Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (22 commits)
  ALSA: hda - Cirrus Logic CS421x support
  ALSA: Make pcm.h self-contained
  ALSA: hda - Allow codec-specific set_power_state ops
  ALSA: hda - Add post_suspend patch ops
  ALSA: hda - Make CONFIG_SND_HDA_POWER_SAVE depending on CONFIG_PM
  ALSA: hda - Make sure mute led reflects master mute state
  ALSA: hda - Fix invalid mute led state on resume of IDT codecs
  ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
  ALSA: hda - Add support of the 4 internal speakers on certain HP laptops
  ALSA: Make snd_pcm_debug_name usable outside pcm_lib
  ALSA: hda - Fix DAC filling for multi-connection pins in Realtek parser
  ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
  ASoC: SAMSUNG: Add I2S0 internal dma driver
  ASoC: SAMSUNG: Modify I2S driver to support idma
  ASoC: davinci: add missing break statement
  ASoC: davinci: fix codec start and stop functions
  ASoC: dapm - add DAPM macro for external enum widgets
  ASoC: Acknowledge WM8962 interrupts before acting on them
  ASoC: sgtl5000: guide user when regulator support is needed
  ASoC: sgtl5000: refactor registering internal ldo
  ...

include/sound/pcm.h

@@ -507,6 +507,18 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream);
 void snd_pcm_vma_notify_data(void *client, void *data);
 int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area);
 
+
+#ifdef CONFIG_SND_DEBUG
+void snd_pcm_debug_name(struct snd_pcm_substream *substream,
+			   char *name, size_t len);
+#else
+static inline void
+snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
+{
+	*buf = 0;
+}
+#endif
+
 /*
  *  PCM library
  */
@@ -749,17 +761,18 @@ static inline const struct snd_interval *hw_param_interval_c(const struct snd_pc
 	return &params->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL];
 }
 
-#define params_access(p) ((__force snd_pcm_access_t)snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS)))
-#define params_format(p) ((__force snd_pcm_format_t)snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_FORMAT)))
-#define params_subformat(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_SUBFORMAT))
-#define params_channels(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_CHANNELS)->min
-#define params_rate(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_RATE)->min
-#define params_period_size(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_PERIOD_SIZE)->min
-#define params_period_bytes(p) ((params_period_size(p)*snd_pcm_format_physical_width(params_format(p))*params_channels(p))/8)
-#define params_periods(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_PERIODS)->min
-#define params_buffer_size(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min
-#define params_buffer_bytes(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min
-
+#define params_channels(p) \
+	(hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_CHANNELS)->min)
+#define params_rate(p) \
+	(hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_RATE)->min)
+#define params_period_size(p) \
+	(hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_PERIOD_SIZE)->min)
+#define params_periods(p) \
+	(hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_PERIODS)->min)
+#define params_buffer_size(p) \
+	(hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min)
+#define params_buffer_bytes(p) \
+	(hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min)
 
 int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v);
 void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c);

include/sound/pcm_params.h

@@ -337,5 +337,19 @@ static inline unsigned int sub(unsigned int a, unsigned int b)
 	return 0;
 }
 
-#endif /* __SOUND_PCM_PARAMS_H */
+#define params_access(p) ((__force snd_pcm_access_t)\
+		snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_ACCESS)))
+#define params_format(p) ((__force snd_pcm_format_t)\
+		snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_FORMAT)))
+#define params_subformat(p) \
+	snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_SUBFORMAT))
 
+static inline unsigned int
+params_period_bytes(const struct snd_pcm_hw_params *p)
+{
+	return (params_period_size(p) *
+		snd_pcm_format_physical_width(params_format(p)) *
+		params_channels(p)) / 8;
+}
+
+#endif /* __SOUND_PCM_PARAMS_H */

include/sound/soc-dapm.h

@@ -266,6 +266,12 @@
 	.get = snd_soc_dapm_get_enum_virt, \
 	.put = snd_soc_dapm_put_enum_virt, \
 	.private_value = (unsigned long)&xenum }
+#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = snd_soc_info_enum_double, \
+	.get = xget, \
+	.put = xput, \
+	.private_value = (unsigned long)&xenum }
 #define SOC_DAPM_VALUE_ENUM(xname, xenum) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_enum_double, \

sound/core/pcm_lib.c

@@ -128,7 +128,8 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram
 	}
 }
 
-static void pcm_debug_name(struct snd_pcm_substream *substream,
+#ifdef CONFIG_SND_DEBUG
+void snd_pcm_debug_name(struct snd_pcm_substream *substream,
 			   char *name, size_t len)
 {
 	snprintf(name, len, "pcmC%dD%d%c:%d",
@@ -137,6 +138,8 @@ static void pcm_debug_name(struct snd_pcm_substream *substream,
 		 substream->stream ? 'c' : 'p',
 		 substream->number);
 }
+EXPORT_SYMBOL(snd_pcm_debug_name);
+#endif
 
 #define XRUN_DEBUG_BASIC	(1<<0)
 #define XRUN_DEBUG_STACK	(1<<1)	/* dump also stack */
@@ -168,7 +171,7 @@ static void xrun(struct snd_pcm_substream *substream)
 	snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
 	if (xrun_debug(substream, XRUN_DEBUG_BASIC)) {
 		char name[16];
-		pcm_debug_name(substream, name, sizeof(name));
+		snd_pcm_debug_name(substream, name, sizeof(name));
 		snd_printd(KERN_DEBUG "XRUN: %s\n", name);
 		dump_stack_on_xrun(substream);
 	}
@@ -243,7 +246,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream)
 		return;
 	if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit)
 		return;
-	pcm_debug_name(substream, name, sizeof(name));
+	snd_pcm_debug_name(substream, name, sizeof(name));
 	for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) {
 		entry = &log->entries[idx];
 		if (entry->period_size == 0)
@@ -319,7 +322,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	if (pos >= runtime->buffer_size) {
 		if (printk_ratelimit()) {
 			char name[16];
-			pcm_debug_name(substream, name, sizeof(name));
+			snd_pcm_debug_name(substream, name, sizeof(name));
 			xrun_log_show(substream);
 			snd_printd(KERN_ERR  "BUG: %s, pos = %ld, "
 				   "buffer size = %ld, period size = %ld\n",
@@ -364,7 +367,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	if (xrun_debug(substream, in_interrupt ?
 			XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) {
 		char name[16];
-		pcm_debug_name(substream, name, sizeof(name));
+		snd_pcm_debug_name(substream, name, sizeof(name));
 		snd_printd("%s_update: %s: pos=%u/%u/%u, "
 			   "hwptr=%ld/%ld/%ld/%ld\n",
 			   in_interrupt ? "period" : "hwptr",

sound/pci/asihpi/asihpi.c

@@ -41,31 +41,10 @@
 #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
-/* copied from pcm_lib.c, hope later patch will make that version public
-and this copy can be removed */
-static inline void
-snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
-{
-	snprintf(buf, size, "pcmC%dD%d%c:%d",
-		 substream->pcm->card->number,
-		 substream->pcm->device,
-		 substream->stream ? 'c' : 'p',
-		 substream->number);
-}
-#else
-static inline void
-snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
-{
-	*buf = 0;
-}
-#endif
-
 #if defined CONFIG_SND_DEBUG_VERBOSE
 /**
  * snd_printddd - very verbose debug printk

sound/pci/hda/Kconfig

@@ -243,6 +243,7 @@ config SND_HDA_GENERIC
 
 config SND_HDA_POWER_SAVE
 	bool "Aggressive power-saving on HD-audio"
+	depends on PM
 	help
 	  Say Y here to enable more aggressive power-saving mode on
 	  HD-audio driver.  The power-saving timeout can be configured

sound/pci/hda/hda_codec.c

@@ -91,8 +91,10 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void hda_power_work(struct work_struct *work);
 static void hda_keep_power_on(struct hda_codec *codec);
+#define hda_codec_is_power_on(codec)	((codec)->power_on)
 #else
 static inline void hda_keep_power_on(struct hda_codec *codec) {}
+#define hda_codec_is_power_on(codec)	1
 #endif
 
