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-rw-r--r--sound/arm/pxa2xx-ac97-lib.c26
-rw-r--r--sound/oss/pas2_card.c5
-rw-r--r--sound/pci/au88x0/au88x0_synth.c2
-rw-r--r--sound/pci/hda/hda_codec.c2
-rw-r--r--sound/pci/hda/patch_realtek.c30
-rw-r--r--sound/soc/codecs/arizona.c9
-rw-r--r--sound/soc/codecs/arizona.h18
-rw-r--r--sound/soc/codecs/cs4271.c6
-rw-r--r--sound/soc/codecs/cs42l52.c4
-rw-r--r--sound/soc/codecs/lm49453.c106
-rw-r--r--sound/soc/codecs/wm5102.c48
-rw-r--r--sound/soc/soc-core.c35
-rw-r--r--sound/soc/soc-pcm.c1
-rw-r--r--sound/usb/midi.c4
-rw-r--r--sound/usb/quirks-table.h24
-rw-r--r--sound/usb/quirks.c16
-rw-r--r--sound/usb/usbaudio.h1
17 files changed, 224 insertions, 113 deletions
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 6fc0ae90e5b1..fff7753e35c1 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -18,6 +18,7 @@
#include <linux/delay.h>
#include <linux/module.h>
#include <linux/io.h>
+#include <linux/gpio.h>
#include <sound/ac97_codec.h>
#include <sound/pxa2xx-lib.h>
@@ -148,6 +149,8 @@ static inline void pxa_ac97_warm_pxa27x(void)
static inline void pxa_ac97_cold_pxa27x(void)
{
+ unsigned int timeout;
+
GCR &= GCR_COLD_RST; /* clear everything but nCRST */
GCR &= ~GCR_COLD_RST; /* then assert nCRST */
@@ -157,8 +160,10 @@ static inline void pxa_ac97_cold_pxa27x(void)
clk_enable(ac97conf_clk);
udelay(5);
clk_disable(ac97conf_clk);
- GCR = GCR_COLD_RST;
- udelay(50);
+ GCR = GCR_COLD_RST | GCR_WARM_RST;
+ timeout = 100; /* wait for the codec-ready bit to be set */
+ while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
+ mdelay(1);
}
#endif
@@ -340,8 +345,21 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
}
if (cpu_is_pxa27x()) {
- /* Use GPIO 113 as AC97 Reset on Bulverde */
+ /*
+ * This gpio is needed for a work-around to a bug in the ac97
+ * controller during warm reset. The direction and level is set
+ * here so that it is an output driven high when switching from
+ * AC97_nRESET alt function to generic gpio.
+ */
+ ret = gpio_request_one(reset_gpio, GPIOF_OUT_INIT_HIGH,
+ "pxa27x ac97 reset");
+ if (ret < 0) {
+ pr_err("%s: gpio_request_one() failed: %d\n",
+ __func__, ret);
+ goto err_conf;
+ }
pxa27x_assert_ac97reset(reset_gpio, 0);
+
ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
if (IS_ERR(ac97conf_clk)) {
ret = PTR_ERR(ac97conf_clk);
@@ -384,6 +402,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe);
void pxa2xx_ac97_hw_remove(struct platform_device *dev)
{
+ if (cpu_is_pxa27x())
+ gpio_free(reset_gpio);
GCR |= GCR_ACLINK_OFF;
free_irq(IRQ_AC97, NULL);
if (ac97conf_clk) {
diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c
index dabf8a871dcc..7004e24d209f 100644
--- a/sound/oss/pas2_card.c
+++ b/sound/oss/pas2_card.c
@@ -333,6 +333,11 @@ static void __init attach_pas_card(struct address_info *hw_config)
{
char temp[100];
+ if (pas_model < 0 ||
+ pas_model >= ARRAY_SIZE(pas_model_names)) {
+ printk(KERN_ERR "pas2 unrecognized model.\n");
+ return;
+ }
sprintf(temp,
"%s rev %d", pas_model_names[(int) pas_model],
pas_read(0x2789));
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index 2805e34bd41d..8bef47311e45 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -58,7 +58,7 @@ static void vortex_wt_setdsout(vortex_t * vortex, u32 wt, int en)
if (en)
temp |= (1 << (wt & 0x1f));
else
- temp &= (1 << ~(wt & 0x1f));
+ temp &= ~(1 << (wt & 0x1f));
hwwrite(vortex->mmio, WT_DSREG((wt >= 0x20) ? 1 : 0), temp);
}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8353c77536ac..b8fb0a5adb9b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2531,7 +2531,7 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char * const texts[] = {
- "Off", "On", "Follow Master"
+ "On", "Off", "Follow Master"
};
unsigned int index;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6ee34593774a..71ae23dd7103 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5992,6 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec,
spec->automute_hook = alc269_quanta_automute;
}
+/* update mute-LED according to the speaker mute state via mic1 VREF pin */
+static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ unsigned int pinval = AC_PINCTL_IN_EN + (enabled ?
+ AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80);
+ snd_hda_set_pin_ctl_cache(codec, 0x18, pinval);
+}
+
+static void alc269_fixup_mic1_mute(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ switch (action) {
+ case ALC_FIXUP_ACT_BUILD:
+ spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook;
+ snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true);
+ /* fallthru */
+ case ALC_FIXUP_ACT_INIT:
+ snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+ break;
+ }
+}
+
/* update mute-LED according to the speaker mute state via mic2 VREF pin */
static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled)
{
@@ -6043,6 +6067,7 @@ enum {
ALC269_FIXUP_DMIC,
ALC269VB_FIXUP_AMIC,
ALC269VB_FIXUP_DMIC,
+ ALC269_FIXUP_MIC1_MUTE_LED,
ALC269_FIXUP_MIC2_MUTE_LED,
ALC269_FIXUP_INV_DMIC,
ALC269_FIXUP_LENOVO_DOCK,
@@ -6171,6 +6196,10 @@ static const struct alc_fixup alc269_fixups[] = {
{ }
},
},
+ [ALC269_FIXUP_MIC1_MUTE_LED] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_mic1_mute,
+ },
[ALC269_FIXUP_MIC2_MUTE_LED] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc269_fixup_mic2_mute,
@@ -6215,6 +6244,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
+ SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index adf397b9d0e6..1d8bb5917594 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
mode = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- mode = 1;
- break;
case SND_SOC_DAIFMT_I2S:
mode = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- mode = 3;
- break;
default:
arizona_aif_err(dai, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
- ARIZONA_AIF1_RATE_MASK, 8);
+ ARIZONA_AIF1_RATE_MASK,
+ 8 << ARIZONA_AIF1_RATE_SHIFT);
break;
default:
arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 41dae1ed3b71..4deebeb07177 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -34,15 +34,15 @@
#define ARIZONA_FLL_SRC_MCLK1 0
#define ARIZONA_FLL_SRC_MCLK2 1
-#define ARIZONA_FLL_SRC_SLIMCLK 2
-#define ARIZONA_FLL_SRC_FLL1 3
-#define ARIZONA_FLL_SRC_FLL2 4
-#define ARIZONA_FLL_SRC_AIF1BCLK 5
-#define ARIZONA_FLL_SRC_AIF2BCLK 6
-#define ARIZONA_FLL_SRC_AIF3BCLK 7
-#define ARIZONA_FLL_SRC_AIF1LRCLK 8
-#define ARIZONA_FLL_SRC_AIF2LRCLK 9
-#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+#define ARIZONA_FLL_SRC_SLIMCLK 3
+#define ARIZONA_FLL_SRC_FLL1 4
+#define ARIZONA_FLL_SRC_FLL2 5
+#define ARIZONA_FLL_SRC_AIF1BCLK 8
+#define ARIZONA_FLL_SRC_AIF2BCLK 9
+#define ARIZONA_FLL_SRC_AIF3BCLK 10
+#define ARIZONA_FLL_SRC_AIF1LRCLK 12
+#define ARIZONA_FLL_SRC_AIF2LRCLK 13
+#define ARIZONA_FLL_SRC_AIF3LRCLK 14
#define ARIZONA_MIXER_VOL_MASK 0x00FE
#define ARIZONA_MIXER_VOL_SHIFT 1
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 4f1127935fdf..ac8742a1f25a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec)
struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
int ret;
int gpio_nreset = -EINVAL;
- int amutec_eq_bmutec = 0;
+ bool amutec_eq_bmutec = false;
#ifdef CONFIG_OF
if (of_match_device(cs4271_dt_ids, codec->dev)) {
gpio_nreset = of_get_named_gpio(codec->dev->of_node,
"reset-gpio", 0);
- if (!of_get_property(codec->dev->of_node,
+ if (of_get_property(codec->dev->of_node,
"cirrus,amutec-eq-bmutec", NULL))
- amutec_eq_bmutec = 1;
+ amutec_eq_bmutec = true;
}
#endif
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 99bb1c69499e..9811a5478c87 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = {
static int cs42l52_get_clk(int mclk, int rate)
{
- int i, ret = 0;
+ int i, ret = -EINVAL;
u_int mclk1, mclk2 = 0;
for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
@@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate)
}
}
}
- if (ret > ARRAY_SIZE(clk_map_table))
- return -EINVAL;
return ret;
}
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index d75257d40a49..e19490cfb3a8 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = {
{ 101, 0x00 },
{ 102, 0x00 },
{ 103, 0x01 },
- { 105, 0x01 },
- { 106, 0x00 },
- { 107, 0x01 },
+ { 104, 0x01 },
+ { 105, 0x00 },
+ { 106, 0x01 },
{ 107, 0x00 },
{ 108, 0x00 },
{ 109, 0x00 },
@@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = {
{ 184, 0x00 },
{ 185, 0x00 },
{ 186, 0x00 },
- { 189, 0x00 },
+ { 187, 0x00 },
{ 188, 0x00 },
- { 194, 0x00 },
- { 195, 0x00 },
- { 196, 0x00 },
- { 197, 0x00 },
- { 200, 0x00 },
- { 201, 0x00 },
- { 202, 0x00 },
- { 203, 0x00 },
- { 204, 0x00 },
- { 205, 0x00 },
- { 208, 0x00 },
+ { 189, 0x00 },
+ { 208, 0x06 },
{ 209, 0x00 },
- { 210, 0x00 },
- { 211, 0x00 },
- { 213, 0x00 },
- { 214, 0x00 },
- { 215, 0x00 },
- { 216, 0x00 },
- { 217, 0x00 },
- { 218, 0x00 },
- { 219, 0x00 },
+ { 210, 0x08 },
+ { 211, 0x54 },
+ { 212, 0x14 },
+ { 213, 0x0d },
+ { 214, 0x0d },
+ { 215, 0x14 },
+ { 216, 0x60 },
{ 221, 0x00 },
{ 222, 0x00 },
+ { 223, 0x00 },
{ 224, 0x00 },
