summaryrefslogtreecommitdiff
path: root/sound
diff options
context:
space:
mode:
Diffstat (limited to 'sound')
-rw-r--r--sound/core/info.c4
-rw-r--r--sound/core/pcm_misc.c4
-rw-r--r--sound/firewire/amdtp.c11
-rw-r--r--sound/firewire/amdtp.h1
-rw-r--r--sound/firewire/dice.c29
-rw-r--r--sound/pci/ctxfi/ct20k1reg.h4
-rw-r--r--sound/pci/hda/ca0132_regs.h2
-rw-r--r--sound/pci/hda/patch_conexant.c9
-rw-r--r--sound/pci/hda/patch_realtek.c20
-rw-r--r--sound/soc/codecs/cs4265.c12
-rw-r--r--sound/soc/codecs/da732x.h2
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/codecs/rt5677.c8
-rw-r--r--sound/soc/generic/simple-card.c8
-rw-r--r--sound/soc/omap/omap-twl4030.c2
-rw-r--r--sound/soc/sh/rcar/gen.c2
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h2
18 files changed, 89 insertions, 34 deletions
diff --git a/sound/core/info.c b/sound/core/info.c
index 051d55b05521..9f404e965ea2 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -684,7 +684,7 @@ int snd_info_card_free(struct snd_card *card)
* snd_info_get_line - read one line from the procfs buffer
* @buffer: the procfs buffer
* @line: the buffer to store
- * @len: the max. buffer size - 1
+ * @len: the max. buffer size
*
* Reads one line from the buffer and stores the string.
*
@@ -704,7 +704,7 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
buffer->stop = 1;
if (c == '\n')
break;
- if (len) {
+ if (len > 1) {
len--;
*line++ = c;
}
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 4560ca0e5651..2c6fd80e0bd1 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -142,11 +142,11 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
},
[SNDRV_PCM_FORMAT_DSD_U8] = {
.width = 8, .phys = 8, .le = 1, .signd = 0,
- .silence = {},
+ .silence = { 0x69 },
},
[SNDRV_PCM_FORMAT_DSD_U16_LE] = {
.width = 16, .phys = 16, .le = 1, .signd = 0,
- .silence = {},
+ .silence = { 0x69, 0x69 },
},
/* FIXME: the following three formats are not defined properly yet */
[SNDRV_PCM_FORMAT_MPEG] = {
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index f96bf4c7c232..95fc2eaf11dc 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -507,7 +507,16 @@ static void amdtp_pull_midi(struct amdtp_stream *s,
static void update_pcm_pointers(struct amdtp_stream *s,
struct snd_pcm_substream *pcm,
unsigned int frames)
-{ unsigned int ptr;
+{
+ unsigned int ptr;
+
+ /*
+ * In IEC 61883-6, one data block represents one event. In ALSA, one
+ * event equals to one PCM frame. But Dice has a quirk to transfer
+ * two PCM frames in one data block.
+ */
+ if (s->double_pcm_frames)
+ frames *= 2;
ptr = s->pcm_buffer_pointer + frames;
if (ptr >= pcm->runtime->buffer_size)
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index d8ee7b0e9386..4823c08196ac 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -125,6 +125,7 @@ struct amdtp_stream {
unsigned int pcm_buffer_pointer;
unsigned int pcm_period_pointer;
bool pointer_flush;
+ bool double_pcm_frames;
struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c
index a9a30c0161f1..e3a04d69c853 100644
--- a/sound/firewire/dice.c
+++ b/sound/firewire/dice.c
@@ -567,10 +567,14 @@ static int dice_hw_params(struct snd_pcm_substream *substream,
return err;
/*
- * At rates above 96 kHz, pretend that the stream runs at half the
- * actual sample rate with twice the number of channels; two samples
- * of a channel are stored consecutively in the packet. Requires
- * blocking mode and PCM buffer size should be aligned to SYT_INTERVAL.
+ * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in
+ * one data block of AMDTP packet. Thus sampling transfer frequency is
+ * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are
+ * transferred on AMDTP packets at 96 kHz. Two successive samples of a
+ * channel are stored consecutively in the packet. This quirk is called
+ * as 'Dual Wire'.
