| lists.openwall.net | lists / announce owl-users owl-dev john-users john-dev passwdqc-users yescrypt popa3d-users / oss-security kernel-hardening musl sabotage tlsify passwords / crypt-dev xvendor / Bugtraq Full-Disclosure linux-kernel linux-netdev linux-ext4 linux-hardening linux-cve-announce PHC | |
|
Open Source and information security mailing list archives
| ||
|
Date: Fri, 21 Jan 2011 08:40:12 +0100
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Jaroslav Kysela <perex@...ex.cz>,
Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lrg@...mlogic.co.uk>,
Andrew Morton <akpm@...ux-foundation.org>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes
Linus,
please pull sound fixes for v2.6.38-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,
};
--
To unsubscribe from this list: send the line "unsubscribe linux-kernel" in
the body of a message to majordomo@...r.kernel.org
More majordomo info at http://vger.kernel.org/majordomo-info.html
Please read the FAQ at http://www.tux.org/lkml/
Powered by blists - more mailing lists