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-rw-r--r--sound/core/control.c4
-rw-r--r--sound/drivers/opl3/opl3_midi.c2
-rw-r--r--sound/firewire/iso-resources.c3
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c5
-rw-r--r--sound/isa/msnd/msnd_pinnacle_mixer.c3
-rw-r--r--sound/pci/hda/hda_controller.c2
-rw-r--r--sound/pci/hda/hda_generic.c30
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_conexant.c11
-rw-r--r--sound/pci/hda/patch_realtek.c7
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c68
-rw-r--r--sound/soc/cirrus/Kconfig2
-rw-r--r--sound/soc/codecs/Kconfig2
-rw-r--r--sound/soc/codecs/max98357a.c12
-rw-r--r--sound/soc/codecs/rt5670.c7
-rw-r--r--sound/soc/codecs/rt5677.c32
-rw-r--r--sound/soc/codecs/sta32x.c6
-rw-r--r--sound/soc/fsl/fsl_spdif.c4
-rw-r--r--sound/soc/fsl/fsl_ssi.c11
-rw-r--r--sound/soc/generic/simple-card.c5
-rw-r--r--sound/soc/intel/sst-atom-controls.h2
-rw-r--r--sound/soc/intel/sst/sst.c10
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c2
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c3
-rw-r--r--sound/soc/omap/omap-mcbsp.c11
-rw-r--r--sound/soc/omap/omap-pcm.c2
-rw-r--r--sound/soc/samsung/Kconfig10
-rw-r--r--sound/soc/sh/rcar/core.c4
-rw-r--r--sound/usb/line6/playback.c6
-rw-r--r--sound/usb/quirks-table.h30
30 files changed, 203 insertions, 95 deletions
diff --git a/sound/core/control.c b/sound/core/control.c
index 35324a8e83c8..eeb691d1911f 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
if (info->count < 1)
return -EINVAL;
+ if (!*info->id.name)
+ return -EINVAL;
+ if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name))
+ return -EINVAL;
access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
(info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
SNDRV_CTL_ELEM_ACCESS_INACTIVE|
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index f62780ed64ad..7821b07415a7 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -105,6 +105,8 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum,
int pitchbend = chan->midi_pitchbend;
int segment;
+ if (pitchbend < -0x2000)
+ pitchbend = -0x2000;
if (pitchbend > 0x1FFF)
pitchbend = 0x1FFF;
diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c
index 5f17b77ee152..f0e4d502d604 100644
--- a/sound/firewire/iso-resources.c
+++ b/sound/firewire/iso-resources.c
@@ -26,7 +26,7 @@
int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
{
r->channels_mask = ~0uLL;
- r->unit = fw_unit_get(unit);
+ r->unit = unit;
mutex_init(&r->mutex);
r->allocated = false;
@@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r)
{
WARN_ON(r->allocated);
mutex_destroy(&r->mutex);
- fw_unit_put(r->unit);
}
EXPORT_SYMBOL(fw_iso_resources_destroy);
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 29ccb3637164..e6757cd85724 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -171,9 +171,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
}
/* Wait first packet */
- err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT);
- if (err < 0)
+ if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) {
stop_stream(oxfw, stream);
+ err = -ETIMEDOUT;
+ }
end:
return err;
}
diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
index 17e49a071af4..b408540798c1 100644
--- a/sound/isa/msnd/msnd_pinnacle_mixer.c
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card)
spin_lock_init(&chip->mixer_lock);
strcpy(card->mixername, "MSND Pinnacle Mixer");
- for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++)
+ for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) {
err = snd_ctl_add(card,
snd_ctl_new1(snd_msnd_controls + idx, chip));
if (err < 0)
return err;
+ }
return 0;
}
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a2ce773bdc62..17c2637d842c 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
}
}
- if (!bus->no_response_fallback)
+ if (bus->no_response_fallback)
return -1;
if (!chip->polling_mode && chip->poll_count < 2) {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b680b4ec6331..fe18071bf93a 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)
{
unsigned int caps = query_amp_caps(codec, nid, dir);
int val = get_amp_val_to_activate(codec, nid, dir, caps, false);
- snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+
+ if (get_wcaps(codec, nid) & AC_WCAP_STEREO)
+ snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+ else
+ snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val);
+}
+
+/* update the amp, doing in stereo or mono depending on NID */
+static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx,
+ unsigned int mask, unsigned int val)
+{
+ if (get_wcaps(codec, nid) & AC_WCAP_STEREO)
+ return snd_hda_codec_amp_stereo(codec, nid, dir, idx,
+ mask, val);
+ else
+ return snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+ mask, val);
}
/* calculate amp value mask we can modify;