- { 225, 0x00 },
- { 226, 0x00 },
- { 227, 0x00 },
- { 228, 0x00 },
- { 229, 0x00 },
- { 230, 0x13 },
- { 231, 0x00 },
- { 232, 0x80 },
- { 233, 0x0C },
- { 234, 0xDD },
- { 235, 0x00 },
- { 236, 0x04 },
- { 237, 0x00 },
- { 238, 0x00 },
- { 239, 0x00 },
- { 240, 0x00 },
- { 241, 0x00 },
- { 242, 0x00 },
- { 243, 0x00 },
- { 244, 0x00 },
- { 245, 0x00 },
{ 248, 0x00 },
{ 249, 0x00 },
- { 254, 0x00 },
+ { 250, 0x00 },
{ 255, 0x00 },
};
@@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
};
/* TLV Declarations */
-static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
-static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0);
+static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0);
static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
/* Sidetone supports mono only */
SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
};
static const struct snd_kcontrol_new lm49453_snd_controls[] = {
/* mic1 and mic2 supports mono only */
- SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
- 0, digital_tlv),
- SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
- 0, digital_tlv),
+ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv),
+ SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv),
+
+ SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63,
+ 0, adc_dac_tlv),
+ SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63,
+ 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
- LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
- LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
@@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = {
2, 1, 0),
SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
- LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
- LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
- LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
- LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
- 0, 6, 0, digital_tlv),
+ 0, 63, 0, adc_dac_tlv),
SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
0, 3, 0, port_tlv),
@@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
}
snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
- LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+ LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5),
(aif_val | mode | clk_phase));
snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 688ade080589..7a9048dad1cd 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,6 +36,9 @@
struct wm5102_priv {
struct arizona_priv core;
struct arizona_fll fll[2];
+
+ unsigned int spk_ena:2;
+ unsigned int spk_ena_pending:1;
};
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
};
+static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+ if (arizona->rev < 1)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (!wm5102->spk_ena) {
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ wm5102->spk_ena_pending = true;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ if (wm5102->spk_ena_pending) {
+ msleep(75);
+ snd_soc_write(codec, 0x4f5, 0xda);
+ wm5102->spk_ena_pending = false;
+ wm5102->spk_ena++;
+ }
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ wm5102->spk_ena--;
+ if (!wm5102->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ if (!wm5102->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x0da);
+ break;
+ }
+
+ return 0;
+}
+
+
ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 91d592ff67b7..2370063b5824 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card,
INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
ret = device_add(rtd->dev);
if (ret < 0) {
+ /* calling put_device() here to free the rtd->dev */
+ put_device(rtd->dev);
dev_err(card->dev,
"ASoC: failed to register runtime device: %d\n", ret);
return ret;
@@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
/* unregister the rtd device */
if (rtd->dev_registered) {
device_remove_file(rtd->dev, &dev_attr_codec_reg);
- device_del(rtd->dev);
+ device_unregister(rtd->dev);
rtd->dev_registered = 0;
}
@@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
platform_max = mc->platform_max;
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- uinfo->count = 1;
+ uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = platform_max - min;
@@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
+ unsigned int rreg = mc->rreg;
unsigned int shift = mc->shift;
int min = mc->min;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
unsigned int val, val_mask;
+ int ret;
val = ((ucontrol->value.integer.value[0] + min) & mask);
if (invert)
@@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
val_mask = mask << shift;
val = val << shift;
- return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+ ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
+ if (ret != 0)
+ return ret;
+
+ if (snd_soc_volsw_is_stereo(mc)) {