 /**
@@ -1101,7 +1103,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
 static void restore_shutup_pins(struct hda_codec *codec)
 {
@@ -1499,7 +1501,7 @@ static void purify_inactive_streams(struct hda_codec *codec)
 	}
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /* clean up all streams; called from suspend */
 static void hda_cleanup_all_streams(struct hda_codec *codec)
 {
@@ -1838,7 +1840,7 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo);
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /**
  * snd_hda_codec_resume_amp - Resume all AMP commands from the cache
  * @codec: HD-audio codec
@@ -1868,7 +1870,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec)
 	}
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp);
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
 static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir,
 			     unsigned int ofs)
@@ -3082,7 +3084,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 }
 EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls);
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /*
  * command cache
  */
@@ -3199,53 +3201,32 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec,
 					  seq->param);
 }
 EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache);
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
-/*
- * set power state of the codec
- */
-static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
-				unsigned int power_state)
+void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
+				    unsigned int power_state,
+				    bool eapd_workaround)
 {
-	hda_nid_t nid;
+	hda_nid_t nid = codec->start_nid;
 	int i;
 
-	/* this delay seems necessary to avoid click noise at power-down */
-	if (power_state == AC_PWRST_D3)
-		msleep(100);
-	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
-			    power_state);
-	/* partial workaround for "azx_get_response timeout" */
-	if (power_state == AC_PWRST_D0 &&
-	    (codec->vendor_id & 0xffff0000) == 0x14f10000)
-		msleep(10);
-
-	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		unsigned int wcaps = get_wcaps(codec, nid);
-		if (wcaps & AC_WCAP_POWER) {
-			unsigned int wid_type = get_wcaps_type(wcaps);
-			if (power_state == AC_PWRST_D3 &&
-			    wid_type == AC_WID_PIN) {
-				unsigned int pincap;
-				/*
-				 * don't power down the widget if it controls
-				 * eapd and EAPD_BTLENABLE is set.
-				 */
-				pincap = snd_hda_query_pin_caps(codec, nid);
-				if (pincap & AC_PINCAP_EAPD) {
-					int eapd = snd_hda_codec_read(codec,
-						nid, 0,
+		if (!(wcaps & AC_WCAP_POWER))
+			continue;
+		/* don't power down the widget if it controls eapd and
+		 * EAPD_BTLENABLE is set.
+		 */
+		if (eapd_workaround && power_state == AC_PWRST_D3 &&
+		    get_wcaps_type(wcaps) == AC_WID_PIN &&
+		    (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) {
+			int eapd = snd_hda_codec_read(codec, nid, 0,
 						AC_VERB_GET_EAPD_BTLENABLE, 0);
-					eapd &= 0x02;
-					if (eapd)
-						continue;
-				}
-			}
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    power_state);
+			if (eapd & 0x02)
+				continue;
 		}
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE,
+				    power_state);
 	}
 
 	if (power_state == AC_PWRST_D0) {
@@ -3262,6 +3243,26 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 		} while (time_after_eq(end_time, jiffies));
 	}
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all);
+
+/*
+ * set power state of the codec
+ */
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state)
+{
+	if (codec->patch_ops.set_power_state) {
+		codec->patch_ops.set_power_state(codec, fg, power_state);
+		return;
+	}
+
+	/* this delay seems necessary to avoid click noise at power-down */
+	if (power_state == AC_PWRST_D3)
+		msleep(100);
+	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+			    power_state);
+	snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+}
 
 #ifdef CONFIG_SND_HDA_HWDEP
 /* execute additional init verbs */
@@ -3274,7 +3275,7 @@ static void hda_exec_init_verbs(struct hda_codec *codec)
 static inline void hda_exec_init_verbs(struct hda_codec *codec) {}
 #endif
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 /*
  * call suspend and power-down; used both from PM and power-save
  */
@@ -3315,7 +3316,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 		snd_hda_codec_resume_cache(codec);
 	}
 }
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
 
 /**
@@ -4071,9 +4072,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec,
 EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls);
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
-				unsigned int power_state);
-
 static void hda_power_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
@@ -4376,11 +4374,8 @@ void snd_hda_bus_reboot_notify(struct hda_bus *bus)
 	if (!bus)
 		return;
 	list_for_each_entry(codec, &bus->codec_list, list) {
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-		if (!codec->power_on)
-			continue;
-#endif
-		if (codec->patch_ops.reboot_notify)
+		if (hda_codec_is_power_on(codec) &&
+		    codec->patch_ops.reboot_notify)
 			codec->patch_ops.reboot_notify(codec);
 	}
 }
@@ -5079,11 +5074,10 @@ int snd_hda_suspend(struct hda_bus *bus)
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-		if (!codec->power_on)
-			continue;
-#endif
-		hda_call_codec_suspend(codec);
+		if (hda_codec_is_power_on(codec))
+			hda_call_codec_suspend(codec);
+		if (codec->patch_ops.post_suspend)
+			codec->patch_ops.post_suspend(codec);
 	}
 	return 0;
 }
@@ -5103,6 +5097,8 @@ int snd_hda_resume(struct hda_bus *bus)
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
+		if (codec->patch_ops.pre_resume)
+			codec->patch_ops.pre_resume(codec);
 		if (snd_hda_codec_needs_resume(codec))
 			hda_call_codec_resume(codec);
 	}

sound/pci/hda/hda_codec.h

@@ -26,10 +26,6 @@
 #include <sound/pcm.h>
 #include <sound/hwdep.h>
 
-#if defined(CONFIG_PM) || defined(CONFIG_SND_HDA_POWER_SAVE)
-#define SND_HDA_NEEDS_RESUME	/* resume control code is required */
-#endif
-
 /*
  * nodes
  */
@@ -704,8 +700,12 @@ struct hda_codec_ops {
 	int (*init)(struct hda_codec *codec);
 	void (*free)(struct hda_codec *codec);
 	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
-#ifdef SND_HDA_NEEDS_RESUME
+	void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state);
+#ifdef CONFIG_PM
 	int (*suspend)(struct hda_codec *codec, pm_message_t state);
+	int (*post_suspend)(struct hda_codec *codec);
+	int (*pre_resume)(struct hda_codec *codec);
 	int (*resume)(struct hda_codec *codec);
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -927,7 +927,7 @@ void snd_hda_sequence_write(struct hda_codec *codec,
 int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex);
 