@@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir,
return;
val &= mask;
- snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val);
+ update_amp(codec, nid, dir, idx, mask, val);
}
static void activate_amp_out(struct hda_codec *codec, struct nid_path *path,
@@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
has_amp = nid_has_mute(codec, mix, HDA_INPUT);
for (i = 0; i < nums; i++) {
if (has_amp)
- snd_hda_codec_amp_stereo(codec, mix,
- HDA_INPUT, i,
- 0xff, HDA_AMP_MUTE);
+ update_amp(codec, mix, HDA_INPUT, i,
+ 0xff, HDA_AMP_MUTE);
else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
- snd_hda_codec_amp_stereo(codec, conn[i],
- HDA_OUTPUT, 0,
- 0xff, HDA_AMP_MUTE);
+ update_amp(codec, conn[i], HDA_OUTPUT, 0,
+ 0xff, HDA_AMP_MUTE);
}
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 1589c9bcce3e..dd2b3d92071f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
+ SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81),
SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
{} /* terminator */
@@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec)
return -ENOMEM;
spec->gen.automute_hook = cs_automute;
+ codec->single_adc_amp = 1;
snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
cs420x_fixups);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fd3ed18670e9..da67ea8645a6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -223,6 +223,7 @@ enum {
CXT_PINCFG_LENOVO_TP410,
CXT_PINCFG_LEMOTE_A1004,
CXT_PINCFG_LEMOTE_A1205,
+ CXT_PINCFG_COMPAQ_CQ60,
CXT_FIXUP_STEREO_DMIC,
CXT_FIXUP_INC_MIC_BOOST,
CXT_FIXUP_HEADPHONE_MIC_PIN,
@@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = {
.type = HDA_FIXUP_PINS,
.v.pins = cxt_pincfg_lemote,
},
+ [CXT_PINCFG_COMPAQ_CQ60] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ /* 0x17 was falsely set up as a mic, it should 0x1d */
+ { 0x17, 0x400001f0 },
+ { 0x1d, 0x97a70120 },
+ { }
+ }
+ },
[CXT_FIXUP_STEREO_DMIC] = {
.type = HDA_FIXUP_FUNC,
.v.func = cxt_fixup_stereo_dmic,
@@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = {
};
static const struct snd_pci_quirk cxt5051_fixups[] = {
+ SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b2b24a8b3dac..526398a4a442 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5209,6 +5209,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x17, 0x40000000},
{0x1d, 0x40700001},
{0x21, 0x02211040}),
+ SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC255_STANDARD_PINS,
+ {0x12, 0x90a60170},
+ {0x14, 0x90170140},
+ {0x17, 0x40000000},
+ {0x1d, 0x40700001},
+ {0x21, 0x02211050}),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
{0x13, 0x40000000},
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f5ad214663f9..8de836165cf2 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -46,8 +46,6 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <asm/mach-types.h>
-
#include "../codecs/wm8731.h"
#include "atmel-pcm.h"
#include "atmel_ssc_dai.h"
@@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
int ret;
if (!np) {
- if (!(machine_is_at91sam9g20ek() ||
- machine_is_at91sam9g20ek_2mmc()))
- return -ENODEV;
+ return -ENODEV;
}
ret = atmel_ssc_set_audio(0);
@@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
/* Parse device node info */
- if (np) {
- ret = snd_soc_of_parse_card_name(card, "atmel,model");
- if (ret)
- goto err;
-
- ret = snd_soc_of_parse_audio_routing(card,
- "atmel,audio-routing");
- if (ret)
- goto err;
-
- /* Parse codec info */
- at91sam9g20ek_dai.codec_name = NULL;
- codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
- if (!codec_np) {
- dev_err(&pdev->dev, "codec info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.codec_of_node = codec_np;
-
- /* Parse dai and platform info */
- at91sam9g20ek_dai.cpu_dai_name = NULL;
- at91sam9g20ek_dai.platform_name = NULL;
- cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
- if (!cpu_np) {
- dev_err(&pdev->dev, "dai and pcm info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.cpu_of_node = cpu_np;
- at91sam9g20ek_dai.platform_of_node = cpu_np;
-
- of_node_put(codec_np);
- of_node_put(cpu_np);
+ ret = snd_soc_of_parse_card_name(card, "atmel,model");
+ if (ret)
+ goto err;
+
+ ret = snd_soc_of_parse_audio_routing(card,
+ "atmel,audio-routing");
+ if (ret)
+ goto err;
+
+ /* Parse codec info */
+ at91sam9g20ek_dai.codec_name = NULL;
+ codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "codec info missing\n");
+ return -EINVAL;
+ }
+ at91sam9g20ek_dai.codec_of_node = codec_np;
+
+ /* Parse dai and platform info */
+ at91sam9g20ek_dai.cpu_dai_name = NULL;
+ at91sam9g20ek_dai.platform_name = NULL;
+ cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "dai and pcm info missing\n");
+ return -EINVAL;
}
+ at91sam9g20ek_dai.cpu_of_node = cpu_np;
+ at91sam9g20ek_dai.platform_of_node = cpu_np;