+ val = ((ucontrol->value.integer.value[1] + min) & mask);
+ if (invert)
+ val = max - val;
+ val_mask = mask << shift;
+ val = val << shift;
+
+ ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val);
+ }
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
@@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
+ unsigned int rreg = mc->rreg;
unsigned int shift = mc->shift;
int min = mc->min;
int max = mc->max;
@@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
ucontrol->value.integer.value[0] =
ucontrol->value.integer.value[0] - min;
+ if (snd_soc_volsw_is_stereo(mc)) {
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, rreg) >> shift) & mask;
+ if (invert)
+ ucontrol->value.integer.value[1] =
+ max - ucontrol->value.integer.value[1];
+ ucontrol->value.integer.value[1] =
+ ucontrol->value.integer.value[1] - min;
+ }
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d7711fce119b..cf191e6aebbe 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
continue;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index c183d34842ac..34b9bb7fe87c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -2181,10 +2181,6 @@ int snd_usbmidi_create(struct snd_card *card,
umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
break;
- case QUIRK_MIDI_MBOX2:
- umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops;
- err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
- break;
case QUIRK_MIDI_RAW_BYTES:
umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
/*
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index cdcf6b45e8a8..78e845ec65da 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -50,6 +50,28 @@
}
},
+{
+ /* Creative BT-D1 */
+ USB_DEVICE(0x041e, 0x0005),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x03,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .attributes = 0,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ }
+ }
+},
+
/* Creative/Toshiba Multimedia Center SB-0500 */
{
USB_DEVICE(0x041e, 0x3048),
@@ -2993,7 +3015,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
},
{
.ifnum = 6,
- .type = QUIRK_MIDI_MBOX2,
+ .type = QUIRK_MIDI_MIDIMAN,
.data = &(const struct snd_usb_midi_endpoint_info) {
.out_ep = 0x02,
.out_cables = 0x0001,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index f104c68fe1e0..acc12f004c23 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -306,7 +306,6 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_MIDI_YAMAHA] = create_any_midi_quirk,
[QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk,
[QUIRK_MIDI_NOVATION] = create_any_midi_quirk,
- [QUIRK_MIDI_MBOX2] = create_any_midi_quirk,
[QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk,
[QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
[QUIRK_MIDI_CME] = create_any_midi_quirk,
@@ -528,11 +527,11 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev)
#define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */
#define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */
-int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
+static int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
{
struct usb_host_config *config = dev->actconfig;
int err;
- u8 bootresponse;
+ u8 bootresponse[12];
int fwsize;
int count;
@@ -546,20 +545,20 @@ int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
snd_printd("usb-audio: Sending Digidesign Mbox 2 boot sequence...\n");
count = 0;
- bootresponse = MBOX2_BOOT_LOADING;
- while ((bootresponse == MBOX2_BOOT_LOADING) && (count < 10)) {
+ bootresponse[0] = MBOX2_BOOT_LOADING;
+ while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)) {
msleep(500); /* 0.5 second delay */
snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
/* Control magic - load onboard firmware */
0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012);
- if (bootresponse == MBOX2_BOOT_READY)
+ if (bootresponse[0] == MBOX2_BOOT_READY)
break;
snd_printd("usb-audio: device not ready, resending boot sequence...\n");
count++;
}
- if (bootresponse != MBOX2_BOOT_READY) {
- snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse);
+ if (bootresponse[0] != MBOX2_BOOT_READY) {
+ snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse[0]);
return -ENODEV;
}
@@ -660,7 +659,6 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
return 0; /* keep this altsetting */
}
-
static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip,
int iface, int altno)
{
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index a8172c119796..1ac3fd9cc5a6 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -76,7 +76,6 @@ enum quirk_type {
QUIRK_MIDI_YAMAHA,
QUIRK_MIDI_MIDIMAN,
QUIRK_MIDI_NOVATION,
- QUIRK_MIDI_MBOX2,
QUIRK_MIDI_RAW_BYTES,
QUIRK_MIDI_EMAGIC,
QUIRK_MIDI_CME,
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-22 17:22:05 +0000