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authorLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 23:51:23 +0400
committerLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 23:51:23 +0400
commit046e7d685bc370fd4c879ab6635ad3f69e6673d1 (patch)
tree36b981f8d1f2bfd348c1479acbe3a9426d35c377 /sound/pci/hda/patch_conexant.c
parentfe504c5c745aeb767d978fbedeb94775fd4cb69c (diff)
parent6eb827d23577a4efec2b10a9c4cc9ded268a1d1c (diff)
downloadlinux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.tar.xz
Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This update contains a fairly wide range of changes all over in sound subdirectory, mainly because of UAPI header moves by David and __dev* annotation removals by Bill. Other highlights are: - Introduced the support for wallclock timestamps in ALSA PCM core - Add the poll loop implementation for HD-audio jack detection - Yet more VGA-switcheroo fixes for HD-audio - New VIA HD-audio codec support - More fixes on resource management in USB audio and MIDI drivers - More quirks for USB-audio ASUS Xonar U3, Reloop Play, Focusrite, Roland VG-99, etc - Add support for FastTrack C400 usb-audio - Clean ups in many drivers regarding firmware loading - Add PSC724 Ultiimate Edge support to ice1712 - A few hdspm driver updates - New Stanton SCS.1d/1m FireWire driver - Standardisation of the logging in ASoC codes - DT and dmaengine support for ASoC Atmel - Support for Wolfson ADSP cores - New drivers for Freescale/iVeia P1022 and Maxim MAX98090 - Lots of other ASoC driver fixes and developments" Fix up trivial conflicts. And go out on a limb and assume the dts file 'status' field of one of the conflicting things was supposed to be "disabled", not "disable" like in pretty much all other cases. * tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (341 commits) ALSA: hda - Move runtime PM check to runtime_idle callback ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522 ALSA: hda - Avoid doubly suspend after vga switcheroo ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3 ALSA: hda - Check validity of CORB/RIRB WP reads ALSA: hda - use usleep_range in link reset and change timeout check ALSA: HDA: VIA: Add support for codec VT1808. ALSA: HDA: VIA Add support for codec VT1705CF. ASoC: codecs: remove __dev* attributes ASoC: utils: remove __dev* attributes ASoC: ux500: remove __dev* attributes ASoC: txx9: remove __dev* attributes ASoC: tegra: remove __dev* attributes ASoC: spear: remove __dev* attributes ASoC: sh: remove __dev* attributes ASoC: s6000: remove __dev* attributes ASoC: OMAP: remove __dev* attributes ASoC: nuc900: remove __dev* attributes ASoC: mxs: remove __dev* attributes ASoC: kirkwood: remove __dev* attributes ...
Diffstat (limited to 'sound/pci/hda/patch_conexant.c')
-rw-r--r--sound/pci/hda/patch_conexant.c99
1 files changed, 78 insertions, 21 deletions
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 03b1dc317ff0..60890bfecc19 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -337,6 +337,8 @@ static const struct hda_pcm_stream cx5051_pcm_analog_capture = {
},
};
+static bool is_2_1_speaker(struct conexant_spec *spec);
+
static int conexant_build_pcms(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@@ -351,6 +353,9 @@ static int conexant_build_pcms(struct hda_codec *codec)
spec->multiout.max_channels;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
spec->multiout.dac_nids[0];
+ if (is_2_1_speaker(spec))
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap =
+ snd_pcm_2_1_chmaps;
if (spec->capture_stream)
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream;
else {
@@ -472,7 +477,7 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = {
#endif
static const char * const slave_pfxs[] = {
- "Headphone", "Speaker", "Front", "Surround", "CLFE",
+ "Headphone", "Speaker", "Bass Speaker", "Front", "Surround", "CLFE",
NULL
};
@@ -3430,28 +3435,13 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct conexant_spec *spec = codec->spec;
- static const char * const texts2[] = {
- "Disabled", "Enabled"
- };
static const char * const texts3[] = {
"Disabled", "Speaker Only", "Line Out+Speaker"
};
- const char * const *texts;