+
+ of_node_put(codec_np);
+ of_node_put(cpu_np);
ret = snd_soc_register_card(card);
if (ret) {
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 7b7fbcd49e5e..c7cd60f009e9 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97
config SND_EP93XX_SOC_SNAPPERCL15
tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
- depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
+ depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C
select SND_EP93XX_SOC_I2S
select SND_SOC_TLV320AIC23_I2C
help
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 064e6c18e109..ea9f0e31f9d4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98090 if I2C
select SND_SOC_MAX98095 if I2C
- select SND_SOC_MAX98357A
+ select SND_SOC_MAX98357A if GPIOLIB
select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 1806333ea29e..e9e6efbc21dd 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -12,9 +12,19 @@
* max98357a.c -- MAX98357A ALSA SoC Codec driver
*/
-#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/err.h>
#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/kernel.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
#define DRV_NAME "max98357a"
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index e1a4a45c57e2..fd102613d20d 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg)
case RT5670_ADC_EQ_CTRL1:
case RT5670_EQ_CTRL1:
case RT5670_ALC_CTRL_1:
- case RT5670_IRQ_CTRL1:
case RT5670_IRQ_CTRL2:
case RT5670_INT_IRQ_ST:
case RT5670_IL_CMD:
@@ -2703,6 +2702,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
regmap_write(rt5670->regmap, RT5670_RESET, 0);
+ regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val);
+ if (val >= 4)
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980);
+ else
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00);
+
ret = regmap_register_patch(rt5670->regmap, init_list,
ARRAY_SIZE(init_list));
if (ret != 0)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5d0bb8748dd1..fb9c20eace3f 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB45 Bypass Mux", "Bypass", "IB45 Mux" },
{ "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" },
- { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "IB6 Mux", "SLB DAC 6", "SLB DAC6" },
{ "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" },
{ "IB6 Mux", "IF4 DAC L", "IF4 DAC L" },
@@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" },
{ "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" },
- { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "IB7 Mux", "SLB DAC 7", "SLB DAC7" },
{ "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" },
{ "IB7 Mux", "IF4 DAC R", "IF4 DAC R" },
@@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC1 FS", NULL, "DAC1 MIXL" },
{ "DAC1 FS", NULL, "DAC1 MIXR" },
- { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" },
- { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" },
+ { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" },
+ { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" },
{ "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" },
{ "DAC2 L Mux", "OB 2", "OutBound2" },
- { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" },
- { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" },
+ { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" },
+ { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" },
{ "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" },
@@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC2 R Mux", "Haptic Generator", "Haptic Generator" },
{ "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" },
- { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" },
- { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" },
+ { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" },
+ { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" },
{ "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" },
{ "DAC3 L Mux", "OB 4", "OutBound4" },
- { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" },
- { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" },
+ { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" },
+ { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" },
{ "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" },
{ "DAC3 R Mux", "OB 5", "OutBound5" },
- { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" },
{ "DAC4 L Mux", "OB 6", "OutBound6" },
- { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" },
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 3a1343fa109b..007a0e3bc273 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = {
};
static const struct regmap_range sta32x_write_regs_range[] = {
- regmap_reg_range(STA32X_CONFA, STA32X_AUTO2),
- regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2),
+ regmap_reg_range(STA32X_CONFA, STA32X_FDRC2),
};
