/gpio.h>
+#ifdef CONFIG_ARCH_AT91
+#include <mach/hardware.h>
+#endif
+
#include "ac97c.h"
enum {
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 6117595..573594b 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
snd_azf3328_dbgcallenter();
switch (bitrate) {
-#define AZF_FMT_XLATE(in_freq, out_bits) \
- do { \
- case AZF_FREQ_ ## in_freq: \
- freq = SOUNDFORMAT_FREQ_ ## out_bits; \
- break; \
- } while (0);
- AZF_FMT_XLATE(4000, SUSPECTED_4000)
- AZF_FMT_XLATE(4800, SUSPECTED_4800)
- /* the AZF3328 names it "5510" for some strange reason: */
- AZF_FMT_XLATE(5512, 5510)
- AZF_FMT_XLATE(6620, 6620)
- AZF_FMT_XLATE(8000, 8000)
- AZF_FMT_XLATE(9600, 9600)
- AZF_FMT_XLATE(11025, 11025)
- AZF_FMT_XLATE(13240, SUSPECTED_13240)
- AZF_FMT_XLATE(16000, 16000)
- AZF_FMT_XLATE(22050, 22050)
- AZF_FMT_XLATE(32000, 32000)
+ case AZF_FREQ_4000: freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+ case AZF_FREQ_4800: freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+ case AZF_FREQ_5512:
+ /* the AZF3328 names it "5510" for some strange reason */
+ freq = SOUNDFORMAT_FREQ_5510; break;
+ case AZF_FREQ_6620: freq = SOUNDFORMAT_FREQ_6620; break;
+ case AZF_FREQ_8000: freq = SOUNDFORMAT_FREQ_8000; break;
+ case AZF_FREQ_9600: freq = SOUNDFORMAT_FREQ_9600; break;
+ case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break;
+ case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+ case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break;
+ case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break;
+ case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
default:
snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
/* fall-through */
- AZF_FMT_XLATE(44100, 44100)
- AZF_FMT_XLATE(48000, 48000)
- AZF_FMT_XLATE(66200, SUSPECTED_66200)
-#undef AZF_FMT_XLATE
+ case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
+ case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
+ case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
}
/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 4a66347..74b0560 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
snd_print_pcm_rates(a->rates, buf, sizeof(buf));
if (a->format == AUDIO_CODING_TYPE_LPCM)
- snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
+ snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
else if (a->max_bitrate)
snprintf(buf2, sizeof(buf2),
", max bitrate = %d", a->max_bitrate);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index be4df4c..2fa9ed9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14954,9 +14954,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
- SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
{}
diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c
index 9f72d42..2527191 100644
--- a/sound/pci/oxygen/xonar_cs43xx.c
+++ b/sound/pci/oxygen/xonar_cs43xx.c
@@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip,
unsigned int i;
snd_iprintf(buffer, "\nCS4398: 7?");
- for (i = 2; i <= 8; ++i)
+ for (i = 2; i < 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
snd_iprintf(buffer, "\n");
dump_cs4362a_registers(data, buffer);
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e..5e4d499 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
.cpu_dai_name = "atmel-ssc-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "atmel_pcm-audio",
- .codec_name = "tlv320aic23-codec.0-0x1a",
+ .codec_name = "tlv320aic23-codec.0-001a",
.init = afeb9260_tlv320aic23_init,
.ops = &afeb9260_ops,
};
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24..ad28663 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
.cpu_dai_name = "bf5xx-i2s",
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bf5xx-pcm-audio",
- .codec_name = "ssm2602-codec.0-0x1b",
+ .codec_name = "ssm2602-codec.0-001b",
.ops = &bf5xx_ssm2602_ops,
};
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 247a6a9..3351f77 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2386,7 +2386,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
#define WM8994_RATES SNDRV_PCM_RATE_8000_96000
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 6045cbd..608c84c 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
/* The size in bits of the FLL divide multiplied by 10
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index c466982..613df5d 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
static void calibrate_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ s8 offset;
u16 reg, reg_l, reg_r, dcs_cfg;
/* If we're using a digital only path and have a previously
@@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
hubs->dcs_codes);
/* HPOUT1L */
- if (reg_l + hubs->dcs_codes > 0 &&
- reg_l + hubs->dcs_codes < 0xff)
- reg_l += hubs->dcs_codes;
- dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ offset = reg_l;
+ offset += hubs->dcs_codes;
+ dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1R */
- if (reg_r + hubs->dcs_codes > 0 &&
- reg_r + hubs->dcs_codes < 0xff)
- reg_r += hubs->dcs_codes;
- dcs_cfg |= reg_r;
+ offset = reg_r;
+ offset += hubs->dcs_codes;
+ dcs_cfg |= (u8)offset;
dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6ba..b36f0b3 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-0018",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0..784cff5 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
- .cpu_dai_name = "pxa-is2-dai",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec-0.001a",
+ .codec_name = "wm8731-codec-0.001b",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee1..a7d4999 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec.0-001a",
+ .codec_name = "wm8731-codec.0-001b",
.init = poodle_wm8731_init,
.ops = &poodle_ops,
};
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69..8e15713 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link spitz_dai = {
.name = "wm8750",
.stream_name = "WM8750",
- .cpu_dai_name = "pxa-is2",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8750-codec.0-001a",
+ .codec_name = "wm8750-codec.0-001b",
.init = spitz_wm8750_init,
.ops = &spitz_ops,
};
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 3eec610..0d0ae2b 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -397,11 +397,11 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
.stream_name = "WM8753 HiFi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
.init = neo1973_gta02_wm8753_init,
.platform_name = "samsung-audio",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_gta02_hifi_ops,
},
{ /* Voice via BT */
@@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
.ops = &neo1973_gta02_voice_ops,
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.platform_name = "samsung-audio",
},
};
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a2451..d20815d 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -559,9 +559,9 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.name = "WM8753",
.stream_name = "WM8753 HiFi",
.platform_name = "samsung-audio",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index bb4292e..08fcaaa 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -94,8 +94,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic33 = {
.name = "tlv320aic33",
.stream_name = "TLV320AIC33",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .codec_name = "tlv320aic3x-codec.0-001a",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index fbba4e3..116e3e6 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -85,8 +85,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic23 = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .codec_name = "tlv320aic3x-codec.0-001a",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index cdc8ecb..2c09e93 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
.stream_name = "UDA134X",
.codec_name = "uda134x-hifi",
.codec_dai_name = "uda134x-hifi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.ops = &s3c24xx_uda134x_ops,
.platform_name = "samsung-audio",
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2011-01-21 7:40 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2011-01-21 7:40 UTC (permalink / raw)
To: Linus Torvalds
Cc: Jaroslav Kysela, Mark Brown, Liam Girdwood, Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.38-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Mostly trivial fixes in HD-audio and ASoC. The only large changes are
documentation updates.
Thanks!
Takashi
===
Anisse Astier (1):
ALSA: hda - Fix EAPD to low on CZC P10T tablet computer with ALC662
Barry Song (2):
ASoC: Blackfin TDM: use external frame syncs
ASoC: Blackfin: fix DAI/SPORT config dependency issues
Brian Bloniarz (1):
ALSA: ice1712 delta - initialize SPI clock
David Henningsson (1):
ALSA: HDA: Add SKU ignore for another Thinkpad Edge 14
Mike Frysinger (2):
ASoC: Blackfin TDM: fix missed snd_soc_dai_get_drvdata update
ASoC: Blackfin AC97: fix build error after multi-component update
Seungwhan Youn (1):
ASoC: documentation updates
Takashi Iwai (3):
Revert "ALSA: HDA: Create mixers on ALC887"
ALSA: hda - Add quirk for HP Z-series workstation
ALSA: hda - Fix "unused variable" compile warning
Vasily Khoruzhick (2):
ASoC: PXA: Fix jack detection on Zipit Z2
ASoC: PXA: Fix codec address on Zipit Z2
---
Documentation/sound/alsa/soc/codec.txt | 45 +++++++++++++---------------
Documentation/sound/alsa/soc/machine.txt | 38 ++++++------------------
Documentation/sound/alsa/soc/platform.txt | 12 ++++++-
sound/pci/hda/patch_realtek.c | 26 +++++++++++------
sound/pci/ice1712/delta.c | 7 ++++
sound/soc/blackfin/Kconfig | 11 ++++---
sound/soc/blackfin/bf5xx-ac97.c | 4 +-
sound/soc/blackfin/bf5xx-tdm.c | 10 +++---
sound/soc/pxa/z2.c | 3 +-
9 files changed, 79 insertions(+), 77 deletions(-)
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index 37ba3a7..bce23a4 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -27,42 +27,38 @@ ASoC Codec driver breakdown
1 - Codec DAI and PCM configuration
-----------------------------------
-Each codec driver must have a struct snd_soc_codec_dai to define its DAI and
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
PCM capabilities and operations. This struct is exported so that it can be
registered with the core by your machine driver.
e.g.
-struct snd_soc_codec_dai wm8731_dai = {
- .name = "WM8731",
- /* playback capabilities */
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+ .prepare = wm8731_pcm_prepare,
+ .hw_params = wm8731_hw_params,
+ .shutdown = wm8731_shutdown,
+ .digital_mute = wm8731_mute,
+ .set_sysclk = wm8731_set_dai_sysclk,
+ .set_fmt = wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+ .name = "wm8731-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8731_RATES,
.formats = WM8731_FORMATS,},
- /* capture capabilities */
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8731_RATES,
.formats = WM8731_FORMATS,},
- /* pcm operations - see section 4 below */
- .ops = {
- .prepare = wm8731_pcm_prepare,
- .hw_params = wm8731_hw_params,
- .shutdown = wm8731_shutdown,
- },
- /* DAI operations - see DAI.txt */
- .dai_ops = {
- .digital_mute = wm8731_mute,
- .set_sysclk = wm8731_set_dai_sysclk,
- .set_fmt = wm8731_set_dai_fmt,
- }
+ .ops = &wm8731_dai_ops,
+ .symmetric_rates = 1,
};
-EXPORT_SYMBOL_GPL(wm8731_dai);
2 - Codec control IO
@@ -186,13 +182,14 @@ when the mute is applied or freed.
i.e.
-static int wm8974_mute(struct snd_soc_codec *codec,
- struct snd_soc_codec_dai *dai, int mute)
+static int wm8974_mute(struct snd_soc_dai *dai, int mute)
{
- u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
- if(mute)
- wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+ if (mute)
+ snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
else
- wm8974_write(codec, WM8974_DAC, mute_reg);
+ snd_soc_write(codec, WM8974_DAC, mute_reg);
return 0;
}
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index 2524c75..3e2ec9c 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -12,6 +12,8 @@ the following struct:-
struct snd_soc_card {
char *name;
+ ...
+
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
@@ -22,12 +24,13 @@ struct snd_soc_card {
int (*resume_pre)(struct platform_device *pdev);
int (*resume_post)(struct platform_device *pdev);
- /* machine stream operations */
- struct snd_soc_ops *ops;
+ ...
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;
+
+ ...
};
probe()/remove()
@@ -42,11 +45,6 @@ of any machine audio tasks that have to be done before or after the codec, DAIs
and DMA is suspended and resumed. Optional.
-Machine operations
-------------------
-The machine specific audio operations can be set here. Again this is optional.
-
-
Machine DAI Configuration
-------------------------
The machine DAI configuration glues all the codec and CPU DAIs together. It can
@@ -61,8 +59,10 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
- .cpu_dai = &pxa_i2s_dai,
- .codec_dai = &wm8731_dai,
+ .cpu_dai_name = "pxa-is2-dai",
+ .codec_dai_name = "wm8731-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8713-codec.0-001a",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
@@ -77,26 +77,6 @@ static struct snd_soc_card snd_soc_corgi = {
};
-Machine Audio Subsystem
------------------------
-
-The machine soc device glues the platform, machine and codec driver together.
-Private data can also be set here. e.g.
-
-/* corgi audio private data */
-static struct wm8731_setup_data corgi_wm8731_setup = {
- .i2c_address = 0x1b,
-};
-
-/* corgi audio subsystem */
-static struct snd_soc_device corgi_snd_devdata = {
- .machine = &snd_soc_corgi,
- .platform = &pxa2xx_soc_platform,
- .codec_dev = &soc_codec_dev_wm8731,
- .codec_data = &corgi_wm8731_setup,
-};
-
-
Machine Power Map
-----------------
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index 06d8359..d57efad 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -20,9 +20,10 @@ struct snd_soc_ops {
int (*trigger)(struct snd_pcm_substream *, int);
};
-The platform driver exports its DMA functionality via struct snd_soc_platform:-
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
-struct snd_soc_platform {
+struct snd_soc_platform_driver {
char *name;
int (*probe)(struct platform_device *pdev);
@@ -34,6 +35,13 @@ struct snd_soc_platform {
int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
void (*pcm_free)(struct snd_pcm *);
+ /*
+ * For platform caused delay reporting.
+ * Optional.
+ */
+ snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+
/* platform stream ops */
struct snd_pcm_ops *pcm_ops;
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 269dbff..be4df4c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1721,7 +1721,9 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
{
struct alc_spec *spec = codec->spec;
int id = spec->fixup_id;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
const char *modelname = spec->fixup_name;
+#endif
int depth = 0;
if (!spec->fixup_list)
@@ -10930,9 +10932,6 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
return 0;
}
-static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg);
-
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
@@ -10950,10 +10949,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
- if (codec->vendor_id == 0x10ec0887)
- err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
- else
- err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
@@ -12635,6 +12631,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
ALC262_HP_BPC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
+ ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -14957,6 +14955,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -17134,7 +17133,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
#define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c)
/* add playback controls from the parsed DAC table */
-/* Based on ALC880 version. But ALC861VD and ALC887 have separate,
+/* Based on ALC880 version. But ALC861VD has separate,
* different NIDs for mute/unmute switch and volume control */
static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
@@ -19461,6 +19460,7 @@ enum {
ALC662_FIXUP_ASPIRE,
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
+ ALC662_FIXUP_CZC_P10T,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -19481,7 +19481,14 @@ static const struct alc_fixup alc662_fixups[] = {
[ALC272_FIXUP_MARIO] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc272_fixup_mario,
- }
+ },
+ [ALC662_FIXUP_CZC_P10T] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0},
+ {}
+ }
+ },
};
static struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -19489,6 +19496,7 @@ static struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
+ SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
{}
};
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index 7b62de0..20c6b07 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -580,6 +580,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
{
int err;
struct snd_akm4xxx *ak;
+ unsigned char tmp;
if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA1010 &&
ice->eeprom.gpiodir == 0x7b)
@@ -622,6 +623,12 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
break;
}
+ /* initialize the SPI clock to high */
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+ tmp |= ICE1712_DELTA_AP_CCLK;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+
/* initialize spdif */
switch (ice->eeprom.subvendor) {
case ICE1712_SUBDEVICE_AUDIOPHILE:
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 3abeedd..ae40359 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,6 +1,7 @@
config SND_BF5XX_I2S
tristate "SoC I2S Audio for the ADI BF5xx chip"
depends on BLACKFIN
+ select SND_BF5XX_SOC_SPORT
help
Say Y or M if you want to add support for codecs attached to
the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -35,6 +36,7 @@ config SND_BFIN_AD73311_SE
config SND_BF5XX_TDM
tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
depends on (BLACKFIN && SND_SOC)
+ select SND_BF5XX_SOC_SPORT
help
Say Y or M if you want to add support for codecs attached to
the Blackfin SPORT (synchronous serial ports) interface in TDM
@@ -61,6 +63,10 @@ config SND_BF5XX_SOC_AD193X
config SND_BF5XX_AC97
tristate "SoC AC97 Audio for the ADI BF5xx chip"
depends on BLACKFIN
+ select AC97_BUS
+ select SND_SOC_AC97_BUS
+ select SND_BF5XX_SOC_SPORT
+ select SND_BF5XX_SOC_AC97
help
Say Y or M if you want to add support for codecs attached to
the Blackfin SPORT (synchronous serial ports) interface in slot 16
@@ -122,17 +128,12 @@ config SND_BF5XX_SOC_SPORT
config SND_BF5XX_SOC_I2S
tristate
- select SND_BF5XX_SOC_SPORT
config SND_BF5XX_SOC_TDM
tristate
- select SND_BF5XX_SOC_SPORT
config SND_BF5XX_SOC_AC97
tristate
- select AC97_BUS
- select SND_SOC_AC97_BUS
- select SND_BF5XX_SOC_SPORT
config SND_BF5XX_SPORT_NUM
int "Set a SPORT for Sound chip"
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index c5f856e..ffbac26 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -260,9 +260,9 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
pr_debug("%s : sport %d\n", __func__, dai->id);
if (!dai->active)
return 0;
- if (dai->capture.active)
+ if (dai->capture_active)
sport_rx_stop(sport);
- if (dai->playback.active)
+ if (dai->playback_active)
sport_tx_stop(sport);
return 0;
}
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 1251239..5515ac9 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -210,7 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai,
#ifdef CONFIG_PM
static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
{
- struct sport_device *sport = dai->private_data;
+ struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
if (!dai->active)
return 0;
@@ -235,13 +235,13 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai)
ret = -EBUSY;
}
- ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0);
+ ret = sport_config_rx(sport, 0, 0x1F, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
ret = -EBUSY;
}
- ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0);
+ ret = sport_config_tx(sport, 0, 0x1F, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
ret = -EBUSY;
@@ -303,14 +303,14 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev)
goto sport_config_err;
}
- ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0);
+ ret = sport_config_rx(sport_handle, 0, 0x1F, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
ret = -EBUSY;
goto sport_config_err;
}
- ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0);
+ ret = sport_config_tx(sport_handle, 0, 0x1F, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
ret = -EBUSY;
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 2d4f896..3ceaef6 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -104,6 +104,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = {
.name = "hsdet-gpio",
.report = SND_JACK_HEADSET,
.debounce_time = 200,
+ .invert = 1,
},
};
@@ -192,7 +193,7 @@ static struct snd_soc_dai_link z2_dai = {
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8750-codec.0-001a",
+ .codec_name = "wm8750-codec.0-001b",
.init = z2_wm8750_init,
.ops = &z2_ops,
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-10-11 16:21 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-10-11 16:21 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.36 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing only safe changes: a trivial fix for the old OSS error
path, a quirk entry for HD-audio, and changes of MAINTAINERS file.
Thanks!
Takashi
===
Dan Carpenter (1):
OSS: soundcard: locking bug in sound_ioctl()
Luke Yelavich (1):
ALSA: hda - Add another HP DV6 quirk
Mark Brown (3):
ASoC: Add Jassi Brar as Samsung maintainer
ASoC: Add Dimitris Papastamos to Wolfson maintainers
ASoC: Update links for Wolfson MAINTAINERS entry
---
MAINTAINERS | 10 +++++++++-
sound/oss/soundcard.c | 4 ++--
sound/pci/hda/patch_sigmatel.c | 2 ++
3 files changed, 13 insertions(+), 3 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index d458281..7679bf3 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5030,6 +5030,12 @@ F: drivers/media/common/saa7146*
F: drivers/media/video/*7146*
F: include/media/*7146*
+SAMSUNG AUDIO (ASoC) DRIVERS
+M: Jassi Brar <jassi.brar@samsung.com>
+L: alsa-devel@alsa-project.org (moderated for non-subscribers)
+S: Supported
+F: sound/soc/s3c24xx
+
TLG2300 VIDEO4LINUX-2 DRIVER
M: Huang Shijie <shijie8@gmail.com>
M: Kang Yong <kangyong@telegent.com>
@@ -6472,8 +6478,10 @@ F: include/linux/wm97xx.h
WOLFSON MICROELECTRONICS DRIVERS
M: Mark Brown <broonie@opensource.wolfsonmicro.com>
M: Ian Lartey <ian@opensource.wolfsonmicro.com>
+M: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
-W: http://opensource.wolfsonmicro.com/node/8
+W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
S: Supported
F: Documentation/hwmon/wm83??
F: drivers/leds/leds-wm83*.c
diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c
index 92aa762..07f803e 100644
--- a/sound/oss/soundcard.c
+++ b/sound/oss/soundcard.c
@@ -391,11 +391,11 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
case SND_DEV_DSP:
case SND_DEV_DSP16:
case SND_DEV_AUDIO:
- return audio_ioctl(dev, file, cmd, p);
+ ret = audio_ioctl(dev, file, cmd, p);
break;
case SND_DEV_MIDIN:
- return MIDIbuf_ioctl(dev, file, cmd, p);
+ ret = MIDIbuf_ioctl(dev, file, cmd, p);
break;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 95148e5..c16c5ba 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1747,6 +1747,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"HP dv6", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061,
"HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x363e,
+ "HP DV6", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010,
"HP", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-10-04 18:09 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-10-04 18:09 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.36-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/misc
Just two small obvious fixes below.
Thanks!
Takashi
===
Dan Rosenberg (1):
ALSA: prevent heap corruption in snd_ctl_new()
Takashi Iwai (1):
ALSA: i2c/other/ak4xx-adda: Fix a compile warning with CONFIG_PROCFS=n
---
sound/core/control.c | 5 +++++
sound/i2c/other/ak4xxx-adda.c | 2 +-
2 files changed, 6 insertions(+), 1 deletions(-)
diff --git a/sound/core/control.c b/sound/core/control.c
index 070aab4..45a8180 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -31,6 +31,7 @@
/* max number of user-defined controls */
#define MAX_USER_CONTROLS 32
+#define MAX_CONTROL_COUNT 1028
struct snd_kctl_ioctl {
struct list_head list; /* list of all ioctls */
@@ -195,6 +196,10 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control,
if (snd_BUG_ON(!control || !control->count))
return NULL;
+
+ if (control->count > MAX_CONTROL_COUNT)
+ return NULL;
+
kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL);
if (kctl == NULL) {
snd_printk(KERN_ERR "Cannot allocate control instance\n");
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index 1adb8a3..42d7844 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -900,7 +900,7 @@ static int proc_init(struct snd_akm4xxx *ak)
return 0;
}
#else /* !CONFIG_PROC_FS */
-static int proc_init(struct snd_akm4xxx *ak) {}
+static int proc_init(struct snd_akm4xxx *ak) { return 0; }
#endif
int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-09-25 16:11 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-09-25 16:11 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.36-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All small and local fixes as below.
Thanks!
Takashi
===
Dan Rosenberg (1):
ALSA: sound/pci/rme9652: prevent reading uninitialized stack memory
Dimitris Papastamos (1):
ASoC: Fix soc-cache buffer overflow bug
Erik J. Staab (1):
ALSA: oxygen: fix analog capture on Claro halo cards
Guennadi Liakhovetski (1):
ASoC: fix clkdev API usage in sh/migor.c
Luke Yelavich (1):
ALSA: hda - Add Dell Latitude E6400 model quirk
Mark Brown (1):
ASoC: Fix multi-componentism
Takashi Iwai (1):
ALSA: hda - Fix auto-parse of SPDIF input of Realtek codecs
---
sound/pci/hda/patch_analog.c | 1 +
sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++------
sound/pci/oxygen/oxygen.c | 4 ++++
sound/pci/rme9652/hdsp.c | 1 +
sound/pci/rme9652/hdspm.c | 1 +
sound/soc/sh/migor.c | 15 +++++++++++++--
sound/soc/soc-cache.c | 5 +++--
7 files changed, 39 insertions(+), 10 deletions(-)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index b697fd2..10bbbaf 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3641,6 +3641,7 @@ static struct snd_pci_quirk ad1984_cfg_tbl[] = {
/* Lenovo Thinkpad T61/X61 */
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
+ SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a1312a6..a432e6e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1594,12 +1594,22 @@ static void alc_auto_parse_digital(struct hda_codec *codec)
}
if (spec->autocfg.dig_in_pin) {
- hda_nid_t dig_nid;
- err = snd_hda_get_connections(codec,
- spec->autocfg.dig_in_pin,
- &dig_nid, 1);
- if (err > 0)
- spec->dig_in_nid = dig_nid;
+ dig_nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, dig_nid++) {
+ unsigned int wcaps = get_wcaps(codec, dig_nid);
+ if (get_wcaps_type(wcaps) != AC_WID_AUD_IN)
+ continue;
+ if (!(wcaps & AC_WCAP_DIGITAL))
+ continue;
+ if (!(wcaps & AC_WCAP_CONN_LIST))
+ continue;
+ err = get_connection_index(codec, dig_nid,
+ spec->autocfg.dig_in_pin);
+ if (err >= 0) {
+ spec->dig_in_nid = dig_nid;
+ break;
+ }
+ }
}
}
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 289cb4d..6c0a11a 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -543,6 +543,10 @@ static int __devinit get_oxygen_model(struct oxygen *chip,
chip->model.suspend = claro_suspend;
chip->model.resume = claro_resume;
chip->model.set_adc_params = set_ak5385_params;
+ chip->model.device_config = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_1_FROM_SPDIF;
break;
}
if (id->driver_data == MODEL_MERIDIAN ||
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index b92adef..d6fa7bf 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -4609,6 +4609,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
if (err < 0)
return err;
+ memset(&info, 0, sizeof(info));
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);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 547b713..0c98ef9 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -4127,6 +4127,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO:
+ memset(&info, 0, sizeof(info));
spin_lock_irq(&hdspm->lock);
info.pref_sync_ref = hdspm_pref_sync_ref(hdspm);
info.wordclock_sync_check = hdspm_wc_sync_check(hdspm);
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index b823a5c..87e2b7f 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -12,6 +12,7 @@
#include <linux/firmware.h>
#include <linux/module.h>
+#include <asm/clkdev.h>
#include <asm/clock.h>
#include <cpu/sh7722.h>
@@ -40,12 +41,12 @@ static struct clk_ops siumckb_clk_ops = {
};
static struct clk siumckb_clk = {
- .name = "siumckb_clk",
- .id = -1,
.ops = &siumckb_clk_ops,
.rate = 0, /* initialised at run-time */
};
+static struct clk_lookup *siumckb_lookup;
+
static int migor_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -180,6 +181,13 @@ static int __init migor_init(void)
if (ret < 0)
return ret;
+ siumckb_lookup = clkdev_alloc(&siumckb_clk, "siumckb_clk", NULL);
+ if (!siumckb_lookup) {
+ ret = -ENOMEM;
+ goto eclkdevalloc;
+ }
+ clkdev_add(siumckb_lookup);
+
/* Port number used on this machine: port B */
migor_snd_device = platform_device_alloc("soc-audio", 1);
if (!migor_snd_device) {
@@ -200,12 +208,15 @@ static int __init migor_init(void)
epdevadd:
platform_device_put(migor_snd_device);
epdevalloc:
+ clkdev_drop(siumckb_lookup);
+eclkdevalloc:
clk_unregister(&siumckb_clk);
return ret;
}
static void __exit migor_exit(void)
{
+ clkdev_drop(siumckb_lookup);
clk_unregister(&siumckb_clk);
platform_device_unregister(migor_snd_device);
}
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index adbc68c..f6b0d28 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -203,8 +203,9 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
data[1] = (value >> 8) & 0xff;
data[2] = value & 0xff;
- if (!snd_soc_codec_volatile_register(codec, reg))
- reg_cache[reg] = value;
+ if (!snd_soc_codec_volatile_register(codec, reg)
+ && reg < codec->reg_cache_size)
+ reg_cache[reg] = value;
if (codec->cache_only) {
codec->cache_sync = 1;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-09-17 17:39 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-09-17 17:39 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.36-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All commits below are simple fixes.
Thanks!
Takashi
===
Anisse Astier (1):
ALSA: hda - Add quirk for Toshiba C650D using a Conexant CX20585
David Henningsson (1):
ALSA: HDA: Enable internal speaker on Dell M101z
Joe Perches (1):
sound: Remove pr_<level> uses of KERN_<level>
Seth Heasley (1):
ALSA: hda_intel: ALSA HD Audio patch for Intel Patsburg DeviceIDs
Stephen Warren (1):
ALSA: patch_nvhdmi.c: Fix supported sample rate list.
Takashi Iwai (2):
ALSA: pcm - Fix unbalanced pm_qos_request
ALSA: pcm - Fix race with proc files
---
sound/core/pcm.c | 33 ++++++++++++++++++++++++---------
sound/core/pcm_native.c | 2 ++
sound/pci/hda/hda_intel.c | 3 +++
sound/pci/hda/patch_conexant.c | 1 +
sound/pci/hda/patch_nvhdmi.c | 2 +-
sound/pci/hda/patch_realtek.c | 10 ++++++++++
sound/ppc/snd_ps3.c | 2 +-
sound/soc/s3c24xx/s3c-dma.c | 3 +--
8 files changed, 43 insertions(+), 13 deletions(-)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 204af48..ac242a3 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -372,14 +372,17 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_pcm_substream *substream = entry->private_data;
- struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_runtime *runtime;
+
+ mutex_lock(&substream->pcm->open_mutex);
+ runtime = substream->runtime;
if (!runtime) {
snd_iprintf(buffer, "closed\n");
- return;
+ goto unlock;
}
if (runtime->status->state == SNDRV_PCM_STATE_OPEN) {
snd_iprintf(buffer, "no setup\n");
- return;
+ goto unlock;
}
snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access));
snd_iprintf(buffer, "format: %s\n", snd_pcm_format_name(runtime->format));
@@ -398,20 +401,25 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry,
snd_iprintf(buffer, "OSS period frames: %lu\n", (unsigned long)runtime->oss.period_frames);
}
#endif
+ unlock:
+ mutex_unlock(&substream->pcm->open_mutex);
}
static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_pcm_substream *substream = entry->private_data;
- struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_runtime *runtime;
+
+ mutex_lock(&substream->pcm->open_mutex);
+ runtime = substream->runtime;
if (!runtime) {
snd_iprintf(buffer, "closed\n");
- return;
+ goto unlock;
}
if (runtime->status->state == SNDRV_PCM_STATE_OPEN) {
snd_iprintf(buffer, "no setup\n");
- return;
+ goto unlock;
}
snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode));
snd_iprintf(buffer, "period_step: %u\n", runtime->period_step);
@@ -421,24 +429,29 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry,
snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold);
snd_iprintf(buffer, "silence_size: %lu\n", runtime->silence_size);
snd_iprintf(buffer, "boundary: %lu\n", runtime->boundary);
+ unlock:
+ mutex_unlock(&substream->pcm->open_mutex);
}
static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_pcm_substream *substream = entry->private_data;
- struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_runtime *runtime;
struct snd_pcm_status status;
int err;
+
+ mutex_lock(&substream->pcm->open_mutex);
+ runtime = substream->runtime;
if (!runtime) {
snd_iprintf(buffer, "closed\n");
- return;
+ goto unlock;
}
memset(&status, 0, sizeof(status));
err = snd_pcm_status(substream, &status);
if (err < 0) {
snd_iprintf(buffer, "error %d\n", err);
- return;
+ goto unlock;
}
snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state));
snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid));
@@ -452,6 +465,8 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry,
snd_iprintf(buffer, "-----\n");
snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr);
snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr);
+ unlock:
+ mutex_unlock(&substream->pcm->open_mutex);
}
#ifdef CONFIG_SND_PCM_XRUN_DEBUG
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 134fc6c..d4eb2ef 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1992,6 +1992,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream)
substream->ops->close(substream);
substream->hw_opened = 0;
}
+ if (pm_qos_request_active(&substream->latency_pm_qos_req))
+ pm_qos_remove_request(&substream->latency_pm_qos_req);
if (substream->pcm_release) {
substream->pcm_release(substream);
substream->pcm_release = NULL;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1053fff..34940a0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -126,6 +126,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, ICH10},"
"{Intel, PCH},"
"{Intel, CPT},"
+ "{Intel, PBG},"
"{Intel, SCH},"
"{ATI, SB450},"
"{ATI, SB600},"
@@ -2749,6 +2750,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ PCI_DEVICE(0x8086, 0x3b57), .driver_data = AZX_DRIVER_ICH },
/* CPT */
{ PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH },
+ /* PBG */
+ { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH },
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
/* ATI SB 450/600 */
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 71f9d64..972e7c4 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3092,6 +3092,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
+ SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 69b950d..baa108b 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -84,7 +84,7 @@ static struct hda_verb nvhdmi_basic_init_7x[] = {
#else
/* support all rates and formats */
#define SUPPORTED_RATES \
- (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
SNDRV_PCM_RATE_192000)
#define SUPPORTED_MAXBPS 24
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bcbf916..a1312a6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14453,6 +14453,7 @@ static void alc269_auto_init(struct hda_codec *codec)
enum {
ALC269_FIXUP_SONY_VAIO,
+ ALC269_FIXUP_DELL_M101Z,
};
static const struct hda_verb alc269_sony_vaio_fixup_verbs[] = {
@@ -14464,11 +14465,20 @@ static const struct alc_fixup alc269_fixups[] = {
[ALC269_FIXUP_SONY_VAIO] = {
.verbs = alc269_sony_vaio_fixup_verbs
},
+ [ALC269_FIXUP_DELL_M101Z] = {
+ .verbs = (const struct hda_verb[]) {
+ /* Enables internal speaker */
+ {0x20, AC_VERB_SET_COEF_INDEX, 13},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x4040},
+ {}
+ }
+ },
};
static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+ SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
{}
};
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 2f12da4..581a670 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -579,7 +579,7 @@ static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream,
rate * delay_ms / 1000)
* substream->runtime->channels;
- pr_debug(KERN_ERR "%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n",
+ pr_debug("%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n",
__func__,
delay_ms,
rate,
diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c
index 1b61c23..f1b1bc4 100644
--- a/sound/soc/s3c24xx/s3c-dma.c
+++ b/sound/soc/s3c24xx/s3c-dma.c
@@ -94,8 +94,7 @@ static void s3c_dma_enqueue(struct snd_pcm_substream *substream)
if ((pos + len) > prtd->dma_end) {
len = prtd->dma_end - pos;
- pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n",
- __func__, len);
+ pr_debug("%s: corrected dma len %ld\n", __func__, len);
}
ret = s3c2410_dma_enqueue(prtd->params->channel,
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-09-10 13:27 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-09-10 13:27 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.36-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
In addition to previous pull request, a few more fixes came in.
