Module Name: src
Committed By: jmcneill
Date: Sun May 13 01:01:37 UTC 2018
Modified Files:
src/sys/arch/arm/sunxi: sun50i_a64_acodec.c
Log Message:
Mixer improvements:
- outputs.master switches between line and hp volume control, depending
on jack sense
- add outputs.source to override jack sense output routing
- (playback) line and hp controls are in the input class
- (capture) line and mic controls are in the record class
To generate a diff of this commit:
cvs rdiff -u -r1.4 -r1.5 src/sys/arch/arm/sunxi/sun50i_a64_acodec.c
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/arch/arm/sunxi/sun50i_a64_acodec.c
diff -u src/sys/arch/arm/sunxi/sun50i_a64_acodec.c:1.4 src/sys/arch/arm/sunxi/sun50i_a64_acodec.c:1.5
--- src/sys/arch/arm/sunxi/sun50i_a64_acodec.c:1.4 Fri May 11 23:05:41 2018
+++ src/sys/arch/arm/sunxi/sun50i_a64_acodec.c Sun May 13 01:01:37 2018
@@ -1,4 +1,4 @@
-/* $NetBSD: sun50i_a64_acodec.c,v 1.4 2018/05/11 23:05:41 jmcneill Exp $ */
+/* $NetBSD: sun50i_a64_acodec.c,v 1.5 2018/05/13 01:01:37 jmcneill Exp $ */
/*-
* Copyright (c) 2018 Jared McNeill <jmcne...@invisible.ca>
@@ -27,7 +27,7 @@
*/
#include <sys/cdefs.h>
-__KERNEL_RCSID(0, "$NetBSD: sun50i_a64_acodec.c,v 1.4 2018/05/11 23:05:41 jmcneill Exp $");
+__KERNEL_RCSID(0, "$NetBSD: sun50i_a64_acodec.c,v 1.5 2018/05/13 01:01:37 jmcneill Exp $");
#include <sys/param.h>
#include <sys/bus.h>
@@ -57,6 +57,7 @@ __KERNEL_RCSID(0, "$NetBSD: sun50i_a64_a
#define A64_LINEOUT_CTRL0 0x05
#define A64_LINEOUT_LEFT_EN __BIT(7)
#define A64_LINEOUT_RIGHT_EN __BIT(6)
+#define A64_LINEOUT_EN (A64_LINEOUT_LEFT_EN|A64_LINEOUT_RIGHT_EN)
#define A64_LINEOUT_CTRL1 0x06
#define A64_LINEOUT_VOL __BITS(4,0)
#define A64_MIC1_CTRL 0x07
@@ -101,6 +102,7 @@ struct a64_acodec_softc {
int sc_phandle;
struct audio_dai_device sc_dai;
+ int sc_master_dev;
};
enum a64_acodec_mixer_ctrl {
@@ -109,35 +111,38 @@ enum a64_acodec_mixer_ctrl {
A64_CODEC_RECORD_CLASS,
A64_CODEC_OUTPUT_MASTER_VOLUME,
- A64_CODEC_OUTPUT_HP_VOLUME,
- A64_CODEC_INPUT_DAC_VOLUME,
- A64_CODEC_INPUT_LINEIN_VOLUME,
- A64_CODEC_INPUT_MIC1_VOLUME,
- A64_CODEC_INPUT_MIC2_VOLUME,
+ A64_CODEC_OUTPUT_SOURCE,
+ A64_CODEC_INPUT_LINE_VOLUME,
+ A64_CODEC_INPUT_HP_VOLUME,
+ A64_CODEC_RECORD_LINE_VOLUME,
+ A64_CODEC_RECORD_MIC1_VOLUME,
+ A64_CODEC_RECORD_MIC2_VOLUME,
A64_CODEC_RECORD_AGC_VOLUME,
A64_CODEC_RECORD_SOURCE,
A64_CODEC_MIXER_CTRL_LAST
};
+#define A64_OUTPUT_SOURCE_LINE __BIT(0)
+#define A64_OUTPUT_SOURCE_HP __BIT(1)
+
