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-rw-r--r--sound/arm/pxa2xx-ac97-lib.c26
-rw-r--r--sound/pci/hda/hda_codec.c5
-rw-r--r--sound/pci/hda/hda_intel.c13
-rw-r--r--sound/pci/hda/patch_conexant.c25
-rw-r--r--sound/pci/hda/patch_hdmi.c2
-rw-r--r--sound/pci/hda/patch_realtek.c26
-rw-r--r--sound/pci/rme9652/hdspm.c17
-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/sgtl5000.c4
-rw-r--r--sound/soc/codecs/sta529.c9
-rw-r--r--sound/soc/codecs/wm2000.c4
-rw-r--r--sound/soc/codecs/wm2200.c8
-rw-r--r--sound/soc/codecs/wm5100.c6
-rw-r--r--sound/soc/codecs/wm5102.c48
-rw-r--r--sound/soc/codecs/wm_adsp.c23
-rw-r--r--sound/soc/soc-core.c35
-rw-r--r--sound/soc/soc-pcm.c1
-rw-r--r--sound/usb/mixer_maps.c13
-rw-r--r--sound/usb/mixer_quirks.c2
-rw-r--r--sound/usb/pcm.c10
-rw-r--r--sound/usb/quirks-table.h4
-rw-r--r--sound/usb/quirks.c19
26 files changed, 288 insertions, 155 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/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b8fb0a5adb9b..822df971972c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3654,6 +3654,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
hda_set_power_state(codec, AC_PWRST_D0);
restore_shutup_pins(codec);
hda_exec_init_verbs(codec);
+ snd_hda_jack_set_dirty_all(codec);
if (codec->patch_ops.resume)
codec->patch_ops.resume(codec);
else {
@@ -3665,10 +3666,8 @@ static void hda_call_codec_resume(struct hda_codec *codec)
if (codec->jackpoll_interval)
hda_jackpoll_work(&codec->jackpoll_work.work);
- else {
- snd_hda_jack_set_dirty_all(codec);
+ else
snd_hda_jack_report_sync(codec);
- }
codec->in_pm = 0;
snd_hda_power_down(codec); /* flag down before returning */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cca87277baf0..0b6aebacc56b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -573,9 +573,12 @@ enum {
#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */
/* quirks for Intel PCH */
-#define AZX_DCAPS_INTEL_PCH \
+#define AZX_DCAPS_INTEL_PCH_NOPM \
(AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \
- AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME)
+ AZX_DCAPS_COUNT_LPIB_DELAY)
+
+#define AZX_DCAPS_INTEL_PCH \
+ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME)
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci)
static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
/* CPT */
{ PCI_DEVICE(0x8086, 0x1c20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
/* PBG */
{ PCI_DEVICE(0x8086, 0x1d20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
/* Panther Point */
{ PCI_DEVICE(0x8086, 0x1e20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
/* Lynx Point */
{ PCI_DEVICE(0x8086, 0x8c20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 60890bfecc19..009b77a693cf 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec)
return 0;
}
-#ifdef CONFIG_PM
-static int conexant_suspend(struct hda_codec *codec)
-{
- snd_hda_shutup_pins(codec);
- return 0;
-}
-#endif
-
static const struct hda_codec_ops conexant_patch_ops = {
.build_controls = conexant_build_controls,
.build_pcms = conexant_build_pcms,
.init = conexant_init,
.free = conexant_free,
.set_power_state = conexant_set_power,
-#ifdef CONFIG_PM
- .suspend = conexant_suspend,
-#endif
- .reboot_notify = snd_hda_shutup_pins,
};
#ifdef CONFIG_SND_HDA_INPUT_BEEP
@@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
.init = cx_auto_init,
.free = conexant_free,
.unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
- .suspend = conexant_suspend,
-#endif
- .reboot_notify = snd_hda_shutup_pins,
};
/*
@@ -4652,6 +4636,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_conexant_auto },
{ .id = 0x14f15111, .name = "CX20753/4",
.patch = patch_conexant_auto },
+ { .id = 0x14f15113, .name = "CX20755",
+ .patch = patch_conexant_auto },
+ { .id = 0x14f15114, .name = "CX20756",
+ .patch = patch_conexant_auto },
+ { .id = 0x14f15115, .name = "CX20757",
+ .patch = patch_conexant_auto },
{} /* terminator */
};
@@ -4675,6 +4665,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150b9");