Slightly large chunks are found in HD-audio codec drivers, but they
are either the addition of errata verbs or the quirk for one laptop
model, so pretty safe. Others are mostly small fixes.
Thanks!
Takashi
===
Brian Austin (1):
ALSA: hda - Add errata initverb sequence for CS42xx codecs
Clemens Ladisch (3):
ALSA: usb-audio: fix detection of vendor-specific device protocol settings
ALSA: virtuoso: work around missing reset in the Xonar DS Windows driver
ALSA: virtuoso: fix setting of Xonar DS line-in/mic-in controls
Dan Carpenter (1):
ALSA: rawmidi: fix the get next midi device ioctl
Daniel Mack (1):
ALSA: usb-audio: Assume first control interface is for audio
David Henningsson (1):
ALSA: hda - Add a new hp-laptop model for Conexant 5066, tested on HP G60
Takashi Iwai (6):
ALSA: hda - Add quirk for Lenovo T400s
ALSA: usb - Release capture substream URBs properly
ALSA: hda - Enable PC-beep for EeePC with ALC269 codec
ALSA: msnd-classic: Fix invalid cfg parameter
ALSA: seq/oss - Fix double-free at error path of snd_seq_oss_open()
ALSA: hda - Fix wrong HP pin detection in snd_hda_parse_pin_def_config()
---
Documentation/sound/alsa/HD-Audio-Models.txt | 1 +
sound/core/rawmidi.c | 2 +
sound/core/seq/oss/seq_oss_init.c | 9 ++--
sound/isa/msnd/msnd_pinnacle.c | 8 +++-
sound/pci/hda/hda_codec.c | 2 +-
sound/pci/hda/patch_cirrus.c | 50 ++++++++++++++++++++++
sound/pci/hda/patch_conexant.c | 58 ++++++++++++++++++++++++++
sound/pci/hda/patch_realtek.c | 1 +
sound/pci/oxygen/oxygen.h | 1 +
sound/pci/oxygen/oxygen_lib.c | 21 ++++++++-
sound/pci/oxygen/virtuoso.c | 1 +
sound/pci/oxygen/xonar_wm87x6.c | 22 +++++++++-
sound/usb/card.c | 19 ++++++---
sound/usb/clock.c | 3 +-
sound/usb/endpoint.c | 11 +++--
sound/usb/format.c | 22 ++++-----
sound/usb/mixer.c | 10 ++++-
sound/usb/pcm.c | 3 +-
18 files changed, 204 insertions(+), 40 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index ce46fa1..37c6aad 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -296,6 +296,7 @@ Conexant 5051
Conexant 5066
=============
laptop Basic Laptop config (default)
+ hp-laptop HP laptops, e g G60
dell-laptop Dell laptops
dell-vostro Dell Vostro
olpc-xo-1_5 OLPC XO 1.5
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index eb68326..a7868ad 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -829,6 +829,8 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card,
if (get_user(device, (int __user *)argp))
return -EFAULT;
+ if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */
+ device = SNDRV_RAWMIDI_DEVICES - 1;
mutex_lock(®ister_mutex);
device = device < 0 ? 0 : device + 1;
while (device < SNDRV_RAWMIDI_DEVICES) {
diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c
index 6857122..69cd7b3 100644
--- a/sound/core/seq/oss/seq_oss_init.c
+++ b/sound/core/seq/oss/seq_oss_init.c
@@ -281,13 +281,10 @@ snd_seq_oss_open(struct file *file, int level)
return 0;
_error:
- snd_seq_oss_writeq_delete(dp->writeq);
- snd_seq_oss_readq_delete(dp->readq);
snd_seq_oss_synth_cleanup(dp);
snd_seq_oss_midi_cleanup(dp);
- delete_port(dp);
delete_seq_queue(dp->queue);
- kfree(dp);
+ delete_port(dp);
return rc;
}
@@ -350,8 +347,10 @@ create_port(struct seq_oss_devinfo *dp)
static int
delete_port(struct seq_oss_devinfo *dp)
{
- if (dp->port < 0)
+ if (dp->port < 0) {
+ kfree(dp);
return 0;
+ }
debug_printk(("delete_port %i\n", dp->port));
return snd_seq_event_port_detach(dp->cseq, dp->port);
diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c
index 5f3e684..91d6023 100644
--- a/sound/isa/msnd/msnd_pinnacle.c
+++ b/sound/isa/msnd/msnd_pinnacle.c
@@ -764,9 +764,9 @@ static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+#ifndef MSND_CLASSIC
static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
-#ifndef MSND_CLASSIC
/* Extra Peripheral Configuration (Default: Disable) */
static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
@@ -894,7 +894,11 @@ static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx)
struct snd_card *card;
struct snd_msnd *chip;
- if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) {
+ if (has_isapnp(idx)
+#ifndef MSND_CLASSIC
+ || cfg[idx] == SNDRV_AUTO_PORT
+#endif
+ ) {
printk(KERN_INFO LOGNAME ": Assuming PnP mode\n");
return -ENODEV;
}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3827092..1482921 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -4536,7 +4536,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
cfg->hp_outs--;
memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
- memmove(sequences_hp + i - 1, sequences_hp + i,
+ memmove(sequences_hp + i, sequences_hp + i + 1,
sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
}
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 4ef5efa..488fd9a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -972,6 +972,53 @@ static struct hda_verb cs_coef_init_verbs[] = {
{} /* terminator */
};
+/* Errata: CS4207 rev C0/C1/C2 Silicon
+ *
+ * http://www.cirrus.com/en/pubs/errata/ER880C3.pdf
+ *
+ * 6. At high temperature (TA > +85°C), the digital supply current (IVD)
+ * may be excessive (up to an additional 200 μA), which is most easily
+ * observed while the part is being held in reset (RESET# active low).
+ *
+ * Root Cause: At initial powerup of the device, the logic that drives
+ * the clock and write enable to the S/PDIF SRC RAMs is not properly
+ * initialized.
+ * Certain random patterns will cause a steady leakage current in those
+ * RAM cells. The issue will resolve once the SRCs are used (turned on).
+ *
+ * Workaround: The following verb sequence briefly turns on the S/PDIF SRC
+ * blocks, which will alleviate the issue.
+ */
+
+static struct hda_verb cs_errata_init_verbs[] = {
+ {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */
+ {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */
+
+ {0x11, AC_VERB_SET_COEF_INDEX, 0x0008},
+ {0x11, AC_VERB_SET_PROC_COEF, 0x9999},
+ {0x11, AC_VERB_SET_COEF_INDEX, 0x0017},
+ {0x11, AC_VERB_SET_PROC_COEF, 0xa412},
+ {0x11, AC_VERB_SET_COEF_INDEX, 0x0001},
+ {0x11, AC_VERB_SET_PROC_COEF, 0x0009},
+
+ {0x07, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Rx: D0 */
+ {0x08, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Tx: D0 */
+
+ {0x11, AC_VERB_SET_COEF_INDEX, 0x0017},
+ {0x11, AC_VERB_SET_PROC_COEF, 0x2412},
+ {0x11, AC_VERB_SET_COEF_INDEX, 0x0008},
+ {0x11, AC_VERB_SET_PROC_COEF, 0x0000},
+ {0x11, AC_VERB_SET_COEF_INDEX, 0x0001},
+ {0x11, AC_VERB_SET_PROC_COEF, 0x0008},
+ {0x11, AC_VERB_SET_PROC_STATE, 0x00},
+
+ {0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */
+ {0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */
+ /*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */
+
+ {} /* terminator */
+};
+
/* SPDIF setup */
static void init_digital(struct hda_codec *codec)
{
@@ -991,6 +1038,9 @@ static int cs_init(struct hda_codec *codec)
{
struct cs_spec *spec = codec->spec;
+ /* init_verb sequence for C0/C1/C2 errata*/
+ snd_hda_sequence_write(codec, cs_errata_init_verbs);
+
snd_hda_sequence_write(codec, cs_coef_init_verbs);
if (spec->gpio_mask) {
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 5cdb80e..71f9d64 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -116,6 +116,7 @@ struct conexant_spec {
unsigned int dell_vostro:1;
unsigned int ideapad:1;
unsigned int thinkpad:1;
+ unsigned int hp_laptop:1;
unsigned int ext_mic_present;
unsigned int recording;
@@ -2299,6 +2300,18 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
}
}
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_jack_detect(codec, 0x1b);
+ snd_printdd("CXT5066: external microphone present=%d\n", present);
+ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 1 : 3);
+}
+
+
/* toggle input of built-in digital mic and mic jack appropriately
order is: external mic -> dock mic -> interal mic */
static void cxt5066_thinkpad_automic(struct hda_codec *codec)
@@ -2408,6 +2421,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
}
/* unsolicited event for jack sensing */
+static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+{
+ snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+ switch (res >> 26) {
+ case CONEXANT_HP_EVENT:
+ cxt5066_hp_automute(codec);
+ break;
+ case CONEXANT_MIC_EVENT:
+ cxt5066_hp_laptop_automic(codec);
+ break;
+ }
+}
+
+/* unsolicited event for jack sensing */
static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
{
snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
@@ -2989,6 +3016,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
{ } /* end */
};
+
+static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+ { } /* end */
+};
+
/* initialize jack-sensing, too */
static int cxt5066_init(struct hda_codec *codec)
{
@@ -3004,6 +3039,8 @@ static int cxt5066_init(struct hda_codec *codec)
cxt5066_ideapad_automic(codec);
else if (spec->thinkpad)
cxt5066_thinkpad_automic(codec);
+ else if (spec->hp_laptop)
+ cxt5066_hp_laptop_automic(codec);
}
cxt5066_set_mic_boost(codec);
return 0;
@@ -3031,6 +3068,7 @@ enum {
CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */
CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */
CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */
+ CXT5066_HP_LAPTOP, /* HP Laptop */
CXT5066_MODELS
};
@@ -3041,6 +3079,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = {
[CXT5066_DELL_VOSTO] = "dell-vostro",
[CXT5066_IDEAPAD] = "ideapad",
[CXT5066_THINKPAD] = "thinkpad",
+ [CXT5066_HP_LAPTOP] = "hp-laptop",
};
static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
@@ -3052,8 +3091,10 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
+ SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
@@ -3116,6 +3157,23 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->num_init_verbs++;
spec->dell_automute = 1;
break;
+ case CXT5066_HP_LAPTOP:
+ codec->patch_ops.init = cxt5066_init;
+ codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+ spec->init_verbs[spec->num_init_verbs] =
+ cxt5066_init_verbs_hp_laptop;
+ spec->num_init_verbs++;
+ spec->hp_laptop = 1;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+ /* no S/PDIF out */
+ spec->multiout.dig_out_nid = 0;
+ /* input source automatically selected */
+ spec->input_mux = NULL;
+ spec->port_d_mode = 0;
+ spec->mic_boost = 3; /* default 30dB gain */
+ break;
+
case CXT5066_OLPC_XO_1_5:
codec->patch_ops.init = cxt5066_olpc_init;
codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 627bf99..bcbf916 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5334,6 +5334,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
static struct snd_pci_quirk beep_white_list[] = {
SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
{}
};
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index 6147216..a3409ed 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -155,6 +155,7 @@ void oxygen_pci_remove(struct pci_dev *pci);
int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
int oxygen_pci_resume(struct pci_dev *pci);
#endif
+void oxygen_pci_shutdown(struct pci_dev *pci);
/* oxygen_mixer.c */
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index fad03d6..7e93cf8 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -519,16 +519,21 @@ static void oxygen_init(struct oxygen *chip)
}
}
-static void oxygen_card_free(struct snd_card *card)
+static void oxygen_shutdown(struct oxygen *chip)
{
- struct oxygen *chip = card->private_data;
-
spin_lock_irq(&chip->reg_lock);
chip->interrupt_mask = 0;
chip->pcm_running = 0;
oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
spin_unlock_irq(&chip->reg_lock);
+}
+
+static void oxygen_card_free(struct snd_card *card)
+{
+ struct oxygen *chip = card->private_data;
+
+ oxygen_shutdown(chip);
if (chip->irq >= 0)
free_irq(chip->irq, chip);
flush_scheduled_work();
@@ -778,3 +783,13 @@ int oxygen_pci_resume(struct pci_dev *pci)
}
EXPORT_SYMBOL(oxygen_pci_resume);
#endif /* CONFIG_PM */
+
+void oxygen_pci_shutdown(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct oxygen *chip = card->private_data;
+
+ oxygen_shutdown(chip);
+ chip->model.cleanup(chip);
+}
+EXPORT_SYMBOL(oxygen_pci_shutdown);
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index f03a2f2..06c863e 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -95,6 +95,7 @@ static struct pci_driver xonar_driver = {
.suspend = oxygen_pci_suspend,
.resume = oxygen_pci_resume,
#endif
+ .shutdown = oxygen_pci_shutdown,
};
static int __init alsa_card_xonar_init(void)
diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c
index dbc4b89..b82c1cf 100644
--- a/sound/pci/oxygen/xonar_wm87x6.c
+++ b/sound/pci/oxygen/xonar_wm87x6.c
@@ -53,6 +53,8 @@ struct xonar_wm87x6 {
struct xonar_generic generic;
u16 wm8776_regs[0x17];
u16 wm8766_regs[0x10];
+ struct snd_kcontrol *line_adcmux_control;
+ struct snd_kcontrol *mic_adcmux_control;
struct snd_kcontrol *lc_controls[13];
};
@@ -193,6 +195,7 @@ static void xonar_ds_init(struct oxygen *chip)
static void xonar_ds_cleanup(struct oxygen *chip)
{
xonar_disable_output(chip);
+ wm8776_write(chip, WM8776_RESET, 0);
}
static void xonar_ds_suspend(struct oxygen *chip)
@@ -603,6 +606,7 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl,
{
struct oxygen *chip = ctl->private_data;
struct xonar_wm87x6 *data = chip->model_data;
+ struct snd_kcontrol *other_ctl;
unsigned int mux_bit = ctl->private_value;
u16 reg;
int changed;
@@ -610,8 +614,18 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl,
mutex_lock(&chip->mutex);
reg = data->wm8776_regs[WM8776_ADCMUX];
if (value->value.integer.value[0]) {
- reg &= ~0x003;
reg |= mux_bit;
+ /* line-in and mic-in are exclusive */
+ mux_bit ^= 3;
+ if (reg & mux_bit) {
+ reg &= ~mux_bit;
+ if (mux_bit == 1)
+ other_ctl = data->line_adcmux_control;
+ else
+ other_ctl = data->mic_adcmux_control;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &other_ctl->id);
+ }
} else
reg &= ~mux_bit;
changed = reg != data->wm8776_regs[WM8776_ADCMUX];
@@ -963,7 +977,13 @@ static int xonar_ds_mixer_init(struct oxygen *chip)
err = snd_ctl_add(chip->card, ctl);
if (err < 0)
return err;
+ if (!strcmp(ctl->id.name, "Line Capture Switch"))
+ data->line_adcmux_control = ctl;
+ else if (!strcmp(ctl->id.name, "Mic Capture Switch"))
+ data->mic_adcmux_control = ctl;
}
+ if (!data->line_adcmux_control || !data->mic_adcmux_control)
+ return -ENXIO;
BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls));
for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) {
ctl = snd_ctl_new1(&lc_controls[i], chip);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 9feb00c..4eabafa 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -126,7 +126,7 @@ static void snd_usb_stream_disconnect(struct list_head *head)
for (idx = 0; idx < 2; idx++) {
subs = &as->substream[idx];
if (!subs->num_formats)
- return;
+ continue;
snd_usb_release_substream_urbs(subs, 1);
subs->interface = -1;
}
@@ -216,6 +216,11 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
}
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n",
+ protocol);
+ /* fall through */
+
case UAC_VERSION_1: {
struct uac1_ac_header_descriptor *h1 = control_header;
@@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
break;
}
-
- default:
- snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol);
- return -EINVAL;
}
return 0;
@@ -465,7 +466,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
goto __error;
}
- chip->ctrl_intf = alts;
+ /*
+ * For devices with more than one control interface, we assume the
+ * first contains the audio controls. We might need a more specific
+ * check here in the future.
+ */
+ if (!chip->ctrl_intf)
+ chip->ctrl_intf = alts;
if (err > 0) {
/* create normal USB audio interfaces */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index b853f8d..7754a10 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -295,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
switch (altsd->bInterfaceProtocol) {
case UAC_VERSION_1:
+ default:
return set_sample_rate_v1(chip, iface, alts, fmt, rate);
case UAC_VERSION_2:
return set_sample_rate_v2(chip, iface, alts, fmt, rate);
}
-
- return -EINVAL;
}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 1a701f1..ef0a07e 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -275,6 +275,12 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
/* get audio formats */
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n",
+ dev->devnum, iface_no, altno, protocol);
+ protocol = UAC_VERSION_1;
+ /* fall through */
+
case UAC_VERSION_1: {
struct uac1_as_header_descriptor *as =
snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
@@ -336,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
dev->devnum, iface_no, altno, as->bTerminalLink);
continue;
}
-
- default:
- snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n",
- dev->devnum, iface_no, altno, protocol);
- continue;
}
/* get format type */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 3a13754..6914821 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
u64 pcm_formats;
switch (protocol) {
- case UAC_VERSION_1: {
+ case UAC_VERSION_1:
+ default: {
struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
sample_width = fmt->bBitResolution;
sample_bytes = fmt->bSubframeSize;
@@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
format <<= 1;
break;
}
-
- default:
- return -EINVAL;
}
pcm_formats = 0;
@@ -384,6 +382,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
* audio class v2 uses class specific EP0 range requests for that.
*/
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ /* fall through */
case UAC_VERSION_1:
fp->channels = fmt->bNrChannels;
ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7);
@@ -392,10 +394,6 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
/* fp->channels is already set in this case */
ret = parse_audio_format_rates_v2(chip, fp);
break;
- default:
- snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n",
- chip->dev->devnum, fp->iface, fp->altsetting, protocol);
- return -EINVAL;
}
if (fp->channels < 1) {
@@ -438,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip,
fp->channels = 1;
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ /* fall through */
case UAC_VERSION_1: {
struct uac_format_type_ii_discrete_descriptor *fmt = _fmt;
brate = le16_to_cpu(fmt->wMaxBitRate);
@@ -456,10 +458,6 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip,
ret = parse_audio_format_rates_v2(chip, fp);
break;
}
- default:
- snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n",
- chip->dev->devnum, fp->iface, fp->altsetting, protocol);
- return -EINVAL;
}
return ret;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c166db0..3ed3901 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2175,7 +2175,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
}
host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
- mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol;
+ switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+ case UAC_VERSION_1:
+ default:
+ mixer->protocol = UAC_VERSION_1;
+ break;
+ case UAC_VERSION_2:
+ mixer->protocol = UAC_VERSION_2;
+ break;
+ }
if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
(err = snd_usb_mixer_status_create(mixer)) < 0)
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 3634ced..3b5135c 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
switch (altsd->bInterfaceProtocol) {
case UAC_VERSION_1:
+ default:
return init_pitch_v1(chip, iface, alts, fmt);
case UAC_VERSION_2:
return init_pitch_v2(chip, iface, alts, fmt);
}
-
- return -EINVAL;
}
/*
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-09-03 20:54 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-09-03 20:54 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.36-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Two USB-audio fixes and one HD-audio quirk. A bit longer patches
than usual at this time, but they look sane enough.
thanks,
Takashi
===
Clemens Ladisch (1):
ALSA: usb-audio: fix detection of vendor-specific device protocol settings
Daniel Mack (1):
ALSA: usb-audio: Assume first control interface is for audio
David Henningsson (1):
ALSA: hda - Add a new hp-laptop model for Conexant 5066, tested on HP G60
---
Documentation/sound/alsa/HD-Audio-Models.txt | 1 +
sound/pci/hda/patch_conexant.c | 57 ++++++++++++++++++++++++++
sound/usb/card.c | 17 +++++--
sound/usb/clock.c | 3 +-
sound/usb/endpoint.c | 11 +++--
sound/usb/format.c | 22 +++++-----
sound/usb/mixer.c | 10 ++++-
sound/usb/pcm.c | 3 +-
8 files changed, 97 insertions(+), 27 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index ce46fa1..37c6aad 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -296,6 +296,7 @@ Conexant 5051
Conexant 5066
=============
laptop Basic Laptop config (default)
+ hp-laptop HP laptops, e g G60
dell-laptop Dell laptops
dell-vostro Dell Vostro
olpc-xo-1_5 OLPC XO 1.5
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 5cdb80e..4f06199 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -116,6 +116,7 @@ struct conexant_spec {
unsigned int dell_vostro:1;
unsigned int ideapad:1;
unsigned int thinkpad:1;
+ unsigned int hp_laptop:1;
unsigned int ext_mic_present;
unsigned int recording;
@@ -2299,6 +2300,18 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
}
}
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_jack_detect(codec, 0x1b);
+ snd_printdd("CXT5066: external microphone present=%d\n", present);
+ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 1 : 3);
+}
+
+
/* toggle input of built-in digital mic and mic jack appropriately
order is: external mic -> dock mic -> interal mic */
static void cxt5066_thinkpad_automic(struct hda_codec *codec)
@@ -2408,6 +2421,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
}
/* unsolicited event for jack sensing */
+static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+{
+ snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+ switch (res >> 26) {
+ case CONEXANT_HP_EVENT:
+ cxt5066_hp_automute(codec);
+ break;
+ case CONEXANT_MIC_EVENT:
+ cxt5066_hp_laptop_automic(codec);
+ break;
+ }
+}
+
+/* unsolicited event for jack sensing */
static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
{
snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
@@ -2989,6 +3016,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
{ } /* end */
};
+
+static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+ { } /* end */
+};
+
/* initialize jack-sensing, too */
static int cxt5066_init(struct hda_codec *codec)
{
@@ -3004,6 +3039,8 @@ static int cxt5066_init(struct hda_codec *codec)
cxt5066_ideapad_automic(codec);
else if (spec->thinkpad)
cxt5066_thinkpad_automic(codec);
+ else if (spec->hp_laptop)
+ cxt5066_hp_laptop_automic(codec);
}
cxt5066_set_mic_boost(codec);
return 0;
@@ -3031,6 +3068,7 @@ enum {
CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */
CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */
CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */
+ CXT5066_HP_LAPTOP, /* HP Laptop */
CXT5066_MODELS
};
@@ -3041,6 +3079,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = {
[CXT5066_DELL_VOSTO] = "dell-vostro",
[CXT5066_IDEAPAD] = "ideapad",
[CXT5066_THINKPAD] = "thinkpad",
+ [CXT5066_HP_LAPTOP] = "hp-laptop",
};
static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
@@ -3052,6 +3091,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
@@ -3116,6 +3156,23 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->num_init_verbs++;
spec->dell_automute = 1;
break;
+ case CXT5066_HP_LAPTOP:
+ codec->patch_ops.init = cxt5066_init;
+ codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+ spec->init_verbs[spec->num_init_verbs] =
+ cxt5066_init_verbs_hp_laptop;
+ spec->num_init_verbs++;
+ spec->hp_laptop = 1;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+ /* no S/PDIF out */
+ spec->multiout.dig_out_nid = 0;
+ /* input source automatically selected */
+ spec->input_mux = NULL;
+ spec->port_d_mode = 0;
+ spec->mic_boost = 3; /* default 30dB gain */
+ break;
+
case CXT5066_OLPC_XO_1_5:
codec->patch_ops.init = cxt5066_olpc_init;
codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 9feb00c..32e4be8 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -216,6 +216,11 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
}
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n",
+ protocol);
+ /* fall through */
+
case UAC_VERSION_1: {
struct uac1_ac_header_descriptor *h1 = control_header;
@@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
break;
}
-
- default:
- snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol);
- return -EINVAL;
}
return 0;
@@ -465,7 +466,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
goto __error;
}
- chip->ctrl_intf = alts;
+ /*
+ * For devices with more than one control interface, we assume the
+ * first contains the audio controls. We might need a more specific
+ * check here in the future.
+ */
+ if (!chip->ctrl_intf)
+ chip->ctrl_intf = alts;
if (err > 0) {
/* create normal USB audio interfaces */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index b853f8d..7754a10 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -295,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
switch (altsd->bInterfaceProtocol) {
case UAC_VERSION_1:
+ default:
return set_sample_rate_v1(chip, iface, alts, fmt, rate);
case UAC_VERSION_2:
return set_sample_rate_v2(chip, iface, alts, fmt, rate);
}
-
- return -EINVAL;
}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 1a701f1..ef0a07e 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -275,6 +275,12 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
/* get audio formats */
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n",
+ dev->devnum, iface_no, altno, protocol);
+ protocol = UAC_VERSION_1;
+ /* fall through */
+
case UAC_VERSION_1: {
struct uac1_as_header_descriptor *as =
snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
@@ -336,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
dev->devnum, iface_no, altno, as->bTerminalLink);
continue;
}
-
- default:
- snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n",
- dev->devnum, iface_no, altno, protocol);
- continue;
}
/* get format type */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 3a13754..6914821 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
u64 pcm_formats;
switch (protocol) {
- case UAC_VERSION_1: {
+ case UAC_VERSION_1:
+ default: {
struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
sample_width = fmt->bBitResolution;
sample_bytes = fmt->bSubframeSize;
@@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
format <<= 1;
break;
}
-
- default:
- return -EINVAL;
}
pcm_formats = 0;
@@ -384,6 +382,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
* audio class v2 uses class specific EP0 range requests for that.
*/
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ /* fall through */
case UAC_VERSION_1:
fp->channels = fmt->bNrChannels;
ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7);
@@ -392,10 +394,6 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
/* fp->channels is already set in this case */
ret = parse_audio_format_rates_v2(chip, fp);
break;
- default:
- snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n",
- chip->dev->devnum, fp->iface, fp->altsetting, protocol);
- return -EINVAL;
}
if (fp->channels < 1) {
@@ -438,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip,
fp->channels = 1;
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ /* fall through */
case UAC_VERSION_1: {
struct uac_format_type_ii_discrete_descriptor *fmt = _fmt;
brate = le16_to_cpu(fmt->wMaxBitRate);
@@ -456,10 +458,6 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip,
ret = parse_audio_format_rates_v2(chip, fp);
break;
}
- default:
- snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n",
- chip->dev->devnum, fp->iface, fp->altsetting, protocol);
- return -EINVAL;
}
return ret;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c166db0..3ed3901 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2175,7 +2175,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
}
host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
- mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol;
+ switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+ case UAC_VERSION_1:
+ default:
+ mixer->protocol = UAC_VERSION_1;
+ break;
+ case UAC_VERSION_2:
+ mixer->protocol = UAC_VERSION_2;
+ break;
+ }
if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
(err = snd_usb_mixer_status_create(mixer)) < 0)
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 3634ced..3b5135c 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
switch (altsd->bInterfaceProtocol) {
case UAC_VERSION_1:
+ default:
return init_pitch_v1(chip, iface, alts, fmt);
case UAC_VERSION_2:
return init_pitch_v2(chip, iface, alts, fmt);
}
-
- return -EINVAL;
}
/*
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-08-28 19:48 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-08-28 19:48 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.36-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All are small and trivial fixes.
Thanks!
Takashi
===
Akinobu Mita (1):
sound: oss: fix uninitialized spinlock
Axel Lin (1):
ASoC: soc-core: fix debugfs_pop_time file permissions
Dan Carpenter (1):
ALSA: pcm: add more format names
David Henningsson (1):
ALSA: hda - Add Sony VAIO quirk for ALC269
Eliot Blennerhassett (1):
ALSA: asihpi - Return hw error directly from oustream_write.
---
sound/core/pcm.c | 6 ++++++
sound/oss/sound_timer.c | 2 +-
sound/pci/asihpi/hpi6205.c | 7 +++++--
sound/pci/hda/patch_realtek.c | 1 +
sound/soc/soc-core.c | 2 +-
5 files changed, 14 insertions(+), 4 deletions(-)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index cbe815d..204af48 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -203,10 +203,16 @@ static char *snd_pcm_format_names[] = {
FORMAT(S18_3BE),
FORMAT(U18_3LE),
FORMAT(U18_3BE),
+ FORMAT(G723_24),
+ FORMAT(G723_24_1B),
+ FORMAT(G723_40),
+ FORMAT(G723_40_1B),
};
const char *snd_pcm_format_name(snd_pcm_format_t format)
{
+ if (format >= ARRAY_SIZE(snd_pcm_format_names))
+ return "Unknown";
return snd_pcm_format_names[format];
}
EXPORT_SYMBOL_GPL(snd_pcm_format_name);
diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c
index f0f0c19..48cda6c 100644
--- a/sound/oss/sound_timer.c
+++ b/sound/oss/sound_timer.c
@@ -26,7 +26,7 @@ static unsigned long prev_event_time;
static volatile unsigned long usecs_per_tmr; /* Length of the current interval */
static struct sound_lowlev_timer *tmr;
-static spinlock_t lock;
+static DEFINE_SPINLOCK(lock);
static unsigned long tmr2ticks(int tmr_value)
{
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 3b44134..22c5fc6 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -941,8 +941,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao,
}
-static u32 outstream_get_space_available(struct hpi_hostbuffer_status
- *status)
+static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status)
{
return status->size_in_bytes - (status->host_index -
status->dSP_index);
@@ -987,6 +986,10 @@ static void outstream_write(struct hpi_adapter_obj *pao,
/* write it */
phm->function = HPI_OSTREAM_WRITE;
hw_message(pao, phm, phr);
+
+ if (phr->error)
+ return;
+
/* update status information that the DSP would typically
* update (and will update next time the DSP
* buffer update task reads data from the host BBM buffer)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a4dd045..627bf99 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14467,6 +14467,7 @@ static const struct alc_fixup alc269_fixups[] = {
static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+ SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
{}
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 844ae82..acc91da 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -251,7 +251,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
printk(KERN_WARNING
"ASoC: Failed to create codec register debugfs file\n");
- codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
+ codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
codec->debugfs_codec_root,
&codec->pop_time);
if (!codec->debugfs_pop_time)
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-05-13 8:16 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-05-13 8:16 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.34-final from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This last-minute fixes became bigger than I've thought, but they are
fairly safe to apply as small regression fixes and a MIPS-specific fix.