 /* cached write */
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
 			      int direct, unsigned int verb, unsigned int parm);
 void snd_hda_sequence_write_cache(struct hda_codec *codec,
@@ -1008,6 +1008,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
  */
 void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen);
 void snd_hda_bus_reboot_notify(struct hda_bus *bus);
+void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
+				    unsigned int power_state,
+				    bool eapd_workaround);
 
 /*
  * power management

sound/pci/hda/hda_local.h

@@ -131,7 +131,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 			     int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
 			     int dir, int idx, int mask, int val);
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 void snd_hda_codec_resume_amp(struct hda_codec *codec);
 #endif
 

sound/pci/hda/patch_analog.c

@@ -563,7 +563,7 @@ static void ad198x_free(struct hda_codec *codec)
 	snd_hda_detach_beep_device(codec);
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	ad198x_shutup(codec);
@@ -579,7 +579,7 @@ static const struct hda_codec_ops ad198x_patch_ops = {
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	.check_power_status = ad198x_check_power_status,
 #endif
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.suspend = ad198x_suspend,
 #endif
 	.reboot_notify = ad198x_shutup,

sound/pci/hda/patch_cirrus.c

@@ -25,6 +25,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include <sound/tlv.h>
 
 /*
  */
@@ -61,9 +62,15 @@ struct cs_spec {
 
 	unsigned int hp_detect:1;
 	unsigned int mic_detect:1;
+	/* CS421x */
+	unsigned int spdif_detect:1;
+	unsigned int sense_b:1;
+	hda_nid_t vendor_nid;
+	struct hda_input_mux input_mux;
+	unsigned int last_input;
 };
 
-/* available models */
+/* available models with CS420x */
 enum {
 	CS420X_MBP53,
 	CS420X_MBP55,
@@ -72,6 +79,12 @@ enum {
 	CS420X_MODELS
 };
 
+/* CS421x boards */
+enum {
+	CS421X_CDB4210,
+	CS421X_MODELS
+};
+
 /* Vendor-specific processing widget */
 #define CS420X_VENDOR_NID	0x11
 #define CS_DIG_OUT1_PIN_NID	0x10
@@ -111,21 +124,42 @@ enum {
 /* 0x0009 - 0x0014 -> 12 test regs */
 /* 0x0015 - visibility reg */
 
+/*
+ * Cirrus Logic CS4210
+ *
+ * 1 DAC => HP(sense) / Speakers,
+ * 1 ADC <= LineIn(sense) / MicIn / DMicIn,
+ * 1 SPDIF OUT => SPDIF Trasmitter(sense)
+*/
+#define CS4210_DAC_NID		0x02
+#define CS4210_ADC_NID		0x03
+#define CS421X_VENDOR_NID	0x0B
+#define CS421X_DMIC_PIN_NID	0x09 /* Port E */
+#define CS421X_SPDIF_PIN_NID	0x0A /* Port H */
+
+#define CS421X_IDX_DEV_CFG	0x01
+#define CS421X_IDX_ADC_CFG	0x02
+#define CS421X_IDX_DAC_CFG	0x03
+#define CS421X_IDX_SPK_CTL	0x04
+
+#define SPDIF_EVENT		0x04
 
 static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
 {
-	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+	struct cs_spec *spec = codec->spec;
+	snd_hda_codec_write(codec, spec->vendor_nid, 0,
 			    AC_VERB_SET_COEF_INDEX, idx);
-	return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0,
+	return snd_hda_codec_read(codec, spec->vendor_nid, 0,
 				  AC_VERB_GET_PROC_COEF, 0);
 }
 
 static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx,
 				      unsigned int coef)
 {
-	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+	struct cs_spec *spec = codec->spec;
+	snd_hda_codec_write(codec, spec->vendor_nid, 0,
 			    AC_VERB_SET_COEF_INDEX, idx);
-	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+	snd_hda_codec_write(codec, spec->vendor_nid, 0,
 			    AC_VERB_SET_PROC_COEF, coef);
 }
 
@@ -347,15 +381,12 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		unsigned int type;
-		int idx;
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		idx = snd_hda_get_conn_index(codec, nid, pin, 0);
-		if (idx >= 0) {
-			*idxp = idx;
+		*idxp = snd_hda_get_conn_index(codec, nid, pin, false);
+		if (*idxp >= 0)
 			return nid;
-		}
 	}
 	return 0;
 }
@@ -835,6 +866,8 @@ static int build_digital_input(struct hda_codec *codec)
 
 /*
  * auto-mute and auto-mic switching
+ * CS421x auto-output redirecting
+ * HP/SPK/SPDIF
  */
 
 static void cs_automute(struct hda_codec *codec)
@@ -842,9 +875,25 @@ static void cs_automute(struct hda_codec *codec)
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	unsigned int hp_present;
+	unsigned int spdif_present;
 	hda_nid_t nid;
 	int i;
 
+	spdif_present = 0;
+	if (cfg->dig_outs) {
+		nid = cfg->dig_out_pins[0];
+		if (is_jack_detectable(codec, nid)) {
+			/*
+			TODO: SPDIF output redirect when SENSE_B is enabled.
+			Shared (SENSE_A) jack (e.g HP/mini-TOSLINK)
+			assumed.
+			*/
+			if (snd_hda_jack_detect(codec, nid)
+				/* && spec->sense_b */)
+				spdif_present = 1;
+		}
+	}
+
 	hp_present = 0;
 	for (i = 0; i < cfg->hp_outs; i++) {
 		nid = cfg->hp_pins[i];
@@ -854,11 +903,19 @@ static void cs_automute(struct hda_codec *codec)
 		if (hp_present)
 			break;
 	}
+
+	/* mute speakers if spdif or hp jack is plugged in */
 	for (i = 0; i < cfg->speaker_outs; i++) {
 		nid = cfg->speaker_pins[i];
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL,
 				    hp_present ? 0 : PIN_OUT);
+		/* detect on spdif is specific to CS421x */
+		if (spec->vendor_nid == CS421X_VENDOR_NID) {
+			snd_hda_codec_write(codec, nid, 0,
+					AC_VERB_SET_PIN_WIDGET_CONTROL,
+					spdif_present ? 0 : PIN_OUT);
+		}
 	}
 	if (spec->board_config == CS420X_MBP53 ||
 	    spec->board_config == CS420X_MBP55 ||
@@ -867,21 +924,62 @@ static void cs_automute(struct hda_codec *codec)
 		snd_hda_codec_write(codec, 0x01, 0,
 				    AC_VERB_SET_GPIO_DATA, gpio);
 	}
+
+	/* specific to CS421x */
+	if (spec->vendor_nid == CS421X_VENDOR_NID) {
+		/* mute HPs if spdif jack (SENSE_B) is present */
+		for (i = 0; i < cfg->hp_outs; i++) {
+			nid = cfg->hp_pins[i];
+			snd_hda_codec_write(codec, nid, 0,
+				AC_VERB_SET_PIN_WIDGET_CONTROL,
+				(spdif_present && spec->sense_b) ? 0 : PIN_HP);
+		}
+
+		/* SPDIF TX on/off */
+		if (cfg->dig_outs) {
+			nid = cfg->dig_out_pins[0];
+			snd_hda_codec_write(codec, nid, 0,
+				AC_VERB_SET_PIN_WIDGET_CONTROL,
+				spdif_present ? PIN_OUT : 0);
+
+		}
+		/* Update board GPIOs if neccessary ... */
+	}
 }
 