static const struct a64_acodec_mixer {
const char * name;
enum a64_acodec_mixer_ctrl mixer_class;
u_int reg;
u_int mask;
} a64_acodec_mixers[A64_CODEC_MIXER_CTRL_LAST] = {
- [A64_CODEC_OUTPUT_MASTER_VOLUME] = { AudioNmaster,
- A64_CODEC_OUTPUT_CLASS, A64_LINEOUT_CTRL1, A64_LINEOUT_VOL },
- [A64_CODEC_OUTPUT_HP_VOLUME] = { AudioNheadphone,
- A64_CODEC_OUTPUT_CLASS, A64_HP_CTRL, A64_HPVOL },
- [A64_CODEC_INPUT_DAC_VOLUME] = { AudioNdac,
+ [A64_CODEC_INPUT_LINE_VOLUME] = { AudioNline,
A64_CODEC_INPUT_CLASS, A64_LINEOUT_CTRL1, A64_LINEOUT_VOL },
- [A64_CODEC_INPUT_LINEIN_VOLUME] = { AudioNline,
- A64_CODEC_INPUT_CLASS, A64_LINEIN_CTRL, A64_LINEING },
- [A64_CODEC_INPUT_MIC1_VOLUME] = { "mic1",
- A64_CODEC_INPUT_CLASS, A64_MIC1_CTRL, A64_MIC1G },
- [A64_CODEC_INPUT_MIC2_VOLUME] = { "mic2",
- A64_CODEC_INPUT_CLASS, A64_MIC2_CTRL, A64_MIC2G },
+ [A64_CODEC_INPUT_HP_VOLUME] = { AudioNheadphone,
+ A64_CODEC_INPUT_CLASS, A64_HP_CTRL, A64_HPVOL },
+
+ [A64_CODEC_RECORD_LINE_VOLUME] = { AudioNline,
+ A64_CODEC_RECORD_CLASS, A64_LINEIN_CTRL, A64_LINEING },
+ [A64_CODEC_RECORD_MIC1_VOLUME] = { "mic1",
+ A64_CODEC_RECORD_CLASS, A64_MIC1_CTRL, A64_MIC1G },
+ [A64_CODEC_RECORD_MIC2_VOLUME] = { "mic2",
+ A64_CODEC_RECORD_CLASS, A64_MIC2_CTRL, A64_MIC2G },
[A64_CODEC_RECORD_AGC_VOLUME] = { AudioNagc,
A64_CODEC_RECORD_CLASS, A64_ADC_CTRL, A64_ADCG },
};
@@ -281,17 +286,20 @@ a64_acodec_set_port(void *priv, mixer_ct
struct a64_acodec_softc * const sc = priv;
const struct a64_acodec_mixer *mix;
u_int val, shift;
- int nvol;
+ int nvol, dev;
- switch (mc->dev) {
- case A64_CODEC_OUTPUT_MASTER_VOLUME:
- case A64_CODEC_OUTPUT_HP_VOLUME:
- case A64_CODEC_INPUT_DAC_VOLUME:
- case A64_CODEC_INPUT_LINEIN_VOLUME:
- case A64_CODEC_INPUT_MIC1_VOLUME:
- case A64_CODEC_INPUT_MIC2_VOLUME:
+ dev = mc->dev;
+ if (dev == A64_CODEC_OUTPUT_MASTER_VOLUME)
+ dev = sc->sc_master_dev;
+
+ switch (dev) {
+ case A64_CODEC_INPUT_LINE_VOLUME:
+ case A64_CODEC_INPUT_HP_VOLUME:
+ case A64_CODEC_RECORD_LINE_VOLUME:
+ case A64_CODEC_RECORD_MIC1_VOLUME:
+ case A64_CODEC_RECORD_MIC2_VOLUME:
case A64_CODEC_RECORD_AGC_VOLUME:
- mix = &a64_acodec_mixers[mc->dev];
+ mix = &a64_acodec_mixers[dev];
val = a64_acodec_pr_read(sc, mix->reg);
shift = 8 - fls32(__SHIFTOUT_MASK(mix->mask));
nvol = mc->un.value.level[AUDIO_MIXER_LEVEL_LEFT] >> shift;
@@ -300,6 +308,22 @@ a64_acodec_set_port(void *priv, mixer_ct
a64_acodec_pr_write(sc, mix->reg, val);
return 0;
+ case A64_CODEC_OUTPUT_SOURCE:
+ if (mc->un.mask & A64_OUTPUT_SOURCE_LINE)
+ a64_acodec_pr_set_clear(sc, A64_LINEOUT_CTRL0,
+ A64_LINEOUT_EN, 0);
+ else
+ a64_acodec_pr_set_clear(sc, A64_LINEOUT_CTRL0,
+ 0, A64_LINEOUT_EN);
+
+ if (mc->un.mask & A64_OUTPUT_SOURCE_HP)
+ a64_acodec_pr_set_clear(sc, A64_HP_CTRL,
+ A64_HPPA_EN, 0);
+ else
+ a64_acodec_pr_set_clear(sc, A64_HP_CTRL,
+ 0, A64_HPPA_EN);
+ return 0;
+
case A64_CODEC_RECORD_SOURCE:
a64_acodec_pr_write(sc, A64_L_ADCMIX_SRC, mc->un.mask);
a64_acodec_pr_write(sc, A64_R_ADCMIX_SRC, mc->un.mask);
@@ -315,17 +339,20 @@ a64_acodec_get_port(void *priv, mixer_ct
struct a64_acodec_softc * const sc = priv;
const struct a64_acodec_mixer *mix;
u_int val, shift;
- int nvol;
+ int nvol, dev;
- switch (mc->dev) {
- case A64_CODEC_OUTPUT_MASTER_VOLUME:
- case A64_CODEC_OUTPUT_HP_VOLUME:
- case A64_CODEC_INPUT_DAC_VOLUME:
- case A64_CODEC_INPUT_LINEIN_VOLUME:
- case A64_CODEC_INPUT_MIC1_VOLUME:
- case A64_CODEC_INPUT_MIC2_VOLUME:
+ dev = mc->dev;
+ if (dev == A64_CODEC_OUTPUT_MASTER_VOLUME)
+ dev = sc->sc_master_dev;
+
+ switch (dev) {
+ case A64_CODEC_INPUT_LINE_VOLUME:
+ case A64_CODEC_INPUT_HP_VOLUME:
+ case A64_CODEC_RECORD_LINE_VOLUME:
+ case A64_CODEC_RECORD_MIC1_VOLUME:
+ case A64_CODEC_RECORD_MIC2_VOLUME:
case A64_CODEC_RECORD_AGC_VOLUME:
- mix = &a64_acodec_mixers[mc->dev];
+ mix = &a64_acodec_mixers[dev];
val = a64_acodec_pr_read(sc, mix->reg);
shift = 8 - fls32(__SHIFTOUT_MASK(mix->mask));
nvol = __SHIFTOUT(val, mix->mask) << shift;
@@ -333,6 +360,14 @@ a64_acodec_get_port(void *priv, mixer_ct
mc->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = nvol;
return 0;
+ case A64_CODEC_OUTPUT_SOURCE:
+ mc->un.mask = 0;
+ if (a64_acodec_pr_read(sc, A64_LINEOUT_CTRL0) & A64_LINEOUT_EN)
+ mc->un.mask |= A64_OUTPUT_SOURCE_LINE;
+ if (a64_acodec_pr_read(sc, A64_HP_CTRL) & A64_HPPA_EN)
+ mc->un.mask |= A64_OUTPUT_SOURCE_HP;
+ return 0;
+
case A64_CODEC_RECORD_SOURCE:
mc->un.mask =
a64_acodec_pr_read(sc, A64_L_ADCMIX_SRC) |
@@ -346,6 +381,7 @@ a64_acodec_get_port(void *priv, mixer_ct
static int
a64_acodec_query_devinfo(void *priv, mixer_devinfo_t *di)
{
+ struct a64_acodec_softc * const sc = priv;
const struct a64_acodec_mixer *mix;
switch (di->index) {
@@ -371,11 +407,22 @@ a64_acodec_query_devinfo(void *priv, mix
return 0;
case A64_CODEC_OUTPUT_MASTER_VOLUME:
- case A64_CODEC_OUTPUT_HP_VOLUME:
- case A64_CODEC_INPUT_DAC_VOLUME:
- case A64_CODEC_INPUT_LINEIN_VOLUME:
- case A64_CODEC_INPUT_MIC1_VOLUME:
- case A64_CODEC_INPUT_MIC2_VOLUME:
+ mix = &a64_acodec_mixers[sc->sc_master_dev];
+ di->mixer_class = A64_CODEC_OUTPUT_CLASS;
+ strcpy(di->label.name, AudioNmaster);
+ di->un.v.delta =
+ 256 / (__SHIFTOUT_MASK(mix->mask) + 1);
+ di->type = AUDIO_MIXER_VALUE;
+ di->next = di->prev = AUDIO_MIXER_LAST;
+ di->un.v.num_channels = 2;
+ strcpy(di->un.v.units.name, AudioNvolume);
+ return 0;
+
+ case A64_CODEC_INPUT_LINE_VOLUME:
+ case A64_CODEC_INPUT_HP_VOLUME:
+ case A64_CODEC_RECORD_LINE_VOLUME:
+ case A64_CODEC_RECORD_MIC1_VOLUME:
+ case A64_CODEC_RECORD_MIC2_VOLUME:
case A64_CODEC_RECORD_AGC_VOLUME:
mix = &a64_acodec_mixers[di->index];
di->mixer_class = mix->mixer_class;
@@ -388,6 +435,18 @@ a64_acodec_query_devinfo(void *priv, mix
strcpy(di->un.v.units.name, AudioNvolume);
return 0;
+ case A64_CODEC_OUTPUT_SOURCE:
+ di->mixer_class = A64_CODEC_OUTPUT_CLASS;
+ strcpy(di->label.name, AudioNsource);
+ di->type = AUDIO_MIXER_SET;
+ di->next = di->prev = AUDIO_MIXER_LAST;
+ di->un.s.num_mem = 2;
+ strcpy(di->un.s.member[0].label.name, AudioNline);
+ di->un.s.member[0].mask = A64_OUTPUT_SOURCE_LINE;
+ strcpy(di->un.s.member[1].label.name, AudioNheadphone);
+ di->un.s.member[1].mask = A64_OUTPUT_SOURCE_HP;
+ return 0;
+
case A64_CODEC_RECORD_SOURCE:
di->mixer_class = A64_CODEC_RECORD_CLASS;
strcpy(di->label.name, AudioNsource);
@@ -438,23 +497,27 @@ static int
a64_acodec_dai_jack_detect(audio_dai_tag_t dai, u_int jack, int present)
{
struct a64_acodec_softc * const sc = audio_dai_private(dai);
- const uint32_t lineout_mask = A64_LINEOUT_LEFT_EN | A64_LINEOUT_RIGHT_EN;
- const uint32_t hppa_mask = A64_HPPA_EN;
switch (jack) {
case AUDIO_DAI_JACK_HP:
if (present) {
a64_acodec_pr_set_clear(sc, A64_LINEOUT_CTRL0,
- 0, lineout_mask);
+ 0, A64_LINEOUT_EN);
a64_acodec_pr_set_clear(sc, A64_HP_CTRL,
- hppa_mask, 0);
+ A64_HPPA_EN, 0);
} else {
a64_acodec_pr_set_clear(sc, A64_LINEOUT_CTRL0,
- lineout_mask, 0);
+ A64_LINEOUT_EN, 0);
a64_acodec_pr_set_clear(sc, A64_HP_CTRL,
- 0, hppa_mask);
+ 0, A64_HPPA_EN);
}
+
+ /* Master volume controls either HP or line out */
+ sc->sc_master_dev = present ?
+ A64_CODEC_INPUT_HP_VOLUME : A64_CODEC_INPUT_LINE_VOLUME;
+
break;
+
case AUDIO_DAI_JACK_MIC:
/* XXX TODO */
break;
@@ -506,6 +569,7 @@ a64_acodec_attach(device_t parent, devic
A64_HPPA_EN, 0);
/* Jack detect enable */
+ sc->sc_master_dev = A64_CODEC_INPUT_HP_VOLUME;
a64_acodec_pr_set_clear(sc, A64_JACK_MIC_CTRL,
A64_JACKDETEN | A64_INNERRESEN | A64_AUTOPLEN, 0);