MODULE_ALIAS("snd-hda-codec-id:14f1510f");
MODULE_ALIAS("snd-hda-codec-id:14f15110");
MODULE_ALIAS("snd-hda-codec-id:14f15111");
+MODULE_ALIAS("snd-hda-codec-id:14f15113");
+MODULE_ALIAS("snd-hda-codec-id:14f15114");
+MODULE_ALIAS("snd-hda-codec-id:14f15115");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index b6c21ea187ca..807a2aa1ff38 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1502,7 +1502,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol,
ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
substream = snd_pcm_chmap_substream(info, ctl_idx);
if (!substream || !substream->runtime)
- return -EBADFD;
+ return 0; /* just for avoiding error from alsactl restore */
switch (substream->runtime->status->state) {
case SNDRV_PCM_STATE_OPEN:
case SNDRV_PCM_STATE_SETUP:
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 71ae23dd7103..cf3886171109 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5817,6 +5817,9 @@ enum {
ALC269_TYPE_ALC269VB,
ALC269_TYPE_ALC269VC,
ALC269_TYPE_ALC269VD,
+ ALC269_TYPE_ALC280,
+ ALC269_TYPE_ALC282,
+ ALC269_TYPE_ALC284,
};
/*
@@ -5833,10 +5836,13 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
switch (spec->codec_variant) {
case ALC269_TYPE_ALC269VA:
case ALC269_TYPE_ALC269VC:
+ case ALC269_TYPE_ALC280:
+ case ALC269_TYPE_ALC284:
ssids = alc269va_ssids;
break;
case ALC269_TYPE_ALC269VB:
case ALC269_TYPE_ALC269VD:
+ case ALC269_TYPE_ALC282:
ssids = alc269_ssids;
break;
default:
@@ -6245,6 +6251,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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(0x103c, 0x1977, "HP Pavilion 14", 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),
@@ -6259,6 +6266,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+ SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
@@ -6400,7 +6408,8 @@ static int patch_alc269(struct hda_codec *codec)
alc_auto_parse_customize_define(codec);
- if (codec->vendor_id == 0x10ec0269) {
+ switch (codec->vendor_id) {
+ case 0x10ec0269:
spec->codec_variant = ALC269_TYPE_ALC269VA;
switch (alc_get_coef0(codec) & 0x00f0) {
case 0x0010:
@@ -6425,6 +6434,20 @@ static int patch_alc269(struct hda_codec *codec)
goto error;
spec->init_hook = alc269_fill_coef;
alc269_fill_coef(codec);
+ break;
+
+ case 0x10ec0280:
+ case 0x10ec0290:
+ spec->codec_variant = ALC269_TYPE_ALC280;
+ break;
+ case 0x10ec0282:
+ case 0x10ec0283:
+ spec->codec_variant = ALC269_TYPE_ALC282;
+ break;
+ case 0x10ec0284:
+ case 0x10ec0292:
+ spec->codec_variant = ALC269_TYPE_ALC284;
+ break;
}
/* automatic parse from the BIOS config */
@@ -7129,6 +7152,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
{ .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
{ .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
+ { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
{ .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
{ .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6e02e064d7b4..223c3d9cc69e 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
*/
/* status */
#define HDSPM_AES32_wcLock 0x0200000
+#define HDSPM_AES32_wcSync 0x0100000
#define HDSPM_AES32_wcFreq_bit 22
/* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function
HDSPM_bit2freq */
@@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm)
switch (hdspm->io_type) {
case AES32:
status = hdspm_read(hdspm, HDSPM_statusRegister);
- if (status & HDSPM_wcSync)
- return 2;
- else if (status & HDSPM_wcLock)
- return 1;
+ if (status & HDSPM_AES32_wcLock) {
+ if (status & HDSPM_AES32_wcSync)
+ return 2;
+ else
+ return 1;
+ }
return 0;
break;
@@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
unsigned int status;
unsigned int status2;
unsigned int timecode;
+ unsigned int wcLock, wcSync;
int pref_syncref;
char *autosync_ref;
int x;
@@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
snd_iprintf(buffer, "--- Status:\n");
+ wcLock = status & HDSPM_AES32_wcLock;
+ wcSync = wcLock && (status & HDSPM_AES32_wcSync);
+
snd_iprintf(buffer, "Word: %s Frequency: %d\n",