+ * For this quirk, blocking mode is required and PCM buffer size should
+ * be aligned to SYT_INTERVAL.
*/
channels = params_channels(hw_params);
if (rate_index > 4) {
@@ -579,18 +583,25 @@ static int dice_hw_params(struct snd_pcm_substream *substream,
return err;
}
- for (i = 0; i < channels; i++) {
- dice->stream.pcm_positions[i * 2] = i;
- dice->stream.pcm_positions[i * 2 + 1] = i + channels;
- }
-
rate /= 2;
channels *= 2;
+ dice->stream.double_pcm_frames = true;
+ } else {
+ dice->stream.double_pcm_frames = false;
}
mode = rate_index_to_mode(rate_index);
amdtp_stream_set_parameters(&dice->stream, rate, channels,
dice->rx_midi_ports[mode]);
+ if (rate_index > 4) {
+ channels /= 2;
+
+ for (i = 0; i < channels; i++) {
+ dice->stream.pcm_positions[i] = i * 2;
+ dice->stream.pcm_positions[i + channels] = i * 2 + 1;
+ }
+ }
+
amdtp_stream_set_pcm_format(&dice->stream,
params_format(hw_params));
diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h
index f2e34e3f27ee..5851249f11d9 100644
--- a/sound/pci/ctxfi/ct20k1reg.h
+++ b/sound/pci/ctxfi/ct20k1reg.h
@@ -7,7 +7,7 @@
*/
#ifndef CT20K1REG_H
-#define CT20k1REG_H
+#define CT20K1REG_H
/* 20k1 registers */
#define DSPXRAM_START 0x000000
@@ -632,5 +632,3 @@
#define I2SD_R 0x19L
#endif /* CT20K1REG_H */
-
-
diff --git a/sound/pci/hda/ca0132_regs.h b/sound/pci/hda/ca0132_regs.h
index 07e760937d3c..8371274aa811 100644
--- a/sound/pci/hda/ca0132_regs.h
+++ b/sound/pci/hda/ca0132_regs.h
@@ -20,7 +20,7 @@
*/
#ifndef __CA0132_REGS_H
-#define __CA0312_REGS_H
+#define __CA0132_REGS_H
#define DSP_CHIP_OFFSET 0x100000
#define DSP_DBGCNTL_MODULE_OFFSET 0xE30
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6f2fa838b635..6e5d0cb4e3d7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -217,6 +217,7 @@ enum {
CXT_FIXUP_HEADPHONE_MIC_PIN,
CXT_FIXUP_HEADPHONE_MIC,
CXT_FIXUP_GPIO1,
+ CXT_FIXUP_ASPIRE_DMIC,
CXT_FIXUP_THINKPAD_ACPI,
CXT_FIXUP_OLPC_XO,
CXT_FIXUP_CAP_MIX_AMP,
@@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = {
{ }
},
},
+ [CXT_FIXUP_ASPIRE_DMIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_stereo_dmic,
+ .chained = true,
+ .chain_id = CXT_FIXUP_GPIO1,
+ },
[CXT_FIXUP_THINKPAD_ACPI] = {
.type = HDA_FIXUP_FUNC,
.v.func = hda_fixup_thinkpad_acpi,
@@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = {
static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC),
SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d71270a3f73f..1ba22fb527c2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -328,6 +328,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case 0x10ec0885:
case 0x10ec0887:
/*case 0x10ec0889:*/ /* this causes an SPDIF problem */
+ case 0x10ec0900:
alc889_coef_init(codec);
break;
case 0x10ec0888:
@@ -2350,6 +2351,7 @@ static int patch_alc882(struct hda_codec *codec)
switch (codec->vendor_id) {
case 0x10ec0882:
case 0x10ec0885:
+ case 0x10ec0900:
break;
default:
/* ALC883 and variants */
@@ -4408,6 +4410,7 @@ enum {
ALC292_FIXUP_TPT440_DOCK,
ALC283_FIXUP_BXBT2807_MIC,
ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
+ ALC282_FIXUP_ASPIRE_V5_PINS,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -4855,6 +4858,22 @@ static const struct hda_fixup alc269_fixups[] = {
.chained_before = true,
.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
},
+ [ALC282_FIXUP_ASPIRE_V5_PINS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x90a60130 },
+ { 0x14, 0x90170110 },
+ { 0x17, 0x40000008 },
