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authorTakashi Iwai <tiwai@suse.de>2006-09-06 08:03:14 -0400
committerJaroslav Kysela <perex@suse.cz>2006-09-23 04:46:27 -0400
commita7da6ce564a80952d9c0b210deca5a8cd3474a31 (patch)
treee0520ce4255be7a8692e39241e086334a232b6a8
parent160ea0dc6b86e2c0c4d325c06bf402bfdde7c1c7 (diff)
[ALSA] hda-codec - Add independent headphone volume control
This patch addes the support of the independent 'Headphone' volume control to the generic codec parser. Some codecs (e.g. Conexant) have separate connections to the headphone and the independent amp adjustment is needed. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
-rw-r--r--sound/pci/hda/hda_generic.c71
1 files changed, 52 insertions, 19 deletions
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index dedfc5b1083a..97e9af130b71 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -46,11 +46,18 @@ struct hda_gnode {
46}; 46};
47 47
48/* patch-specific record */ 48/* patch-specific record */
49
50#define MAX_PCM_VOLS 2
51struct pcm_vol {
52 struct hda_gnode *node; /* Node for PCM volume */
53 unsigned int index; /* connection of PCM volume */
54};
55
49struct hda_gspec { 56struct hda_gspec {
50 struct hda_gnode *dac_node[2]; /* DAC node */ 57 struct hda_gnode *dac_node[2]; /* DAC node */
51 struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */ 58 struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */
52 struct hda_gnode *pcm_vol_node[2]; /* Node for PCM volume */ 59 struct pcm_vol pcm_vol[MAX_PCM_VOLS]; /* PCM volumes */
53 unsigned int pcm_vol_index[2]; /* connection of PCM volume */ 60 unsigned int pcm_vol_nodes; /* number of PCM volumes */
54 61
55 struct hda_gnode *adc_node; /* ADC node */ 62 struct hda_gnode *adc_node; /* ADC node */
56 struct hda_gnode *cap_vol_node; /* Node for capture volume */ 63 struct hda_gnode *cap_vol_node; /* Node for capture volume */
@@ -285,9 +292,11 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec,
285 return node == spec->dac_node[dac_idx]; 292 return node == spec->dac_node[dac_idx];
286 } 293 }
287 spec->dac_node[dac_idx] = node; 294 spec->dac_node[dac_idx] = node;
288 if (node->wid_caps & AC_WCAP_OUT_AMP) { 295 if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
289 spec->pcm_vol_node[dac_idx] = node; 296 spec->pcm_vol_nodes < MAX_PCM_VOLS) {
290 spec->pcm_vol_index[dac_idx] = 0; 297 spec->pcm_vol[spec->pcm_vol_nodes].node = node;
298 spec->pcm_vol[spec->pcm_vol_nodes].index = 0;
299 spec->pcm_vol_nodes++;
291 } 300 }
292 return 1; /* found */ 301 return 1; /* found */
293 } 302 }
@@ -307,13 +316,16 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec,
307 select_input_connection(codec, node, i); 316 select_input_connection(codec, node, i);
308 unmute_input(codec, node, i); 317 unmute_input(codec, node, i);
309 unmute_output(codec, node); 318 unmute_output(codec, node);
310 if (! spec->pcm_vol_node[dac_idx]) { 319 if (spec->dac_node[dac_idx] &&
311 if (node->wid_caps & AC_WCAP_IN_AMP) { 320 spec->pcm_vol_nodes < MAX_PCM_VOLS &&
312 spec->pcm_vol_node[dac_idx] = node; 321 !(spec->dac_node[dac_idx]->wid_caps &
313 spec->pcm_vol_index[dac_idx] = i; 322 AC_WCAP_OUT_AMP)) {
314 } else if (node->wid_caps & AC_WCAP_OUT_AMP) { 323 if ((node->wid_caps & AC_WCAP_IN_AMP) ||
315 spec->pcm_vol_node[dac_idx] = node; 324 (node->wid_caps & AC_WCAP_OUT_AMP)) {
316 spec->pcm_vol_index[dac_idx] = 0; 325 int n = spec->pcm_vol_nodes;
326 spec->pcm_vol[n].node = node;
327 spec->pcm_vol[n].index = i;
328 spec->pcm_vol_nodes++;
317 } 329 }
318 } 330 }
319 return 1; 331 return 1;
@@ -370,7 +382,9 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec,
370 /* set PIN-Out enable */ 382 /* set PIN-Out enable */
371 snd_hda_codec_write(codec, node->nid, 0, 383 snd_hda_codec_write(codec, node->nid, 0,
372 AC_VERB_SET_PIN_WIDGET_CONTROL, 384 AC_VERB_SET_PIN_WIDGET_CONTROL,
373 AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); 385 AC_PINCTL_OUT_EN |
386 ((node->pin_caps & AC_PINCAP_HP_DRV) ?
387 AC_PINCTL_HP_EN : 0));
374 return node; 388 return node;
375 } 389 }
376 } 390 }
@@ -745,22 +759,41 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con
745/* 759/*
746 * build output mixer controls 760 * build output mixer controls
747 */ 761 */
748static int build_output_controls(struct hda_codec *codec) 762static int create_output_mixers(struct hda_codec *codec, const char **names)
749{ 763{
750 struct hda_gspec *spec = codec->spec; 764 struct hda_gspec *spec = codec->spec;
751 static const char *types[2] = { "Master", "Headphone" };
752 int i, err; 765 int i, err;
753 766
754 for (i = 0; i < 2 && spec->pcm_vol_node[i]; i++) { 767 for (i = 0; i < spec->pcm_vol_nodes; i++) {
755 err = create_mixer(codec, spec->pcm_vol_node[i], 768 err = create_mixer(codec, spec->pcm_vol[i].node,
756 spec->pcm_vol_index[i], 769 spec->pcm_vol[i].index,
757 types[i], "Playback"); 770 names[i], "Playback");
758 if (err < 0) 771 if (err < 0)
759 return err; 772 return err;
760 } 773 }
761 return 0; 774 return 0;
762} 775}
763 776
777static int build_output_controls(struct hda_codec *codec)
778{
779 struct hda_gspec *spec = codec->spec;
780 static const char *types_speaker[] = { "Speaker", "Headphone" };
781 static const char *types_line[] = { "Front", "Headphone" };
782
783 switch (spec->pcm_vol_nodes) {
784 case 1:
785 return create_mixer(codec, spec->pcm_vol[0].node,
786 spec->pcm_vol[0].index,
787 "Master", "Playback");
788 case 2:
789 if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER)
790 return create_output_mixers(codec, types_speaker);
791 else
792 return create_output_mixers(codec, types_line);
793 }
794 return 0;
795}
796
764/* create capture volume/switch */ 797/* create capture volume/switch */
765static int build_input_controls(struct hda_codec *codec) 798static int build_input_controls(struct hda_codec *codec)
766{ 799{