- (status & HDSPM_AES32_wcLock) ? "Sync " : "No Lock",
+ (wcLock) ? (wcSync ? "Sync " : "Lock ") : "No Lock",
HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF));
for (x = 0; x < 8; x++) {
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/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index cb1675cd8e1c..92bbfec9b107 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
5, 1, 0),
SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
- 0, 4, 0, mic_gain_tlv),
+ 0, 3, 0, mic_gain_tlv),
};
/* mute the codec used by alsa core */
@@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
SGTL5000_HP_ZCD_EN |
SGTL5000_ADC_ZCD_EN);
- snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+ snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
/*
* disable DAP
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index ab355c4f0b2d..40c07be9b581 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -74,9 +74,10 @@
SNDRV_PCM_FMTBIT_S32_LE)
#define S2PC_VALUE 0x98
#define CLOCK_OUT 0x60
-#define LEFT_J_DATA_FORMAT 0x10
-#define I2S_DATA_FORMAT 0x12
-#define RIGHT_J_DATA_FORMAT 0x14
+#define DATA_FORMAT_MSK 0x0E
+#define LEFT_J_DATA_FORMAT 0x00
+#define I2S_DATA_FORMAT 0x02
+#define RIGHT_J_DATA_FORMAT 0x04
#define CODEC_MUTE_VAL 0x80
#define POWER_CNTLMSAK 0x40
@@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
return -EINVAL;
}
- snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+ snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode);
return 0;
}
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 1cbe88f01d63..12bcae63a7f0 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
if (wm2000->speech_clarity)
- ret &= ~WM2000_SPEECH_CLARITY;
- else
ret |= WM2000_SPEECH_CLARITY;
+ else
+ ret &= ~WM2000_SPEECH_CLARITY;
wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index afcf31df77e0..e6cefe1ac677 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1566,15 +1566,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
fmt_val = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- fmt_val = 1;
- break;
case SND_SOC_DAIFMT_I2S:
fmt_val = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- fmt_val = 3;
- break;
default:
dev_err(codec->dev, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -1626,7 +1620,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
lrclk);
snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
- WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+ WM2200_AIF1_FMT_MASK, fmt_val);
return 0;
}
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 5a5f36936235..54397a508073 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
mask = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- mask = 1;
- break;
case SND_SOC_DAIFMT_I2S:
mask = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- mask = 3;
- break;
default:
dev_err(codec->dev, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
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/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ffc89fab96fb..7b198c38f3ef 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
const struct wm_adsp_region *mem;
const char *region_name;
char *file, *text;
+ void *buf;
unsigned int reg;
int regions = 0;
int ret, offset, type, sizes;
@@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp)
}
if (reg) {
- ret = regmap_raw_write(regmap, reg, region->data,
+ buf = kmemdup(region->data, le32_to_cpu(region->len),
+ GFP_KERNEL);
+ if (!buf) {
+ adsp_err(dsp, "Out of memory\n");
+ return -ENOMEM;
+ }
+
+ ret = regmap_raw_write(regmap, reg, buf,
le32_to_cpu(region->len));
+
+ kfree(buf);
+
if (ret != 0) {
adsp_err(dsp,
"%s.%d: Failed to write %d bytes at %d in %s: %d\n",
@@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
const char *region_name;
int ret, pos, blocks, type, offset, reg;
char *file;
+ void *buf;
file = kzalloc(PAGE_SIZE, GFP_KERNEL);
if (file == NULL)
@@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
}
if (reg) {
+ buf = kmemdup(blk->data, le32_to_cpu(blk->len),
+ GFP_KERNEL);
+ if (!buf) {
