[ALSA] hda-intel - Add POWER_SAVE option

Added CONFIG_SND_HDA_POWER_SAVE kconfig.  It's an experimental option
to achieve an aggressive power-saving.  With this option, the driver
will turn on/off the power of each codec and controller chip dynamically
on demand.
The patch introduces a new module option 'power_save'.  It specifies
the second of time-out for automatic power-down.  As default, it's
10 seconds.  Setting 0 means to suppress the power-saving feature.
The codec may have analog-input loopbacks, which are usually represented
by mixer elements such as 'Mic Playback Switch' or 'CD Playback Switch'.
When these are on, we cannot turn off the mixer and the codec chip has
to be kept on.  For bookkeeping these states, a new codec-callback is
introduced.
For the bus-controller side, a new callback pm_notify is introduced,
which can be used to turn on/off the contoller appropriately.
Note that this power-saving might cause slight click-noise at
power-on/off.  Also, it might take some time to wake up the codec, and
might even drop some tones at the very beginning.  This seems to be the
side-effect of turning off the controller chip.
This turn-off of the controller can be disabled by undefining
HDA_POWER_SAVE_RESET_CONTOLLER in hda_intel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>

1 files changed, 50 insertions, 5 deletions

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 91cd9b9e..819c804 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -70,6 +70,13 @@ struct hda_gspec {
 	struct hda_pcm pcm_rec;		/* PCM information */
 
 	struct list_head nid_list;	/* list of widgets */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define MAX_LOOPBACK_AMPS	7
+	struct hda_loopback_check loopback;
+	int num_loopbacks;
+	struct hda_amp_list loopback_list[MAX_LOOPBACK_AMPS + 1];
+#endif
 };
 
 /*
@@ -682,11 +689,33 @@ static int parse_input(struct hda_codec *codec)
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void add_input_loopback(struct hda_codec *codec, hda_nid_t nid,
+			       int dir, int idx)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct hda_amp_list *p;
+
+	if (spec->num_loopbacks >= MAX_LOOPBACK_AMPS) {
+		snd_printk(KERN_ERR "hda_generic: Too many loopback ctls\n");
+		return;
+	}
+	p = &spec->loopback_list[spec->num_loopbacks++];
+	p->nid = nid;
+	p->dir = dir;
+	p->idx = idx;
+	spec->loopback.amplist = spec->loopback_list;
+}
+#else
+#define add_input_loopback(codec,nid,dir,idx)
+#endif
+
 /*
  * create mixer controls if possible
  */
 static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
-			unsigned int index, const char *type, const char *dir_sfx)
+			unsigned int index, const char *type,
+			const char *dir_sfx, int is_loopback)
 {
 	char name[32];
 	int err;
@@ -700,6 +729,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
 	if ((node->wid_caps & AC_WCAP_IN_AMP) &&
 	    (node->amp_in_caps & AC_AMPCAP_MUTE)) {
 		knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT);
+		if (is_loopback)
+			add_input_loopback(codec, node->nid, HDA_INPUT, index);
 		snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
 		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
 			return err;
@@ -707,6 +738,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
 	} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
 		   (node->amp_out_caps & AC_AMPCAP_MUTE)) {
 		knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT);
+		if (is_loopback)
+			add_input_loopback(codec, node->nid, HDA_OUTPUT, 0);
 		snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
 		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
 			return err;
@@ -765,7 +798,7 @@ static int create_output_mixers(struct hda_codec *codec, const char **names)
 	for (i = 0; i < spec->pcm_vol_nodes; i++) {
 		err = create_mixer(codec, spec->pcm_vol[i].node,
 				   spec->pcm_vol[i].index,
-				   names[i], "Playback");
+				   names[i], "Playback", 0);
 		if (err < 0)
 			return err;
 	}
@@ -782,7 +815,7 @@ static int build_output_controls(struct hda_codec *codec)
 	case 1:
 		return create_mixer(codec, spec->pcm_vol[0].node,
 				    spec->pcm_vol[0].index,
-				    "Master", "Playback");
+				    "Master", "Playback", 0);
 	case 2:
 		if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER)
 			return create_output_mixers(codec, types_speaker);
@@ -818,7 +851,7 @@ static int build_input_controls(struct hda_codec *codec)
 	if (spec->input_mux.num_items == 1) {
 		err = create_mixer(codec, adc_node,
 				   spec->input_mux.items[0].index,
-				   NULL, "Capture");
+				   NULL, "Capture", 0);
 		if (err < 0)
 			return err;
 		return 0;
@@ -884,7 +917,8 @@ static int parse_loopback_path(struct hda_codec *codec, struct hda_gspec *spec,
 			return err;
 		else if (err >= 1) {
 			if (err == 1) {
-				err = create_mixer(codec, node, i, type, "Playback");
+				err = create_mixer(codec, node, i, type,
+						   "Playback", 1);
 				if (err < 0)
 					return err;
 				if (err > 0)
@@ -1020,6 +1054,14 @@ static int build_generic_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int generic_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct hda_gspec *spec = codec->spec;
+	return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
 
 /*
  */
@@ -1027,6 +1069,9 @@ static struct hda_codec_ops generic_patch_ops = {
 	.build_controls = build_generic_controls,
 	.build_pcms = build_generic_pcms,
 	.free = snd_hda_generic_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	.check_power_status = generic_check_power_status,
+#endif
 };
 
 /*