Thanks!
Takashi
===
Clemens Ladisch (1):
ALSA: virtuoso: fix Xonar D1/DX front panel microphone
Daniel T Chen (1):
ALSA: hda: Fix 0 dB for Lenovo models using Conexant CX20549 (Venice)
Takashi Iwai (4):
ALSA: hda - Fix mute-LED GPIO pin for HP dv series
ALSA: hda - Add hp-dv4 model for IDT 92HD71bx
ALSA: pcm - Use pgprot_noncached() for MIPS non-coherent archs
ALSA: ice1724 - Fix ESI Maya44 capture source control
---
sound/core/pcm_native.c | 7 +++++++
sound/pci/hda/patch_conexant.c | 7 ++++---
sound/pci/hda/patch_sigmatel.c | 15 +++++++++++++--
sound/pci/ice1712/maya44.c | 6 +++---
sound/pci/oxygen/xonar_cs43xx.c | 3 +++
5 files changed, 30 insertions(+), 8 deletions(-)
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 8728876..20b5982 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -36,6 +36,9 @@
#include <sound/timer.h>
#include <sound/minors.h>
#include <asm/io.h>
+#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT)
+#include <dma-coherence.h>
+#endif
/*
* Compatibility
@@ -3184,6 +3187,10 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream,
substream->runtime->dma_area,
substream->runtime->dma_addr,
area->vm_end - area->vm_start);
+#elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT)
+ if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV &&
+ !plat_device_is_coherent(substream->dma_buffer.dev.dev))
+ area->vm_page_prot = pgprot_noncached(area->vm_page_prot);
#endif /* ARCH_HAS_DMA_MMAP_COHERENT */
/* mmap with fault handler */
area->vm_ops = &snd_pcm_vm_ops_data_fault;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d8213e2..feabb44 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1197,9 +1197,10 @@ static int patch_cxt5045(struct hda_codec *codec)
case 0x103c:
case 0x1631:
case 0x1734:
- /* HP, Packard Bell, & Fujitsu-Siemens laptops have really bad
- * sound over 0dB on NID 0x17. Fix max PCM level to 0 dB
- * (originally it has 0x2b steps with 0dB offset 0x14)
+ case 0x17aa:
+ /* HP, Packard Bell, Fujitsu-Siemens & Lenovo laptops have
+ * really bad sound over 0dB on NID 0x17. Fix max PCM level to
+ * 0 dB (originally it has 0x2b steps with 0dB offset 0x14)
*/
snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
(0x14 << AC_AMPCAP_OFFSET_SHIFT) |
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 12825aa..a0e06d8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -104,6 +104,7 @@ enum {
STAC_DELL_M4_2,
STAC_DELL_M4_3,
STAC_HP_M4,
+ STAC_HP_DV4,
STAC_HP_DV5,
STAC_HP_HDX,
STAC_HP_DV4_1222NR,
@@ -1691,6 +1692,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
[STAC_DELL_M4_2] = dell_m4_2_pin_configs,
[STAC_DELL_M4_3] = dell_m4_3_pin_configs,
[STAC_HP_M4] = NULL,
+ [STAC_HP_DV4] = NULL,
[STAC_HP_DV5] = NULL,
[STAC_HP_HDX] = NULL,
[STAC_HP_DV4_1222NR] = NULL,
@@ -1703,6 +1705,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
[STAC_DELL_M4_2] = "dell-m4-2",
[STAC_DELL_M4_3] = "dell-m4-3",
[STAC_HP_M4] = "hp-m4",
+ [STAC_HP_DV4] = "hp-dv4",
[STAC_HP_DV5] = "hp-dv5",
[STAC_HP_HDX] = "hp-hdx",
[STAC_HP_DV4_1222NR] = "hp-dv4-1222nr",
@@ -1721,7 +1724,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
"HP", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
- "HP dv4-7", STAC_HP_DV5),
+ "HP dv4-7", STAC_HP_DV4),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600,
"HP dv4-7", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610,
@@ -4766,6 +4769,9 @@ static void set_hp_led_gpio(struct hda_codec *codec)
struct sigmatel_spec *spec = codec->spec;
unsigned int gpio;
+ if (spec->gpio_led)
+ return;
+
gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP);
gpio &= AC_GPIO_IO_COUNT;
if (gpio > 3)
@@ -5675,6 +5681,9 @@ again:
spec->num_smuxes = 1;
spec->num_dmuxes = 1;
/* fallthrough */
+ case STAC_HP_DV4:
+ spec->gpio_led = 0x01;
+ /* fallthrough */
case STAC_HP_DV5:
snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
@@ -5688,6 +5697,7 @@ again:
spec->num_dmics = 1;
spec->num_dmuxes = 1;
spec->num_smuxes = 1;
+ spec->gpio_led = 0x08;
break;
}
@@ -5744,7 +5754,8 @@ again:
}
/* enable bass on HP dv7 */
- if (spec->board_config == STAC_HP_DV5) {
+ if (spec->board_config == STAC_HP_DV4 ||
+ spec->board_config == STAC_HP_DV5) {
unsigned int cap;
cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP);
cap &= AC_GPIO_IO_COUNT;
diff --git a/sound/pci/ice1712/maya44.c b/sound/pci/ice1712/maya44.c
index 3e1c20a..726fd4b 100644
--- a/sound/pci/ice1712/maya44.c
+++ b/sound/pci/ice1712/maya44.c
@@ -347,7 +347,7 @@ static int maya_gpio_sw_put(struct snd_kcontrol *kcontrol,
/* known working input slots (0-4) */
#define MAYA_LINE_IN 1 /* in-2 */
-#define MAYA_MIC_IN 4 /* in-5 */
+#define MAYA_MIC_IN 3 /* in-4 */
static void wm8776_select_input(struct snd_maya44 *chip, int idx, int line)
{
@@ -393,8 +393,8 @@ static int maya_rec_src_put(struct snd_kcontrol *kcontrol,
int changed;
mutex_lock(&chip->mutex);
- changed = maya_set_gpio_bits(chip->ice, GPIO_MIC_RELAY,
- sel ? GPIO_MIC_RELAY : 0);
+ changed = maya_set_gpio_bits(chip->ice, 1 << GPIO_MIC_RELAY,
+ sel ? (1 << GPIO_MIC_RELAY) : 0);
wm8776_select_input(chip, 0, sel ? MAYA_MIC_IN : MAYA_LINE_IN);
mutex_unlock(&chip->mutex);
return changed;
diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c
index 16c226b..7c4986b 100644
--- a/sound/pci/oxygen/xonar_cs43xx.c
+++ b/sound/pci/oxygen/xonar_cs43xx.c
@@ -56,6 +56,7 @@
#include <sound/pcm_params.h>
#include <sound/tlv.h>
#include "xonar.h"
+#include "cm9780.h"
#include "cs4398.h"
#include "cs4362a.h"
@@ -172,6 +173,8 @@ static void xonar_d1_init(struct oxygen *chip)
oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA,
GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE);
+ oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC);
+
xonar_init_cs53x1(chip);
xonar_enable_output(chip);
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-05-10 15:23 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-05-10 15:23 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound (HD-audio) fixes for v2.6.34-final from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/hda
This contain the following three trivial fixes.
Thanks!
Takashi
===
Andrej Gelenberg (1):
ALSA: hda - add support for Lenovo ThinkPad X100e in conexant codec
Stefan Lippers-Hollmann (1):
ALSA: Revert "ALSA: hda/realtek: quirk for D945GCLF2 mainboard"
Wu Fengguang (1):
ALSA: hda - fix DG45ID SPDIF output
---
sound/pci/hda/patch_conexant.c | 1 +
sound/pci/hda/patch_realtek.c | 1 -
sound/pci/hda/patch_sigmatel.c | 6 ++----
3 files changed, 3 insertions(+), 5 deletions(-)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 56e5207..d8213e2 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2846,6 +2846,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
+ SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7404dba..886d8e4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -17871,7 +17871,6 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
ALC663_ASUS_H13),
- SND_PCI_QUIRK(0x8086, 0xd604, "Intel mobo", ALC662_3ST_2ch_DIG),
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7fb7d01..12825aa 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1544,11 +1544,9 @@ static unsigned int alienware_m17x_pin_configs[13] = {
0x904601b0,
};
-static unsigned int intel_dg45id_pin_configs[14] = {
+static unsigned int intel_dg45id_pin_configs[13] = {
0x02214230, 0x02A19240, 0x01013214, 0x01014210,
- 0x01A19250, 0x01011212, 0x01016211, 0x40f000f0,
- 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x014510A0,
- 0x074510B0, 0x40f000f0
+ 0x01A19250, 0x01011212, 0x01016211
};
static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-05-05 8:12 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-05-05 8:12 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.34-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All small and trivial fixes there.
Thanks!
Takashi
===
Anisse Astier (1):
ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
Brian J. Tarricone (1):
ALSA: hda - fix array indexing while creating inputs for Cirrus codecs
Dan Carpenter (1):
ALSA: take tu->qlock with irqs disabled
Daniel T Chen (3):
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice)
Krzysztof Helt (1):
ALSA: es968: fix wrong PnP dma index
---
sound/core/timer.c | 5 +++--
sound/isa/sb/es968.c | 2 +-
sound/pci/hda/patch_cirrus.c | 2 +-
sound/pci/hda/patch_conexant.c | 8 ++++++--
4 files changed, 11 insertions(+), 6 deletions(-)
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7394365..5040c7b 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1160,6 +1160,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri,
{
struct snd_timer_user *tu = timeri->callback_data;
struct snd_timer_tread r1;
+ unsigned long flags;
if (event >= SNDRV_TIMER_EVENT_START &&
event <= SNDRV_TIMER_EVENT_PAUSE)
@@ -1169,9 +1170,9 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri,
r1.event = event;
r1.tstamp = *tstamp;
r1.val = resolution;
- spin_lock(&tu->qlock);
+ spin_lock_irqsave(&tu->qlock, flags);
snd_timer_user_append_to_tqueue(tu, &r1);
- spin_unlock(&tu->qlock);
+ spin_unlock_irqrestore(&tu->qlock, flags);
kill_fasync(&tu->fasync, SIGIO, POLL_IN);
wake_up(&tu->qchange_sleep);
}
diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c
index cafc3a7..ff18286 100644
--- a/sound/isa/sb/es968.c
+++ b/sound/isa/sb/es968.c
@@ -93,7 +93,7 @@ static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard,
return err;
}
port[dev] = pnp_port_start(pdev, 0);
- dma8[dev] = pnp_dma(pdev, 1);
+ dma8[dev] = pnp_dma(pdev, 0);
irq[dev] = pnp_irq(pdev, 0);
return 0;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 7de782a..350ee8a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -766,7 +766,7 @@ static int build_input(struct hda_codec *codec)
for (n = 0; n < AUTO_PIN_LAST; n++) {
if (!spec->adc_nid[n])
continue;
- err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]);
+ err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]);
if (err < 0)
return err;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 61682e1..56e5207 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1195,9 +1195,10 @@ static int patch_cxt5045(struct hda_codec *codec)
switch (codec->subsystem_id >> 16) {
case 0x103c:
+ case 0x1631:
case 0x1734:
- /* HP & Fujitsu-Siemens laptops have really bad sound over 0dB
- * on NID 0x17. Fix max PCM level to 0 dB
+ /* HP, Packard Bell, & Fujitsu-Siemens laptops have really bad
+ * sound over 0dB on NID 0x17. Fix max PCM level to 0 dB
* (originally it has 0x2b steps with 0dB offset 0x14)
*/
snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
@@ -2842,6 +2843,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
CXT5066_DELL_LAPTOP),
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
+ SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
+ SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
{}
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-04-23 15:24 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-04-23 15:24 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.34-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All are fairly trivial fixes. Most of them will go to stable tree
while one commit is a regression fix in 2.6.34.
Thanks!
Takashi
===
Daniel T Chen (4):
ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio XPS 1645
ALSA: hda: Use ALC880_F1734 quirk for Fujitsu Siemens AMILO Xi 1526
ALSA: hda: Use LPIB quirk for DG965OT board version AAD63733-203
ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio 1558
Hans de Goede (2):
ALSA: snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C
ALSA: snd-meastro3: Ignore spurious HV interrupts during suspend / resume
Takashi Iwai (1):
ALSA: hda - Fix resume from StR of HP 2510p with docking-station
---
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_analog.c | 9 ---------
sound/pci/hda/patch_realtek.c | 2 +-
sound/pci/hda/patch_sigmatel.c | 4 ++++
sound/pci/maestro3.c | 9 +++++++++
5 files changed, 15 insertions(+), 10 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f669442..cec6815 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2273,6 +2273,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
{}
};
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index af34606..e9fdfc4 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -519,14 +519,6 @@ static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
ad198x_power_eapd(codec);
return 0;
}
-
-static int ad198x_resume(struct hda_codec *codec)
-{
- ad198x_init(codec);
- snd_hda_codec_resume_amp(codec);
- snd_hda_codec_resume_cache(codec);
- return 0;
-}
#endif
static struct hda_codec_ops ad198x_patch_ops = {
@@ -539,7 +531,6 @@ static struct hda_codec_ops ad198x_patch_ops = {
#endif
#ifdef SND_HDA_NEEDS_RESUME
.suspend = ad198x_suspend,
- .resume = ad198x_resume,
#endif
.reboot_notify = ad198x_shutup,
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index aad1627..7404dba 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4143,7 +4143,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
- SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL),
+ SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734),
SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c4be3fa..7fb7d01 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1607,6 +1607,10 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
"Dell Studio 1555", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
"Dell Studio 1557", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe,
+ "Dell Studio XPS 1645", STAC_DELL_M6_BOTH),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413,
+ "Dell Studio 1558", STAC_DELL_M6_BOTH),
{} /* terminator */
};
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index b64e781..b56e336 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -849,6 +849,7 @@ struct snd_m3 {
struct snd_kcontrol *master_switch;
struct snd_kcontrol *master_volume;
struct tasklet_struct hwvol_tq;
+ unsigned int in_suspend;
#ifdef CONFIG_PM
u16 *suspend_mem;
@@ -884,6 +885,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = {
MODULE_DEVICE_TABLE(pci, snd_m3_ids);
static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c),
SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d),
SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d),
SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03),
@@ -1613,6 +1615,11 @@ static void snd_m3_update_hw_volume(unsigned long private_data)
outb(0x88, chip->iobase + SHADOW_MIX_REG_MASTER);
outb(0x88, chip->iobase + HW_VOL_COUNTER_MASTER);
+ /* Ignore spurious HV interrupts during suspend / resume, this avoids
+ mistaking them for a mute button press. */
+ if (chip->in_suspend)
+ return;
+
if (!chip->master_switch || !chip->master_volume)
return;
@@ -2424,6 +2431,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state)
if (chip->suspend_mem == NULL)
return 0;
+ chip->in_suspend = 1;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
snd_pcm_suspend_all(chip->pcm);
snd_ac97_suspend(chip->ac97);
@@ -2497,6 +2505,7 @@ static int m3_resume(struct pci_dev *pci)
snd_m3_hv_init(chip);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ chip->in_suspend = 0;
return 0;
}
#endif /* CONFIG_PM */
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-04-16 9:14 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-04-16 9:14 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.34-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This update again includes a bit bigger changes than usual.
But all changes are pretty local, and mostly safe to apply.
Thanks!
Takashi
===
Huang Weiyi (1):
ASoC: wm2000: remove unused #include <linux/version.h>
Joerg Schirottke (1):
ALSA: hda - add a quirk for Clevo M570U laptop
Kailang Yang (2):
ALSA: hda - Fix auto-parser of ALC269vb for HP pin NID 0x21
ALSA: hda - Fix setup for ALC269vb amic and dmic models
Maurus Cuelenaere (1):
ALSA: hda: Add support for Medion WIM2160
Philby John (1):
ALSA: aaci - Fix alignment faults on ARM Cortex introduced by commit 29a4f2d3
Sascha Hauer (5):
ASoC: imx-ssi: honor IMX_SSI_DMA flag
ASoC: imx-pcm-dma-mx2: restart DMA after an error
ASoC: imx-ssi: Use a hrtimer in FIQ mode
ASoC: imx-ssi: increase minimum periods to 4
ASoC: imx-ssi: do not call hrtimer_disable in trigger function
Takashi Iwai (7):
ALSA: hda - Fix initial capture source connections of ALC880/260
ALSA: usb - Fix Oops after usb-midi disconnection
ALSA: hda - Enhance fix-up table for Realtek codecs
ALSA: hda - Add fix-up for Sony VAIO with ALC269
ALSA: hda - Fix control element allocations in VIA codec parser
ALSA: hda - Avoid invalid "Independent HP" control for VIA codecs
ALSA: hda - Add position_fix quirk for Biostar mobo
---
sound/arm/aaci.c | 7 +-
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++++++++++++++++----
sound/pci/hda/patch_via.c | 41 ++++++----
sound/soc/codecs/wm2000.c | 1 -
sound/soc/imx/imx-pcm-dma-mx2.c | 15 +++-
sound/soc/imx/imx-pcm-fiq.c | 55 +++++++------
sound/soc/imx/imx-ssi.c | 3 +-
sound/usb/usbmidi.c | 24 ++++--
9 files changed, 247 insertions(+), 70 deletions(-)
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 656e474..91acc9a 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -863,7 +863,6 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
struct snd_ac97 *ac97;
int ret;
- writel(0, aaci->base + AC97_POWERDOWN);
/*
* Assert AACIRESET for 2us
*/
@@ -1047,7 +1046,11 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
writel(0x1fff, aaci->base + AACI_INTCLR);
writel(aaci->maincr, aaci->base + AACI_MAINCR);
-
+ /*
+ * Fix: ac97 read back fail errors by reading
+ * from any arbitrary aaci register.
+ */
+ readl(aaci->base + AACI_CSCH1);
ret = aaci_probe_ac97(aaci);
if (ret)
goto out;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f8fd586..f669442 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2272,6 +2272,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c7730db..aad1627 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -230,6 +230,7 @@ enum {
ALC888_ACER_ASPIRE_7730G,
ALC883_MEDION,
ALC883_MEDION_MD2,
+ ALC883_MEDION_WIM2160,
ALC883_LAPTOP_EAPD,
ALC883_LENOVO_101E_2ch,
ALC883_LENOVO_NB0763,
@@ -1389,22 +1390,31 @@ struct alc_fixup {
static void alc_pick_fixup(struct hda_codec *codec,
const struct snd_pci_quirk *quirk,
- const struct alc_fixup *fix)
+ const struct alc_fixup *fix,
+ int pre_init)
{
const struct alc_pincfg *cfg;
quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
if (!quirk)
return;
-
fix += quirk->value;
cfg = fix->pins;
- if (cfg) {
+ if (pre_init && cfg) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n",
+ codec->chip_name, quirk->name);
+#endif
for (; cfg->nid; cfg++)
snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
}
- if (fix->verbs)
+ if (!pre_init && fix->verbs) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n",
+ codec->chip_name, quirk->name);
+#endif
add_verb(codec->spec, fix->verbs);
+ }
}
static int alc_read_coef_idx(struct hda_codec *codec,
@@ -4808,6 +4818,25 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
}
}
+static void alc880_auto_init_input_src(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int c;
+
+ for (c = 0; c < spec->num_adc_nids; c++) {
+ unsigned int mux_idx;
+ const struct hda_input_mux *imux;
+ mux_idx = c >= spec->num_mux_defs ? 0 : c;
+ imux = &spec->input_mux[mux_idx];
+ if (!imux->num_items && mux_idx > 0)
+ imux = &spec->input_mux[0];
+ if (imux)
+ snd_hda_codec_write(codec, spec->adc_nids[c], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ imux->items[0].index);
+ }
+}
+
/* parse the BIOS configuration and set up the alc_spec */
/* return 1 if successful, 0 if the proper config is not found,
* or a negative error code
@@ -4886,6 +4915,7 @@ static void alc880_auto_init(struct hda_codec *codec)
alc880_auto_init_multi_out(codec);
alc880_auto_init_extra_out(codec);
alc880_auto_init_analog_input(codec);
+ alc880_auto_init_input_src(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
@@ -6397,6 +6427,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec)
}
}
+#define alc260_auto_init_input_src alc880_auto_init_input_src
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -6483,6 +6515,7 @@ static void alc260_auto_init(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
alc260_auto_init_multi_out(codec);
alc260_auto_init_analog_input(codec);
+ alc260_auto_init_input_src(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
@@ -8455,6 +8488,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc883_medion_wim2160_verbs[] = {
+ /* Unmute front mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Set speaker pin to front mixer */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Init headphone pin */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_medion_wim2160_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1a;
+ spec->autocfg.speaker_pins[0] = 0x15;
+}
+
static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -9164,6 +9233,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
[ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g",
[ALC883_MEDION] = "medion",
[ALC883_MEDION_MD2] = "medion-md2",
+ [ALC883_MEDION_WIM2160] = "medion-wim2160",
[ALC883_LAPTOP_EAPD] = "laptop-eapd",
[ALC883_LENOVO_101E_2ch] = "lenovo-101e",
[ALC883_LENOVO_NB0763] = "lenovo-nb0763",
@@ -9280,6 +9350,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
@@ -9818,6 +9889,21 @@ static struct alc_config_preset alc882_presets[] = {
.setup = alc883_medion_md2_setup,
.init_hook = alc_automute_amp,
},
+ [ALC883_MEDION_WIM2160] = {
+ .mixers = { alc883_medion_wim2160_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .adc_nids = alc883_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .setup = alc883_medion_wim2160_setup,
+ .init_hook = alc_automute_amp,
+ },
[ALC883_LAPTOP_EAPD] = {
.mixers = { alc883_base_mixer },
.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
@@ -10363,7 +10449,8 @@ static int patch_alc882(struct hda_codec *codec)
board_config = ALC882_AUTO;
}
- alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups);
+ if (board_config == ALC882_AUTO)
+ alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1);
if (board_config == ALC882_AUTO) {
/* automatic parse from the BIOS config */
@@ -10436,6 +10523,9 @@ static int patch_alc882(struct hda_codec *codec)
set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+ if (board_config == ALC882_AUTO)
+ alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0);
+
spec->vmaster_nid = 0x0c;
codec->patch_ops = alc_patch_ops;
@@ -12816,6 +12906,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
dac = 0x02;
break;
case 0x15:
+ case 0x21: /* ALC269vb has this pin, too */
dac = 0x03;
break;
default:
@@ -13735,19 +13826,19 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec,
}
}
-static void alc269_laptop_dmic_setup(struct hda_codec *codec)
+static void alc269_laptop_amic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 5;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
-static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
+static void alc269_laptop_dmic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
@@ -13755,14 +13846,14 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 6;
+ spec->int_mic.mux_idx = 5;
spec->auto_mic = 1;
}
-static void alc269_laptop_amic_setup(struct hda_codec *codec)
+static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.hp_pins[0] = 0x21;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
@@ -13771,6 +13862,18 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec)
spec->auto_mic = 1;
}
+static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x12;
+ spec->int_mic.mux_idx = 6;
+ spec->auto_mic = 1;
+}
+
static void alc269_laptop_inithook(struct hda_codec *codec)
{
alc269_speaker_automute(codec);
@@ -13975,6 +14078,27 @@ static void alc269_auto_init(struct hda_codec *codec)
alc_inithook(codec);
}
+enum {
+ ALC269_FIXUP_SONY_VAIO,
+};
+
+const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = {
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
+ {}
+};
+
+static const struct alc_fixup alc269_fixups[] = {
+ [ALC269_FIXUP_SONY_VAIO] = {
+ .verbs = alc269_sony_vaio_fixup_verbs
+ },
+};
+
+static struct snd_pci_quirk alc269_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+ {}
+};
+
+
/*
* configuration and preset
*/
@@ -14034,7 +14158,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = {
ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
- SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC),
+ SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
@@ -14108,7 +14232,7 @@ static struct alc_config_preset alc269_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc269_laptop_unsol_event,
- .setup = alc269_laptop_amic_setup,
+ .setup = alc269vb_laptop_amic_setup,
.init_hook = alc269_laptop_inithook,
},
[ALC269VB_DMIC] = {
@@ -14188,6 +14312,9 @@ static int patch_alc269(struct hda_codec *codec)
board_config = ALC269_AUTO;
}
+ if (board_config == ALC269_AUTO)
+ alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1);
+
if (board_config == ALC269_AUTO) {
/* automatic parse from the BIOS config */
err = alc269_parse_auto_config(codec);
@@ -14240,6 +14367,9 @@ static int patch_alc269(struct hda_codec *codec)
set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+ if (board_config == ALC269_AUTO)
+ alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0);
+
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
@@ -15328,7 +15458,8 @@ static int patch_alc861(struct hda_codec *codec)
board_config = ALC861_AUTO;
}
- alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups);
+ if (board_config == ALC861_AUTO)
+ alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1);
if (board_config == ALC861_AUTO) {
/* automatic parse from the BIOS config */
@@ -15365,6 +15496,9 @@ static int patch_alc861(struct hda_codec *codec)
spec->vmaster_nid = 0x03;
+ if (board_config == ALC861_AUTO)
+ alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0);
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO) {
spec->init_hook = alc861_auto_init;
@@ -16299,7 +16433,8 @@ static int patch_alc861vd(struct hda_codec *codec)
board_config = ALC861VD_AUTO;
}
- alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups);
+ if (board_config == ALC861VD_AUTO)
+ alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1);
if (board_config == ALC861VD_AUTO) {
/* automatic parse from the BIOS config */
@@ -16347,6 +16482,9 @@ static int patch_alc861vd(struct hda_codec *codec)
spec->vmaster_nid = 0x02;
+ if (board_config == ALC861VD_AUTO)
+ alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0);
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861VD_AUTO)
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 9ddc373..7345381 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -476,7 +476,7 @@ static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec,
knew->name = kstrdup(tmpl->name, GFP_KERNEL);
if (!knew->name)
return NULL;
- return 0;
+ return knew;
}
static void via_free_kctls(struct hda_codec *codec)
@@ -1215,14 +1215,13 @@ static struct snd_kcontrol_new via_hp_mixer[2] = {
},
};
-static int via_hp_build(struct via_spec *spec)
+static int via_hp_build(struct hda_codec *codec)
{
+ struct via_spec *spec = codec->spec;
struct snd_kcontrol_new *knew;
hda_nid_t nid;
-
- knew = via_clone_control(spec, &via_hp_mixer[0]);
- if (knew == NULL)
- return -ENOMEM;
+ int nums;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
switch (spec->codec_type) {
case VT1718S:
@@ -1239,6 +1238,14 @@ static int via_hp_build(struct via_spec *spec)
break;
}
+ 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)
+ return -ENOMEM;
+
knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
knew->private_value = nid;
@@ -2561,7 +2568,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
via_smart51_build(spec);
return 1;
@@ -3087,7 +3094,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
via_smart51_build(spec);
return 1;
@@ -3654,7 +3661,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
via_smart51_build(spec);
return 1;
@@ -4140,7 +4147,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
via_smart51_build(spec);
return 1;
@@ -4510,7 +4517,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
return 1;
}
@@ -4930,7 +4937,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
via_smart51_build(spec);
@@ -5425,7 +5432,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
via_smart51_build(spec);
@@ -5781,7 +5788,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
return 1;
}
@@ -6000,12 +6007,12 @@ static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec,
/* Line-Out: PortE */
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
+ "Front Playback Volume",
HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
- "Master Front Playback Switch",
+ "Front Playback Switch",
HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
@@ -6130,7 +6137,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec)
spec->input_mux = &spec->private_imux[0];
if (spec->hp_mux)
- via_hp_build(spec);
+ via_hp_build(codec);
return 1;
}
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index a34cbcf..002e289 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -23,7 +23,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/firmware.h>
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 2e79d71..2b31ac6 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -71,7 +71,12 @@ static void imx_ssi_dma_callback(int channel, void *data)
static void snd_imx_dma_err_callback(int channel, void *data, int err)
{
- pr_err("DMA error callback called\n");
+ struct snd_pcm_substream *substream = data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+ int ret;
pr_err("DMA timeout on channel %d -%s%s%s%s\n",
channel,
@@ -79,6 +84,14 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err)
err & IMX_DMA_ERR_REQUEST ? " request" : "",
err & IMX_DMA_ERR_TRANSFER ? " transfer" : "",
err & IMX_DMA_ERR_BUFFER ? " buffer" : "");
+
+ imx_dma_disable(iprtd->dma);
+ ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
+ IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ DMA_MODE_WRITE : DMA_MODE_READ);
+ if (!ret)
+ imx_dma_enable(iprtd->dma);
}
static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index f96a373..6b518e0 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -39,23 +39,24 @@ struct imx_pcm_runtime_data {
unsigned long offset;
unsigned long last_offset;
unsigned long size;
- struct timer_list timer;
- int poll_time;
+ struct hrtimer hrt;
+ int poll_time_ns;
+ struct snd_pcm_substream *substream;
+ atomic_t running;
};
-static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd)
+static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
{
- iprtd->timer.expires = jiffies + iprtd->poll_time;
-}
-
-static void imx_ssi_timer_callback(unsigned long data)
-{
- struct snd_pcm_substream *substream = (void *)data;
+ struct imx_pcm_runtime_data *iprtd =
+ container_of(hrt, struct imx_pcm_runtime_data, hrt);
+ struct snd_pcm_substream *substream = iprtd->substream;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct imx_pcm_runtime_data *iprtd = runtime->private_data;
struct pt_regs regs;
unsigned long delta;
+ if (!atomic_read(&iprtd->running))
+ return HRTIMER_NORESTART;
+
get_fiq_regs(®s);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -72,16 +73,14 @@ static void imx_ssi_timer_callback(unsigned long data)
/* If we've transferred at least a period then report it and
* reset our poll time */
- if (delta >= runtime->period_size) {
+ if (delta >= iprtd->period) {
snd_pcm_period_elapsed(substream);
iprtd->last_offset = iprtd->offset;
-
- imx_ssi_set_next_poll(iprtd);
}
- /* Restart the timer; if we didn't report we'll run on the next tick */
- add_timer(&iprtd->timer);
+ hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns));
+ return HRTIMER_RESTART;
}
static struct fiq_handler fh = {
@@ -99,8 +98,8 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
iprtd->period = params_period_bytes(params) ;
iprtd->offset = 0;
iprtd->last_offset = 0;
- iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params));
-
+ iprtd->poll_time_ns = 1000000000 / params_rate(params) *
+ params_period_size(params);
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
return 0;
@@ -135,8 +134,9 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- imx_ssi_set_next_poll(iprtd);
- add_timer(&iprtd->timer);
+ atomic_set(&iprtd->running, 1);
+ hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns),
+ HRTIMER_MODE_REL);
if (++fiq_enable == 1)
enable_fiq(imx_pcm_fiq);
@@ -145,11 +145,11 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- del_timer(&iprtd->timer);
+ atomic_set(&iprtd->running, 0);
+
if (--fiq_enable == 0)
disable_fiq(imx_pcm_fiq);
-
break;
default:
return -EINVAL;
@@ -180,7 +180,7 @@ static struct snd_pcm_hardware snd_imx_hardware = {
.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
.period_bytes_min = 128,
.period_bytes_max = 16 * 1024,
- .periods_min = 2,
+ .periods_min = 4,
.periods_max = 255,
.fifo_size = 0,
};
@@ -194,9 +194,11 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
runtime->private_data = iprtd;
- init_timer(&iprtd->timer);
- iprtd->timer.data = (unsigned long)substream;
- iprtd->timer.function = imx_ssi_timer_callback;
+ iprtd->substream = substream;
+
+ atomic_set(&iprtd->running, 0);
+ hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ iprtd->hrt.function = snd_hrtimer_callback;
ret = snd_pcm_hw_constraint_integer(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
@@ -212,7 +214,8 @@ static int snd_imx_close(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- del_timer_sync(&iprtd->timer);
+ hrtimer_cancel(&iprtd->hrt);
+
kfree(iprtd);
return 0;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 0bcc6d7..80b4fee 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -656,7 +656,8 @@ static int imx_ssi_probe(struct platform_device *pdev)
dai->private_data = ssi;
if ((cpu_is_mx27() || cpu_is_mx21()) &&
- !(ssi->flags & IMX_SSI_USE_AC97)) {
+ !(ssi->flags & IMX_SSI_USE_AC97) &&
+ (ssi->flags & IMX_SSI_DMA)) {
ssi->flags |= IMX_SSI_DMA;
platform = imx_ssi_dma_mx2_init(pdev, ssi);
} else
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 2c59afd..9e28b20 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream)
DEFINE_WAIT(wait);
long timeout = msecs_to_jiffies(50);
+ if (ep->umidi->disconnected)
+ return;
/*
* The substream buffer is empty, but some data might still be in the
* currently active URBs, so we have to wait for those to complete.