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- if (spec->automute_hp_lo) {
- uinfo->value.enumerated.items = 3;
- texts = texts3;
- } else {
- uinfo->value.enumerated.items = 2;
- texts = texts2;
- }
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
- return 0;
+ if (spec->automute_hp_lo)
+ return snd_hda_enum_helper_info(kcontrol, uinfo, 3, texts3);
+ return snd_hda_enum_bool_helper_info(kcontrol, uinfo);
}
static int cx_automute_mode_get(struct snd_kcontrol *kcontrol,
@@ -4116,11 +4106,26 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
return 0;
}
+static bool is_2_1_speaker(struct conexant_spec *spec)
+{
+ int i, type, num_spk = 0;
+
+ for (i = 0; i < spec->dac_info_filled; i++) {
+ type = spec->dac_info[i].type;
+ if (type == AUTO_PIN_LINE_OUT)
+ type = spec->autocfg.line_out_type;
+ if (type == AUTO_PIN_SPEAKER_OUT)
+ num_spk++;
+ }
+ return (num_spk == 2 && spec->autocfg.line_out_type != AUTO_PIN_LINE_OUT);
+}
+
static int cx_auto_build_output_controls(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
int i, err;
int num_line = 0, num_hp = 0, num_spk = 0;
+ bool speaker_2_1;
static const char * const texts[3] = { "Front", "Surround", "CLFE" };
if (spec->dac_info_filled == 1)
@@ -4128,6 +4133,8 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
spec->dac_info[0].pin,
"Master", 0);
+ speaker_2_1 = is_2_1_speaker(spec);
+
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
@@ -4146,8 +4153,13 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
idx = num_hp++;
break;
case AUTO_PIN_SPEAKER_OUT:
- label = "Speaker";
- idx = num_spk++;
+ if (speaker_2_1) {
+ label = num_spk++ ? "Bass Speaker" : "Speaker";
+ idx = 0;
+ } else {
+ label = "Speaker";
+ idx = num_spk++;
+ }
break;
}
err = try_add_pb_volume(codec, dac,
@@ -4405,7 +4417,10 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
enum {
CXT_PINCFG_LENOVO_X200,
CXT_PINCFG_LENOVO_TP410,
+ CXT_PINCFG_LEMOTE_A1004,
+ CXT_PINCFG_LEMOTE_A1205,
CXT_FIXUP_STEREO_DMIC,
+ CXT_FIXUP_INC_MIC_BOOST,
};
static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -4415,6 +4430,19 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
spec->fixup_stereo_dmic = 1;
}
+static void cxt5066_increase_mic_boost(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action != HDA_FIXUP_ACT_PRE_PROBE)
+ return;
+
+ snd_hda_override_amp_caps(codec, 0x17, HDA_OUTPUT,
+ (0x3 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x4 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (0 << AC_AMPCAP_MUTE_SHIFT));
+}
+
/* ThinkPad X200 & co with cxt5051 */
static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -4432,6 +4460,18 @@ static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = {
{}
};
+/* Lemote A1004/A1205 with cxt5066 */
+static const struct hda_pintbl cxt_pincfg_lemote[] = {
+ { 0x1a, 0x90a10020 }, /* Internal mic */
+ { 0x1b, 0x03a11020 }, /* External mic */
+ { 0x1d, 0x400101f0 }, /* Not used */
+ { 0x1e, 0x40a701f0 }, /* Not used */
+ { 0x20, 0x404501f0 }, /* Not used */
+ { 0x22, 0x404401f0 }, /* Not used */
+ { 0x23, 0x40a701f0 }, /* Not used */
+ {}
+};
+
static const struct hda_fixup cxt_fixups[] = {
[CXT_PINCFG_LENOVO_X200] = {
.type = HDA_FIXUP_PINS,
@@ -4441,10 +4481,24 @@ static const struct hda_fixup cxt_fixups[] = {
.type = HDA_FIXUP_PINS,
.v.pins = cxt_pincfg_lenovo_tp410,
},
+ [CXT_PINCFG_LEMOTE_A1004] = {
+ .type = HDA_FIXUP_PINS,
+ .chained = true,
+ .chain_id = CXT_FIXUP_INC_MIC_BOOST,
+ .v.pins = cxt_pincfg_lemote,
+ },
+ [CXT_PINCFG_LEMOTE_A1205] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = cxt_pincfg_lemote,
+ },
[CXT_FIXUP_STEREO_DMIC] = {
.type = HDA_FIXUP_FUNC,
.v.func = cxt_fixup_stereo_dmic,
},
+ [CXT_FIXUP_INC_MIC_BOOST] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt5066_increase_mic_boost,
+ },
};
static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -4453,6 +4507,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
};
static const struct snd_pci_quirk cxt5066_fixups[] = {
+ SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -4461,6 +4516,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004),
+ SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205),
{}
};