+ adsp_err(dsp, "Out of memory\n");
+ return -ENOMEM;
+ }
+
ret = regmap_raw_write(regmap, reg, blk->data,
le32_to_cpu(blk->len));
if (ret != 0) {
@@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
"%s.%d: Failed to write to %x in %s\n",
file, blocks, reg, region_name);
}
+
+ kfree(buf);
}
pos += le32_to_cpu(blk->len) + sizeof(*blk);
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/mixer_maps.c b/sound/usb/mixer_maps.c
index e71fe55cebef..0e2ed3d05c45 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -179,6 +179,15 @@ static struct usbmix_name_map audigy2nx_map[] = {
{ 0 } /* terminator */
};
+static struct usbmix_selector_map c400_selectors[] = {
+ {
+ .id = 0x80,
+ .count = 2,
+ .names = (const char*[]) {"Internal", "SPDIF"}
+ },
+ { 0 } /* terminator */
+};
+
static struct usbmix_selector_map audigy2nx_selectors[] = {
{
.id = 14, /* Capture Source */
@@ -367,6 +376,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
.map = hercules_usb51_map,
},
{
+ .id = USB_ID(0x0763, 0x2030),
+ .selector_map = c400_selectors,
+ },
+ {
.id = USB_ID(0x08bb, 0x2702),
.map = linex_map,
.ignore_ctl_error = 1,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 0422b1360af3..15520de1df56 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer)
* are valid they presents mono controls as L and R channels of
* stereo. So we provide a good mixer here.
*/
-struct std_mono_table ebox44_table[] = {
+static struct std_mono_table ebox44_table[] = {
{
.unitid = 4,
.control = 1,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index c6593101c049..d82e378d37cb 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -511,6 +511,16 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs)
struct snd_usb_substream *sync_subs =
&subs->stream->substream[subs->direction ^ 1];
+ if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA ||
+ !subs->stream)
+ return snd_usb_endpoint_set_params(subs->sync_endpoint,
+ subs->pcm_format,
+ subs->channels,
+ subs->period_bytes,
+ subs->cur_rate,
+ subs->cur_audiofmt,
+ NULL);
+
/* Try to find the best matching audioformat. */
list_for_each_entry(fp, &sync_subs->fmt_list, list) {
int score = match_endpoint_audioformats(fp, subs->cur_audiofmt,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 78e845ec65da..64d25a7a4d59 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2289,7 +2289,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.rate_table = (unsigned int[]) {
44100, 48000, 88200, 96000
},
- .clock = 0x81,
+ .clock = 0x80,
}
},
/* Capture */
@@ -2315,7 +2315,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.rate_table = (unsigned int[]) {
44100, 48000, 88200, 96000
},
- .clock = 0x81,
+ .clock = 0x80,
}
},
/* MIDI */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index acc12f004c23..2c971858d6b7 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -387,11 +387,13 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev)
* rules
*/
err = usb_driver_set_configuration(dev, 2);
- if (err < 0) {
+ if (err < 0)
snd_printdd("error usb_driver_set_configuration: %d\n",
err);
- return -ENODEV;
- }
+ /* Always return an error, so that we stop creating a device
+ that will just be destroyed and recreated with a new
+ configuration */
+ return -ENODEV;
} else
snd_printk(KERN_INFO "usb-audio: Fast Track Pro config OK\n");
@@ -859,6 +861,17 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep)
if ((le16_to_cpu(ep->chip->dev->descriptor.idVendor) == 0x23ba) &&
ep->type == SND_USB_ENDPOINT_TYPE_SYNC)
ep->skip_packets = 4;
+
+ /*
+ * M-Audio Fast Track C400 - when packets are not skipped, real world
+ * latency varies by approx. +/- 50 frames (at 96KHz) each time the
+ * stream is (re)started. When skipping packets 16 at endpoint start
+ * up, the real world latency is stable within +/- 1 frame (also
+ * across power cycles).
+ */
+ if (ep->chip->usb_id == USB_ID(0x0763, 0x2030) &&
+ ep->type == SND_USB_ENDPOINT_TYPE_DATA)
+ ep->skip_packets = 16;
}
void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-18 03:45:37 +0300