@@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi,
* Frees an output endpoint.
* May be called when ep hasn't been initialized completely.
*/
-static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep)
{
unsigned int i;
for (i = 0; i < OUTPUT_URBS; ++i)
- if (ep->urbs[i].urb)
+ if (ep->urbs[i].urb) {
free_urb_and_buffer(ep->umidi, ep->urbs[i].urb,
ep->max_transfer);
+ ep->urbs[i].urb = NULL;
+ }
+}
+
+static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep)
+{
+ snd_usbmidi_out_endpoint_clear(ep);
kfree(ep);
}
@@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p)
usb_kill_urb(ep->out->urbs[j].urb);
if (umidi->usb_protocol_ops->finish_out_endpoint)
umidi->usb_protocol_ops->finish_out_endpoint(ep->out);
+ ep->out->active_urbs = 0;
+ if (ep->out->drain_urbs) {
+ ep->out->drain_urbs = 0;
+ wake_up(&ep->out->drain_wait);
+ }
}
if (ep->in)
for (j = 0; j < INPUT_URBS; ++j)
usb_kill_urb(ep->in->urbs[j]);
/* free endpoints here; later call can result in Oops */
- if (ep->out) {
- snd_usbmidi_out_endpoint_delete(ep->out);
- ep->out = NULL;
- }
+ if (ep->out)
+ snd_usbmidi_out_endpoint_clear(ep->out);
if (ep->in) {
snd_usbmidi_in_endpoint_delete(ep->in);
ep->in = NULL;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-04-07 8:11 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-04-07 8:11 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.34-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This became a larger update than usual in the end. The biggest chunks
are Realtek ALC269 codec parser update, ASoC WM8994 codec update, and
ASoC wm_hubs DC servo fixes. Also, dma_data corruption fix for ASoC is
found in ASoC changes.
Other than that, most of commits are small spotty fixes, including the
security fix for invalid procfs read of mixart driver.
Thanks!
Takashi
===
Dan Carpenter (3):
ASoC: wm8994: playback => capture
ALSA: i2c: cleanup: change parameter to pointer
ALSA: mixart: range checking proc file
Daniel Mack (1):
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
Daniel T Chen (1):
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
Graham Gower (1):
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
Janusz Krzysztofik (1):
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
Julia Lawall (1):
ALSA: echoaudio - Eliminate use after free
Mark Brown (5):
ASoC: Avoid wraparound in wm_hubs DC servo correction
ASoC: Support second DC servo readback method for wm_hubs
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
ASoC: Only do WM8994 bias off transition from standby
Takashi Iwai (6):
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALSA: hda - Add missing printk argument in previous patch
ALSA: hda - Update document about MSI and interrupts
ALSA: hda - Add MSI blacklist for Aopen MZ915-M
ALSA: hda - Fix a wrong array range check in patch_realtek.c
Tony Vroon (1):
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
---
Documentation/sound/alsa/HD-Audio.txt | 16 +++-
include/sound/ak4113.h | 2 +-
include/sound/soc-dai.h | 18 ++++-
include/sound/soc.h | 1 +
sound/i2c/other/ak4113.c | 2 +-
sound/pci/echoaudio/echoaudio.c | 5 +-
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_analog.c | 8 ++
sound/pci/hda/patch_realtek.c | 164 +++++++++++++++++++++++----------
sound/pci/mixart/mixart.c | 24 +++--
sound/soc/atmel/atmel-pcm.c | 2 +-
sound/soc/atmel/atmel_ssc_dai.c | 6 +-
sound/soc/codecs/ac97.c | 15 ++-
sound/soc/codecs/wm8994.c | 58 +++++++-----
sound/soc/codecs/wm_hubs.c | 83 ++++++++++-------
sound/soc/codecs/wm_hubs.h | 1 +
sound/soc/davinci/davinci-i2s.c | 3 +-
sound/soc/davinci/davinci-mcasp.c | 3 +-
sound/soc/davinci/davinci-pcm.c | 4 +-
sound/soc/imx/imx-pcm-dma-mx2.c | 8 +-
sound/soc/imx/imx-ssi.c | 7 +-
sound/soc/omap/omap-mcbsp.c | 4 +-
sound/soc/omap/omap-mcpdm.c | 3 +-
sound/soc/omap/omap-pcm.c | 21 ++--
sound/soc/pxa/pxa-ssp.c | 23 +++--
sound/soc/pxa/pxa2xx-ac97.c | 17 +++-
sound/soc/pxa/pxa2xx-i2s.c | 7 +-
sound/soc/pxa/pxa2xx-pcm.c | 4 +-
sound/soc/s3c24xx/s3c-ac97.c | 21 +++--
sound/soc/s3c24xx/s3c-dma.c | 4 +-
sound/soc/s3c24xx/s3c-i2s-v2.c | 13 ++-
sound/soc/s3c24xx/s3c-pcm.c | 7 +-
sound/soc/s3c24xx/s3c24xx-i2s.c | 19 ++--
sound/soc/s6000/s6000-i2s.c | 3 +-
sound/soc/s6000/s6000-pcm.c | 40 ++++++--
sound/soc/soc-core.c | 3 +-
36 files changed, 408 insertions(+), 212 deletions(-)
^ permalink raw reply [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-03-29 9:02 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-03-29 9:02 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.34-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
A couple of one-liner quirks, in addition to a fix of a regression
introduced in 2.6.34-rc1 with CONFIG_SND_DEBUG_*=n kconfig.
Thanks!
Takashi
===
Daniel Chen (1):
ALSA: ac97: Add Toshiba P500 to ac97 jack sense blacklist
Daniel T Chen (2):
ALSA: ac97: Add IBM ThinkPad R40e to Headphone/Line Jack Sense blacklist
ALSA: hda: Use LPIB for ga-ma770-ud3 board
Jarkko Nikula (1):
ALSA: pcm_lib - fix xrun functionality
Takashi Iwai (1):
ALSA: hda - Don't set invalid connection index in Realtek initialiaiton
---
sound/core/pcm_lib.c | 6 ++++--
sound/pci/ac97/ac97_patch.c | 2 ++
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_realtek.c | 5 ++++-
4 files changed, 11 insertions(+), 3 deletions(-)
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index b546ac2..a2ff861 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -148,6 +148,9 @@ static void pcm_debug_name(struct snd_pcm_substream *substream,
#define xrun_debug(substream, mask) \
((substream)->pstr->xrun_debug & (mask))
+#else
+#define xrun_debug(substream, mask) 0
+#endif
#define dump_stack_on_xrun(substream) do { \
if (xrun_debug(substream, XRUN_DEBUG_STACK)) \
@@ -169,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream)
}
}
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
#define hw_ptr_error(substream, fmt, args...) \
do { \
if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \
@@ -255,8 +259,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream)
#else /* ! CONFIG_SND_PCM_XRUN_DEBUG */
-#define xrun_debug(substream, mask) 0
-#define xrun(substream) do { } while (0)
#define hw_ptr_error(substream, fmt, args...) do { } while (0)
#define xrun_log(substream, pos) do { } while (0)
#define xrun_log_show(substream) do { } while (0)
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 1caf5e3..e68c98e 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1852,12 +1852,14 @@ static unsigned int ad1981_jacks_blacklist[] = {
0x10140523, /* Thinkpad R40 */
0x10140534, /* Thinkpad X31 */
0x10140537, /* Thinkpad T41p */
+ 0x1014053e, /* Thinkpad R40e */
0x10140554, /* Thinkpad T42p/R50p */
0x10140567, /* Thinkpad T43p 2668-G7U */
0x10140581, /* Thinkpad X41-2527 */
0x10280160, /* Dell Dimension 2400 */
0x104380b0, /* Asus A7V8X-MX */
0x11790241, /* Toshiba Satellite A-15 S127 */
+ 0x1179ff10, /* Toshiba P500 */
0x144dc01a, /* Samsung NP-X20C004/SEG */
0 /* end */
};
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8b29156..4bb9067 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2269,6 +2269,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 053d53d..9a23444 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10043,8 +10043,11 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
alc_set_pin_output(codec, nid, pin_type);
if (spec->multiout.dac_nids[dac_idx] == 0x25)
idx = 4;
- else
+ else {
+ if (spec->multiout.num_dacs >= dac_idx)
+ return;
idx = spec->multiout.dac_nids[dac_idx] - 2;
+ }
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-03-24 7:07 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-03-24 7:07 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.34-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing following small changes.
Thanks!
Takashi
===
Clemens Ladisch (1):
ALSA: cmipci: work around invalid PCM pointer
Daniel Mack (1):
ASoC: pxa-pcm-lib: initialize DMA channel to -1
Daniel T Chen (1):
ALSA: hda: Fix 0 dB offset for HP laptops using CX20551 (Waikiki)
Derek Kelly (1):
ALSA: hda - Add support of Nvidia GT220 HDMI
Guennadi Liakhovetski (1):
ASoC: SIU driver shall select FW_LOADER
Julia Lawall (1):
sound/oss/vidc.c: change the field used with DMA_ACTIVE
Kunal Gangakhedkar (1):
ALSA: hda - Add PCI quirk for HP dv6-1110ax.
Mark Brown (2):
ASoC: Bail out of wm_hubs DC servo if calibration fails
ASoC: Remove BROKEN from i.MX audio after dependencies merged
Peter Ujfalusi (2):
ASoC: tlv320dac33: Fix DSP modes
ASoC: tlv320dac33: Internal clocking changes
Takashi Iwai (2):
ALSA: hda - Sort codec entry list of Nvidia HDMI
ALSA: hda - Fix access-after-free in patch_realtek.c
---
sound/arm/pxa2xx-pcm-lib.c | 1 +
sound/oss/vidc.c | 4 ++--
sound/pci/cmipci.c | 14 +++++++++++---
sound/pci/hda/patch_conexant.c | 15 +++++++++++++++
sound/pci/hda/patch_nvhdmi.c | 15 +++++++++------
sound/pci/hda/patch_realtek.c | 5 +++--
sound/pci/hda/patch_sigmatel.c | 2 ++
sound/soc/codecs/tlv320dac33.c | 10 +++-------
sound/soc/codecs/wm_hubs.c | 2 +-
sound/soc/imx/Kconfig | 2 +-
sound/soc/sh/Kconfig | 1 +
11 files changed, 49 insertions(+), 22 deletions(-)
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 743ac6a..fd51fa8 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -205,6 +205,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
if (!rtd->dma_desc_array)
goto err1;
+ rtd->dma_ch = -1;
runtime->private_data = rtd;
return 0;
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
index 725fef0..a4127ba 100644
--- a/sound/oss/vidc.c
+++ b/sound/oss/vidc.c
@@ -363,13 +363,13 @@ static void vidc_audio_trigger(int dev, int enable_bits)
struct audio_operations *adev = audio_devs[dev];
if (enable_bits & PCM_ENABLE_OUTPUT) {
- if (!(adev->flags & DMA_ACTIVE)) {
+ if (!(adev->dmap_out->flags & DMA_ACTIVE)) {
unsigned long flags;
local_irq_save(flags);
/* prevent recusion */
- adev->flags |= DMA_ACTIVE;
+ adev->dmap_out->flags |= DMA_ACTIVE;
dma_interrupt = vidc_audio_dma_interrupt;
vidc_sound_dma_irq(0, NULL);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 1ded64e..329968e 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -941,13 +941,21 @@ static snd_pcm_uframes_t snd_cmipci_pcm_pointer(struct cmipci *cm, struct cmipci
struct snd_pcm_substream *substream)
{
size_t ptr;
- unsigned int reg;
+ unsigned int reg, rem, tries;
+
if (!rec->running)
return 0;
#if 1 // this seems better..
reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
- ptr = rec->dma_size - (snd_cmipci_read_w(cm, reg) + 1);
- ptr >>= rec->shift;
+ for (tries = 0; tries < 3; tries++) {
+ rem = snd_cmipci_read_w(cm, reg);
+ if (rem < rec->dma_size)
+ goto ok;
+ }
+ printk(KERN_ERR "cmipci: invalid PCM pointer: %#x\n", rem);
+ return SNDRV_PCM_POS_XRUN;
+ok:
+ ptr = (rec->dma_size - (rem + 1)) >> rec->shift;
#else
reg = rec->ch ? CM_REG_CH1_FRAME1 : CM_REG_CH0_FRAME1;
ptr = snd_cmipci_read(cm, reg) - rec->offset;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 194a28c..61682e1 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1591,6 +1591,21 @@ static int patch_cxt5047(struct hda_codec *codec)
#endif
}
spec->vmaster_nid = 0x13;
+
+ switch (codec->subsystem_id >> 16) {
+ case 0x103c:
+ /* HP laptops have really bad sound over 0 dB on NID 0x10.
+ * Fix max PCM level to 0 dB (originally it has 0x1e steps
+ * with 0 dB offset 0x17)
+ */
+ snd_hda_override_amp_caps(codec, 0x10, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
+ break;
+ }
+
return 0;
}
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 70669a2..3c10c0b 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -538,8 +538,6 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
* patch entries
*/
static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
- { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
- { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch },
{ .id = 0x10de0002, .name = "MCP77/78 HDMI",
.patch = patch_nvhdmi_8ch_7x },
{ .id = 0x10de0003, .name = "MCP77/78 HDMI",
@@ -550,12 +548,16 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
.patch = patch_nvhdmi_8ch_7x },
{ .id = 0x10de0007, .name = "MCP79/7A HDMI",
.patch = patch_nvhdmi_8ch_7x },
- { .id = 0x10de000c, .name = "MCP89 HDMI",
+ { .id = 0x10de000a, .name = "GT220 HDMI",
.patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de000b, .name = "GT21x HDMI",
.patch = patch_nvhdmi_8ch_89 },
+ { .id = 0x10de000c, .name = "MCP89 HDMI",
+ .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de000d, .name = "GT240 HDMI",
.patch = patch_nvhdmi_8ch_89 },
+ { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
+ { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch },
{} /* terminator */
};
@@ -564,11 +566,12 @@ MODULE_ALIAS("snd-hda-codec-id:10de0003");
MODULE_ALIAS("snd-hda-codec-id:10de0005");
MODULE_ALIAS("snd-hda-codec-id:10de0006");
MODULE_ALIAS("snd-hda-codec-id:10de0007");
-MODULE_ALIAS("snd-hda-codec-id:10de0067");
-MODULE_ALIAS("snd-hda-codec-id:10de8001");
-MODULE_ALIAS("snd-hda-codec-id:10de000c");
+MODULE_ALIAS("snd-hda-codec-id:10de000a");
MODULE_ALIAS("snd-hda-codec-id:10de000b");
+MODULE_ALIAS("snd-hda-codec-id:10de000c");
MODULE_ALIAS("snd-hda-codec-id:10de000d");
+MODULE_ALIAS("snd-hda-codec-id:10de0067");
+MODULE_ALIAS("snd-hda-codec-id:10de8001");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("NVIDIA HDMI HD-audio codec");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4ec5763..053d53d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2532,8 +2532,6 @@ static int alc_build_controls(struct hda_codec *codec)
return err;
}
- alc_free_kctls(codec); /* no longer needed */
-
/* assign Capture Source enums to NID */
kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
if (!kctl)
@@ -2602,6 +2600,9 @@ static int alc_build_controls(struct hda_codec *codec)
}
}
}
+
+ alc_free_kctls(codec); /* no longer needed */
+
return 0;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8c416bb..c4be3fa 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1730,6 +1730,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"HP HDX", STAC_HP_HDX), /* HDX16 */
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620,
"HP dv6", STAC_HP_DV5),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061,
+ "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010,
"HP", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index f9f367d..d50f169 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -778,7 +778,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
if (dac33->fifo_mode) {
/* Generic for all FIFO modes */
/* 50-51 : ASRC Control registers */
- dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */
+ dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCLKDIV(1));
dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */
/* Write registers 0x34 and 0x35 (MSB, LSB) */
@@ -1038,11 +1038,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai,
case SND_SOC_DAIFMT_DSP_A:
aictrl_a |= DAC33_AFMT_DSP;
aictrl_b &= ~DAC33_DATA_DELAY_MASK;
- aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */
- break;
- case SND_SOC_DAIFMT_DSP_B:
- aictrl_a |= DAC33_AFMT_DSP;
- aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */
+ aictrl_b |= DAC33_DATA_DELAY(0);
break;
case SND_SOC_DAIFMT_RIGHT_J:
aictrl_a |= DAC33_AFMT_RIGHT_J;
@@ -1066,7 +1062,7 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
{
/* 44-46: DAC Control Registers */
/* A : DAC sample rate Fsref/1.5 */
- dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1));
+ dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0));
/* B : DAC src=normal, not muted */
dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT |
DAC33_DACSRCL_LEFT);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 0ad9f5d..486bdd2 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -74,7 +74,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
msleep(1);
reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0);
dev_dbg(codec->dev, "DC servo: %x\n", reg);
- } while (reg & WM8993_DCS_DATAPATH_BUSY);
+ } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400);
if (reg & WM8993_DCS_DATAPATH_BUSY)
dev_err(codec->dev, "Timed out waiting for DC Servo\n");
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index c7d0fd9..7174b4c 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -1,6 +1,6 @@
config SND_IMX_SOC
tristate "SoC Audio for Freescale i.MX CPUs"
- depends on ARCH_MXC && BROKEN
+ depends on ARCH_MXC
select SND_PCM
select FIQ
select SND_SOC_AC97_BUS
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 1066749..f07f6d8 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -32,6 +32,7 @@ config SND_SOC_SH4_SIU
select DMA_ENGINE
select DMADEVICES
select SH_DMAE
+ select FW_LOADER
##
## Boards
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-03-17 8:07 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-03-17 8:07 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.34-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following fixes. Most of them are one- (or a few) liners.
Thanks!
Takashi
===
Anisse Astier (1):
ALSA: hda - Add PCI quirks for MSI NetOn AP1900 and Wind Top AE2220
Dan Carpenter (1):
sound: sequencer: clean up remove bogus check
Daniel T Chen (1):
ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212
Takashi Iwai (3):
ALSA: hda - Add an error message for invalid mapping NID
ALSA: hda - Fix secondary ADC of ALC260 basic model
ALSA: hda - Disable MSI for Nvidia controller
Vitaliy Kulikov (1):
ALSA: hda - New Intel HDA controller
---
sound/oss/sequencer.c | 2 --
sound/pci/hda/hda_codec.c | 4 +++-
sound/pci/hda/hda_intel.c | 9 +++++++++
sound/pci/hda/patch_realtek.c | 6 ++++--
4 files changed, 16 insertions(+), 5 deletions(-)
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index c798746..e85789e 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -1631,8 +1631,6 @@ unsigned long compute_finetune(unsigned long base_freq, int bend, int range,
}
semitones = bend / 100;
- if (semitones > 99)
- semitones = 99;
cents = bend % 100;
amount = (int) (semitone_tuning[semitones] * multiplier * cent_tuning[cents]) / 10000;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5bd7cf4..0e76ac2 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1806,6 +1806,8 @@ int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl,
item->nid = nid;
return 0;
}
+ printk(KERN_ERR "hda-codec: no NID for mapping control %s:%d:%d\n",
+ kctl->id.name, kctl->id.index, index);
return -EINVAL;
}
EXPORT_SYMBOL_HDA(snd_hda_add_nid);
@@ -2884,7 +2886,7 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus)
list_for_each_entry(codec, &bus->codec_list, list) {
int err = snd_hda_codec_build_controls(codec);
if (err < 0) {
- printk(KERN_ERR "hda_codec: cannot build controls"
+ printk(KERN_ERR "hda_codec: cannot build controls "
"for #%d (error %d)\n", codec->addr, err);
err = snd_hda_codec_reset(codec);
if (err < 0) {
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index da1ac90..8b29156 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2271,6 +2271,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
{}
};
@@ -2378,6 +2379,13 @@ static void __devinit check_msi(struct azx *chip)
"hda_intel: msi for device %04x:%04x set to %d\n",
q->subvendor, q->subdevice, q->value);
chip->msi = q->value;
+ return;
+ }
+
+ /* NVidia chipsets seem to cause troubles with MSI */
+ if (chip->driver_type == AZX_DRIVER_NVIDIA) {
+ printk(KERN_INFO "hda_intel: Disable MSI for Nvidia chipset\n");
+ chip->msi = 0;
}
}
@@ -2706,6 +2714,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH },
/* PCH */
{ PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x3b57), .driver_data = AZX_DRIVER_ICH },
/* CPT */
{ PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH },
/* SCH */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a83719..4ec5763 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6477,7 +6477,7 @@ static struct alc_config_preset alc260_presets[] = {
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_adc_nids,
+ .adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
@@ -9195,6 +9195,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
@@ -9204,6 +9205,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
@@ -9235,7 +9237,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL),
SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL),
SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL),
- SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
+ SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG),
{}
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-03-11 21:41 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-03-11 21:41 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.34-rc1 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following small fixes.
Thanks!
Takashi
===
Ralf Gerbig (1):
ALSA: hda - Sound MSI fallout on a Asus mobo NVIDIA MCP55
Randy Dunlap (1):
sound: fix opti92x-ad1848 build
Takashi Iwai (2):
ALSA: hda - Fix wrong model range check for ALC268
ALSA: hda - Fix input source elements of secondary ADCs on Realtek
---
sound/isa/opti9xx/opti92x-ad1848.c | 3 ++-
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_realtek.c | 8 +++++++-
3 files changed, 10 insertions(+), 2 deletions(-)
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index becd90d..4d2d040 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -217,8 +217,9 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip,
if (isapnp && chip->mc_base)
/* PnP resource gives the least 10 bits */
chip->mc_base |= 0xc00;
+ else
#endif /* CONFIG_PNP */
- else {
+ {
chip->mc_base = 0xf8c;
chip->mc_base_size = opti9xx_mc_size[hardware];
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 43b7cfb..da1ac90 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2358,6 +2358,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
static struct snd_pci_quirk msi_black_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
+ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */
SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5d2fbb8..3a83719 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -411,6 +411,8 @@ static int alc_mux_enum_info(struct snd_kcontrol *kcontrol,
unsigned int mux_idx = snd_ctl_get_ioffidx(kcontrol, &uinfo->id);
if (mux_idx >= spec->num_mux_defs)
mux_idx = 0;
+ if (!spec->input_mux[mux_idx].num_items && mux_idx > 0)
+ mux_idx = 0;
return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo);
}
@@ -439,6 +441,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
imux = &spec->input_mux[mux_idx];
+ if (!imux->num_items && mux_idx > 0)
+ imux = &spec->input_mux[0];
type = get_wcaps_type(get_wcaps(codec, nid));
if (type == AC_WID_AUD_MIX) {
@@ -10105,6 +10109,8 @@ static void alc882_auto_init_input_src(struct hda_codec *codec)
continue;
mux_idx = c >= spec->num_mux_defs ? 0 : c;
imux = &spec->input_mux[mux_idx];
+ if (!imux->num_items && mux_idx > 0)
+ imux = &spec->input_mux[0];
for (idx = 0; idx < conns; idx++) {
/* if the current connection is the selected one,
* unmute it as default - otherwise mute it
@@ -13201,7 +13207,7 @@ static int patch_alc268(struct hda_codec *codec)
if (board_config < 0 || board_config >= ALC268_MODEL_LAST)
board_config = snd_hda_check_board_codec_sid_config(codec,
- ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
+ ALC268_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-02-15 17:40 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-02-15 17:40 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fix for v2.6.33 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/hda
This contains the following trivial fix for a wrong PCI SSID.
Thanks!
Takashi
===
Takashi Iwai (1):
ALSA: hda - Correct ASUA blacklist for MSI brokenness
---
sound/pci/hda/hda_intel.c | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3600e9c..ff6da6f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2350,7 +2350,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
*/
static struct snd_pci_quirk msi_black_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
- SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */
+ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
{}
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
2010-02-12 17:03 ` Takashi Iwai
@ 2010-02-12 17:37 ` Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-02-12 17:37 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull a sound fix for v2.6.33-rc8 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/hda
Thanks!
Takashi
===
Takashi Iwai (1):
ALSA: hda - use WARN_ON_ONCE() for zero-division detection
---
sound/pci/hda/hda_intel.c | 9 +++------
1 files changed, 3 insertions(+), 6 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e767c3f..3600e9c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1893,12 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
if (!bdl_pos_adj[chip->dev_index])
return 1; /* no delayed ack */
- if (azx_dev->period_bytes == 0) {
- printk(KERN_WARNING
- "hda-intel: Divide by zero was avoided "
- "in azx_dev->period_bytes.\n");
- return 0;
- }
+ if (WARN_ONCE(!azx_dev->period_bytes,
+ "hda-intel: zero azx_dev->period_bytes"))
+ return 0; /* this shouldn't happen! */
if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
return 0; /* NG - it's below the period boundary */
return 1; /* OK, it's fine */
^ permalink raw reply related [flat|nested] 60+ messages in thread
* Re: [GIT PULL] sound fixes
2010-02-12 16:51 ` Linus Torvalds
@ 2010-02-12 17:03 ` Takashi Iwai
2010-02-12 17:37 ` Takashi Iwai
0 siblings, 1 reply; 60+ messages in thread
From: Takashi Iwai @ 2010-02-12 17:03 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
At Fri, 12 Feb 2010 08:51:29 -0800 (PST),
Linus Torvalds wrote:
>
>
>
> On Fri, 12 Feb 2010, Takashi Iwai wrote:
> >
> > Jody Bruchon (1):
> > ALSA: hda-intel: Avoid divide by zero crash
>
> I pulled, but I really think it should look more like:
>
> >
> > ---
> > sound/pci/hda/hda_intel.c | 6 ++++++
> > 1 files changed, 6 insertions(+), 0 deletions(-)
> >
> > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
> > index b8faa6d..e767c3f 100644
> > --- a/sound/pci/hda/hda_intel.c
> > +++ b/sound/pci/hda/hda_intel.c
> > @@ -1893,6 +1893,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
> >
> > if (!bdl_pos_adj[chip->dev_index])
> > return 1; /* no delayed ack */
> > + if (azx_dev->period_bytes == 0) {
> > + printk(KERN_WARNING
> > + "hda-intel: Divide by zero was avoided "
> > + "in azx_dev->period_bytes.\n");
> > + return 0;
> > + }
>
> if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")
> return 0;
>
> or something. It really doesn't want to warn multiple times, and by using
> a WARN_ONCE() it will also get picked up by kerneloops etc.
Ah, right.
> Or if that zero period_bytes is normal under certain circumstances, then
> there shouldn't be a printout at all.
This zero division shouldn't happen at all. I've never seen this, and
this is the first report. I suspect it being some bogus IRQ, but the
fix itself doesn't do wrong, so picked it up.
I'm going to fix to WARN_ONCE() quickly and send a pull request again.
Thanks!
Takashi
^ permalink raw reply [flat|nested] 60+ messages in thread
* Re: [GIT PULL] sound fixes
2010-02-12 15:33 Takashi Iwai
@ 2010-02-12 16:51 ` Linus Torvalds
2010-02-12 17:03 ` Takashi Iwai
0 siblings, 1 reply; 60+ messages in thread
From: Linus Torvalds @ 2010-02-12 16:51 UTC (permalink / raw)
To: Takashi Iwai; +Cc: Andrew Morton, linux-kernel
On Fri, 12 Feb 2010, Takashi Iwai wrote:
>
> Jody Bruchon (1):
> ALSA: hda-intel: Avoid divide by zero crash
I pulled, but I really think it should look more like:
>
> ---
> sound/pci/hda/hda_intel.c | 6 ++++++
> 1 files changed, 6 insertions(+), 0 deletions(-)
>
> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
> index b8faa6d..e767c3f 100644
> --- a/sound/pci/hda/hda_intel.c
> +++ b/sound/pci/hda/hda_intel.c
> @@ -1893,6 +1893,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
>
> if (!bdl_pos_adj[chip->dev_index])
> return 1; /* no delayed ack */
> + if (azx_dev->period_bytes == 0) {
> + printk(KERN_WARNING
> + "hda-intel: Divide by zero was avoided "
> + "in azx_dev->period_bytes.\n");
> + return 0;
> + }
if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")
return 0;
or something. It really doesn't want to warn multiple times, and by using
a WARN_ONCE() it will also get picked up by kerneloops etc.
Or if that zero period_bytes is normal under certain circumstances, then
there shouldn't be a printout at all.
Linus
^ permalink raw reply [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-02-12 15:33 Takashi Iwai
2010-02-12 16:51 ` Linus Torvalds
0 siblings, 1 reply; 60+ messages in thread
From: Takashi Iwai @ 2010-02-12 15:33 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.33-rc8 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/hda
Hopefully this is the last pull request for 2.6.33...
Thanks!
Takashi
===
Jody Bruchon (1):
ALSA: hda-intel: Avoid divide by zero crash
---
sound/pci/hda/hda_intel.c | 6 ++++++
1 files changed, 6 insertions(+), 0 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b8faa6d..e767c3f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1893,6 +1893,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
if (!bdl_pos_adj[chip->dev_index])
return 1; /* no delayed ack */
+ if (azx_dev->period_bytes == 0) {
+ printk(KERN_WARNING
+ "hda-intel: Divide by zero was avoided "
+ "in azx_dev->period_bytes.\n");
+ return 0;
+ }
if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
return 0; /* NG - it's below the period boundary */
return 1; /* OK, it's fine */
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-02-05 19:05 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-02-05 19:05 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.33-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Large changes are for a long-standing bug of ctxfi driver.
The others are rather sparse and small changes.
Thanks!
Takashi
===
Grazvydas Ignotas (1):
ASoC: pandora: Add APLL supply to fix audio output
Jaroslav Kysela (2):
ALSA: ctxfi - fix PTP address initialization
ALSA: ice1724 - aureon - fix wm8770 volume offset
Maxim Levitsky (2):
ALSA: hda - Delay switching to polling mode if an interrupt was missing
ALSA: cosmetic: make hda intel interrupt name consistent with others
---
sound/pci/ctxfi/ctatc.c | 15 ++-------------
sound/pci/ctxfi/ctvmem.c | 38 ++++++++++++++++++--------------------
sound/pci/ctxfi/ctvmem.h | 8 +++++---
sound/pci/hda/hda_intel.c | 21 ++++++++++++++++++---
sound/pci/ice1712/aureon.c | 12 +++++++-----
sound/soc/omap/omap3pandora.c | 1 +
6 files changed, 51 insertions(+), 44 deletions(-)
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index cb65bd0..459c1f6 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -166,18 +166,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index)
{
- struct ct_vm *vm;
- void *kvirt_addr;
- unsigned long phys_addr;
-
- vm = atc->vm;
- kvirt_addr = vm->get_ptp_virt(vm, index);
- if (kvirt_addr == NULL)
- phys_addr = (~0UL);
- else
- phys_addr = virt_to_phys(kvirt_addr);
-
- return phys_addr;
+ return atc->vm->get_ptp_phys(atc->vm, index);
}
static unsigned int convert_format(snd_pcm_format_t snd_format)
@@ -1669,7 +1658,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
}
/* Set up device virtual memory management object */
- err = ct_vm_create(&atc->vm);
+ err = ct_vm_create(&atc->vm, pci);
if (err < 0)
goto error1;
diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c
index 6b78752..65da6e4 100644
--- a/sound/pci/ctxfi/ctvmem.c
+++ b/sound/pci/ctxfi/ctvmem.c
@@ -138,7 +138,7 @@ ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size)
return NULL;
}
- ptp = vm->ptp[0];
+ ptp = (unsigned long *)vm->ptp[0].area;
pte_start = (block->addr >> CT_PAGE_SHIFT);
pages = block->size >> CT_PAGE_SHIFT;
for (i = 0; i < pages; i++) {
@@ -158,25 +158,25 @@ static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block)
}
/* *
- * return the host (kmalloced) addr of the @index-th device
- * page talbe page on success, or NULL on failure.
