[PATCH] [UBUNTU:sound/pci/hda/] #59715: Fix headphone volume control for generic HDA codec parser
crimsun at fungus.sh.nu
crimsun at fungus.sh.nu
Sat Sep 16 03:54:23 UTC 2006
From 600f5ba6084d8e94785c67f561b9a5a6c9126035 Mon Sep 17 00:00:00 2001
From: Daniel T. Chen <crimsun at garnish.localdomain>
Date: Fri, 15 Sep 2006 23:53:59 -0400
Subject: [PATCH] [UBUNTU:sound/pci/hda/] #59715: Fix headphone volume control for generic HDA codec parser
UpstreamStatus: Added in upstream alsa-kernel hg changeset:
b674990c217c [http://hg-mirror.alsa-project.org/alsa-kernel?cmd=changeset;node=b674990c217c3bb6e7da95319d8c10590a2a64f3;style=raw]
References: ALSA #2412
Ubuntu #59715
This patch from Takashi Iwai supports an independent "Headphone" volume
control that is useful for Conexant and other codecs that separate
connections to headphone, resulting in needing to adjust the amp
explicitly.
This commit is applicable to both Dapper and Edgy linux-source.
This commit closes Ubuntu #59715.
Signed-off-by: Daniel T Chen <crimsun at ubuntu.com>
---
sound/pci/hda/hda_generic.c | 71 +++++++++++++++++++++++++++++++------------
1 files changed, 52 insertions(+), 19 deletions(-)
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b7954c8..6a96b64 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -46,11 +46,18 @@ struct hda_gnode {
};
/* patch-specific record */
+
+#define MAX_PCM_VOLS 2
+struct pcm_vol {
+ struct hda_gnode *node; /* Node for PCM volume */
+ unsigned int index; /* connection of PCM volume */
+};
+
struct hda_gspec {
struct hda_gnode *dac_node[2]; /* DAC node */
struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */
- struct hda_gnode *pcm_vol_node[2]; /* Node for PCM volume */
- unsigned int pcm_vol_index[2]; /* connection of PCM volume */
+ struct pcm_vol pcm_vol[MAX_PCM_VOLS]; /* PCM volumes */
+ unsigned int pcm_vol_nodes; /* number of PCM volumes */
struct hda_gnode *adc_node; /* ADC node */
struct hda_gnode *cap_vol_node; /* Node for capture volume */
@@ -285,9 +292,11 @@ static int parse_output_path(struct hda_
return node == spec->dac_node[dac_idx];
}
spec->dac_node[dac_idx] = node;
- if (node->wid_caps & AC_WCAP_OUT_AMP) {
- spec->pcm_vol_node[dac_idx] = node;
- spec->pcm_vol_index[dac_idx] = 0;
+ if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
+ spec->pcm_vol_nodes < MAX_PCM_VOLS) {
+ spec->pcm_vol[spec->pcm_vol_nodes].node = node;
+ spec->pcm_vol[spec->pcm_vol_nodes].index = 0;
+ spec->pcm_vol_nodes++;
}
return 1; /* found */
}
@@ -307,13 +316,16 @@ static int parse_output_path(struct hda_
select_input_connection(codec, node, i);
unmute_input(codec, node, i);
unmute_output(codec, node);
- if (! spec->pcm_vol_node[dac_idx]) {
- if (node->wid_caps & AC_WCAP_IN_AMP) {
- spec->pcm_vol_node[dac_idx] = node;
- spec->pcm_vol_index[dac_idx] = i;
- } else if (node->wid_caps & AC_WCAP_OUT_AMP) {
- spec->pcm_vol_node[dac_idx] = node;
- spec->pcm_vol_index[dac_idx] = 0;
+ if (spec->dac_node[dac_idx] &&
+ spec->pcm_vol_nodes < MAX_PCM_VOLS &&
+ !(spec->dac_node[dac_idx]->wid_caps &
+ AC_WCAP_OUT_AMP)) {
+ if ((node->wid_caps & AC_WCAP_IN_AMP) ||
+ (node->wid_caps & AC_WCAP_OUT_AMP)) {
+ int n = spec->pcm_vol_nodes;
+ spec->pcm_vol[n].node = node;
+ spec->pcm_vol[n].index = i;
+ spec->pcm_vol_nodes++;
}
}
return 1;
@@ -370,7 +382,9 @@ static struct hda_gnode *parse_output_ja
/* set PIN-Out enable */
snd_hda_codec_write(codec, node->nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
- AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ AC_PINCTL_OUT_EN |
+ ((node->pin_caps & AC_PINCAP_HP_DRV) ?
+ AC_PINCTL_HP_EN : 0));
return node;
}
}
@@ -722,22 +736,41 @@ static int check_existing_control(struct
/*
* build output mixer controls
*/
-static int build_output_controls(struct hda_codec *codec)
+static int create_output_mixers(struct hda_codec *codec, const char **names)
{
struct hda_gspec *spec = codec->spec;
- static const char *types[2] = { "Master", "Headphone" };
int i, err;
- for (i = 0; i < 2 && spec->pcm_vol_node[i]; i++) {
- err = create_mixer(codec, spec->pcm_vol_node[i],
- spec->pcm_vol_index[i],
- types[i], "Playback");
+ 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");
if (err < 0)
return err;
}
return 0;
}
+static int build_output_controls(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ static const char *types_speaker[] = { "Speaker", "Headphone" };
+ static const char *types_line[] = { "Front", "Headphone" };
+
+ switch (spec->pcm_vol_nodes) {
+ case 1:
+ return create_mixer(codec, spec->pcm_vol[0].node,
+ spec->pcm_vol[0].index,
+ "Master", "Playback");
+ case 2:
+ if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER)
+ return create_output_mixers(codec, types_speaker);
+ else
+ return create_output_mixers(codec, types_line);
+ }
+ return 0;
+}
+
/* create capture volume/switch */
static int build_input_controls(struct hda_codec *codec)
{
--
1.4.1
--
Daniel T. Chen crimsun at ubuntu.com
GPG key: 0xC88ABDA3
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