+/*
+ * Auto-input redirect for CS421x
+ * Switch max 3 inputs of a single ADC (nid 3)
+*/
+
 static void cs_automic(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	hda_nid_t nid;
 	unsigned int present;
-	
+
 	nid = cfg->inputs[spec->automic_idx].pin;
 	present = snd_hda_jack_detect(codec, nid);
-	if (present)
-		change_cur_input(codec, spec->automic_idx, 0);
-	else
-		change_cur_input(codec, !spec->automic_idx, 0);
+
+	/* specific to CS421x, single ADC */
+	if (spec->vendor_nid == CS421X_VENDOR_NID) {
+		if (present) {
+			spec->last_input = spec->cur_input;
+			spec->cur_input = spec->automic_idx;
+		} else  {
+			spec->cur_input = spec->last_input;
+		}
+
+		snd_hda_codec_write_cache(codec, spec->cur_adc, 0,
+					AC_VERB_SET_CONNECT_SEL,
+					spec->adc_idx[spec->cur_input]);
+	} else {
+		if (present)
+			change_cur_input(codec, spec->automic_idx, 0);
+		else
+			change_cur_input(codec, !spec->automic_idx, 0);
+	}
 }
 
 /*
@@ -911,23 +1009,28 @@ static void init_output(struct hda_codec *codec)
 	for (i = 0; i < cfg->line_outs; i++)
 		snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	/* HP */
 	for (i = 0; i < cfg->hp_outs; i++) {
 		hda_nid_t nid = cfg->hp_pins[i];
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
 		if (!cfg->speaker_outs)
 			continue;
-		if (is_jack_detectable(codec, nid)) {
+		if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
 			snd_hda_codec_write(codec, nid, 0,
 					    AC_VERB_SET_UNSOLICITED_ENABLE,
 					    AC_USRSP_EN | HP_EVENT);
 			spec->hp_detect = 1;
 		}
 	}
+
+	/* Speaker */
 	for (i = 0; i < cfg->speaker_outs; i++)
 		snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-	if (spec->hp_detect)
+
+	/* SPDIF is enabled on presence detect for CS421x */
+	if (spec->hp_detect || spec->spdif_detect)
 		cs_automute(codec);
 }
 
@@ -961,19 +1064,31 @@ static void init_input(struct hda_codec *codec)
 					    AC_VERB_SET_UNSOLICITED_ENABLE,
 					    AC_USRSP_EN | MIC_EVENT);
 	}
-	change_cur_input(codec, spec->cur_input, 1);
-	if (spec->mic_detect)
-		cs_automic(codec);
-
-	coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
-	if (is_active_pin(codec, CS_DMIC2_PIN_NID))
-		coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
-	if (is_active_pin(codec, CS_DMIC1_PIN_NID))
-		coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 
-				 * No effect if SPDIF_OUT2 is selected in 
-				 * IDX_SPDIF_CTL.
-				  */
-	cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+	/* specific to CS421x */
+	if (spec->vendor_nid == CS421X_VENDOR_NID) {
+		if (spec->mic_detect)
+			cs_automic(codec);
+		else  {
+			spec->cur_adc = spec->adc_nid[spec->cur_input];
+			snd_hda_codec_write(codec, spec->cur_adc, 0,
+					AC_VERB_SET_CONNECT_SEL,
+					spec->adc_idx[spec->cur_input]);
+		}
+	} else {
+		change_cur_input(codec, spec->cur_input, 1);
+		if (spec->mic_detect)
+			cs_automic(codec);
+
+		coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
+		if (is_active_pin(codec, CS_DMIC2_PIN_NID))
+			coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */
+		if (is_active_pin(codec, CS_DMIC1_PIN_NID))
+			coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off
+					 * No effect if SPDIF_OUT2 is
+					 * selected in IDX_SPDIF_CTL.
+					*/
+		cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+	}
 }
 
 static const struct hda_verb cs_coef_init_verbs[] = {
@@ -1221,16 +1336,16 @@ static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
 	[CS420X_IMAC27] = imac27_pincfgs,
 };
 
-static void fix_pincfg(struct hda_codec *codec, int model)
+static void fix_pincfg(struct hda_codec *codec, int model,
+		       const struct cs_pincfg **pin_configs)
 {
-	const struct cs_pincfg *cfg = cs_pincfgs[model];
+	const struct cs_pincfg *cfg = pin_configs[model];
 	if (!cfg)
 		return;
 	for (; cfg->nid; cfg++)
 		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
 }
 
-
 static int patch_cs420x(struct hda_codec *codec)
 {
 	struct cs_spec *spec;
@@ -1241,11 +1356,13 @@ static int patch_cs420x(struct hda_codec *codec)
 		return -ENOMEM;
 	codec->spec = spec;
 
+	spec->vendor_nid = CS420X_VENDOR_NID;
+
 	spec->board_config =
 		snd_hda_check_board_config(codec, CS420X_MODELS,
 					   cs420x_models, cs420x_cfg_tbl);
 	if (spec->board_config >= 0)
-		fix_pincfg(codec, spec->board_config);
+		fix_pincfg(codec, spec->board_config, cs_pincfgs);
 
 	switch (spec->board_config) {
 	case CS420X_IMAC27:
@@ -1272,6 +1389,562 @@ static int patch_cs420x(struct hda_codec *codec)
 	return err;
 }
 