- * The first returned NULL indicates the termination.
+ * return the host physical addr of the @index-th device
+ * page table page on success, or ~0UL on failure.
+ * The first returned ~0UL indicates the termination.
* */
-static void *
-ct_get_ptp_virt(struct ct_vm *vm, int index)
+static dma_addr_t
+ct_get_ptp_phys(struct ct_vm *vm, int index)
{
- void *addr;
+ dma_addr_t addr;
- addr = (index >= CT_PTP_NUM) ? NULL : vm->ptp[index];
+ addr = (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr;
return addr;
}
-int ct_vm_create(struct ct_vm **rvm)
+int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci)
{
struct ct_vm *vm;
struct ct_vm_block *block;
- int i;
+ int i, err = 0;
*rvm = NULL;
@@ -188,23 +188,21 @@ int ct_vm_create(struct ct_vm **rvm)
/* Allocate page table pages */
for (i = 0; i < CT_PTP_NUM; i++) {
- vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL);
- if (!vm->ptp[i])
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(pci),
+ PAGE_SIZE, &vm->ptp[i]);
+ if (err < 0)
break;
}
- if (!i) {
+ if (err < 0) {
/* no page table pages are allocated */
- kfree(vm);
+ ct_vm_destroy(vm);
return -ENOMEM;
}
vm->size = CT_ADDRS_PER_PAGE * i;
- /* Initialise remaining ptps */
- for (; i < CT_PTP_NUM; i++)
- vm->ptp[i] = NULL;
-
vm->map = ct_vm_map;
vm->unmap = ct_vm_unmap;
- vm->get_ptp_virt = ct_get_ptp_virt;
+ vm->get_ptp_phys = ct_get_ptp_phys;
INIT_LIST_HEAD(&vm->unused);
INIT_LIST_HEAD(&vm->used);
block = kzalloc(sizeof(*block), GFP_KERNEL);
@@ -242,7 +240,7 @@ void ct_vm_destroy(struct ct_vm *vm)
/* free allocated page table pages */
for (i = 0; i < CT_PTP_NUM; i++)
- kfree(vm->ptp[i]);
+ snd_dma_free_pages(&vm->ptp[i]);
vm->size = 0;
diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h
index 01e4fd0..b23adfc 100644
--- a/sound/pci/ctxfi/ctvmem.h
+++ b/sound/pci/ctxfi/ctvmem.h
@@ -22,6 +22,8 @@
#include <linux/mutex.h>
#include <linux/list.h>
+#include <linux/pci.h>
+#include <sound/memalloc.h>
/* The chip can handle the page table of 4k pages
* (emu20k1 can handle even 8k pages, but we don't use it right now)
@@ -41,7 +43,7 @@ struct snd_pcm_substream;
/* Virtual memory management object for card device */
struct ct_vm {
- void *ptp[CT_PTP_NUM]; /* Device page table pages */
+ struct snd_dma_buffer ptp[CT_PTP_NUM]; /* Device page table pages */
unsigned int size; /* Available addr space in bytes */
struct list_head unused; /* List of unused blocks */
struct list_head used; /* List of used blocks */
@@ -52,10 +54,10 @@ struct ct_vm {
int size);
/* Unmap device logical addr area. */
void (*unmap)(struct ct_vm *, struct ct_vm_block *block);
- void *(*get_ptp_virt)(struct ct_vm *vm, int index);
+ dma_addr_t (*get_ptp_phys)(struct ct_vm *vm, int index);
};
-int ct_vm_create(struct ct_vm **rvm);
+int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci);
void ct_vm_destroy(struct ct_vm *vm);
#endif /* CTVMEM_H */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 565de38..b8faa6d 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -426,6 +426,7 @@ struct azx {
/* flags */
int position_fix;
+ int poll_count;
unsigned int running :1;
unsigned int initialized :1;
unsigned int single_cmd :1;
@@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = {
#define get_azx_dev(substream) (substream->runtime->private_data)
static int azx_acquire_irq(struct azx *chip, int do_disconnect);
-
+static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
/*
* Interface for HD codec
*/
@@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
{
struct azx *chip = bus->private_data;
unsigned long timeout;
+ int do_poll = 0;
again:
timeout = jiffies + msecs_to_jiffies(1000);
for (;;) {
- if (chip->polling_mode) {
+ if (chip->polling_mode || do_poll) {
spin_lock_irq(&chip->reg_lock);
azx_update_rirb(chip);
spin_unlock_irq(&chip->reg_lock);
@@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
if (!chip->rirb.cmds[addr]) {
smp_rmb();
bus->rirb_error = 0;
+
+ if (!do_poll)
+ chip->poll_count = 0;
return chip->rirb.res[addr]; /* the last value */
}
if (time_after(jiffies, timeout))
@@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
}
}
+ if (!chip->polling_mode && chip->poll_count < 2) {
+ snd_printdd(SFX "azx_get_response timeout, "
+ "polling the codec once: last cmd=0x%08x\n",
+ chip->last_cmd[addr]);
+ do_poll = 1;
+ chip->poll_count++;
+ goto again;
+ }
+
+
if (!chip->polling_mode) {
snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
"switching to polling mode: last cmd=0x%08x\n",
@@ -2043,7 +2058,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
{
if (request_irq(chip->pci->irq, azx_interrupt,
chip->msi ? 0 : IRQF_SHARED,
- "HDA Intel", chip)) {
+ "hda_intel", chip)) {
printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
"disabling device\n", chip->pci->irq);
if (do_disconnect)
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 765d7bd..9e66f6d 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -703,11 +703,13 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho
{
unsigned char nvol;
- if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+ if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) {
nvol = 0;
- else
+ } else {
nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) /
WM_VOL_MAX;
+ nvol += 0x1b;
+ }
wm_put(ice, index, nvol);
wm_put_nocache(ice, index, 0x180 | nvol);
@@ -778,7 +780,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
for (ch = 0; ch < 2; ch++) {
unsigned int vol = ucontrol->value.integer.value[ch];
if (vol > WM_VOL_MAX)
- continue;
+ vol = WM_VOL_MAX;
vol |= spec->master[ch] & WM_VOL_MUTE;
if (vol != spec->master[ch]) {
int dac;
@@ -834,8 +836,8 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *
for (i = 0; i < voices; i++) {
unsigned int vol = ucontrol->value.integer.value[i];
if (vol > WM_VOL_MAX)
- continue;
- vol |= spec->vol[ofs+i];
+ vol = WM_VOL_MAX;
+ vol |= spec->vol[ofs+i] & WM_VOL_MUTE;
if (vol != spec->vol[ofs+i]) {
spec->vol[ofs+i] = vol;
idx = WM_DAC_ATTEN + ofs + i;
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 71b2c16..68980c1 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+ {"PCM DAC", NULL, "APLL Enable"},
{"Headphone Amplifier", NULL, "PCM DAC"},
{"Line Out", NULL, "PCM DAC"},
{"Headphone Jack", NULL, "Headphone Amplifier"},
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-01-31 13:45 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-01-31 13:45 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.33-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing only the following trivial fixes.
Thanks!
Takashi
===
Anuj Aggarwal (2):
ASoC: AIC23: Fixing writes to non-existing registers in resume function
ASoC: AM3517: ASoC driver not getting compiled
Takashi Iwai (1):
ALSA: hda - Add an ASUS mobo to MSI blacklist
---
sound/pci/hda/hda_intel.c | 1 +
sound/soc/codecs/tlv320aic23.c | 2 +-
sound/soc/omap/Makefile | 2 +-
3 files changed, 3 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ec9c348..565de38 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2332,6 +2332,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
*/
static struct snd_pci_quirk msi_black_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
+ SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */
{}
};
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a9dc5fb..da589d8 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
u16 reg;
/* Sync reg_cache with the hardware */
- for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
+ for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3db8a6c..19283e5 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-01-25 19:41 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-01-25 19:41 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.33-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Just a few trivial HD-audio and ASoC fixes there as below.
Thanks!
Takashi
===
Guennadi Liakhovetski (1):
ASoC: fix a memory-leak in wm8903
Takashi Iwai (3):
ALSA: hda - Fix parsing pin node 0x21 on ALC259
ALSA: hda - Turn on EAPD only if available for Realtek codecs
ALSA: hda - Fix HP T5735 automute
Łukasz Wojniłowicz (1):
ALSA: hda - add possibility to choose speakers configuration for 4930g
---
sound/pci/hda/patch_realtek.c | 37 ++++++++++++++++++++++---------------
sound/soc/codecs/wm8903.c | 3 ++-
2 files changed, 24 insertions(+), 16 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3f92def..da34095 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1093,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec)
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
}
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+ if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+ return;
+ if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ on ? 2 : 0);
+}
+
static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
unsigned int tmp;
@@ -1110,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case ALC_INIT_DEFAULT:
switch (codec->vendor_id) {
case 0x10ec0260:
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
- snd_hda_codec_write(codec, 0x10, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
+ set_eapd(codec, 0x0f, 1);
+ set_eapd(codec, 0x10, 1);
break;
case 0x10ec0262:
case 0x10ec0267:
case 0x10ec0268:
case 0x10ec0269:
+ case 0x10ec0270:
case 0x10ec0272:
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
case 0x10ec0862:
case 0x10ec0889:
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
- snd_hda_codec_write(codec, 0x15, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
+ set_eapd(codec, 0x14, 1);
+ set_eapd(codec, 0x15, 1);
break;
}
switch (codec->vendor_id) {
@@ -1836,10 +1843,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
static void alc889_power_eapd(struct hda_codec *codec, int power)
{
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
- snd_hda_codec_write(codec, 0x15, 0,
- AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+ set_eapd(codec, 0x14, power);
+ set_eapd(codec, 0x15, power);
}
#endif
@@ -9473,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
+ .const_channel_count = 6,
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
@@ -10377,7 +10383,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+ spec->autocfg.speaker_pins[0] = 0x14;
}
static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -11788,9 +11794,9 @@ static struct alc_config_preset alc262_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
- .unsol_event = alc_automute_amp_unsol_event,
+ .unsol_event = alc_sku_unsol_event,
.setup = alc262_hp_t5735_setup,
- .init_hook = alc_automute_amp,
+ .init_hook = alc_inithook,
},
[ALC262_HP_RP5700] = {
.mixers = { alc262_hp_rp5700_mixer },
@@ -12541,6 +12547,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
dac = 0x02;
break;
case 0x15:
+ case 0x21:
dac = 0x03;
break;
default:
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce5515e..3595bd5 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
struct i2c_client *i2c = codec->control_data;
int i;
u16 *reg_cache = codec->reg_cache;
- u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+ u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
GFP_KERNEL);
/* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
if (tmp_cache[i] != reg_cache[i])
snd_soc_write(codec, i, tmp_cache[i]);
+ kfree(tmp_cache);
} else {
dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
}
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-01-18 17:08 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-01-18 17:08 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.33-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
In addition to usual a few HD-audio fixes, changes of module_init()
to subsys_initcall() are applied for the messy dependency between the
sound subsystem and platform driver like thinkpad-acpi.
Thanks!
Takashi
===
Alex Murray (1):
ALSA: hda - Improved MacBook (Pro) 5,1 / 5,2 support
Kunal Gangakhedkar (1):
ALSA: hda - Fix mute led GPIO on HP dv-series notebooks
Takashi Iwai (4):
ALSA: hda - Fix Toshiba NB20x quirk entry
ALSA: hda - Fix missing capture mixer for ALC861/660 codecs
ALSA: hda - Fix capture on Sony VAIO with single input
ALSA: Remove warning message for invalid OSS minor ranges
Thadeu Lima de Souza Cascardo (1):
ALSA: use subsys_initcall for sound core instead of module_init
---
sound/core/sound.c | 4 +-
sound/core/sound_oss.c | 2 +-
sound/pci/hda/patch_realtek.c | 94 +++++++++++++++++++++++++++++++++++-----
sound/pci/hda/patch_sigmatel.c | 61 +++++++++++++++++++-------
sound/sound_core.c | 2 +-
5 files changed, 132 insertions(+), 31 deletions(-)
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 7872a02..563d196 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void)
unregister_chrdev(major, "alsa");
}
-module_init(alsa_sound_init)
-module_exit(alsa_sound_exit)
+subsys_initcall(alsa_sound_init);
+module_exit(alsa_sound_exit);
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 7fe1226..0c164e5 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
default:
return -EINVAL;
}
- if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+ if (minor < 0 || minor >= SNDRV_OSS_MINORS)
return -EINVAL;
return minor;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e3caa78..3f92def 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1230,6 +1230,8 @@ static void alc_init_auto_mic(struct hda_codec *codec)
return; /* invalid entry */
}
}
+ if (!ext || !fixed)
+ return;
if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
return; /* no unsol support */
snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
@@ -4812,6 +4814,49 @@ static void fixup_automic_adc(struct hda_codec *codec)
spec->auto_mic = 0; /* disable auto-mic to be sure */
}
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+ int i;
+
+ /* search for the input pin; there must be only one */
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (spec->autocfg.input_pins[i]) {
+ pin = spec->autocfg.input_pins[i];
+ break;
+ }
+ }
+ if (!pin)
+ return;
+
+ /* set the default connection to that pin */
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ hda_nid_t cap = spec->capsrc_nids ?
+ spec->capsrc_nids[i] : spec->adc_nids[i];
+ int idx;
+
+ idx = get_connection_index(codec, cap, pin);
+ if (idx < 0)
+ continue;
+ /* use only this ADC */
+ if (spec->capsrc_nids)
+ spec->capsrc_nids += i;
+ spec->adc_nids += i;
+ spec->num_adc_nids = 1;
+ /* select or unmute this route */
+ if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+ snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+ HDA_AMP_MUTE, 0);
+ } else {
+ snd_hda_codec_write_cache(codec, cap, 0,
+ AC_VERB_SET_CONNECT_SEL, idx);
+ }
+ return;
+ }
+}
+
static void set_capture_mixer(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -4824,14 +4869,15 @@ static void set_capture_mixer(struct hda_codec *codec)
alc_capture_mixer3 },
};
if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
- int mux;
- if (spec->auto_mic) {
- mux = 0;
+ int mux = 0;
+ if (spec->auto_mic)
fixup_automic_adc(codec);
- } else if (spec->input_mux && spec->input_mux->num_items > 1)
- mux = 1;
- else
- mux = 0;
+ else if (spec->input_mux) {
+ if (spec->input_mux->num_items > 1)
+ mux = 1;
+ else if (spec->input_mux->num_items == 1)
+ fixup_single_adc(codec);
+ }
spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
}
}
@@ -7094,8 +7140,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7496,6 +7542,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -7680,6 +7727,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
}
+static void alc885_mb5_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+
+}
+
+static void alc885_mb5_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Headphone insertion or removal. */
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc885_mb5_automute(codec);
+}
+
static void alc885_imac91_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -9126,6 +9194,8 @@ static struct alc_config_preset alc882_presets[] = {
.input_mux = &mb5_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc885_mb5_unsol_event,
+ .init_hook = alc885_mb5_automute,
},
[ALC885_MACPRO] = {
.mixers = { alc882_macpro_mixer },
@@ -11179,7 +11249,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
}
#define alc262_auto_create_input_ctls \
- alc880_auto_create_input_ctls
+ alc882_auto_create_input_ctls
/*
* generic initialization of ADC, input mixers and output mixers
@@ -14855,6 +14925,8 @@ static int patch_alc861(struct hda_codec *codec)
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
+ if (!spec->cap_mixer)
+ set_capture_mixer(codec);
set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
spec->vmaster_nid = 0x03;
@@ -17251,7 +17323,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
ALC662_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
+ SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
ALC662_3ST_6ch_DIG),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2291a83..799ba25 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4730,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
}
}
+static int hp_blike_system(u32 subsystem_id);
+
+static void set_hp_led_gpio(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ switch (codec->vendor_id) {
+ case 0x111d7608:
+ /* GPIO 0 */
+ spec->gpio_led = 0x01;
+ break;
+ case 0x111d7600:
+ case 0x111d7601:
+ case 0x111d7602:
+ case 0x111d7603:
+ /* GPIO 3 */
+ spec->gpio_led = 0x08;
+ break;
+ }
+}
+
/*
* This method searches for the mute LED GPIO configuration
* provided as OEM string in SMBIOS. The format of that string
@@ -4741,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
*
* So, HP B-series like systems may have HP_Mute_LED_0 (current models)
* or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings
+ *
+ *
+ * The dv-series laptops don't seem to have the HP_Mute_LED* strings in
+ * SMBIOS - at least the ones I have seen do not have them - which include
+ * my own system (HP Pavilion dv6-1110ax) and my cousin's
+ * HP Pavilion dv9500t CTO.
+ * Need more information on whether it is true across the entire series.
+ * -- kunal
*/
static int find_mute_led_gpio(struct hda_codec *codec)
{
@@ -4751,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec)
while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
NULL, dev))) {
if (sscanf(dev->name, "HP_Mute_LED_%d_%d",
- &spec->gpio_led_polarity,
- &spec->gpio_led) == 2) {
+ &spec->gpio_led_polarity,
+ &spec->gpio_led) == 2) {
spec->gpio_led = 1 << spec->gpio_led;
return 1;
}
if (sscanf(dev->name, "HP_Mute_LED_%d",
- &spec->gpio_led_polarity) == 1) {
- switch (codec->vendor_id) {
- case 0x111d7608:
- /* GPIO 0 */
- spec->gpio_led = 0x01;
- return 1;
- case 0x111d7600:
- case 0x111d7601:
- case 0x111d7602:
- case 0x111d7603:
- /* GPIO 3 */
- spec->gpio_led = 0x08;
- return 1;
- }
+ &spec->gpio_led_polarity) == 1) {
+ set_hp_led_gpio(codec);
+ return 1;
}
}
+
+ /*
+ * Fallback case - if we don't find the DMI strings,
+ * we statically set the GPIO - if not a B-series system.
+ */
+ if (!hp_blike_system(codec->subsystem_id)) {
+ set_hp_led_gpio(codec);
+ spec->gpio_led_polarity = 1;
+ return 1;
+ }
}
return 0;
}
@@ -5548,6 +5575,8 @@ again:
spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
+ snd_printdd("Found board config: %d\n", spec->board_config);
+
switch (spec->board_config) {
case STAC_HP_M4:
/* enable internal microphone */
diff --git a/sound/sound_core.c b/sound/sound_core.c
index dbca7c9..7c2d677 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void)
class_destroy(sound_class);
}
-module_init(init_soundcore);
+subsys_initcall(init_soundcore);
module_exit(cleanup_soundcore);
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2010-01-12 17:05 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2010-01-12 17:05 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.33-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Most of them are trivial one or few liners.
Thanks!
Takashi
===
Dan Carpenter (1):
sound: oss: off by one bug
Daniel T Chen (2):
ALSA: atiixp: Specify codec for Foxconn RC4107MA-RS2
ALSA: ac97: Add Dell Dimension 2400 to Headphone/Line Jack Sense blacklist
Krzysztof Helt (2):
sbawe: fix memory detection part 2
ALSA: ac97: add AC97 STMicroelectronics' codecs
Mark Brown (1):
ASoC: Fix WM8350 DSP mode B configuration
Roel Kluin (1):
ALSA: test off by one in setsamplerate()
Takashi Iwai (2):
ALSA: usb-audio - Avoid Oops after disconnect
ALSA: hda - Fix ALC861-VD capture source mixer
---
sound/isa/sb/emu8000.c | 11 +++++++----
sound/oss/dev_table.c | 16 +++++++++-------
sound/oss/sound_config.h | 2 ++
sound/oss/soundcard.c | 4 ++--
sound/pci/ac97/ac97_codec.c | 10 ++++++++++
sound/pci/ac97/ac97_id.h | 2 ++
sound/pci/ac97/ac97_patch.c | 1 +
sound/pci/atiixp.c | 1 +
sound/pci/hda/patch_realtek.c | 2 +-
sound/pci/riptide/riptide.c | 2 +-
sound/soc/codecs/wm8350.c | 2 +-
sound/usb/usbaudio.c | 2 +-
12 files changed, 38 insertions(+), 17 deletions(-)
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 751762f..0c40951 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -377,12 +377,13 @@ init_arrays(struct snd_emu8000 *emu)
static void __devinit
size_dram(struct snd_emu8000 *emu)
{
- int i, size;
+ int i, size, detected_size;
if (emu->dram_checked)
return;
size = 0;
+ detected_size = 0;
/* write out a magic number */
snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE);
@@ -393,6 +394,8 @@ size_dram(struct snd_emu8000 *emu)
while (size < EMU8000_MAX_DRAM) {
+ size += 512 * 1024; /* increment 512kbytes */
+
/* Write a unique data on the test address.
* if the address is out of range, the data is written on
* 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is
@@ -414,7 +417,7 @@ size_dram(struct snd_emu8000 *emu)
if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2)
break; /* no memory at this address */
- size += 512 * 1024; /* increment 512kbytes */
+ detected_size = size;
snd_emu8000_read_wait(emu);
@@ -442,9 +445,9 @@ size_dram(struct snd_emu8000 *emu)
snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE);
snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n",
- emu->port1, size/1024);
+ emu->port1, detected_size/1024);
- emu->mem_size = size;
+ emu->mem_size = detected_size;
emu->dram_checked = 1;
}
diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c
index 08274c9..727bdb9 100644
--- a/sound/oss/dev_table.c
+++ b/sound/oss/dev_table.c
@@ -67,14 +67,15 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
return -(EBUSY);
}
d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver)));
-
- if (sound_nblocks < 1024)
- sound_nblocks++;
+ sound_nblocks++;
+ if (sound_nblocks >= MAX_MEM_BLOCKS)
+ sound_nblocks = MAX_MEM_BLOCKS - 1;
op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations)));
+ sound_nblocks++;
+ if (sound_nblocks >= MAX_MEM_BLOCKS)
+ sound_nblocks = MAX_MEM_BLOCKS - 1;
- if (sound_nblocks < 1024)
- sound_nblocks++;
if (d == NULL || op == NULL) {
printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name);
sound_unload_audiodev(num);
@@ -128,9 +129,10 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
until you unload sound! */
op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations)));
+ sound_nblocks++;
+ if (sound_nblocks >= MAX_MEM_BLOCKS)
+ sound_nblocks = MAX_MEM_BLOCKS - 1;
- if (sound_nblocks < 1024)
- sound_nblocks++;
if (op == NULL) {
printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
return -ENOMEM;
diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h
index 55271fb..9d35c4c 100644
--- a/sound/oss/sound_config.h
+++ b/sound/oss/sound_config.h
@@ -142,4 +142,6 @@ static inline int translate_mode(struct file *file)
#define TIMER_ARMED 121234
#define TIMER_NOT_ARMED 1
+#define MAX_MEM_BLOCKS 1024
+
#endif
diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c
index 61aaeda..c625309 100644
--- a/sound/oss/soundcard.c
+++ b/sound/oss/soundcard.c
@@ -56,7 +56,7 @@
/*
* Table for permanently allocated memory (used when unloading the module)
*/
-void * sound_mem_blocks[1024];
+void * sound_mem_blocks[MAX_MEM_BLOCKS];
int sound_nblocks = 0;
/* Persistent DMA buffers */
@@ -574,7 +574,7 @@ static int __init oss_init(void)
NULL, "%s%d", dev_list[i].name, j);
}
- if (sound_nblocks >= 1024)
+ if (sound_nblocks >= MAX_MEM_BLOCKS - 1)
printk(KERN_ERR "Sound warning: Deallocation table was too small.\n");
return 0;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index c119206..a7630e9 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -83,6 +83,7 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = {
{ 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL },
{ 0x50534300, 0xffffff00, "Philips", NULL, NULL },
{ 0x53494c00, 0xffffff00, "Silicon Laboratory", NULL, NULL },
+{ 0x53544d00, 0xffffff00, "STMicroelectronics", NULL, NULL },
{ 0x54524100, 0xffffff00, "TriTech", NULL, NULL },
{ 0x54584e00, 0xffffff00, "Texas Instruments", NULL, NULL },
{ 0x56494100, 0xffffff00, "VIA Technologies", NULL, NULL },
@@ -161,6 +162,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
{ 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix
{ 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL },
{ 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH },
+{ 0x53544d02, 0xffffffff, "ST7597", NULL, NULL },
{ 0x54524102, 0xffffffff, "TR28022", NULL, NULL },
{ 0x54524103, 0xffffffff, "TR28023", NULL, NULL },
{ 0x54524106, 0xffffffff, "TR28026", NULL, NULL },
@@ -213,6 +215,14 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg)
{
/* filter some registers for buggy codecs */
switch (ac97->id) {
+ case AC97_ID_ST_AC97_ID4:
+ if (reg == 0x08)
+ return 0;
+ /* fall through */
+ case AC97_ID_ST7597:
+ if (reg == 0x22 || reg == 0x7a)
+ return 1;
+ /* fall through */
case AC97_ID_AK4540:
case AC97_ID_AK4542:
if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c)
diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h
index c129492..d603147 100644
--- a/sound/pci/ac97/ac97_id.h
+++ b/sound/pci/ac97/ac97_id.h
@@ -62,3 +62,5 @@
#define AC97_ID_CM9761_78 0x434d4978
#define AC97_ID_CM9761_82 0x434d4982
#define AC97_ID_CM9761_83 0x434d4983
+#define AC97_ID_ST7597 0x53544d02
+#define AC97_ID_ST_AC97_ID4 0x53544d04
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 139cf3b..d9266ba 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1870,6 +1870,7 @@ static unsigned int ad1981_jacks_blacklist[] = {
0x10140554, /* Thinkpad T42p/R50p */
0x10140567, /* Thinkpad T43p 2668-G7U */
0x10140581, /* Thinkpad X41-2527 */
+ 0x10280160, /* Dell Dimension 2400 */
0x104380b0, /* Asus A7V8X-MX */
0x11790241, /* Toshiba Satellite A-15 S127 */
0x144dc01a, /* Samsung NP-X20C004/SEG */
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index d6752df..42b4fbb 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -297,6 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = {
MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
static struct snd_pci_quirk atiixp_quirks[] __devinitdata = {
+ SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0),
SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0),
{ } /* terminator */
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c746505..e3caa78 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -15493,7 +15493,7 @@ static struct alc_config_preset alc861vd_presets[] = {
static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+ return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0);
}
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index b5ca02e..e66ef2b 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate)
rptr.retwords[2] != M &&
rptr.retwords[3] != N &&
i++ < MAX_WRITE_RETRY);
- if (i == MAX_WRITE_RETRY) {
+ if (i > MAX_WRITE_RETRY) {
snd_printdd("sent samplerate %d: %d failed\n",
*intdec, rate);
return -EIO;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ebbf11b..718ef91 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
iface |= 0x3 << 8;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x3 << 8; /* lg not sure which mode */
+ iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
break;
default:
return -EINVAL;
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 4963def..9edef46 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -1936,7 +1936,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
struct snd_usb_substream *subs = &as->substream[direction];
- if (subs->interface >= 0) {
+ if (!as->chip->shutdown && subs->interface >= 0) {
usb_set_interface(subs->dev, subs->interface, 0);
subs->interface = -1;
}
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-12-27 13:03 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-12-27 13:03 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.33-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing a few regression fixes and trivial patches.
Thanks!
Takashi
===
Guennadi Liakhovetski (1):
ALSA: Fix indentation in pcm_native.c
Masanari Iida (1):
ALSA: Fix a typo in Procfile.txt
Peter Huewe (1):
ALSA: sound/arm: Fix build failure caused by missing struct aaci definition
Takashi Iwai (4):
ALSA: hda - Disable tigger at pin-sensing on AD codecs
ALSA: hda - use snd_hda_jack_detect() again in patch_sigmatel.c
ALSA: hda - Don't cache beep controls
ALSA: hda - Fix Oops at reloading beep devices
Wu Fengguang (1):
ALSA: hda - HDMI sticky stream tag support
---
Documentation/sound/alsa/Procfile.txt | 2 +-
sound/arm/aaci.c | 3 +--
sound/core/pcm_native.c | 8 ++++----
sound/pci/hda/hda_beep.c | 16 ++++++++++------
sound/pci/hda/hda_codec.c | 10 ++++++----
sound/pci/hda/hda_codec.h | 1 +
sound/pci/hda/hda_intel.c | 21 +++++++++++++++------
sound/pci/hda/patch_analog.c | 16 ++++++++++++++++
sound/pci/hda/patch_sigmatel.c | 18 ++++++++++--------
9 files changed, 64 insertions(+), 31 deletions(-)
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index 719a819..07301de 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -95,7 +95,7 @@ card*/pcm*/xrun_debug
It takes an integer value, can be changed by writing to this
file, such as
- # cat 5 > /proc/asound/card0/pcm0p/xrun_debug
+ # echo 5 > /proc/asound/card0/pcm0p/xrun_debug
The value consists of the following bit flags:
bit 0 = Enable XRUN/jiffies debug messages
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index c569986..656e474 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -441,6 +441,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
int err;
+ struct aaci *aaci = substream->private_data;
aaci_pcm_hw_free(substream);
if (aacirun->pcm_open) {
@@ -560,7 +561,6 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream)
static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
unsigned int channels = params_channels(params);
int ret;
@@ -659,7 +659,6 @@ static struct snd_pcm_ops aaci_playback_ops = {
static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
int ret;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 29ab46a..25b0641 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1918,13 +1918,13 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
hw->rate_min, hw->rate_max);
- if (err < 0)
- return err;
+ if (err < 0)
+ return err;
err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
hw->period_bytes_min, hw->period_bytes_max);
- if (err < 0)
- return err;
+ if (err < 0)
+ return err;
err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS,
hw->periods_min, hw->periods_max);
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 5fe34a8..e4581a4 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -42,7 +42,7 @@ static void snd_hda_generate_beep(struct work_struct *work)
return;
/* generate tone */
- snd_hda_codec_write_cache(codec, beep->nid, 0,
+ snd_hda_codec_write(codec, beep->nid, 0,
AC_VERB_SET_BEEP_CONTROL, beep->tone);
}
@@ -119,7 +119,7 @@ static void snd_hda_do_detach(struct hda_beep *beep)
beep->dev = NULL;
cancel_work_sync(&beep->beep_work);
/* turn off beep for sure */
- snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+ snd_hda_codec_write(beep->codec, beep->nid, 0,
AC_VERB_SET_BEEP_CONTROL, 0);
}
@@ -192,7 +192,7 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
beep->enabled = enable;
if (!enable) {
/* turn off beep */
- snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+ snd_hda_codec_write(beep->codec, beep->nid, 0,
AC_VERB_SET_BEEP_CONTROL, 0);
}
if (beep->mode == HDA_BEEP_MODE_SWREG) {
@@ -239,8 +239,12 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
mutex_init(&beep->mutex);
if (beep->mode == HDA_BEEP_MODE_ON) {
- beep->enabled = 1;
- snd_hda_do_register(&beep->register_work);
+ int err = snd_hda_do_attach(beep);
+ if (err < 0) {
+ kfree(beep);
+ codec->beep = NULL;
+ return err;
+ }
}
return 0;
@@ -253,7 +257,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
if (beep) {
cancel_work_sync(&beep->register_work);
cancel_delayed_work(&beep->unregister_work);
- if (beep->enabled)
+ if (beep->dev)
snd_hda_do_detach(beep);
codec->beep = NULL;
kfree(beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 950ee5c..f98b47c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1327,11 +1327,13 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
*/
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
{
- u32 pincap = snd_hda_query_pin_caps(codec, nid);
-
- if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
- snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ u32 pincap;
+ if (!codec->no_trigger_sense) {
+ pincap = snd_hda_query_pin_caps(codec, nid);
+ if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ }
return snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 1d541b7..0a770a2 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -817,6 +817,7 @@ struct hda_codec {
unsigned int pin_amp_workaround:1; /* pin out-amp takes index
* (e.g. Conexant codecs)
*/
+ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ff8ad46..ec9c348 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -356,6 +356,7 @@ struct azx_dev {
*/
unsigned char stream_tag; /* assigned stream */
unsigned char index; /* stream index */
+ int device; /* last device number assigned to */
unsigned int opened :1;
unsigned int running :1;
@@ -1441,10 +1442,13 @@ static int __devinit azx_codec_configure(struct azx *chip)
*/
/* assign a stream for the PCM */
-static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
+static inline struct azx_dev *
+azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
{
int dev, i, nums;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ struct azx_dev *res = NULL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dev = chip->playback_index_offset;
nums = chip->playback_streams;
} else {
@@ -1453,10 +1457,15 @@ static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
}
for (i = 0; i < nums; i++, dev++)
if (!chip->azx_dev[dev].opened) {
- chip->azx_dev[dev].opened = 1;
- return &chip->azx_dev[dev];
+ res = &chip->azx_dev[dev];
+ if (res->device == substream->pcm->device)
+ break;
}
- return NULL;
+ if (res) {
+ res->opened = 1;
+ res->device = substream->pcm->device;
+ }
+ return res;
}
/* release the assigned stream */
@@ -1505,7 +1514,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
int err;
mutex_lock(&chip->open_mutex);
- azx_dev = azx_assign_device(chip, substream->stream);
+ azx_dev = azx_assign_device(chip, substream);
if (azx_dev == NULL) {
mutex_unlock(&chip->open_mutex);
return -EBUSY;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1a36137..69a941c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1186,6 +1186,8 @@ static int patch_ad1986a(struct hda_codec *codec)
*/
spec->multiout.no_share_stream = 1;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -1371,6 +1373,8 @@ static int patch_ad1983(struct hda_codec *codec)
codec->patch_ops = ad198x_patch_ops;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -1813,6 +1817,9 @@ static int patch_ad1981(struct hda_codec *codec)
codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
break;
}
+
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -3118,6 +3125,8 @@ static int patch_ad1988(struct hda_codec *codec)
#endif
spec->vmaster_nid = 0x04;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -3330,6 +3339,8 @@ static int patch_ad1884(struct hda_codec *codec)
codec->patch_ops = ad198x_patch_ops;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -4287,6 +4298,8 @@ static int patch_ad1884a(struct hda_codec *codec)
break;
}
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -4623,6 +4636,9 @@ static int patch_ad1882(struct hda_codec *codec)
spec->mixers[2] = ad1882_6stack_mixers;
break;
}
+
+ codec->no_trigger_sense = 1;
+
return 0;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index eeda7be..2291a83 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4453,14 +4453,7 @@ static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
{
if (!nid)
return 0;
- /* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT
- * codecs behave wrongly when SET_PIN_SENSE is triggered, although
- * the pincap gives TRIG_REQ bit.