static const struct regmap_range sta32x_read_regs_range[] = {
- regmap_reg_range(STA32X_CONFA, STA32X_AUTO2),
- regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2),
+ regmap_reg_range(STA32X_CONFA, STA32X_FDRC2),
};
static const struct regmap_range sta32x_volatile_regs_range[] = {
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 75870c0ea2c9..91eb3aef7f02 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1049,7 +1049,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
enum spdif_txrate index, bool round)
{
const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
- bool is_sysclk = clk == spdif_priv->sysclk;
+ bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk);
u64 rate_ideal, rate_actual, sub;
u32 sysclk_dfmin, sysclk_dfmax;
u32 txclk_df, sysclk_df, arate;
@@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
spdif_priv->txclk_src[index], rate[index]);
dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",
spdif_priv->txclk_df[index], rate[index]);
- if (spdif_priv->txclk[index] == spdif_priv->sysclk)
+ if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk))
dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",
spdif_priv->sysclk_df[index], rate[index]);
dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n",
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2595611e8a6d..b9fabbf69db6 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
factor = (div2 + 1) * (7 * psr + 1) * 2;
for (i = 0; i < 255; i++) {
- /* The bclk rate must be smaller than 1/5 sysclk rate */
- if (factor * (i + 1) < 5)
- continue;
-
tmprate = freq * factor * (i + 2);
if (baudclk_is_used)
@@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+ continue;
+
clkrate /= factor;
afreq = clkrate / (i + 1);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index f7c6734bd5da..fb550b5869d2 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
strlen(dai_link->cpu_dai_name) +
strlen(dai_link->codec_dai_name) + 2,
GFP_KERNEL);
+ if (!name) {
+ ret = -ENOMEM;
+ goto dai_link_of_err;
+ }
+
sprintf(name, "%s-%s", dai_link->cpu_dai_name,
dai_link->codec_dai_name);
dai_link->name = dai_link->stream_name = name;
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index dfebfdd5eb2a..daecc58f28af 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -150,7 +150,7 @@ enum sst_cmd_type {
enum sst_task {
SST_TASK_SBA = 1,
- SST_TASK_MMX,
+ SST_TASK_MMX = 3,
};
enum sst_type {
diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 8a8d56a146e7..11c578651c1c 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx,
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
- shim_regs->imrx = sst_shim_read64(shim, SST_IMRX),
+ shim_regs->imrx = sst_shim_read64(shim, SST_IMRX);
+ shim_regs->csr = sst_shim_read64(shim, SST_CSR);
+
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
@@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx,
*/
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
sst_shim_write64(shim, SST_IMRX, shim_regs->imrx),
+ sst_shim_write64(shim, SST_CSR, shim_regs->csr),
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
@@ -379,6 +382,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx)
* initially active. So change the state to active before
* enabling the pm
*/
+
+ if (!acpi_disabled)
+ pm_runtime_set_active(ctx->dev);
+
pm_runtime_enable(ctx->dev);
if (acpi_disabled)
@@ -409,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev)
synchronize_irq(ctx->irq_num);
flush_workqueue(ctx->post_msg_wq);
+ ctx->ops->reset(ctx);
/* save the shim registers because PMC doesn't save state */
sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64);
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index def7d8260c4e..d19483081f9b 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -579,7 +579,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
return -EPROBE_DEFER;
} else {
- if (priv->extclk == priv->clk) {
+ if (clk_is_match(priv->extclk, priv->clk)) {
devm_clk_put(&pdev->dev, priv->extclk);
priv->extclk = ERR_PTR(-EINVAL);
} else {
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index ccfb41c22e53..f7eb42aa3f38 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
return ret;
card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
card->name = devm_kasprintf(dev, GFP_KERNEL,
"HDMI %s", dev_name(ad->dssdev));
card->owner = THIS_MODULE;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index c7eb9dd67f60..fd99d89de6a8 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case OMAP_MCBSP_SYSCLK_CLKX_EXT:
regs->srgr2 |= CLKSM;
+ regs->pcr0 |= SCLKME;
+ /*
+ * If McBSP is master but yet the CLKX/CLKR pin drives the SRG,
+ * disable output on those pins. This enables to inject the
+ * reference clock through CLKX/CLKR. For this to work
+ * set_dai_sysclk() _needs_ to be called after set_dai_fmt().