+/*
+ * Cirrus Logic CS4210
+ *
+ * 1 DAC => HP(sense) / Speakers,
+ * 1 ADC <= LineIn(sense) / MicIn / DMicIn,
+ * 1 SPDIF OUT => SPDIF Trasmitter(sense)
+*/
+
+/* CS4210 board names */
+static const char *cs421x_models[CS421X_MODELS] = {
+	[CS421X_CDB4210] = "cdb4210",
+};
+
+static const struct snd_pci_quirk cs421x_cfg_tbl[] = {
+	/* Test Intel board + CDB2410  */
+	SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210),
+	{} /* terminator */
+};
+
+/* CS4210 board pinconfigs */
+/* Default CS4210 (CDB4210)*/
+static const struct cs_pincfg cdb4210_pincfgs[] = {
+	{ 0x05, 0x0321401f },
+	{ 0x06, 0x90170010 },
+	{ 0x07, 0x03813031 },
+	{ 0x08, 0xb7a70037 },
+	{ 0x09, 0xb7a6003e },
+	{ 0x0a, 0x034510f0 },
+	{} /* terminator */
+};
+
+static const struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = {
+	[CS421X_CDB4210] = cdb4210_pincfgs,
+};
+
+static const struct hda_verb cs421x_coef_init_verbs[] = {
+	{0x0B, AC_VERB_SET_PROC_STATE, 1},
+	{0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG},
+	/*
+	    Disable Coefficient Index Auto-Increment(DAI)=1,
+	    PDREF=0
+	*/
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x0001 },
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG},
+	/* ADC SZCMode = Digital Soft Ramp */
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x0002 },
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG},
+	{0x0B, AC_VERB_SET_PROC_COEF,
+	 (0x0002 /* DAC SZCMode = Digital Soft Ramp */
+	  | 0x0004 /* Mute DAC on FIFO error */
+	  | 0x0008 /* Enable DAC High Pass Filter */
+	  )},
+	{} /* terminator */
+};
+
+/* Errata: CS4210 rev A1 Silicon
+ *
+ * http://www.cirrus.com/en/pubs/errata/
+ *
+ * Description:
+ * 1. Performance degredation is present in the ADC.
+ * 2. Speaker output is not completely muted upon HP detect.
+ * 3. Noise is present when clipping occurs on the amplified
+ *    speaker outputs.
+ *
+ * Workaround:
+ * The following verb sequence written to the registers during
+ * initialization will correct the issues listed above.
+ */
+
+static const struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = {
+	{0x0B, AC_VERB_SET_PROC_STATE, 0x01},  /* VPW: processing on */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x0006},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x000A},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x0011},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x001A},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */
+
+	{0x0B, AC_VERB_SET_COEF_INDEX, 0x001B},
+	{0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */
+
+	{} /* terminator */
+};
+
+/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */
+static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0);
+
+static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 3;
+	return 0;
+}
+
+static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003;
+	return 0;
+}
+
+static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	unsigned int vol = ucontrol->value.integer.value[0];
+	unsigned int coef =
+		cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL);
+	unsigned int original_coef = coef;
+
+	coef &= ~0x0003;
+	coef |= (vol & 0x0003);
+	if (original_coef == coef)
+		return 0;
+	else {
+		cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef);
+		return 1;
+	}
+}
+
+static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = {
+
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.name = "Speaker Boost Playback Volume",
+	.info = cs421x_boost_vol_info,
+	.get = cs421x_boost_vol_get,
+	.put = cs421x_boost_vol_put,
+	.tlv = { .p = cs421x_speaker_boost_db_scale },
+};
+
+static void cs421x_pinmux_init(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	unsigned int def_conf, coef;
+
+	/* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */
+	coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG);
+
+	if (spec->gpio_mask)
+		coef |= 0x0008; /* B1,B2 are GPIOs */
+	else
+		coef &= ~0x0008;
+
+	if (spec->sense_b)
+		coef |= 0x0010; /* B2 is SENSE_B, not inverted  */
+	else
+		coef &= ~0x0010;
+
+	cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef);
+
+	if ((spec->gpio_mask || spec->sense_b) &&
+	    is_active_pin(codec, CS421X_DMIC_PIN_NID)) {
+
+		/*
+		    GPIO or SENSE_B forced - disconnect the DMIC pin.
+		*/
+		def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID);
+		def_conf &= ~AC_DEFCFG_PORT_CONN;
+		def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT);
+		snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf);
+	}
+}
+
+static void init_cs421x_digital(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+
+	for (i = 0; i < cfg->dig_outs; i++) {
+		hda_nid_t nid = cfg->dig_out_pins[i];
+		if (!cfg->speaker_outs)
+			continue;
+		if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
+
+			snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_UNSOLICITED_ENABLE,
+				    AC_USRSP_EN | SPDIF_EVENT);
+			spec->spdif_detect = 1;
+		}
+	}
+}
+
+static int cs421x_init(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+
+	snd_hda_sequence_write(codec, cs421x_coef_init_verbs);
+	snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes);
+
+	cs421x_pinmux_init(codec);
+
+	if (spec->gpio_mask) {
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK,
+				    spec->gpio_mask);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION,
+				    spec->gpio_dir);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+				    spec->gpio_data);
+	}
+
+	init_output(codec);
+	init_input(codec);
+	init_cs421x_digital(codec);
+
+	return 0;
+}
+
+/*
+ * CS4210 Input MUX (1 ADC)
+ */
+static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+
+	return snd_hda_input_mux_info(&spec->input_mux, uinfo);
+}
+
+static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_input;
+	return 0;
+}
+
+static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+
+	return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol,
+				spec->adc_nid[0], &spec->cur_input);
+
+}
+
+static struct snd_kcontrol_new cs421x_capture_source = {
+
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Capture Source",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = cs421x_mux_enum_info,
+	.get = cs421x_mux_enum_get,
+	.put = cs421x_mux_enum_put,
+};
+
+static int cs421x_add_input_volume_control(struct hda_codec *codec, int item)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	const struct hda_input_mux *imux = &spec->input_mux;
+	hda_nid_t pin = cfg->inputs[item].pin;
+	struct snd_kcontrol *kctl;
+	u32 caps;
+
+	if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP))
+		return 0;
+
+	caps = query_amp_caps(codec, pin, HDA_INPUT);
+	caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+	if (caps <= 1)
+		return 0;
+
+	return add_volume(codec,  imux->items[item].label, 0,
+			  HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl);
+}
+
+/* add a (input-boost) volume control to the given input pin */
+static int build_cs421x_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	struct hda_input_mux *imux = &spec->input_mux;
+	int i, err, type_idx;
+	const char *label;
+
+	if (!spec->num_inputs)
+		return 0;
+
+	/* make bind-capture */
+	spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw);
+	spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol);
+	for (i = 0; i < 2; i++) {
+		struct snd_kcontrol *kctl;
+		int n;
+		if (!spec->capture_bind[i])
+			return -ENOMEM;
+		kctl = snd_ctl_new1(&cs_capture_ctls[i], codec);
+		if (!kctl)
+			return -ENOMEM;
+		kctl->private_value = (long)spec->capture_bind[i];
+		err = snd_hda_ctl_add(codec, 0, kctl);
+		if (err < 0)
+			return err;
+		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[n]);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	/* Add Input MUX Items + Capture Volume/Switch */
+	for (i = 0; i < spec->num_inputs; i++) {
+		label = hda_get_autocfg_input_label(codec, cfg, i);
+		snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx);
+
+		err = cs421x_add_input_volume_control(codec, i);
+		if (err < 0)
+			return err;
+	}
+
+	/*
+	    Add 'Capture Source' Switch if
+		* 2 inputs and no mic detec
+		* 3 inputs
+	*/
+	if ((spec->num_inputs == 2 && !spec->mic_detect) ||
+	    (spec->num_inputs == 3)) {
+
+		err = snd_hda_ctl_add(codec, spec->adc_nid[0],
+			      snd_ctl_new1(&cs421x_capture_source, codec));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* Single DAC (Mute/Gain) */
+static int build_cs421x_output(struct hda_codec *codec)
+{
+	hda_nid_t dac = CS4210_DAC_NID;
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	struct snd_kcontrol *kctl;
+	int err;
+	char *name = "HP/Speakers";
+
+	fix_volume_caps(codec, dac);
+	if (!spec->vmaster_sw) {
+		err = add_vmaster(codec, dac);
+		if (err < 0)
+			return err;
+	}
+
+	err = add_mute(codec, name, 0,
+			HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+	if (err < 0)
+		return err;
+	err = snd_ctl_add_slave(spec->vmaster_sw, kctl);
+	if (err < 0)
+		return err;
+
+	err = add_volume(codec, name, 0,
+			HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+	if (err < 0)
+		return err;
+	err = snd_ctl_add_slave(spec->vmaster_vol, kctl);
+	if (err < 0)
+		return err;
+
+	if (cfg->speaker_outs) {
+		err = snd_hda_ctl_add(codec, 0,
+			snd_ctl_new1(&cs421x_speaker_bost_ctl, codec));
+		if (err < 0)
+			return err;
+	}
+	return err;
+}
+
+static int cs421x_build_controls(struct hda_codec *codec)
+{
+	int err;
+
+	err = build_cs421x_output(codec);
+	if (err < 0)
+		return err;
+	err = build_cs421x_input(codec);
+	if (err < 0)
+		return err;
+	err = build_digital_output(codec);
+	if (err < 0)
+		return err;
+	return cs421x_init(codec);
+}
+
+static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	switch ((res >> 26) & 0x3f) {
+	case HP_EVENT:
+	case SPDIF_EVENT:
+		cs_automute(codec);
+		break;
+
+	case MIC_EVENT:
+		cs_automic(codec);
+		break;
+	}
+}
+
+static int parse_cs421x_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	for (i = 0; i < cfg->num_inputs; i++) {
+		hda_nid_t pin = cfg->inputs[i].pin;
+		spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]);
+		spec->cur_input = spec->last_input = i;
+		spec->num_inputs++;
+
+		/* check whether the automatic mic switch is available */
+		if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) {
+			spec->mic_detect = 1;
+			spec->automic_idx = i;
+		}
+	}
+	return 0;
+}
+
+static int cs421x_parse_auto_config(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+	if (err < 0)
+		return err;
+	err = parse_output(codec);
+	if (err < 0)
+		return err;
+	err = parse_cs421x_input(codec);
+	if (err < 0)
+		return err;
+	err = parse_digital_output(codec);
+	if (err < 0)
+		return err;
+	return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+	Manage PDREF, when transitioning to D3hot
+	(DAC,ADC) -> D3, PDREF=1, AFG->D3
+*/
+static int cs421x_suspend(struct hda_codec *codec, pm_message_t state)
+{
+	unsigned int coef;
+
+	snd_hda_shutup_pins(codec);
+
+	snd_hda_codec_write(codec, CS4210_DAC_NID, 0,
+			    AC_VERB_SET_POWER_STATE,  AC_PWRST_D3);
+	snd_hda_codec_write(codec, CS4210_ADC_NID, 0,
+			    AC_VERB_SET_POWER_STATE,  AC_PWRST_D3);
+
+	coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG);
+	coef |= 0x0004; /* PDREF */
+	cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef);
+
+	return 0;
+}
+#endif
+
+static struct hda_codec_ops cs4210_patch_ops = {
+	.build_controls = cs421x_build_controls,
+	.build_pcms = cs_build_pcms,
+	.init = cs421x_init,
+	.free = cs_free,
+	.unsol_event = cs421x_unsol_event,
+#ifdef CONFIG_PM
+	.suspend = cs421x_suspend,
+#endif
+};
+
+static int patch_cs421x(struct hda_codec *codec)
+{
+	struct cs_spec *spec;
+	int err;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+
+	spec->vendor_nid = CS421X_VENDOR_NID;
+
+	spec->board_config =
+		snd_hda_check_board_config(codec, CS421X_MODELS,
+					   cs421x_models, cs421x_cfg_tbl);
+	if (spec->board_config >= 0)
+		fix_pincfg(codec, spec->board_config, cs421x_pincfgs);
+	/*
+	    Setup GPIO/SENSE for each board (if used)
+	*/
+	switch (spec->board_config) {
+	case CS421X_CDB4210:
+		snd_printd("CS4210 board: %s\n",
+			cs421x_models[spec->board_config]);
+/*		spec->gpio_mask = 3;
+		spec->gpio_dir = 3;
+		spec->gpio_data = 3;
+*/
+		spec->sense_b = 1;
+
+		break;
+	}
+
+	/*
+	    Update the GPIO/DMIC/SENSE_B pinmux before the configuration
+	    is auto-parsed. If GPIO or SENSE_B is forced, DMIC input
+	    is disabled.
+	*/
+	cs421x_pinmux_init(codec);
+
+	err = cs421x_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
+
+	codec->patch_ops = cs4210_patch_ops;
+
+	return 0;
+
+ error:
+	kfree(codec->spec);
+	codec->spec = NULL;
+	return err;
+}
+
 