- */
- if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE)
- return 1;
- return 0;
+ return snd_hda_jack_detect(codec, nid);
}
static void stac92xx_line_out_detect(struct hda_codec *codec,
@@ -4962,6 +4955,7 @@ static int patch_stac9200(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac9200_pin_nids);
spec->pin_nids = stac9200_pin_nids;
@@ -5024,6 +5018,7 @@ static int patch_stac925x(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac925x_pin_nids);
spec->pin_nids = stac925x_pin_nids;
@@ -5108,6 +5103,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->slave_dig_outs = stac92hd73xx_slave_dig_outs;
spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids);
@@ -5255,6 +5251,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
spec->digbeep_nid = 0x21;
@@ -5418,6 +5415,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->patch_ops = stac92xx_patch_ops;
spec->num_pins = STAC92HD71BXX_NUM_PINS;
@@ -5661,6 +5659,7 @@ static int patch_stac922x(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac922x_pin_nids);
spec->pin_nids = stac922x_pin_nids;
@@ -5764,6 +5763,7 @@ static int patch_stac927x(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->slave_dig_outs = stac927x_slave_dig_outs;
spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
@@ -5898,6 +5898,7 @@ static int patch_stac9205(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac9205_pin_nids);
spec->pin_nids = stac9205_pin_nids;
@@ -6053,6 +6054,7 @@ static int patch_stac9872(struct hda_codec *codec)
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
spec->pin_nids = stac9872_pin_nids;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-12-23 17:54 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-12-23 17:54 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull another sound fixes for v2.6.33-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All fixes below are mostly trivial and individual.
Thanks!
Takashi
===
Anisse Astier (1):
ALSA: hda - Add STAC9205 PCI_QUIRK for Dell Vostro 1700
Eric Millbrandt (1):
ASoC: Do not write to invalid registers on the wm9712.
Florian Fainelli (1):
ALSA: sound/core/pcm_timer.c: use lib/gcd.c
Guennadi Liakhovetski (1):
ASoC: add missing parameter to mx27vis_hifi_hw_free()
Rafael Avila de Espindola (1):
ALSA: hda - Add support for the new 27 inch IMacs
Takashi Iwai (3):
ALSA: hda - Fix NULL dereference with enable_beep=0 option
ALSA: hda - Add MSI blacklist
ALSA: hda - Set mixer name after codec patch
Uwe Kleine-König (1):
ASoC: sh: FSI:: don't check platform_get_irq's return value against zero
---
Documentation/sound/alsa/HD-Audio-Models.txt | 1 +
sound/core/Kconfig | 1 +
sound/core/pcm_timer.c | 17 +----------------
sound/pci/hda/hda_codec.c | 10 +++++-----
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_cirrus.c | 22 +++++++++++++++++++++-
sound/pci/hda/patch_sigmatel.c | 22 +++++++++++++---------
sound/soc/codecs/wm9712.c | 3 ++-
sound/soc/imx/mx27vis_wm8974.c | 3 ++-
sound/soc/sh/fsi.c | 2 +-
10 files changed, 48 insertions(+), 34 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index e93afff..e72cee9 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -403,4 +403,5 @@ STAC9872
Cirrus Logic CS4206/4207
========================
mbp55 MacBook Pro 5,5
+ imac27 IMac 27 Inch
auto BIOS setup (default)
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index c15682a..475455c 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -5,6 +5,7 @@ config SND_TIMER
config SND_PCM
tristate
select SND_TIMER
+ select GCD
config SND_HWDEP
tristate
diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c
index ca8068b..b01d948 100644
--- a/sound/core/pcm_timer.c
+++ b/sound/core/pcm_timer.c
@@ -20,6 +20,7 @@
*/
#include <linux/time.h>
+#include <linux/gcd.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/timer.h>
@@ -28,22 +29,6 @@
* Timer functions
*/
-/* Greatest common divisor */
-static unsigned long gcd(unsigned long a, unsigned long b)
-{
- unsigned long r;
- if (a < b) {
- r = a;
- a = b;
- b = r;
- }
- while ((r = a % b) != 0) {
- a = b;
- b = r;
- }
- return b;
-}
-
void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream)
{
unsigned long rate, mult, fsize, l, post;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 9cfdb77..950ee5c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1086,11 +1086,6 @@ int snd_hda_codec_configure(struct hda_codec *codec)
if (err < 0)
return err;
}
- /* audio codec should override the mixer name */
- if (codec->afg || !*codec->bus->card->mixername)
- snprintf(codec->bus->card->mixername,
- sizeof(codec->bus->card->mixername),
- "%s %s", codec->vendor_name, codec->chip_name);
if (is_generic_config(codec)) {
err = snd_hda_parse_generic_codec(codec);
@@ -1109,6 +1104,11 @@ int snd_hda_codec_configure(struct hda_codec *codec)
patched:
if (!err && codec->patch_ops.unsol_event)
err = init_unsol_queue(codec->bus);
+ /* audio codec should override the mixer name */
+ if (!err && (codec->afg || !*codec->bus->card->mixername))
+ snprintf(codec->bus->card->mixername,
+ sizeof(codec->bus->card->mixername),
+ "%s %s", codec->vendor_name, codec->chip_name);
return err;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9b56f93..ff8ad46 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2322,6 +2322,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
* white/black-list for enable_msi
*/
static struct snd_pci_quirk msi_black_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
{}
};
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 4b200da..fe0423c 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -66,6 +66,7 @@ struct cs_spec {
/* available models */
enum {
CS420X_MBP55,
+ CS420X_IMAC27,
CS420X_AUTO,
CS420X_MODELS
};
@@ -827,7 +828,8 @@ static void cs_automute(struct hda_codec *codec)
AC_VERB_SET_PIN_WIDGET_CONTROL,
hp_present ? 0 : PIN_OUT);
}
- if (spec->board_config == CS420X_MBP55) {
+ if (spec->board_config == CS420X_MBP55 ||
+ spec->board_config == CS420X_IMAC27) {
unsigned int gpio = hp_present ? 0x02 : 0x08;
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, gpio);
@@ -1069,12 +1071,14 @@ static int cs_parse_auto_config(struct hda_codec *codec)
static const char *cs420x_models[CS420X_MODELS] = {
[CS420X_MBP55] = "mbp55",
+ [CS420X_IMAC27] = "imac27",
[CS420X_AUTO] = "auto",
};
static struct snd_pci_quirk cs420x_cfg_tbl[] = {
SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
+ SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),
{} /* terminator */
};
@@ -1097,8 +1101,23 @@ static struct cs_pincfg mbp55_pincfgs[] = {
{} /* terminator */
};
+static struct cs_pincfg imac27_pincfgs[] = {
+ { 0x09, 0x012b4050 },
+ { 0x0a, 0x90100140 },
+ { 0x0b, 0x90100142 },
+ { 0x0c, 0x018b3020 },
+ { 0x0d, 0x90a00110 },
+ { 0x0e, 0x400000f0 },
+ { 0x0f, 0x01cbe030 },
+ { 0x10, 0x014be060 },
+ { 0x12, 0x01ab9070 },
+ { 0x15, 0x400000f0 },
+ {} /* terminator */
+};
+
static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
[CS420X_MBP55] = mbp55_pincfgs,
+ [CS420X_IMAC27] = imac27_pincfgs,
};
static void fix_pincfg(struct hda_codec *codec, int model)
@@ -1128,6 +1147,7 @@ static int patch_cs420x(struct hda_codec *codec)
fix_pincfg(codec, spec->board_config);
switch (spec->board_config) {
+ case CS420X_IMAC27:
case CS420X_MBP55:
/* GPIO1 = headphones */
/* GPIO3 = speakers */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3d59f83..eeda7be 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2104,6 +2104,7 @@ static unsigned int ref9205_pin_configs[12] = {
10280204
1028021F
10280228 (Dell Vostro 1500)
+ 10280229 (Dell Vostro 1700)
*/
static unsigned int dell_9205_m42_pin_configs[12] = {
0x0321101F, 0x03A11020, 0x400003FA, 0x90170310,
@@ -2189,6 +2190,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
"Dell Inspiron", STAC_9205_DELL_M44),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
"Dell Vostro 1500", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229,
+ "Dell Vostro 1700", STAC_9205_DELL_M42),
/* Gateway */
SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
@@ -3779,15 +3782,16 @@ 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;
- /* IDT/STAC codecs have linear beep tone parameter */
- codec->beep->linear_tone = 1;
- /* 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;
+ if (codec->beep) {
+ /* IDT/STAC codecs have linear beep tone parameter */
+ codec->beep->linear_tone = 1;
+ /* if no beep switch is available, make its own one */
+ caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+ if (!(caps & AC_AMPCAP_MUTE)) {
+ err = stac92xx_beep_switch_ctl(codec);
+ if (err < 0)
+ return err;
+ }
}
}
#endif
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 0ac1215..e237bf6 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -463,7 +463,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
{
u16 *cache = codec->reg_cache;
- soc_ac97_ops.write(codec->ac97, reg, val);
+ if (reg < 0x7c)
+ soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9712_reg)))
cache[reg] = val;
diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c
index 0267d2d..07d2a24 100644
--- a/sound/soc/imx/mx27vis_wm8974.c
+++ b/sound/soc/imx/mx27vis_wm8974.c
@@ -180,7 +180,8 @@ static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream)
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
/* disable the PLL */
- return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0);
+ return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG,
+ 0, 0);
}
/*
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 9c49c11..42813b8 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -876,7 +876,7 @@ static int fsi_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
irq = platform_get_irq(pdev, 0);
- if (!res || !irq) {
+ if (!res || (int)irq <= 0) {
dev_err(&pdev->dev, "Not enough FSI platform resources.\n");
ret = -ENODEV;
goto exit;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-12-21 16:09 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-12-21 16:09 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.33-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following fixes. Most of them are small individual
fixes. A big chunk is for supporting a few new Realtek codec chips,
which are more or less compatible with older ones.
Thanks!
Takashi
===
Clemens Ladisch (1):
sound: sgio2audio/pdaudiocf/usb-audio: initialize PCM buffer
Daniel T Chen (1):
ALSA: hda: Set Front Mic to input vref 50% for Lenovo 3000 Y410
Einar Rünkaru (2):
ALSA: hda - Fixed internal mic initialization for Dell Vostro 1015
ALSA: hda - Make use of beep device found in Dell Vostro 1015n
Guennadi Liakhovetski (1):
ASoC: wm8974: fix a wrong bit definition
Hector Martin (3):
ALSA: HDA: simplify Aspire 8930G verb array
ALSA: HDA: remove useless mixers on Aspire 8930G
ALSA: HDA: add powersaving hook for Realtek
Jaroslav Kysela (1):
ALSA: hda/realtek: Remove extra .capsrc_nids initialization for ALC889_INTEL
Jon Smirl (1):
ASoC: Fix disable of SPDIF on STAC9766 codec
Julia Lawall (1):
ALSA: Use kzalloc for allocating only one thing
Kailang Yang (1):
ALSA: hda - More ALC663 fixes and support of compatible chips
Krzysztof Helt (2):
ALSA: fix incorrect rounding direction in snd_interval_ratnum()
ALSA: sbawe: fix memory detection
Kuninori Morimoto (1):
ASoC: ak4642: Add default return value in ak4642_modinit
Roel Kluin (1):
sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot()
Russell King (5):
ALSA: AACI: simplify codec rate information
ALSA: AACI: cleanup aaci_pcm_hw_params
ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params
ALSA: AACI: add double-rate support
ALSA: AACI: switch to per-pcm locking
Takashi Iwai (3):
ALSA: hda - Fix missing capsrc_nids for ALC88x
ALSA: hda - Fix quirk for Maxdata obook4-1
ALSA: aaci - Fix a typo
---
sound/arm/aaci.c | 177 +++++----------
sound/arm/aaci.h | 2 +-
sound/core/pcm_lib.c | 4 +-
sound/isa/msnd/msnd_midi.c | 2 +-
sound/isa/sb/emu8000.c | 6 +-
sound/mips/sgio2audio.c | 2 +-
sound/oss/pss.c | 6 +-
sound/pci/hda/patch_conexant.c | 43 +++-
sound/pci/hda/patch_realtek.c | 387 +++++++++++++++++++++++++++++---
sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 2 +-
sound/soc/codecs/ak4642.c | 2 +-
sound/soc/codecs/stac9766.c | 18 +--
sound/soc/codecs/wm8974.c | 2 +-
sound/usb/usbaudio.c | 2 +-
14 files changed, 467 insertions(+), 188 deletions(-)
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1497dce..c569986 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -172,14 +172,15 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
return v;
}
-static inline void aaci_chan_wait_ready(struct aaci_runtime *aacirun)
+static inline void
+aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask)
{
u32 val;
int timeout = 5000;
do {
val = readl(aacirun->base + AACI_SR);
- } while (val & (SR_TXB|SR_RXB) && timeout--);
+ } while (val & mask && timeout--);
}
@@ -208,8 +209,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
writel(0, aacirun->base + AACI_IE);
return;
}
- ptr = aacirun->ptr;
+ spin_lock(&aacirun->lock);
+
+ ptr = aacirun->ptr;
do {
unsigned int len = aacirun->fifosz;
u32 val;
@@ -217,9 +220,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
if (aacirun->bytes <= 0) {
aacirun->bytes += aacirun->period;
aacirun->ptr = ptr;
- spin_unlock(&aaci->lock);
+ spin_unlock(&aacirun->lock);
snd_pcm_period_elapsed(aacirun->substream);
- spin_lock(&aaci->lock);
+ spin_lock(&aacirun->lock);
}
if (!(aacirun->cr & CR_EN))
break;
@@ -245,7 +248,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
ptr = aacirun->start;
}
} while(1);
+
aacirun->ptr = ptr;
+
+ spin_unlock(&aacirun->lock);
}
if (mask & ISR_URINTR) {
@@ -263,6 +269,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
return;
}
+ spin_lock(&aacirun->lock);
+
ptr = aacirun->ptr;
do {
unsigned int len = aacirun->fifosz;
@@ -271,9 +279,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
if (aacirun->bytes <= 0) {
aacirun->bytes += aacirun->period;
aacirun->ptr = ptr;
- spin_unlock(&aaci->lock);
+ spin_unlock(&aacirun->lock);
snd_pcm_period_elapsed(aacirun->substream);
- spin_lock(&aaci->lock);
+ spin_lock(&aacirun->lock);
}
if (!(aacirun->cr & CR_EN))
break;
@@ -301,6 +309,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
} while (1);
aacirun->ptr = ptr;
+
+ spin_unlock(&aacirun->lock);
}
}
@@ -310,7 +320,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
u32 mask;
int i;
- spin_lock(&aaci->lock);
mask = readl(aaci->base + AACI_ALLINTS);
if (mask) {
u32 m = mask;
@@ -320,7 +329,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
}
}
}
- spin_unlock(&aaci->lock);
return mask ? IRQ_HANDLED : IRQ_NONE;
}
@@ -330,63 +338,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
/*
* ALSA support.
*/
-
-struct aaci_stream {
- unsigned char codec_idx;
- unsigned char rate_idx;
-};
-
-static struct aaci_stream aaci_streams[] = {
- [ACSTREAM_FRONT] = {
- .codec_idx = 0,
- .rate_idx = AC97_RATES_FRONT_DAC,
- },
- [ACSTREAM_SURROUND] = {
- .codec_idx = 0,
- .rate_idx = AC97_RATES_SURR_DAC,
- },
- [ACSTREAM_LFE] = {
- .codec_idx = 0,
- .rate_idx = AC97_RATES_LFE_DAC,
- },
-};
-
-static inline unsigned int aaci_rate_mask(struct aaci *aaci, int streamid)
-{
- struct aaci_stream *s = aaci_streams + streamid;
- return aaci->ac97_bus->codec[s->codec_idx]->rates[s->rate_idx];
-}
-
-static unsigned int rate_list[] = {
- 5512, 8000, 11025, 16000, 22050, 32000, 44100,
- 48000, 64000, 88200, 96000, 176400, 192000
-};
-
-/*
- * Double-rate rule: we can support double rate iff channels == 2
- * (unimplemented)
- */
-static int
-aaci_rule_rate_by_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule)
-{
- struct aaci *aaci = rule->private;
- unsigned int rate_mask = SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_5512;
- struct snd_interval *c = hw_param_interval(p, SNDRV_PCM_HW_PARAM_CHANNELS);
-
- switch (c->max) {
- case 6:
- rate_mask &= aaci_rate_mask(aaci, ACSTREAM_LFE);
- case 4:
- rate_mask &= aaci_rate_mask(aaci, ACSTREAM_SURROUND);
- case 2:
- rate_mask &= aaci_rate_mask(aaci, ACSTREAM_FRONT);
- }
-
- return snd_interval_list(hw_param_interval(p, rule->var),
- ARRAY_SIZE(rate_list), rate_list,
- rate_mask);
-}
-
static struct snd_pcm_hardware aaci_hw_info = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
@@ -400,10 +351,7 @@ static struct snd_pcm_hardware aaci_hw_info = {
*/
.formats = SNDRV_PCM_FMTBIT_S16_LE,
- /* should this be continuous or knot? */
- .rates = SNDRV_PCM_RATE_CONTINUOUS,
- .rate_max = 48000,
- .rate_min = 4000,
+ /* rates are setup from the AC'97 codec */
.channels_min = 2,
.channels_max = 6,
.buffer_bytes_max = 64 * 1024,
@@ -423,6 +371,12 @@ static int __aaci_pcm_open(struct aaci *aaci,
aacirun->substream = substream;
runtime->private_data = aacirun;
runtime->hw = aaci_hw_info;
+ runtime->hw.rates = aacirun->pcm->rates;
+ snd_pcm_limit_hw_rates(runtime);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ aacirun->pcm->r[1].slots)
+ snd_ac97_pcm_double_rate_rules(runtime);
/*
* FIXME: ALSA specifies fifo_size in bytes. If we're in normal
@@ -433,17 +387,6 @@ static int __aaci_pcm_open(struct aaci *aaci,
*/
runtime->hw.fifo_size = aaci->fifosize * 2;
- /*
- * Add rule describing hardware rate dependency
- * on the number of channels.
- */
- ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
- aaci_rule_rate_by_channels, aaci,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- SNDRV_PCM_HW_PARAM_RATE, -1);
- if (ret)
- goto out;
-
ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED,
DRIVER_NAME, aaci);
if (ret)
@@ -507,18 +450,22 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
err = snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(params));
- if (err < 0)
- goto out;
+ if (err >= 0) {
+ unsigned int rate = params_rate(params);
+ int dbl = rate > 48000;
- err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
- params_channels(params),
- aacirun->pcm->r[0].slots);
- if (err)
- goto out;
+ err = snd_ac97_pcm_open(aacirun->pcm, rate,
+ params_channels(params),
+ aacirun->pcm->r[dbl].slots);
- aacirun->pcm_open = 1;
+ aacirun->pcm_open = err == 0;
+ aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+ aacirun->fifosz = aaci->fifosize * 4;
+
+ if (aacirun->cr & CR_COMPACT)
+ aacirun->fifosz >>= 1;
+ }
- out:
return err;
}
@@ -527,7 +474,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct aaci_runtime *aacirun = runtime->private_data;
- aacirun->start = (void *)runtime->dma_area;
+ aacirun->start = runtime->dma_area;
aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream);
aacirun->ptr = aacirun->start;
aacirun->period =
@@ -627,14 +574,9 @@ static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
* Enable FIFO, compact mode, 16 bits per sample.
* FIXME: double rate slots?
*/
- if (ret >= 0) {
- aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+ if (ret >= 0)
aacirun->cr |= channels_to_txmask[channels];
- aacirun->fifosz = aaci->fifosize * 4;
- if (aacirun->cr & CR_COMPACT)
- aacirun->fifosz >>= 1;
- }
return ret;
}
@@ -646,7 +588,7 @@ static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun)
ie &= ~(IE_URIE|IE_TXIE);
writel(ie, aacirun->base + AACI_IE);
aacirun->cr &= ~CR_EN;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_TXB);
writel(aacirun->cr, aacirun->base + AACI_TXCR);
}
@@ -654,7 +596,7 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
{
u32 ie;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_TXB);
aacirun->cr |= CR_EN;
ie = readl(aacirun->base + AACI_IE);
@@ -665,12 +607,12 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
unsigned long flags;
int ret = 0;
- spin_lock_irqsave(&aaci->lock, flags);
+ spin_lock_irqsave(&aacirun->lock, flags);
+
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
aaci_pcm_playback_start(aacirun);
@@ -697,7 +639,8 @@ static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cm
default:
ret = -EINVAL;
}
- spin_unlock_irqrestore(&aaci->lock, flags);
+
+ spin_unlock_irqrestore(&aacirun->lock, flags);
return ret;
}
@@ -721,18 +664,10 @@ static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
int ret;
ret = aaci_pcm_hw_params(substream, aacirun, params);
-
- if (ret >= 0) {
- aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
-
+ if (ret >= 0)
/* Line in record: slot 3 and 4 */
aacirun->cr |= CR_SL3 | CR_SL4;
- aacirun->fifosz = aaci->fifosize * 4;
-
- if (aacirun->cr & CR_COMPACT)
- aacirun->fifosz >>= 1;
- }
return ret;
}
@@ -740,7 +675,7 @@ static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun)
{
u32 ie;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_RXB);
ie = readl(aacirun->base + AACI_IE);
ie &= ~(IE_ORIE | IE_RXIE);
@@ -755,7 +690,7 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun)
{
u32 ie;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_RXB);
#ifdef DEBUG
/* RX Timeout value: bits 28:17 in RXCR */
@@ -772,12 +707,11 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun)
static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
unsigned long flags;
int ret = 0;
- spin_lock_irqsave(&aaci->lock, flags);
+ spin_lock_irqsave(&aacirun->lock, flags);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -806,7 +740,7 @@ static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd
ret = -EINVAL;
}
- spin_unlock_irqrestore(&aaci->lock, flags);
+ spin_unlock_irqrestore(&aacirun->lock, flags);
return ret;
}
@@ -889,6 +823,12 @@ static struct ac97_pcm ac97_defs[] __devinitdata = {
(1 << AC97_SLOT_PCM_SRIGHT) |
(1 << AC97_SLOT_LFE),
},
+ [1] = {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT) |
+ (1 << AC97_SLOT_PCM_LEFT_0) |
+ (1 << AC97_SLOT_PCM_RIGHT_0),
+ },
},
},
[1] = { /* PCM in */
@@ -1001,7 +941,6 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
aaci = card->private_data;
mutex_init(&aaci->ac97_sem);
- spin_lock_init(&aaci->lock);
aaci->card = card;
aaci->dev = dev;
@@ -1028,7 +967,7 @@ static int __devinit aaci_init_pcm(struct aaci *aaci)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops);
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
- NULL, 0, 64 * 104);
+ NULL, 0, 64 * 1024);
}
return ret;
@@ -1088,12 +1027,14 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
/*
* Playback uses AACI channel 0
*/
+ spin_lock_init(&aaci->playback.lock);
aaci->playback.base = aaci->base + AACI_CSCH1;
aaci->playback.fifo = aaci->base + AACI_DR1;
/*
* Capture uses AACI channel 0
*/
+ spin_lock_init(&aaci->capture.lock);
aaci->capture.base = aaci->base + AACI_CSCH1;
aaci->capture.fifo = aaci->base + AACI_DR1;
diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h
index 924f69c..6a4a2ee 100644
--- a/sound/arm/aaci.h
+++ b/sound/arm/aaci.h
@@ -202,6 +202,7 @@
struct aaci_runtime {
void __iomem *base;
void __iomem *fifo;
+ spinlock_t lock;
struct ac97_pcm *pcm;
int pcm_open;
@@ -232,7 +233,6 @@ struct aaci {
struct snd_ac97 *ac97;
u32 maincr;
- spinlock_t lock;
struct aaci_runtime playback;
struct aaci_runtime capture;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 30f4108..a27545b 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -758,7 +758,7 @@ int snd_interval_ratnum(struct snd_interval *i,
int diff;
if (q == 0)
q = 1;
- den = div_down(num, q);
+ den = div_up(num, q);
if (den < rats[k].den_min)
continue;
if (den > rats[k].den_max)
@@ -794,7 +794,7 @@ int snd_interval_ratnum(struct snd_interval *i,
i->empty = 1;
return -EINVAL;
}
- den = div_up(num, q);
+ den = div_down(num, q);
if (den > rats[k].den_max)
continue;
if (den < rats[k].den_min)
diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c
index cb9aa4c..4be562b 100644
--- a/sound/isa/msnd/msnd_midi.c
+++ b/sound/isa/msnd/msnd_midi.c
@@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *card, int device)
err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi);
if (err < 0)
return err;
- mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL);
+ mpu = kzalloc(sizeof(*mpu), GFP_KERNEL);
if (mpu == NULL) {
snd_device_free(card, rmidi);
return -ENOMEM;
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 96678d5..751762f 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -393,8 +393,6 @@ size_dram(struct snd_emu8000 *emu)
while (size < EMU8000_MAX_DRAM) {
- size += 512 * 1024; /* increment 512kbytes */
-
/* Write a unique data on the test address.
* if the address is out of range, the data is written on
* 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is
@@ -414,7 +412,9 @@ size_dram(struct snd_emu8000 *emu)
/*snd_emu8000_read_wait(emu);*/
EMU8000_SMLD_READ(emu); /* discard stale data */
if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2)
- break; /* we must have wrapped around */
+ break; /* no memory at this address */
+
+ size += 512 * 1024; /* increment 512kbytes */
snd_emu8000_read_wait(emu);
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 8691f4c..f1d9d16 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -609,7 +609,7 @@ static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
/* alloc virtual 'dma' area */
if (runtime->dma_area)
vfree(runtime->dma_area);
- runtime->dma_area = vmalloc(size);
+ runtime->dma_area = vmalloc_user(size);
if (runtime->dma_area == NULL)
return -ENOMEM;
runtime->dma_bytes = size;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 83f5ee2..e19dd5d 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -269,7 +269,7 @@ static int pss_reset_dsp(pss_confdata * devc)
unsigned long i, limit = jiffies + HZ/10;
outw(0x2000, REG(PSS_CONTROL));
- for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+ for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
inw(REG(PSS_CONTROL));
outw(0x0000, REG(PSS_CONTROL));
return 1;
@@ -369,11 +369,11 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size
outw(0, REG(PSS_DATA));
limit = jiffies + HZ/10;
- for (i = 0; i < 32768 && (limit - jiffies >= 0); i++)
+ for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
val = inw(REG(PSS_STATUS));
limit = jiffies + HZ/10;
- for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+ for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
{
val = inw(REG(PSS_STATUS));
if (val & 0x4000)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a09c03c..c578c28 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -29,6 +29,7 @@
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_beep.h"
#define CXT_PIN_DIR_IN 0x00
#define CXT_PIN_DIR_OUT 0x01
@@ -111,6 +112,7 @@ struct conexant_spec {
unsigned int dell_automute;
unsigned int port_d_mode;
unsigned char ext_mic_bias;
+ unsigned int dell_vostro;
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -476,6 +478,7 @@ static void conexant_free(struct hda_codec *codec)
snd_array_free(&spec->jacks);
}
#endif
+ snd_hda_detach_beep_device(codec);
kfree(codec->spec);
}
@@ -2109,9 +2112,12 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int val;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+ int inout = (kcontrol->private_value & 0x100) ?
+ AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT;
- val = snd_hda_codec_read(codec, 0x17, 0,
- AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT);
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE, inout);
ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN;
return 0;
@@ -2123,6 +2129,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
unsigned int idx;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+ int inout = (kcontrol->private_value & 0x100) ?
+ AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT;
if (!imux->num_items)
return 0;
@@ -2130,9 +2139,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- snd_hda_codec_write_cache(codec, 0x17, 0,
+ snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT |
+ AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout |
imux->items[idx].index);
return 1;
@@ -2201,10 +2210,11 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Analog Mic Boost Capture Enum",
+ .name = "Ext Mic Boost Capture Enum",
.info = cxt5066_mic_boost_mux_enum_info,
.get = cxt5066_mic_boost_mux_enum_get,
.put = cxt5066_mic_boost_mux_enum_put,
+ .private_value = 0x17,
},
HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others),
@@ -2212,6 +2222,19 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
{}
};
+static struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Int Mic Boost Capture Enum",
+ .info = cxt5066_mic_boost_mux_enum_info,
+ .get = cxt5066_mic_boost_mux_enum_get,
+ .put = cxt5066_mic_boost_mux_enum_put,
+ .private_value = 0x23 | 0x100,
+ },
+ HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+ {}
+};
+
static struct hda_verb cxt5066_init_verbs[] = {
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
@@ -2397,11 +2420,16 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
/* initialize jack-sensing, too */
static int cxt5066_init(struct hda_codec *codec)
{
+ struct conexant_spec *spec = codec->spec;
+
snd_printdd("CXT5066: init\n");
conexant_init(codec);
if (codec->patch_ops.unsol_event) {
cxt5066_hp_automute(codec);
- cxt5066_automic(codec);
+ if (spec->dell_vostro)
+ cxt5066_vostro_automic(codec);
+ else
+ cxt5066_automic(codec);
}
return 0;
}
@@ -2500,7 +2528,10 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->init_verbs[0] = cxt5066_init_verbs_vostro;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+ spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers;
spec->port_d_mode = 0;
+ spec->dell_vostro = 1;
+ snd_hda_attach_beep_device(codec, 0x13);
/* no S/PDIF out */
spec->multiout.dig_out_nid = 0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index aeed4cc..c746505 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -131,8 +131,8 @@ enum {
enum {
ALC269_BASIC,
ALC269_QUANTA_FL1,
- ALC269_ASUS_EEEPC_P703,
- ALC269_ASUS_EEEPC_P901,
+ ALC269_ASUS_AMIC,
+ ALC269_ASUS_DMIC,
ALC269_FUJITSU,
ALC269_LIFEBOOK,
ALC269_AUTO,
@@ -188,6 +188,8 @@ enum {
ALC663_ASUS_MODE4,
ALC663_ASUS_MODE5,
ALC663_ASUS_MODE6,
+ ALC663_ASUS_MODE7,
+ ALC663_ASUS_MODE8,
ALC272_DELL,
ALC272_DELL_ZM1,
ALC272_SAMSUNG_NC10,
@@ -335,6 +337,9 @@ struct alc_spec {
/* hooks */
void (*init_hook)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ void (*power_hook)(struct hda_codec *codec, int power);
+#endif
/* for pin sensing */
unsigned int sense_updated: 1;
@@ -386,6 +391,7 @@ struct alc_config_preset {
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_amp_list *loopbacks;
+ void (*power_hook)(struct hda_codec *codec, int power);
#endif
};
@@ -898,6 +904,7 @@ static void setup_preset(struct hda_codec *codec,
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->power_hook = preset->power_hook;
spec->loopback.amplist = preset->loopbacks;
#endif
@@ -1663,9 +1670,6 @@ static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
/* some bit here disables the other DACs. Init=0x4900 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x08},
{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
-/* Enable amplifiers */
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
/* DMIC fix
* This laptop has a stereo digital microphone. The mics are only 1cm apart
* which makes the stereo useless. However, either the mic or the ALC889
@@ -1778,6 +1782,25 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1808,6 +1831,16 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[2] = 0x1b;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void alc889_power_eapd(struct hda_codec *codec, int power)
+{
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+ snd_hda_codec_write(codec, 0x15, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+}
+#endif
+
/*
* ALC880 3-stack model
*
@@ -3601,12 +3634,29 @@ static void alc_free(struct hda_codec *codec)
snd_hda_detach_beep_device(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc_suspend(struct hda_codec *codec, pm_message_t state)
+{
+ struct alc_spec *spec = codec->spec;
+ if (spec && spec->power_hook)
+ spec->power_hook(codec, 0);
+ return 0;
+}
+#endif
+
#ifdef SND_HDA_NEEDS_RESUME
static int alc_resume(struct hda_codec *codec)
{
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct alc_spec *spec = codec->spec;
+#endif
codec->patch_ops.init(codec);
snd_hda_codec_resume_amp(codec);
snd_hda_codec_resume_cache(codec);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (spec && spec->power_hook)
+ spec->power_hook(codec, 1);
+#endif
return 0;
}
#endif
@@ -3623,6 +3673,7 @@ static struct hda_codec_ops alc_patch_ops = {
.resume = alc_resume,
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .suspend = alc_suspend,
.check_power_status = alc_check_power_status,
#endif
};
@@ -8919,7 +8970,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
- SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
@@ -9282,6 +9333,7 @@ static struct alc_config_preset alc882_presets[] = {
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
@@ -9378,10 +9430,11 @@ static struct alc_config_preset alc882_presets[] = {
.init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_8930G] = {
- .mixers = { alc888_base_mixer,
+ .mixers = { alc889_acer_aspire_8930g_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
- alc889_acer_aspire_8930g_verbs },
+ alc889_acer_aspire_8930g_verbs,
+ alc889_eapd_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
@@ -9398,6 +9451,9 @@ static struct alc_config_preset alc882_presets[] = {
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc889_acer_aspire_8930g_setup,
.init_hook = alc_automute_amp,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .power_hook = alc889_power_eapd,
+#endif
},
[ALC888_ACER_ASPIRE_7730G] = {
.mixers = { alc883_3ST_6ch_mixer,
@@ -9428,6 +9484,7 @@ static struct alc_config_preset alc882_presets[] = {
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
@@ -9489,6 +9546,7 @@ static struct alc_config_preset alc882_presets[] = {
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_lenovo_101e_capture_source,
@@ -9668,6 +9726,7 @@ static struct alc_config_preset alc882_presets[] = {
alc880_gpio1_init_verbs },
.adc_nids = alc883_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .capsrc_nids = alc883_capsrc_nids,
.dac_nids = alc883_dac_nids,
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.channel_mode = alc889A_mb31_6ch_modes,
@@ -10678,6 +10737,13 @@ static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
{}
};
+static struct hda_verb alc262_lenovo_3000_init_verbs[] = {
+ /* Front Mic pin: input vref at 50% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {}
+};
+
static struct hda_input_mux alc262_fujitsu_capture_source = {
.num_items = 3,
.items = {
@@ -11720,7 +11786,8 @@ static struct alc_config_preset alc262_presets[] = {
[ALC262_LENOVO_3000] = {
.mixers = { alc262_lenovo_3000_mixer },
.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
- alc262_lenovo_3000_unsol_verbs },
+ alc262_lenovo_3000_unsol_verbs,
+ alc262_lenovo_3000_init_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
@@ -12857,7 +12924,7 @@ static int patch_alc268(struct hda_codec *codec)
int board_config;
int i, has_beep, err;
- spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
@@ -13232,10 +13299,12 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
/* toggle speaker-output according to the hp-jack state */
static void alc269_speaker_automute(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
+ unsigned int nid = spec->autocfg.hp_pins[0];
unsigned int present;
unsigned char bits;
- present = snd_hda_jack_detect(codec, 0x15);
+ present = snd_hda_jack_detect(codec, nid);
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -13460,8 +13529,8 @@ static void alc269_auto_init(struct hda_codec *codec)
static const char *alc269_models[ALC269_MODEL_LAST] = {
[ALC269_BASIC] = "basic",
[ALC269_QUANTA_FL1] = "quanta",
- [ALC269_ASUS_EEEPC_P703] = "eeepc-p703",
- [ALC269_ASUS_EEEPC_P901] = "eeepc-p901",
+ [ALC269_ASUS_AMIC] = "asus-amic",
+ [ALC269_ASUS_DMIC] = "asus-dmic",
[ALC269_FUJITSU] = "fujitsu",
[ALC269_LIFEBOOK] = "lifebook",
[ALC269_AUTO] = "auto",
@@ -13470,18 +13539,41 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
static struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
- ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_EEEPC_P703),
+ ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80JT", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_AMIC),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
- ALC269_ASUS_EEEPC_P901),
+ ALC269_ASUS_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
- ALC269_ASUS_EEEPC_P901),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_EEEPC_P901),
+ ALC269_ASUS_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_ASUS_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_ASUS_DMIC),
SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
{}
@@ -13511,7 +13603,7 @@ static struct alc_config_preset alc269_presets[] = {
.setup = alc269_quanta_fl1_setup,
.init_hook = alc269_quanta_fl1_init_hook,
},
- [ALC269_ASUS_EEEPC_P703] = {
+ [ALC269_ASUS_AMIC] = {
.mixers = { alc269_eeepc_mixer },
.cap_mixer = alc269_epc_capture_mixer,
.init_verbs = { alc269_init_verbs,
@@ -13525,7 +13617,7 @@ static struct alc_config_preset alc269_presets[] = {
.setup = alc269_eeepc_amic_setup,
.init_hook = alc269_eeepc_inithook,
},
- [ALC269_ASUS_EEEPC_P901] = {
+ [ALC269_ASUS_DMIC] = {
.mixers = { alc269_eeepc_mixer },
.cap_mixer = alc269_epc_capture_mixer,
.init_verbs = { alc269_init_verbs,
@@ -16160,6 +16252,52 @@ static struct snd_kcontrol_new alc663_g50v_mixer[] = {
{ } /* end */
};
+static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc663_mode7_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc663_mode8_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
static struct snd_kcontrol_new alc662_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -16447,6 +16585,45 @@ static struct hda_verb alc272_dell_init_verbs[] = {
{}
};
+static struct hda_verb alc663_mode7_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_mode8_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
static struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
@@ -16626,6 +16803,54 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
}
}
+static void alc663_two_hp_m7_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present1, present2;
+
+ present1 = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ present2 = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+
+ if (present1 || present2) {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ } else {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ }
+}
+
+static void alc663_two_hp_m8_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present1, present2;
+
+ present1 = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ present2 = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+
+ if (present1 || present2) {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ } else {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ }
+}
+
static void alc663_m51va_unsol_event(struct hda_codec *codec,
unsigned int res)
{
@@ -16645,7 +16870,7 @@ static void alc663_m51va_setup(struct hda_codec *codec)
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 1;
+ spec->int_mic.mux_idx = 9;
spec->auto_mic = 1;
}
@@ -16657,7 +16882,17 @@ static void alc663_m51va_inithook(struct hda_codec *codec)
/* ***************** Mode1 ******************************/
#define alc663_mode1_unsol_event alc663_m51va_unsol_event
-#define alc663_mode1_setup alc663_m51va_setup
+
+static void alc663_mode1_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
+}
+
#define alc663_mode1_inithook alc663_m51va_inithook
/* ***************** Mode2 ******************************/
@@ -16674,7 +16909,7 @@ static void alc662_mode2_unsol_event(struct hda_codec *codec,
}
}
-#define alc662_mode2_setup alc663_m51va_setup
+#define alc662_mode2_setup alc663_mode1_setup
static void alc662_mode2_inithook(struct hda_codec *codec)
{
@@ -16695,7 +16930,7 @@ static void alc663_mode3_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode3_setup alc663_m51va_setup
+#define alc663_mode3_setup alc663_mode1_setup
static void alc663_mode3_inithook(struct hda_codec *codec)
{
@@ -16716,7 +16951,7 @@ static void alc663_mode4_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode4_setup alc663_m51va_setup
+#define alc663_mode4_setup alc663_mode1_setup
static void alc663_mode4_inithook(struct hda_codec *codec)
{
@@ -16737,7 +16972,7 @@ static void alc663_mode5_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode5_setup alc663_m51va_setup
+#define alc663_mode5_setup alc663_mode1_setup
static void alc663_mode5_inithook(struct hda_codec *codec)
{
@@ -16758,7 +16993,7 @@ static void alc663_mode6_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode6_setup alc663_m51va_setup
+#define alc663_mode6_setup alc663_mode1_setup
static void alc663_mode6_inithook(struct hda_codec *codec)
{
@@ -16766,6 +17001,50 @@ static void alc663_mode6_inithook(struct hda_codec *codec)
alc_mic_automute(codec);
}
+/* ***************** Mode7 ******************************/
+static void alc663_mode7_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_two_hp_m7_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
+}
+
+#define alc663_mode7_setup alc663_mode1_setup
+
+static void alc663_mode7_inithook(struct hda_codec *codec)
+{
+ alc663_two_hp_m7_speaker_automute(codec);
+ alc_mic_automute(codec);
+}
+
+/* ***************** Mode8 ******************************/
+static void alc663_mode8_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_two_hp_m8_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
+}
+
+#define alc663_mode8_setup alc663_m51va_setup
+
+static void alc663_mode8_inithook(struct hda_codec *codec)
+{
+ alc663_two_hp_m8_speaker_automute(codec);
+ alc_mic_automute(codec);
+}
+
static void alc663_g71v_hp_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -16900,6 +17179,8 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
[ALC663_ASUS_MODE4] = "asus-mode4",
[ALC663_ASUS_MODE5] = "asus-mode5",
[ALC663_ASUS_MODE6] = "asus-mode6",
+ [ALC663_ASUS_MODE7] = "asus-mode7",
+ [ALC663_ASUS_MODE8] = "asus-mode8",
[ALC272_DELL] = "dell",
[ALC272_DELL_ZM1] = "dell-zm1",
[ALC272_SAMSUNG_NC10] = "samsung-nc10",
@@ -16916,12 +17197,22 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
+ SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
@@ -17205,6 +17496,36 @@ static struct alc_config_preset alc662_presets[] = {
.setup = alc663_mode6_setup,
.init_hook = alc663_mode6_inithook,
},
+ [ALC663_ASUS_MODE7] = {
+ .mixers = { alc663_mode7_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc663_mode7_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc663_mode7_unsol_event,
+ .setup = alc663_mode7_setup,
+ .init_hook = alc663_mode7_inithook,
+ },
+ [ALC663_ASUS_MODE8] = {
+ .mixers = { alc663_mode8_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc663_mode8_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc663_mode8_unsol_event,
+ .setup = alc663_mode8_setup,
+ .init_hook = alc663_mode8_inithook,
+ },
[ALC272_DELL] = {
.mixers = { alc663_m51va_mixer },
.cap_mixer = alc272_auto_capture_mixer,
@@ -17688,7 +18009,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
{ .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
{ .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
+ { .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 },
{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
+ { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
.patch = patch_alc861 },
{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index d057e64..5cfa608 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -51,7 +51,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s
return 0; /* already enough large */
vfree(runtime->dma_area);
}
- runtime->dma_area = vmalloc_32(size);
+ runtime->dma_area = vmalloc_32_user(size);
if (! runtime->dma_area)
return -ENOMEM;
runtime->dma_bytes = size;
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index b69861d..3ef16bb 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -470,7 +470,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642);
static int __init ak4642_modinit(void)
{
- int ret;
+ int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&ak4642_i2c_driver);
#endif
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index bbc72c2..81b8c9d 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -191,6 +191,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream,
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x1; /* enable variable rate audio */
+ vra &= ~0x4; /* disable SPDIF output */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
@@ -221,22 +222,6 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream,
return stac9766_ac97_write(codec, reg, runtime->rate);
}
-static int ac97_digital_trigger(struct snd_pcm_substream *substream,
- int cmd, struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
- unsigned short vra;
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_STOP:
- vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
- vra &= !0x04;
- stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
- break;
- }
- return 0;
-}
-
static int stac9766_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -315,7 +300,6 @@ static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
.prepare = ac97_digital_prepare,
- .trigger = ac97_digital_trigger,
};
struct snd_soc_dai stac9766_dai[] = {
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 81c57b5..a808675 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -47,7 +47,7 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = {
};
#define WM8974_POWER1_BIASEN 0x08
-#define WM8974_POWER1_BUFIOEN 0x10
+#define WM8974_POWER1_BUFIOEN 0x04
struct wm8974_priv {
struct snd_soc_codec codec;
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index b074a59..4963def 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -752,7 +752,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s
return 0; /* already large enough */
vfree(runtime->dma_area);
}
- runtime->dma_area = vmalloc(size);
+ runtime->dma_area = vmalloc_user(size);
if (!runtime->dma_area)
return -ENOMEM;
runtime->dma_bytes = size;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-12-15 13:55 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-12-15 13:55 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull sound fixes for v2.6.33 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All trivial fixes as below.
Thanks!
Takashi
===
Daniel T Chen (2):
ALSA: hda: Use ALC260_WILL quirk for another Acer model (0x1025007f)
ALSA: hda: Fix max PCM level to 0 dB for AD1981_HP
Julia Lawall (2):
ALSA: sound/isa/gus: Correct code taking the size of a pointer
ASoC: Correct code taking the size of a pointer
Stefan Ringel (1):
ALSA: hda - Add PCI IDs for Nvidia G2xx-series
Steve Soule (1):
ALSA: ac97_codec - increase timeout for analog sections to 5 second
---
sound/isa/gus/gus_mem.c | 3 ++-
sound/pci/ac97/ac97_codec.c | 2 +-
sound/pci/hda/hda_intel.c | 3 +++
sound/pci/hda/patch_analog.c | 8 ++++++++
sound/pci/hda/patch_realtek.c | 1 +
sound/soc/codecs/wm8900.c | 2 +-
6 files changed, 16 insertions(+), 3 deletions(-)
diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c
index 661205c..af888a0 100644
--- a/sound/isa/gus/gus_mem.c
+++ b/sound/isa/gus/gus_mem.c
@@ -127,7 +127,8 @@ static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc,
!share_id[2] && !share_id[3])
return NULL;
for (block = alloc->first; block; block = block->next)
- if (!memcmp(share_id, block->share_id, sizeof(share_id)))
+ if (!memcmp(share_id, block->share_id,
+ sizeof(block->share_id)))
return block;
return NULL;
}
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 20cb60a..c119206 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
}
/* nothing should be in powerdown mode */
snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0);
- end_time = jiffies + msecs_to_jiffies(120);
+ end_time = jiffies + msecs_to_jiffies(5000);
do {
if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f)
goto __ready_ok;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e54420e..9b56f93 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2713,6 +2713,9 @@ static struct pci_device_id azx_ids[] = {
{ PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA },
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 447eda1..1a36137 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1789,6 +1789,14 @@ static int patch_ad1981(struct hda_codec *codec)
codec->patch_ops.init = ad1981_hp_init;
codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
+ /* set the upper-limit for mixer amp to 0dB for avoiding the
+ * possible damage by overloading
+ */
+ snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
break;
case AD1981_THINKPAD:
spec->mixers[0] = ad1981_thinkpad_mixers;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 888b631..aeed4cc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6248,6 +6248,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = {
static struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
+ SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index c9438dd..dbc368c 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -199,7 +199,7 @@ static void wm8900_reset(struct snd_soc_codec *codec)
snd_soc_write(codec, WM8900_REG_RESET, 0);
memcpy(codec->reg_cache, wm8900_reg_defaults,
- sizeof(codec->reg_cache));
+ sizeof(wm8900_reg_defaults));
}
static int wm8900_hp_event(struct snd_soc_dapm_widget *w,
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-08-25 7:13 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-08-25 7:13 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull misc sound fixes for v2.6.31-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/misc
One fix for sound/core/pcm_lib.c looks non-trivial, but actually it's
also a good clean-up. The others are trivial indeed.
Thanks!
Takashi
===
Bartlomiej Zolnierkiewicz (1):
ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
Clemens Ladisch (2):
sound: vx222: fix input level control range check
sound: pcm_lib: fix unsorted list constraint handling
---
sound/core/pcm_lib.c | 39 ++++++++-------------------------------
sound/pci/ali5451/ali5451.c | 18 ++++++++++++------
sound/pci/vx222/vx222_ops.c | 4 ++--
3 files changed, 22 insertions(+), 39 deletions(-)
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 72cfd47..9db60d8 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -943,47 +943,24 @@ static int snd_interval_ratden(struct snd_interval *i,
int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask)
{
unsigned int k;
- int changed = 0;
+ struct snd_interval list_range;
if (!count) {
i->empty = 1;
return -EINVAL;
}
+ snd_interval_any(&list_range);
+ list_range.min = UINT_MAX;
+ list_range.max = 0;
for (k = 0; k < count; k++) {
if (mask && !(mask & (1 << k)))
continue;
- if (i->min == list[k] && !i->openmin)
- goto _l1;
- if (i->min < list[k]) {
- i->min = list[k];
- i->openmin = 0;
- changed = 1;
- goto _l1;
- }
- }
- i->empty = 1;
- return -EINVAL;
- _l1:
- for (k = count; k-- > 0;) {
- if (mask && !(mask & (1 << k)))
+ if (!snd_interval_test(i, list[k]))
continue;
- if (i->max == list[k] && !i->openmax)
- goto _l2;
- if (i->max > list[k]) {
- i->max = list[k];
- i->openmax = 0;
- changed = 1;
- goto _l2;
- }
+ list_range.min = min(list_range.min, list[k]);
+ list_range.max = max(list_range.max, list[k]);
}
- i->empty = 1;
- return -EINVAL;
- _l2:
- if (snd_interval_checkempty(i)) {
- i->empty = 1;
- return -EINVAL;
- }
- return changed;
+ return snd_interval_refine(i, &list_range);
}
EXPORT_SYMBOL(snd_interval_list);
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index c551006..76d76c0 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -310,12 +310,16 @@ static int snd_ali_codec_ready(struct snd_ali *codec,
unsigned int res;
end_time = jiffies + msecs_to_jiffies(250);
- do {
+
+ for (;;) {
res = snd_ali_5451_peek(codec,port);
if (!(res & 0x8000))
return 0;
+ if (!time_after_eq(end_time, jiffies))
+ break;
schedule_timeout_uninterruptible(1);
- } while (time_after_eq(end_time, jiffies));
+ }
+
snd_ali_5451_poke(codec, port, res & ~0x8000);
snd_printdd("ali_codec_ready: codec is not ready.\n ");
return -EIO;
@@ -327,15 +331,17 @@ static int snd_ali_stimer_ready(struct snd_ali *codec)
unsigned long dwChk1,dwChk2;
dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER);
- dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER);
-
end_time = jiffies + msecs_to_jiffies(250);
- do {
+
+ for (;;) {
dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER);
if (dwChk2 != dwChk1)
return 0;
+ if (!time_after_eq(end_time, jiffies))
+ break;
schedule_timeout_uninterruptible(1);
- } while (time_after_eq(end_time, jiffies));
+ }
+
snd_printk(KERN_ERR "ali_stimer_read: stimer is not ready.\n");
return -EIO;
}
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index 6416d3f..a69e774 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -885,10 +885,10 @@ static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
if (ucontrol->value.integer.value[0] < 0 ||
- ucontrol->value.integer.value[0] < MIC_LEVEL_MAX)
+ ucontrol->value.integer.value[0] > MIC_LEVEL_MAX)
return -EINVAL;
if (ucontrol->value.integer.value[1] < 0 ||
- ucontrol->value.integer.value[1] < MIC_LEVEL_MAX)
+ ucontrol->value.integer.value[1] > MIC_LEVEL_MAX)
return -EINVAL;
mutex_lock(&_chip->mixer_mutex);
if (chip->input_level[0] != ucontrol->value.integer.value[0] ||
^ permalink raw reply related [flat|nested] 60+ messages in thread
* Re: [GIT PULL] sound fixes
2009-08-12 15:23 ` Linus Torvalds
@ 2009-08-12 15:24 ` Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-08-12 15:24 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
At Wed, 12 Aug 2009 08:23:19 -0700 (PDT),
Linus Torvalds wrote:
>
>
>
> On Wed, 12 Aug 2009, Takashi Iwai wrote:
> >
> > please pull ALSA fixes for v2.6.31-rc5 from:
> >
> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
> >
> > containing the following trivial fixes.
>
> Nothing there. Forgot to push?
Doh, indeed. Pushed now.
Sorry for inconvenience!
Takashi
^ permalink raw reply [flat|nested] 60+ messages in thread
* Re: [GIT PULL] sound fixes
2009-08-12 6:10 Takashi Iwai
@ 2009-08-12 15:23 ` Linus Torvalds
2009-08-12 15:24 ` Takashi Iwai
0 siblings, 1 reply; 60+ messages in thread
From: Linus Torvalds @ 2009-08-12 15:23 UTC (permalink / raw)
To: Takashi Iwai; +Cc: Andrew Morton, linux-kernel
On Wed, 12 Aug 2009, Takashi Iwai wrote:
>
> please pull ALSA fixes for v2.6.31-rc5 from:
>
> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
>
> containing the following trivial fixes.
Nothing there. Forgot to push?
Linus
^ permalink raw reply [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-08-12 6:10 Takashi Iwai
2009-08-12 15:23 ` Linus Torvalds
0 siblings, 1 reply; 60+ messages in thread
From: Takashi Iwai @ 2009-08-12 6:10 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA fixes for v2.6.31-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
===
Takashi Iwai (3):
ASoC: Add missing DRV_NAME definitions for fsl/* drivers
ALSA: hda - Add missing vmaster initialization for ALC269
ALSA: hda - Don't override ADC definitions for ALC codecs
---
sound/pci/hda/patch_realtek.c | 20 ++++++++++++++------
sound/soc/fsl/efika-audio-fabric.c | 2 ++
sound/soc/fsl/pcm030-audio-fabric.c | 2 ++
3 files changed, 18 insertions(+), 6 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 51c44fd..fea9767 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13563,6 +13563,8 @@ static int patch_alc269(struct hda_codec *codec)
set_capture_mixer(spec);
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC269_AUTO)
spec->init_hook = alc269_auto_init;
@@ -15577,9 +15579,12 @@ static int patch_alc861vd(struct hda_codec *codec)
spec->stream_digital_playback = &alc861vd_pcm_digital_playback;
spec->stream_digital_capture = &alc861vd_pcm_digital_capture;
- spec->adc_nids = alc861vd_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
- spec->capsrc_nids = alc861vd_capsrc_nids;
+ if (!spec->adc_nids) {
+ spec->adc_nids = alc861vd_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
+ }
+ if (!spec->capsrc_nids)
+ spec->capsrc_nids = alc861vd_capsrc_nids;
set_capture_mixer(spec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -17496,9 +17501,12 @@ static int patch_alc662(struct hda_codec *codec)
spec->stream_digital_playback = &alc662_pcm_digital_playback;
spec->stream_digital_capture = &alc662_pcm_digital_capture;
- spec->adc_nids = alc662_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
- spec->capsrc_nids = alc662_capsrc_nids;
+ if (!spec->adc_nids) {
+ spec->adc_nids = alc662_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
+ }
+ if (!spec->capsrc_nids)
+ spec->capsrc_nids = alc662_capsrc_nids;
if (!spec->cap_mixer)
set_capture_mixer(spec);
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 85b0e75..3326e2a 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -30,6 +30,8 @@
#include "mpc5200_psc_ac97.h"
#include "../codecs/stac9766.h"
+#define DRV_NAME "efika-audio-fabric"
+
static struct snd_soc_device device;
static struct snd_soc_card card;
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 8766f7a..b928ef7 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -30,6 +30,8 @@
#include "mpc5200_psc_ac97.h"
#include "../codecs/wm9712.h"
+#define DRV_NAME "pcm030-audio-fabric"
+
static struct snd_soc_device device;
static struct snd_soc_card card;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-07-31 8:25 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-07-31 8:25 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull another ALSA fixes for v2.6.31-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing only the following trivial patches.
Thanks!
Takashi
===
Julia Lawall (1):
ALSA: sound/aoa: Add kmalloc NULL tests
Roel Kluin (3):
ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
sound: aedsp16: Buffer overflow
sound: mpu401.c: Buffer overflow
Takashi Iwai (2):
ALSA: hda - Add quirk for Dell Studio 1555
ALSA: hda - Increase PCM stream name buf in patch_realtek.c
---
sound/aoa/core/gpio-pmf.c | 4 ++++
sound/oss/aedsp16.c | 9 +++++----
sound/oss/mpu401.c | 2 +-
sound/pci/hda/hda_eld.c | 4 ++--
sound/pci/hda/patch_realtek.c | 4 ++--
sound/pci/hda/patch_sigmatel.c | 2 ++
6 files changed, 16 insertions(+), 9 deletions(-)
diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 5ca2220..1dd0c28 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -182,6 +182,10 @@ static int pmf_set_notify(struct gpio_runtime *rt,
if (!old && notify) {
irq_client = kzalloc(sizeof(struct pmf_irq_client),
GFP_KERNEL);
+ if (!irq_client) {
+ err = -ENOMEM;
+ goto out_unlock;
+ }
irq_client->data = notif;
irq_client->handler = pmf_handle_notify_irq;
irq_client->owner = THIS_MODULE;
diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c
index 3ee9900..35b5912 100644
--- a/sound/oss/aedsp16.c
+++ b/sound/oss/aedsp16.c
@@ -325,8 +325,9 @@
/*
* Size of character arrays that store name and version of sound card
*/
-#define CARDNAMELEN 15 /* Size of the card's name in chars */
-#define CARDVERLEN 2 /* Size of the card's version in chars */
+#define CARDNAMELEN 15 /* Size of the card's name in chars */
+#define CARDVERLEN 10 /* Size of the card's version in chars */
+#define CARDVERDIGITS 2 /* Number of digits in the version */
#if defined(CONFIG_SC6600)
/*
@@ -410,7 +411,7 @@
static int soft_cfg __initdata = 0; /* bitmapped config */
static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */
-static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver:
+static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver:
hi->ver[0] lo->ver[1] */
#if defined(CONFIG_SC6600)
@@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port)
* string is finished.
*/
ver[len++] = ret;
- } while (len < CARDVERLEN);
+ } while (len < CARDVERDIGITS);
sprintf(DSPVersion, "%d.%d", ver[0], ver[1]);
DBG(("success.\n"));
diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c
index 1b2316f..734b8f9 100644
--- a/sound/oss/mpu401.c
+++ b/sound/oss/mpu401.c
@@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner)
sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name);
else
sprintf(mpu_synth_info[m].name,
- "MPU-401 %d.%d%c Midi interface #%d",
+ "MPU-401 %d.%d%c MIDI #%d",
(int) (devc->version & 0xf0) >> 4,
devc->version & 0x0f,
revision_char,
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index fcad5ec..9446a5a 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -508,7 +508,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
char name[64];
char *sname;
long long val;
- int n;
+ unsigned int n;
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%s %llx", name, &val) != 2)
@@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
sname++;
n = 10 * n + name[4] - '0';
}
- if (n < 0 || n > 31) /* double the CEA limit */
+ if (n >= ELD_MAX_SAD)
continue;
if (!strcmp(sname, "_coding_type"))
e->sad[n].format = val;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8c8b273..b95df5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,13 +275,13 @@ struct alc_spec {
*/
unsigned int num_init_verbs;
- char stream_name_analog[16]; /* analog PCM stream */
+ char stream_name_analog[32]; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
struct hda_pcm_stream *stream_analog_alt_playback;
struct hda_pcm_stream *stream_analog_alt_capture;
- char stream_name_digital[16]; /* digital PCM stream */
+ char stream_name_digital[32]; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 512f3b9..5383d8c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1809,6 +1809,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
"Dell Studio 1537", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0,
"Dell Studio 17", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
+ "Dell Studio 1555", STAC_DELL_M6_DMIC),
{} /* terminator */
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-07-21 17:08 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-07-21 17:08 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.31-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following small/trivial fixes.
Thanks!
Takashi
===
Frank Roth (1):
ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2
Hao Song (1):
ALSA: hda - Add quirk for Gateway T6834c laptop
Jaroslav Kysela (1):
ALSA: hda_codec: Check for invalid zero connections
Jaswinder Singh Rajput (1):
ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
Julia Lawall (1):
ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
Takashi Iwai (2):
ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
ALSA: ca0106 - Fix the max capture buffer size
---
sound/core/seq/Makefile | 7 ++-----
sound/isa/gus/gus_pcm.c | 4 ++--
sound/pci/ca0106/ca0106_main.c | 4 ++--
sound/pci/ctxfi/ctdaio.c | 4 ++--
sound/pci/hda/hda_codec.c | 6 ++++++
sound/pci/hda/patch_sigmatel.c | 7 +++++--
6 files changed, 19 insertions(+), 13 deletions(-)
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 1bcb360..941f64a 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -3,10 +3,6 @@
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
-ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
- obj-$(CONFIG_SND_SEQUENCER) += oss/
-endif
-
snd-seq-device-objs := seq_device.o
snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
seq_fifo.o seq_prioq.o seq_timer.o \
@@ -19,7 +15,8 @@ snd-seq-virmidi-objs := seq_virmidi.o
obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o
ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+ obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+ obj-$(CONFIG_SND_SEQUENCER) += oss/
endif
obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index edb11ee..2dcf45b 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -795,13 +795,13 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE))
continue;
/* load real volume - better precision */
- spin_lock_irqsave(&gus->reg_lock, flags);
+ spin_lock(&gus->reg_lock);
snd_gf1_select_voice(gus, pvoice->number);
snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol);
pcmp->final_volume = 1;
- spin_unlock_irqrestore(&gus->reg_lock, flags);
+ spin_unlock(&gus->reg_lock);
}
spin_unlock_irqrestore(&gus->voice_alloc, flags);
return change;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index f24bf1e..15e4138 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -325,9 +325,9 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
.rate_max = 192000,
.channels_min = 2,
.channels_max = 2,
- .buffer_bytes_max = ((65536 - 64) * 8),
+ .buffer_bytes_max = 65536 - 128,
.period_bytes_min = 64,
- .period_bytes_max = (65536 - 64),
+ .period_bytes_max = 32768 - 64,
.periods_min = 2,
.periods_max = 2,
.fifo_size = 0,
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 082e35c..deb6cfa 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -57,9 +57,9 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = {
struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
[LINEO1] = {.left = 0x40, .right = 0x41},
- [LINEO2] = {.left = 0x70, .right = 0x71},
+ [LINEO2] = {.left = 0x60, .right = 0x61},
[LINEO3] = {.left = 0x50, .right = 0x51},
- [LINEO4] = {.left = 0x60, .right = 0x61},
+ [LINEO4] = {.left = 0x70, .right = 0x71},
[LINEIM] = {.left = 0x45, .right = 0xc5},
[SPDIFOO] = {.left = 0x00, .right = 0x01},
[SPDIFIO] = {.left = 0x05, .right = 0x85},
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d255d..88480c0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -332,6 +332,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
AC_VERB_GET_CONNECT_LIST, i);
range_val = !!(parm & (1 << (shift-1))); /* ranges */
val = parm & mask;
+ if (val == 0) {
+ snd_printk(KERN_WARNING "hda_codec: "
+ "invalid CONNECT_LIST verb %x[%i]:%x\n",
+ nid, i, parm);
+ return 0;
+ }
parm >>= shift;
if (range_val) {
/* ranges between the previous and this one */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 41b5b3a..da7f9f6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2378,6 +2378,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
"Dell Vostro 1500", STAC_9205_DELL_M42),
/* Gateway */
+ SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
{} /* terminator */
};
@@ -5854,6 +5855,8 @@ static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
};
static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+ SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0,
+ "Sony VAIO F/S", STAC_9872_VAIO),
{} /* terminator */
};
@@ -5866,6 +5869,8 @@ static int patch_stac9872(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+ spec->pin_nids = stac9872_pin_nids;
spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
stac9872_models,
@@ -5877,8 +5882,6 @@ static int patch_stac9872(struct hda_codec *codec)
stac92xx_set_config_regs(codec,
stac9872_brd_tbl[spec->board_config]);
- spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
- spec->pin_nids = stac9872_pin_nids;
spec->multiout.dac_nids = spec->dac_nids;
spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
spec->adc_nids = stac9872_adc_nids;
^ permalink raw reply related [flat|nested] 60+ messages in thread
* Re: [GIT PULL] sound fixes
2009-05-22 20:47 ` Linus Torvalds
@ 2009-05-22 21:25 ` Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-05-22 21:25 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
At Fri, 22 May 2009 13:47:52 -0700 (PDT),
Linus Torvalds wrote:
>
>
>
> On Fri, 22 May 2009, Takashi Iwai wrote:
> >
> > please pull ALSA updates for v2.6.30-rc6 from:
> >
> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
> >
> > containing just the following trivial fixes.
