[GIT PULL] sound fixes

[Takashi Iwai <tiwai@suse.de>]

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,
 };