+ */
+ regs->pcr0 &= ~CLKXM;
+ break;
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
+ /* Disable ouput on CLKR pin in master mode */
+ regs->pcr0 &= ~CLKRM;
break;
default:
err = -ENODEV;
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index f4b05bc23e4b..1343ecbf0bd5 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64));
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
if (ret)
return ret;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 3cebf6ca03df..0632a36852c8 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM
config SND_SOC_SPEYSIDE
tristate "Audio support for Wolfson Speyside"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
@@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY
config SND_SOC_BELLS
tristate "Audio support for Wolfson Bells"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM5102
select SND_SOC_WM5110
@@ -206,7 +206,7 @@ config SND_SOC_LOWLAND
config SND_SOC_LITTLEMILL
tristate "Audio support for Wolfson Littlemill"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C
select SND_SAMSUNG_I2S
select MFD_WM8994
select SND_SOC_WM8994
@@ -223,7 +223,7 @@ config SND_SOC_SNOW
config SND_SOC_ODROIDX2
tristate "Audio support for Odroid-X2 and Odroid-U3"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SOC_MAX98090
select SND_SAMSUNG_I2S
help
@@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2
config SND_SOC_ARNDALE_RT5631_ALC5631
tristate "Audio support for RT5631(ALC5631) on Arndale Board"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SAMSUNG_I2S
select SND_SOC_RT5631
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 1b53605f7154..110577c52317 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1252,6 +1252,8 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_probe;
}
+ dev_set_drvdata(dev, priv);
+
/*
* asoc register
*/
@@ -1268,8 +1270,6 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_soc;
}
- dev_set_drvdata(dev, priv);
-
pm_runtime_enable(dev);
dev_info(dev, "probed\n");
diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 05dee690f487..97ed593f6010 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[],
for (; p < buf_end; ++p) {
short pv = le16_to_cpu(*p);
int val = (pv * volume[chn & 1]) >> 8;
- pv = clamp(val, 0x7fff, -0x8000);
+ pv = clamp(val, -0x8000, 0x7fff);
*p = cpu_to_le16(pv);
++chn;
}
@@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[],
val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16);
val = (val * volume[chn & 1]) >> 8;
- val = clamp(val, 0x7fffff, -0x800000);
+ val = clamp(val, -0x800000, 0x7fffff);
p[0] = val;
p[1] = val >> 8;
p[2] = val >> 16;
@@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal,
short pov = le16_to_cpu(*po);
short piv = le16_to_cpu(*pi);
int val = pov + ((piv * volume) >> 8);
- pov = clamp(val, 0x7fff, -0x8000);
+ pov = clamp(val, -0x8000, 0x7fff);
*po = cpu_to_le16(pov);
}
}
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 67d476548dcf..07f984d5f516 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
+{
+ USB_DEVICE(0x0582, 0x0159),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "Roland", */
+ /* .product_name = "UA-22", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
/* this catches most recent vendor-specific Roland devices */
{
.match_flags = USB_DEVICE_ID_MATCH_VENDOR |
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-20 17:41:29 +0000