>
> Thanks, pulled.
>
> I also note that you've been active on
>
> http://bugzilla.kernel.org/show_bug.cgi?id=13250
>
> but quite frankly, my opinion is that the workaround listed there isn't a
> fix at all. Why isn't such a frontpanel configuration choice exported as a
> switch, so that you can access it with alsamixer, rather than playing
> crazy games with some magic sysfs files?
>
> If it cannot be autodetected, it should still be visible as a mixer
> option, not some magic. No?
Yeah, the method I suggested there is a typical "debug" procedure
and no right "fix" yet.
[Short background:
Basically the HD-audio is designed in a way that BIOS could specify
the proper h/w configuration, and the driver needs just to follow
that. The problem now is when the hardware doesn't match really with
what BIOS assumes -- i.e. the driver is fooled. In the bug above, the
box isn't connected to the front panel as BIOS expected. And, a
recent fix for the Headphone volume control appears now as if a
regression for his use case.]
We can of course add some mixer element to change the DAC assignment.
Some codecs have a mixer "Channel Mode" to change the surround mode.
In a similar way, the front HP / mic could be changed. But, the
implementation in patch_sigmatel.c is a bit complex to put such a
mixer element into the middle, so I'm afraid it would be a bit risky
at this stage.
Alternatively, what we did in most cases for such a case are to create
a specific quirk and let user pass via model option. For this
particular case, this would be a simpler fix because basically we just
need to give a new pin-config map like an (untested) patch below.
Anyway, I'll follow up the bug there with that patch...
thanks,
Takashi
---
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 03b3646..922d05e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -150,6 +150,7 @@ enum {
STAC_D965_REF,
STAC_D965_3ST,
STAC_D965_5ST,
+ STAC_D965_5ST_NO_FP,
STAC_DELL_3ST,
STAC_DELL_BIOS,
STAC_927X_MODELS
@@ -2154,6 +2155,13 @@ static unsigned int d965_5st_pin_configs[14] = {
0x40000100, 0x40000100
};
+static unsigned int d965_5st_no_fp_pin_configs[14] = {
+ 0x40000100, 0x40000100, 0x0181304e, 0x01014010,
+ 0x01a19040, 0x01011012, 0x01016011, 0x40000100,
+ 0x40000100, 0x40000100, 0x40000100, 0x01442070,
+ 0x40000100, 0x40000100
+};
+
static unsigned int dell_3st_pin_configs[14] = {
0x02211230, 0x02a11220, 0x01a19040, 0x01114210,
0x01111212, 0x01116211, 0x01813050, 0x01112214,
@@ -2166,6 +2174,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
[STAC_D965_REF] = ref927x_pin_configs,
[STAC_D965_3ST] = d965_3st_pin_configs,
[STAC_D965_5ST] = d965_5st_pin_configs,
+ [STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs,
[STAC_DELL_3ST] = dell_3st_pin_configs,
[STAC_DELL_BIOS] = NULL,
};
@@ -2176,6 +2185,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
[STAC_D965_REF] = "ref",
[STAC_D965_3ST] = "3stack",
[STAC_D965_5ST] = "5stack",
+ [STAC_D965_5ST_NO_FP] = "5stack-no-fp",
[STAC_DELL_3ST] = "dell-3stack",
[STAC_DELL_BIOS] = "dell-bios",
};
^ permalink raw reply related [flat|nested] 60+ messages in thread
* Re: [GIT PULL] sound fixes
2009-05-22 17:34 Takashi Iwai
@ 2009-05-22 20:47 ` Linus Torvalds
2009-05-22 21:25 ` Takashi Iwai
0 siblings, 1 reply; 60+ messages in thread
From: Linus Torvalds @ 2009-05-22 20:47 UTC (permalink / raw)
To: Takashi Iwai; +Cc: Andrew Morton, linux-kernel
On Fri, 22 May 2009, Takashi Iwai wrote:
>
> please pull ALSA updates for v2.6.30-rc6 from:
>
> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
>
> containing just the following trivial fixes.
Thanks, pulled.
I also note that you've been active on
http://bugzilla.kernel.org/show_bug.cgi?id=13250
but quite frankly, my opinion is that the workaround listed there isn't a
fix at all. Why isn't such a frontpanel configuration choice exported as a
switch, so that you can access it with alsamixer, rather than playing
crazy games with some magic sysfs files?
If it cannot be autodetected, it should still be visible as a mixer
option, not some magic. No?
Linus
^ permalink raw reply [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-05-22 17:34 Takashi Iwai
2009-05-22 20:47 ` Linus Torvalds
0 siblings, 1 reply; 60+ messages in thread
From: Takashi Iwai @ 2009-05-22 17:34 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.30-rc6 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing just the following trivial fixes.
Thanks!
Takashi
===
Adam Williamson (1):
ALSA: hda - fix audio on HP TX25xx series notebooks
Andreas Mohr (1):
ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control name
Takashi Iwai (1):
ALSA: pcsp - fix printk format warning again
---
sound/drivers/pcsp/pcsp_mixer.c | 2 +-
sound/pci/ac97/ac97_patch.c | 7 +++++--
sound/pci/ca0106/ca0106_mixer.c | 2 +-
sound/pci/hda/patch_realtek.c | 1 +
4 files changed, 8 insertions(+), 4 deletions(-)
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 771955a..199b033 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -51,7 +51,7 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
if (uinfo->value.enumerated.item > chip->max_treble)
uinfo->value.enumerated.item = chip->max_treble;
sprintf(uinfo->value.enumerated.name, "%lu",
- PCSP_CALC_RATE(uinfo->value.enumerated.item));
+ (unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item));
return 0;
}
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 81bc93e..7337abd 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97)
}
static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker =
-AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch",
+ AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+/* "Sigmatel " removed due to excessive name length: */
static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert =
-AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
+AC97_SINGLE("Surround Phase Inversion Playback Switch",
+ AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = {
AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0),
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index ad28887..c111efe 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
"Capture Volume",
"External Amplifier",
"Sigmatel 4-Speaker Stereo Playback Switch",
- "Sigmatel Surround Phase Inversion Playback ",
+ "Surround Phase Inversion Playback Switch",
NULL
};
static char *ca0106_rename_ctls[] = {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8a0d3e..bcbb736 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12058,6 +12058,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-05-15 13:44 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-05-15 13:44 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.30-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Relative large number of LOCs found here are for Davici fixes after
upstream sync, so pretty safe to others. Otherwise only trivial
fixes.
Thanks!
Takashi
===
David Brownell (3):
ASoC: davinci-pcm buildfixes
ASoC: DaVinci I2S updates
ASoC: DaVinci EVM board support buildfixes
Mike Rapoport (2):
ASoC: soc-core: fix crash when removing not instantiated card
pxa2xx-ac97: fix reset gpio mode setting
Randy Dunlap (1):
ALSA: pcsp: fix printk format warning
Roel Kluin (1):
ALSA: riptide: postfix increment and off by one
---
arch/arm/mach-davinci/include/mach/asp.h | 25 ++++++++++
sound/arm/pxa2xx-ac97-lib.c | 2 +-
sound/drivers/pcsp/pcsp_mixer.c | 2 +-
sound/pci/riptide/riptide.c | 10 +++--
sound/soc/davinci/Kconfig | 7 ++-
sound/soc/davinci/davinci-evm.c | 63 ++++++++++++++++++++++-----
sound/soc/davinci/davinci-i2s.c | 26 ++++++++++-
sound/soc/davinci/davinci-pcm.c | 71 +++++++++++++++++------------
sound/soc/soc-core.c | 3 +
9 files changed, 157 insertions(+), 52 deletions(-)
create mode 100644 arch/arm/mach-davinci/include/mach/asp.h
diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h
new file mode 100644
index 0000000..e0abc43
--- /dev/null
+++ b/arch/arm/mach-davinci/include/mach/asp.h
@@ -0,0 +1,25 @@
+/*
+ * <mach/asp.h> - DaVinci Audio Serial Port support
+ */
+#ifndef __ASM_ARCH_DAVINCI_ASP_H
+#define __ASM_ARCH_DAVINCI_ASP_H
+
+#include <mach/irqs.h>
+
+/* Bases of register banks */
+#define DAVINCI_ASP0_BASE 0x01E02000
+#define DAVINCI_ASP1_BASE 0x01E04000
+
+/* EDMA channels */
+#define DAVINCI_DMA_ASP0_TX 2
+#define DAVINCI_DMA_ASP0_RX 3
+#define DAVINCI_DMA_ASP1_TX 8
+#define DAVINCI_DMA_ASP1_RX 9
+
+/* Interrupts */
+#define DAVINCI_ASP0_RX_INT IRQ_MBRINT
+#define DAVINCI_ASP0_TX_INT IRQ_MBXINT
+#define DAVINCI_ASP1_RX_INT IRQ_MBRINT
+#define DAVINCI_ASP1_TX_INT IRQ_MBXINT
+
+#endif /* __ASM_ARCH_DAVINCI_ASP_H */
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index a2c12d1..6fdca97 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action)
switch (resetgpio_action) {
case RESETGPIO_NORMAL_ALTFUNC:
if (reset_gpio == 113)
- mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
+ mode = 113 | GPIO_ALT_FN_2_OUT;
if (reset_gpio == 95)
mode = 95 | GPIO_ALT_FN_1_OUT;
break;
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index caeb0f5..771955a 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,7 +50,7 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
uinfo->value.enumerated.items = chip->max_treble + 1;
if (uinfo->value.enumerated.item > chip->max_treble)
uinfo->value.enumerated.item = chip->max_treble;
- sprintf(uinfo->value.enumerated.name, "%d",
+ sprintf(uinfo->value.enumerated.name, "%lu",
PCSP_CALC_RATE(uinfo->value.enumerated.item));
return 0;
}
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 6f10344..e51a5ef 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -889,7 +889,7 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
spin_lock_irqsave(&cif->lock, irqflags);
while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport))
udelay(10);
- if (i >= CMDIF_TIMEOUT) {
+ if (i > CMDIF_TIMEOUT) {
err = -EBUSY;
goto errout;
}
@@ -907,8 +907,10 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
WRITE_PORT_ULONG(cmdport->data1, cmd); /* write cmd */
if ((flags & RESP) && ret) {
while (!IS_DATF(cmdport) &&
- time++ < CMDIF_TIMEOUT)
+ time < CMDIF_TIMEOUT) {
udelay(10);
+ time++;
+ }
if (time < CMDIF_TIMEOUT) { /* read response */
ret->retlongs[0] =
READ_PORT_ULONG(cmdport->data1);
@@ -1454,7 +1456,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd)
SEND_GPOS(cif, 0, data->id, &rptr);
udelay(1);
} while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY);
- if (j >= MAX_WRITE_RETRY)
+ if (j > MAX_WRITE_RETRY)
snd_printk(KERN_ERR "Riptide: Could not stop stream!");
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -1783,7 +1785,7 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg,
SEND_SACR(cif, val, reg);
SEND_RACR(cif, reg, &rptr);
} while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY);
- if (i == MAX_WRITE_RETRY)
+ if (i > MAX_WRITE_RETRY)
snd_printdd("Write AC97 reg failed\n");
}
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c..411a710 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
tristate
config SND_DAVINCI_SOC_EVM
- tristate "SoC Audio support for DaVinci EVM"
- depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+ tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+ depends on SND_DAVINCI_SOC
+ depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
select SND_DAVINCI_SOC_I2S
select SND_SOC_TLV320AIC3X
help
Say Y if you want to add support for SoC audio on TI
- DaVinci EVM platform.
+ DaVinci DM6446 or DM355 EVM platforms.
config SND_DAVINCI_SOC_SFFSDR
tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b34..58fd1cb 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
#include <sound/soc-dapm.h>
#include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
#include "../codecs/tlv320aic3x.h"
#include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
/* evm audio private data */
static struct aic3x_setup_data evm_aic3x_setup = {
- .i2c_bus = 0,
+ .i2c_bus = 1,
.i2c_address = 0x1b,
};
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
.codec_data = &evm_aic3x_setup,
};
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
static struct resource evm_snd_resources[] = {
{
- .start = DAVINCI_MCBSP_BASE,
- .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+ .start = DAVINCI_ASP0_BASE,
+ .end = DAVINCI_ASP0_BASE + SZ_8K - 1,
.flags = IORESOURCE_MEM,
},
};
static struct evm_snd_platform_data evm_snd_data = {
- .tx_dma_ch = DM644X_DMACH_MCBSP_TX,
- .rx_dma_ch = DM644X_DMACH_MCBSP_RX,
+ .tx_dma_ch = DAVINCI_DMA_ASP0_TX,
+ .rx_dma_ch = DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+ {
+ .start = DAVINCI_ASP1_BASE,
+ .end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+ .flags = IORESOURCE_MEM,
+ },
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+ .tx_dma_ch = DAVINCI_DMA_ASP1_TX,
+ .rx_dma_ch = DAVINCI_DMA_ASP1_RX,
};
static struct platform_device *evm_snd_device;
static int __init evm_init(void)
{
+ struct resource *resources;
+ unsigned num_resources;
+ struct evm_snd_platform_data *data;
+ int index;
int ret;
- evm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (machine_is_davinci_evm()) {
+ davinci_cfg_reg(DM644X_MCBSP);
+
+ resources = evm_snd_resources;
+ num_resources = ARRAY_SIZE(evm_snd_resources);
+ data = &evm_snd_data;
+ index = 0;
+ } else if (machine_is_davinci_dm355_evm()) {
+ /* we don't use ASP1 IRQs, or we'd need to mux them ... */
+ davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+ davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+ resources = dm335evm_snd_resources;
+ num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+ data = &dm335evm_snd_data;
+ index = 1;
+ } else
+ return -EINVAL;
+
+ evm_snd_device = platform_device_alloc("soc-audio", index);
if (!evm_snd_device)
return -ENOMEM;
platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
evm_snd_devdata.dev = &evm_snd_device->dev;
- platform_device_add_data(evm_snd_device, &evm_snd_data,
- sizeof(evm_snd_data));
+ platform_device_add_data(evm_snd_device, data, sizeof(*data));
- ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
- ARRAY_SIZE(evm_snd_resources));
+ ret = platform_device_add_resources(evm_snd_device, resources,
+ num_resources);
if (ret) {
platform_device_put(evm_snd_device);
return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb943..b1ea52f 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
#include "davinci-pcm.h"
+
+/*
+ * NOTE: terminology here is confusing.
+ *
+ * - This driver supports the "Audio Serial Port" (ASP),
+ * found on dm6446, dm355, and other DaVinci chips.
+ *
+ * - But it labels it a "Multi-channel Buffered Serial Port"
+ * (McBSP) as on older chips like the dm642 ... which was
+ * backward-compatible, possibly explaining that confusion.
+ *
+ * - OMAP chips have a controller called McBSP, which is
+ * incompatible with the DaVinci flavor of McBSP.
+ *
+ * - Newer DaVinci chips have a controller called McASP,
+ * incompatible with ASP and with either McBSP.
+ *
+ * In short: this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
#define DAVINCI_MCBSP_DRR_REG 0x00
#define DAVINCI_MCBSP_DXR_REG 0x04
#define DAVINCI_MCBSP_SPCR_REG 0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
struct davinci_mcbsp_dev *dev;
struct resource *mem, *ioarea;
struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
cpu_dai->private_data = dev;
- dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+ dev->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(dev->clk)) {
ret = -ENODEV;
goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
struct resource *mem;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b..a059965 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
#include <sound/soc.h>
#include <asm/dma.h>
+#include <mach/edma.h>
#include "davinci-pcm.h"
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
spinlock_t lock;
int period; /* current DMA period */
int master_lch; /* Master DMA channel */
- int slave_lch; /* Slave DMA channel */
+ int slave_lch; /* linked parameter RAM reload slot */
struct davinci_pcm_dma_params *params; /* DMA params */
};
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
dst_bidx = data_type;
}
- davinci_set_dma_src_params(lch, src, INCR, W8BIT);
- davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
- davinci_set_dma_src_index(lch, src_bidx, 0);
- davinci_set_dma_dest_index(lch, dst_bidx, 0);
- davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+ edma_set_src(lch, src, INCR, W8BIT);
+ edma_set_dest(lch, dst, INCR, W8BIT);
+ edma_set_src_index(lch, src_bidx, 0);
+ edma_set_dest_index(lch, dst_bidx, 0);
+ edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
prtd->period++;
if (unlikely(prtd->period >= runtime->periods))
prtd->period = 0;
}
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
{
struct snd_pcm_substream *substream = data;
struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
struct davinci_runtime_data *prtd = substream->runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
- int tcc = TCC_ANY;
+ struct edmacc_param p_ram;
int ret;
if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
prtd->params = dma_data;
/* Request master DMA channel */
- ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+ ret = edma_alloc_channel(prtd->params->channel,
davinci_pcm_dma_irq, substream,
- &prtd->master_lch, &tcc, EVENTQ_0);
- if (ret)
+ EVENTQ_0);
+ if (ret < 0)
return ret;
+ prtd->master_lch = ret;
- /* Request slave DMA channel */
- ret = davinci_request_dma(PARAM_ANY, "Link",
- NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
- if (ret) {
- davinci_free_dma(prtd->master_lch);
+ /* Request parameter RAM reload slot */
+ ret = edma_alloc_slot(EDMA_SLOT_ANY);
+ if (ret < 0) {
+ edma_free_channel(prtd->master_lch);
return ret;
}
-
- /* Link slave DMA channel in loopback */
- davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+ prtd->slave_lch = ret;
+
+ /* Issue transfer completion IRQ when the channel completes a
+ * transfer, then always reload from the same slot (by a kind
+ * of loopback link). The completion IRQ handler will update
+ * the reload slot with a new buffer.
+ *
+ * REVISIT save p_ram here after setting up everything except
+ * the buffer and its length (ccnt) ... use it as a template
+ * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+ */
+ edma_read_slot(prtd->slave_lch, &p_ram);
+ p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+ p_ram.link_bcntrld = prtd->slave_lch << 5;
+ edma_write_slot(prtd->slave_lch, &p_ram);
return 0;
}
@@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- davinci_start_dma(prtd->master_lch);
+ edma_start(prtd->master_lch);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- davinci_stop_dma(prtd->master_lch);
+ edma_stop(prtd->master_lch);
break;
default:
ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
- struct paramentry_descriptor temp;
+ struct edmacc_param temp;
prtd->period = 0;
davinci_pcm_enqueue_dma(substream);
- /* Get slave channel dma params for master channel startup */
- davinci_get_dma_params(prtd->slave_lch, &temp);
- davinci_set_dma_params(prtd->master_lch, &temp);
+ /* Copy self-linked parameter RAM entry into master channel */
+ edma_read_slot(prtd->slave_lch, &temp);
+ edma_write_slot(prtd->master_lch, &temp);
return 0;
}
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
spin_lock(&prtd->lock);
- davinci_dma_getposition(prtd->master_lch, &src, &dst);
+ edma_get_position(prtd->master_lch, &src, &dst);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
count = src - runtime->dma_addr;
else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct davinci_runtime_data *prtd = runtime->private_data;
- davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+ edma_unlink(prtd->slave_lch);
- davinci_free_dma(prtd->slave_lch);
- davinci_free_dma(prtd->master_lch);
+ edma_free_slot(prtd->slave_lch);
+ edma_free_channel(prtd->master_lch);
kfree(prtd);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f6..1cd149b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev)
struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ if (!card->instantiated)
+ return 0;
+
run_delayed_work(&card->delayed_work);
if (platform->remove)
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-05-10 10:15 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-05-10 10:15 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA updates for v2.6.30-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains the following trivial fixes.
Thanks!
Takashi
===
Clemens Ladisch (2):
sound: serial-u16550: fix buffer overflow
sound: via82xx: fix DXS volume range
Jaroslav Kysela (1):
ALSA: Release v1.0.20
Jinyoung Park (1):
ASoC: Fix errors in WM8990
Takashi Iwai (1):
ALSA: hda - Fix line-in on Mac Mini Core2 Duo
---
include/sound/version.h | 2 +-
sound/drivers/serial-u16550.c | 11 +++--------
sound/pci/hda/patch_sigmatel.c | 7 ++++++-
sound/pci/via82xx.c | 2 +-
sound/soc/codecs/wm8990.c | 40 ++++++++++++++++++++--------------------
5 files changed, 31 insertions(+), 31 deletions(-)
diff --git a/include/sound/version.h b/include/sound/version.h
index a7e74e2..456f135 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
/* include/version.h */
-#define CONFIG_SND_VERSION "1.0.19"
+#define CONFIG_SND_VERSION "1.0.20"
#define CONFIG_SND_DATE ""
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index b2b6d50..a25fb7b 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -963,16 +963,11 @@ static int __devinit snd_serial_probe(struct platform_device *devptr)
if (err < 0)
goto _err;
- sprintf(card->longname, "%s at 0x%lx, irq %d speed %d div %d outs %d ins %d adaptor %s droponfull %d",
+ sprintf(card->longname, "%s [%s] at %#lx, irq %d",
card->shortname,
- uart->base,
- uart->irq,
- uart->speed,
- (int)uart->divisor,
- outs[dev],
- ins[dev],
adaptor_names[uart->adaptor],
- uart->drop_on_full);
+ uart->base,
+ uart->irq);
snd_card_set_dev(card, &devptr->dev);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 917bc5d..03b3646 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4079,7 +4079,12 @@ static int stac92xx_init(struct hda_codec *codec)
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)) {
+ /* Also, if both INPUT and OUTPUT are set,
+ * it must be a BIOS bug; need to override, too
+ */
+ if (!(pinctl & AC_PINCTL_IN_EN) ||
+ (pinctl & AC_PINCTL_OUT_EN)) {
+ pinctl &= ~AC_PINCTL_OUT_EN;
pinctl |= AC_PINCTL_IN_EN;
stac92xx_auto_set_pinctl(codec, nid,
pinctl);
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 809b233..1ef58c5 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1687,7 +1687,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol,
return change;
}
-static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1);
static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = {
.name = "PCM Playback Volume",
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e..40cd274 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
&wm8990_dapm_ainlmux_controls, inmixer_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
&wm8990_dapm_ainrmux_controls, inmixer_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"LIN12 PGA", "LIN2 Switch", "LIN2"},
/* LIN34 PGA */
{"LIN34 PGA", "LIN3 Switch", "LIN3"},
- {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+ {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
/* INMIXL */
{"INMIXL", "Record Left Volume", "LOMIX"},
{"INMIXL", "LIN2 Volume", "LIN2"},
{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
- /* AILNMUX */
- {"AILNMUX", "INMIXL Mix", "INMIXL"},
- {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
- {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
- {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
- {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* AINLMUX */
+ {"AINLMUX", "INMIXL Mix", "INMIXL"},
+ {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+ {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+ {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
/* ADC */
- {"Left ADC", NULL, "AILNMUX"},
+ {"Left ADC", NULL, "AINLMUX"},
/* RIN12 PGA */
{"RIN12 PGA", "RIN1 Switch", "RIN1"},
{"RIN12 PGA", "RIN2 Switch", "RIN2"},
/* RIN34 PGA */
{"RIN34 PGA", "RIN3 Switch", "RIN3"},
- {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+ {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
/* INMIXL */
{"INMIXR", "Record Right Volume", "ROMIX"},
{"INMIXR", "RIN2 Volume", "RIN2"},
{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
- /* AIRNMUX */
- {"AIRNMUX", "INMIXR Mix", "INMIXR"},
- {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
- {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
- {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
- {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* AINRMUX */
+ {"AINRMUX", "INMIXR Mix", "INMIXR"},
+ {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+ {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+ {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
/* ADC */
- {"Right ADC", NULL, "AIRNMUX"},
+ {"Right ADC", NULL, "AINRMUX"},
/* LOMIX */
{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
/* OUT3MIX */
- {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+ {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
/* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
/* Output Pins */
{"LON", NULL, "LONMIX"},
{"LOP", NULL, "LOPMIX"},
- {"OUT", NULL, "OUT3MIX"},
+ {"OUT3", NULL, "OUT3MIX"},
{"LOUT", NULL, "LOUT PGA"},
{"SPKN", NULL, "SPKMIX"},
{"ROUT", NULL, "ROUT PGA"},
^ permalink raw reply related [flat|nested] 60+ messages in thread
* [GIT PULL] sound fixes
@ 2009-05-04 15:34 Takashi Iwai
0 siblings, 0 replies; 60+ messages in thread
From: Takashi Iwai @ 2009-05-04 15:34 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Andrew Morton, linux-kernel
Linus,
please pull ALSA fixes for v2.6.30-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following small fixes.
Thanks!
Takashi
===
Giuliano Pochini (1):
ALSA: indigo-express: add missing 64KHz flags
Jon Smirl (1):
ASoC: Set the MPC5200 i2s driver to BROKEN status.
Joonyoung Shim (1):
ASoC: TWL4030: Fix gain control for earpiece amplifier
Mark Brown (1):
ASoC: Fix logic in WM8350 master clocking check
Takashi Iwai (3):
ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some drivers
ALSA: pcm core - Avoid jiffies check for devices with BATCH flag
ASoC: Remove BROKEN from mpc5200 kconfig
---
sound/core/pcm_lib.c | 7 +++++++
sound/isa/msnd/msnd.c | 6 ++++--
sound/pci/bt87x.c | 6 ++++--
sound/pci/echoaudio/indigodjx.c | 1 +
sound/pci/echoaudio/indigoiox.c | 1 +
sound/pci/korg1212/korg1212.c | 6 ++++--
sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 3 ++-
sound/soc/au1x/dbdma2.c | 2 +-
sound/soc/codecs/twl4030.c | 8 +++++++-
sound/soc/codecs/wm8350.c | 2 +-
sound/soc/fsl/mpc5200_psc_i2s.c | 3 ++-
sound/soc/sh/dma-sh7760.c | 3 ++-
sound/sparc/dbri.c | 3 ++-
sound/usb/usx2y/usbusx2yaudio.c | 3 ++-
14 files changed, 40 insertions(+), 14 deletions(-)
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 63d088f..a2a792c 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,12 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
new_hw_ptr = hw_base + pos;
}
}
+ /* Skip the jiffies check for hardwares with BATCH flag.
+ * Such hardware usually just increases the position at each IRQ,
+ * thus it can't give any strange position.
+ */
+ if (runtime->hw.info & SNDRV_PCM_INFO_BATCH)
+ goto no_jiffies_check;
hdelta = new_hw_ptr - old_hw_ptr;
jdelta = jiffies - runtime->hw_ptr_jiffies;
if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
@@ -272,6 +278,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
hw_base -= hw_base % runtime->buffer_size;
delta = 0;
}
+ no_jiffies_check:
if (delta > runtime->period_size + runtime->period_size / 2) {
hw_ptr_error(substream,
"Lost interrupts? "
diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
index 9064544..3a1526a 100644
--- a/sound/isa/msnd/msnd.c
+++ b/sound/isa/msnd/msnd.c
@@ -438,7 +438,8 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip,
static struct snd_pcm_hardware snd_msnd_playback = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
@@ -456,7 +457,8 @@ static struct snd_pcm_hardware snd_msnd_playback = {
static struct snd_pcm_hardware snd_msnd_capture = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index a299340..ce3f2e9 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -349,7 +349,8 @@ static struct snd_pcm_hardware snd_bt87x_digital_hw = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = 0, /* set at runtime */
.channels_min = 2,
@@ -365,7 +366,8 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
.rates = SNDRV_PCM_RATE_KNOT,
.rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX,
diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
index 3482ef6..2e44316 100644
--- a/sound/pci/echoaudio/indigodjx.c
+++ b/sound/pci/echoaudio/indigodjx.c
@@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
.rates = SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000,
.rate_min = 32000,
diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c
index aebee27..eb3819f 100644
--- a/sound/pci/echoaudio/indigoiox.c
+++ b/sound/pci/echoaudio/indigoiox.c
@@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
.rates = SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000,
.rate_min = 32000,
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8b79969..7cc38a1 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1238,7 +1238,8 @@ static struct snd_pcm_hardware snd_korg1212_playback_info =
{
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED),
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = (SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000),
@@ -1258,7 +1259,8 @@ static struct snd_pcm_hardware snd_korg1212_capture_info =
{
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED),
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = (SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000),
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 01066c9..d057e64 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -240,7 +240,8 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
static struct snd_pcm_hardware pdacf_pcm_capture_hw = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_MMAP_VALID),
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a2..594c6c5 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
/* PCM hardware DMA capabilities - platform specific */
static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED,
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
.formats = AU1XPSC_PCM_FMTS,
.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
.period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205..df7c8c2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
/*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
* Capture gain after the ADCs
* from 0 dB to 31 dB in 1 dB steps
*/
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
4, 3, 0, output_tvl),
SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
- TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+ TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
/* Common capture gain controls */
SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d099..0275321 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
* required for LRC in master mode. The DACs or ADCs need a
* valid audio path i.e. pin -> ADC or DAC -> pin before
* the LRC will be enabled in master mode. */
- if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+ if (!master || cmd != SNDRV_PCM_TRIGGER_START)
return 0;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729d..1111c71 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
.rate_min = 8000,
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0..baddb12 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID),
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH),
.formats = DMABRG_FMTS,
.rates = DMABRG_RATES,
.rate_min = 8000,
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index af95ff1..1d2e51b 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -1975,7 +1975,8 @@ static struct snd_pcm_hardware snd_dbri_pcm_hw = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_MU_LAW |
SNDRV_PCM_FMTBIT_A_LAW |
SNDRV_PCM_FMTBIT_U8 |
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 9a608fa..dd1ab61 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -870,7 +870,8 @@ static struct snd_pcm_hardware snd_usX2Y_2c =
{
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID),
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE,
.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.rate_min = 44100,
^ permalink raw reply related [flat|nested] 60+ messages in thread
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