 /*
  * patch entries
@@ -1279,11 +1952,13 @@ static int patch_cs420x(struct hda_codec *codec)
 static const struct hda_codec_preset snd_hda_preset_cirrus[] = {
 	{ .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
 	{ .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
+	{ .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x },
 	{} /* terminator */
 };
 
 MODULE_ALIAS("snd-hda-codec-id:10134206");
 MODULE_ALIAS("snd-hda-codec-id:10134207");
+MODULE_ALIAS("snd-hda-codec-id:10134210");
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");

sound/pci/hda/patch_conexant.c

@@ -446,6 +446,19 @@ static int conexant_init_jacks(struct hda_codec *codec)
 	return 0;
 }
 
+static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg,
+			       unsigned int power_state)
+{
+	if (power_state == AC_PWRST_D3)
+		msleep(100);
+	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+			    power_state);
+	/* partial workaround for "azx_get_response timeout" */
+	if (power_state == AC_PWRST_D0)
+		msleep(10);
+	snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+}
+
 static int conexant_init(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -588,6 +601,7 @@ static const struct hda_codec_ops conexant_patch_ops = {
 	.build_pcms = conexant_build_pcms,
 	.init = conexant_init,
 	.free = conexant_free,
+	.set_power_state = conexant_set_power,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	.suspend = conexant_suspend,
 #endif

sound/pci/hda/patch_realtek.c

@@ -2386,7 +2386,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state)
 }
 #endif
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int alc_resume(struct hda_codec *codec)
 {
 	msleep(150); /* to avoid pop noise */
@@ -2406,7 +2406,7 @@ static const struct hda_codec_ops alc_patch_ops = {
 	.init = alc_init,
 	.free = alc_free,
 	.unsol_event = alc_unsol_event,
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.resume = alc_resume,
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -2801,7 +2801,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	int i;
 
  again:
-	spec->multiout.num_dacs = 0;
+	/* set num_dacs once to full for alc_auto_look_for_dac() */
+	spec->multiout.num_dacs = cfg->line_outs;
 	spec->multiout.hp_nid = 0;
 	spec->multiout.extra_out_nid[0] = 0;
 	memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
@@ -2834,6 +2835,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 		}
 	}
 
+	/* re-count num_dacs and squash invalid entries */
+	spec->multiout.num_dacs = 0;
 	for (i = 0; i < cfg->line_outs; i++) {
 		if (spec->private_dac_nids[i])
 			spec->multiout.num_dacs++;
@@ -4410,7 +4413,7 @@ static void alc269_shutup(struct hda_codec *codec)
 	}
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int alc269_resume(struct hda_codec *codec)
 {
 	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
@@ -4433,7 +4436,7 @@ static int alc269_resume(struct hda_codec *codec)
 	hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
-#endif /* SND_HDA_NEEDS_RESUME */
+#endif /* CONFIG_PM */
 
 static void alc269_fixup_hweq(struct hda_codec *codec,
 			       const struct alc_fixup *fix, int action)
@@ -4725,7 +4728,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->vmaster_nid = 0x02;
 
 	codec->patch_ops = alc_patch_ops;
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	codec->patch_ops.resume = alc269_resume;
 #endif
 	if (board_config == ALC_MODEL_AUTO)

sound/pci/hda/patch_sigmatel.c

@@ -95,6 +95,7 @@ enum {
 	STAC_92HD83XXX_PWR_REF,
 	STAC_DELL_S14,
 	STAC_92HD83XXX_HP,
+	STAC_92HD83XXX_HP_cNB11_INTQUAD,
 	STAC_HP_DV7_4000,
 	STAC_92HD83XXX_MODELS
 };
@@ -1636,10 +1637,17 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = {
 	0x40f000f0, 0x40f000f0,
 };
 
+static const unsigned int hp_cNB11_intquad_pin_configs[10] = {
+	0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110,
+	0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130,
+	0x40f000f0, 0x40f000f0,
+};
+
 static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
 	[STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
 	[STAC_DELL_S14] = dell_s14_pin_configs,
+	[STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs,
 	[STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs,
 };
 
@@ -1649,6 +1657,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
 	[STAC_DELL_S14] = "dell-s14",
 	[STAC_92HD83XXX_HP] = "hp",
+	[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
 	[STAC_HP_DV7_4000] = "hp-dv7-4000",
 };
 
@@ -1661,7 +1670,47 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba,
 		      "unknown Dell", STAC_DELL_S14),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600,
-		      "HP", STAC_92HD83XXX_HP),
+			  "HP", STAC_92HD83XXX_HP),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1657,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1658,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1659,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165A,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165B,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355B,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355C,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355D,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355E,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355F,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3560,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358B,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358C,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358D,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3591,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3592,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593,
+			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
 	{} /* terminator */
 };
 
@@ -4885,7 +4934,18 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer,
 #define stac927x_proc_hook	NULL
 #endif
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
+static int stac92xx_pre_resume(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	/* sync mute LED */
+	if (spec->gpio_led)
+		stac_gpio_set(codec, spec->gpio_mask,
+				spec->gpio_dir, spec->gpio_data);
+	return 0;
+}
+
 static int stac92xx_resume(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -4901,29 +4961,19 @@ static int stac92xx_resume(struct hda_codec *codec)
 			stac_issue_unsol_event(codec,
 					       spec->autocfg.line_out_pins[0]);
 	}
-	/* sync mute LED */
-	if (spec->gpio_led)
-		hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 /*
- * using power check for controlling mute led of HP notebooks
- * check for mute state only on Speakers (nid = 0x10)
- *
- * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise
- * the LED is NOT working properly !
- *
- * Changed name to reflect that it now works for any designated
- * model, not just HP HDX.
+ * For this feature CONFIG_SND_HDA_POWER_SAVE is needed
+ * as mute LED state is updated in check_power_status hook
  */
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static int stac92xx_hp_check_power_status(struct hda_codec *codec,
-					      hda_nid_t nid)
+static int stac92xx_update_led_status(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int i, muted = 1;
+	int i, num_ext_dacs, muted = 1;
+	hda_nid_t nid;
 
 	for (i = 0; i < spec->multiout.num_dacs; i++) {
 		nid = spec->multiout.dac_nids[i];
@@ -4933,6 +4983,22 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 			break;
 		}
 	}
+	if (muted && spec->multiout.hp_nid)
+		if (!(snd_hda_codec_amp_read(codec,
+				spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) &
+					HDA_AMP_MUTE)) {
+			muted = 0; /* HP is not muted */
+		}
+	num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid);
+	for (i = 0; muted && i < num_ext_dacs; i++) {
+		nid = spec->multiout.extra_out_nid[i];
+		if (nid == 0)
+			break;
+		if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
+		      HDA_AMP_MUTE)) {
+			muted = 0; /* extra output is not muted */
+		}
+	}
 	if (muted)
 		spec->gpio_data &= ~spec->gpio_led; /* orange */
 	else
@@ -4946,6 +5012,17 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 	stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
 	return 0;
 }
+
+/*
+ * use power check for controlling mute led of HP notebooks
+ */
+static int stac92xx_check_power_status(struct hda_codec *codec,
+					      hda_nid_t nid)
+{
+	stac92xx_update_led_status(codec);
+
+	return 0;
+}
 #endif
 
 static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
@@ -4953,7 +5030,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 	stac92xx_shutup(codec);
 	return 0;
 }
-#endif
+#endif /* CONFIG_PM */
 
 static const struct hda_codec_ops stac92xx_patch_ops = {
 	.build_controls = stac92xx_build_controls,
@@ -4961,9 +5038,10 @@ static const struct hda_codec_ops stac92xx_patch_ops = {
 	.init = stac92xx_init,
 	.free = stac92xx_free,
 	.unsol_event = stac92xx_unsol_event,
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.suspend = stac92xx_suspend,
 	.resume = stac92xx_resume,
+	.pre_resume = stac92xx_pre_resume,
 #endif
 	.reboot_notify = stac92xx_shutup,
 };
@@ -5482,7 +5560,7 @@ again:
 		spec->gpio_data |= spec->gpio_led;
 		/* register check_power_status callback. */
 		codec->patch_ops.check_power_status =
-			stac92xx_hp_check_power_status;
+			stac92xx_check_power_status;
 	}
 #endif	
 
@@ -5810,7 +5888,7 @@ again:
 		spec->gpio_data |= spec->gpio_led;
 		/* register check_power_status callback. */
 		codec->patch_ops.check_power_status =
-			stac92xx_hp_check_power_status;
+			stac92xx_check_power_status;
 	}
 #endif	
 

sound/pci/hda/patch_via.c

@@ -1708,7 +1708,7 @@ static void via_unsol_event(struct hda_codec *codec,
 		via_gpio_control(codec);
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 static int via_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	struct via_spec *spec = codec->spec;
@@ -1736,7 +1736,7 @@ static const struct hda_codec_ops via_patch_ops = {
 	.init = via_init,
 	.free = via_free,
 	.unsol_event = via_unsol_event,
-#ifdef SND_HDA_NEEDS_RESUME
+#ifdef CONFIG_PM
 	.suspend = via_suspend,
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE

sound/soc/codecs/sgtl5000.c

@@ -907,6 +907,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
 				struct regulator_init_data *init_data,
 				int voltage)
 {
+	dev_err(codec->dev, "this setup needs regulator support in the kernel\n");
 	return -EINVAL;
 }
 
@@ -1218,6 +1219,34 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec)
+{
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	/* set internal ldo to 1.2v */
+	ret = ldo_regulator_register(codec, &ldo_init_data, LDO_VOLTAGE);
+	if (ret) {
+		dev_err(codec->dev,
+			"Failed to register vddd internal supplies: %d\n", ret);
+		return ret;
+	}
+
+	sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+			sgtl5000->supplies);
+
+	if (ret) {
+		ldo_regulator_remove(codec);
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	dev_info(codec->dev, "Using internal LDO instead of VDDD\n");
+	return 0;
+}
+
 static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 {
 	u16 reg;
@@ -1235,30 +1264,9 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 	if (!ret)
 		external_vddd = 1;
 	else {
-		/* set internal ldo to 1.2v */
-		int voltage = LDO_VOLTAGE;
-
-		ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
-		if (ret) {
-			dev_err(codec->dev,
-			"Failed to register vddd internal supplies: %d\n",
-				ret);
-			return ret;
-		}
-
-		sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
-
-		ret = regulator_bulk_get(codec->dev,
-				ARRAY_SIZE(sgtl5000->supplies),
-				sgtl5000->supplies);
-
-		if (ret) {
-			ldo_regulator_remove(codec);
-			dev_err(codec->dev,
-				"Failed to request supplies: %d\n", ret);
-
+		ret = sgtl5000_replace_vddd_with_ldo(codec);
+		if (ret)
 			return ret;
-		}
 	}
 
 	ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
@@ -1287,7 +1295,6 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 	 * roll back to use internal LDO
 	 */
 	if (external_vddd && rev >= 0x11) {
-		int voltage = LDO_VOLTAGE;
 		/* disable all regulator first */
 		regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
 					sgtl5000->supplies);
@@ -1295,23 +1302,10 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 		regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
 					sgtl5000->supplies);
 
-		ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
+		ret = sgtl5000_replace_vddd_with_ldo(codec);
 		if (ret)
 			return ret;
 
-		sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
-
-		ret = regulator_bulk_get(codec->dev,
-				ARRAY_SIZE(sgtl5000->supplies),
-				sgtl5000->supplies);
-		if (ret) {
-			ldo_regulator_remove(codec);
-			dev_err(codec->dev,
-				"Failed to request supplies: %d\n", ret);
-
-			return ret;
-		}
-
 		ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
 		if (ret)

sound/soc/codecs/wm8962.c

@@ -3409,6 +3409,9 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 	active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2);
 	active &= ~mask;
 
+	/* Acknowledge the interrupts */
+	snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active);
+
 	if (active & WM8962_FLL_LOCK_EINT) {
 		dev_dbg(codec->dev, "FLL locked\n");
 		complete(&wm8962->fll_lock);
@@ -3433,9 +3436,6 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 				      msecs_to_jiffies(250));
 	}
 
-	/* Acknowledge the interrupts */
-	snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active);
-
 	return IRQ_HANDLED;
 }
 

sound/soc/davinci/davinci-vcif.c

@@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream)
 	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1);
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0);
 	else
-		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1);
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0);
 
 	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
 }
@@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream)
 	/* Reset transmitter/receiver and sample rate/frame sync generators */
 	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0);
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1);
 	else
-		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0);
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1);
 
 	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
 }
@@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		davinci_vcif_start(substream);
+		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:

sound/soc/samsung/i2s.c

@@ -21,6 +21,7 @@
 #include <plat/audio.h>
 
 #include "dma.h"
+#include "idma.h"
 #include "i2s.h"
 #include "i2s-regs.h"
 
@@ -60,6 +61,7 @@ struct i2s_dai {
 	/* DMA parameters */
 	struct s3c_dma_params dma_playback;
 	struct s3c_dma_params dma_capture;
+	struct s3c_dma_params idma_playback;
 	u32	quirks;
 	u32	suspend_i2smod;
 	u32	suspend_i2scon;
@@ -877,6 +879,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai)
 	if (i2s->quirks & QUIRK_NEED_RSTCLR)
 		writel(CON_RSTCLR, i2s->addr + I2SCON);
 
+	if (i2s->quirks & QUIRK_SEC_DAI)
+		idma_reg_addr_init((void *)i2s->addr,
+					i2s->sec_dai->idma_playback.dma_addr);
+
 probe_exit:
 	/* Reset any constraint on RFS and BFS */
 	i2s->rfs = 0;
@@ -1077,6 +1083,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev)
 		sec_dai->dma_playback.dma_size = 4;
 		sec_dai->base = regs_base;
 		sec_dai->quirks = quirks;
+		sec_dai->idma_playback.dma_addr = i2s_cfg->idma_addr;
 		sec_dai->pri_dai = pri_dai;
 		pri_dai->sec_dai = sec_dai;
 	}

sound/soc/soc-core.c

@@ -577,6 +577,7 @@ int snd_soc_suspend(struct device *dev)
 			case SND_SOC_BIAS_OFF:
 				codec->driver->suspend(codec, PMSG_SUSPEND);
 				codec->suspended = 1;
+				codec->cache_sync = 1;
 				break;
 			default:
 				dev_dbg(codec->dev, "CODEC is on over suspend\n");
@@ -1140,7 +1141,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 			}
 		}
 		cpu_dai->probed = 1;
-		/* mark cpu_dai as probed and add to card cpu_dai list */
+		/* mark cpu_dai as probed and add to card dai list */
 		list_add(&cpu_dai->card_list, &card->dai_dev_list);
 	}
 
@@ -1171,7 +1172,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 			}
 		}
 
-		/* mark cpu_dai as probed and add to card cpu_dai list */
+		/* mark codec_dai as probed and add to card dai list */
 		codec_dai->probed = 1;
 		list_add(&codec_dai->card_list, &card->dai_dev_list);
 	}

sound/soc/soc-dapm.c

@@ -124,6 +124,36 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
 	return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
 }
 
+/* get snd_card from DAPM context */
+static inline struct snd_card *dapm_get_snd_card(
+	struct snd_soc_dapm_context *dapm)
+{
+	if (dapm->codec)
+		return dapm->codec->card->snd_card;
+	else if (dapm->platform)
+		return dapm->platform->card->snd_card;
+	else
+		BUG();
+
+	/* unreachable */
+	return NULL;
+}
+
+/* get soc_card from DAPM context */
+static inline struct snd_soc_card *dapm_get_soc_card(
+		struct snd_soc_dapm_context *dapm)
+{
+	if (dapm->codec)
+		return dapm->codec->card;
+	else if (dapm->platform)
+		return dapm->platform->card;
+	else
+		BUG();
+
+	/* unreachable */
+	return NULL;
+}
+
 static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg)
 {
 	if (w->codec)