+ { 0x18, 0x411111f0 },
+ { 0x19, 0x411111f0 },
+ { 0x1a, 0x411111f0 },
+ { 0x1b, 0x411111f0 },
+ { 0x1d, 0x40f89b2d },
+ { 0x1e, 0x411111f0 },
+ { 0x21, 0x0321101f },
+ { },
+ },
+ },
};
@@ -4866,6 +4885,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
+ SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index a20b30ca52c0..98523209f739 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = {
/*64k*/
{8192000, 64000, 1, 0},
- {1228800, 64000, 1, 1},
- {1693440, 64000, 1, 2},
- {2457600, 64000, 1, 3},
- {3276800, 64000, 1, 4},
+ {12288000, 64000, 1, 1},
+ {16934400, 64000, 1, 2},
+ {24576000, 64000, 1, 3},
+ {32768000, 64000, 1, 4},
/* 88.2k */
{11289600, 88200, 1, 0},
@@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
if (index >= 0) {
snd_soc_update_bits(codec, CS4265_ADC_CTL,
- CS4265_ADC_FM, clk_map_table[index].fm_mode);
+ CS4265_ADC_FM, clk_map_table[index].fm_mode << 6);
snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
CS4265_MCLK_FREQ_MASK,
- clk_map_table[index].mclkdiv);
+ clk_map_table[index].mclkdiv << 4);
} else {
dev_err(codec->dev, "can't get correct mclk\n");
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
index 1dceafeec415..f586cbd30b77 100644
--- a/sound/soc/codecs/da732x.h
+++ b/sound/soc/codecs/da732x.h
@@ -11,7 +11,7 @@
*/
#ifndef __DA732X_H_
-#define __DA732X_H
+#define __DA732X_H_
#include <sound/soc.h>
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 6bc6efdec550..f1ec6e6bd08a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
static const struct regmap_config rt5640_regmap = {
.reg_bits = 8,
.val_bits = 16,
+ .use_single_rw = true,
.max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) *
RT5640_PR_SPACING),
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 67f14556462f..5337c448b5e3 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "BST2", NULL, "IN2P" },
{ "BST2", NULL, "IN2N" },
- { "IN1P", NULL, "micbias1" },
- { "IN1N", NULL, "micbias1" },
- { "IN2P", NULL, "micbias1" },
- { "IN2N", NULL, "micbias1" },
+ { "IN1P", NULL, "MICBIAS1" },
+ { "IN1N", NULL, "MICBIAS1" },
+ { "IN2P", NULL, "MICBIAS1" },
+ { "IN2N", NULL, "MICBIAS1" },
{ "ADC 1", NULL, "BST1" },
{ "ADC 1", NULL, "ADC 1 power" },
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 159e517fa09a..cef7776b712c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(&priv->snd_card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
+ if (ret >= 0)
+ return ret;
err:
asoc_simple_card_unref(pdev);
return ret;
}
+static int asoc_simple_card_remove(struct platform_device *pdev)
+{
+ return asoc_simple_card_unref(pdev);
+}
+
static const struct of_device_id asoc_simple_of_match[] = {
{ .compatible = "simple-audio-card", },
{},
@@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = {
.of_match_table = asoc_simple_of_match,
},
.probe = asoc_simple_card_probe,
+ .remove = asoc_simple_card_remove,
};
module_platform_driver(asoc_simple_card);
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index f8a6adc2d81c..4336d1831485 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
.stream_name = "TWL4030 Voice",
.cpu_dai_name = "omap-mcbsp.3",
.codec_dai_name = "twl4030-voice",
- .platform_name = "omap-mcbsp.2",
+ .platform_name = "omap-mcbsp.3",
.codec_name = "twl4030-codec",
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 3fdf3be7b99a..f95e7ab135e8 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv,
};
/* it shouldn't happen */
- if (use_dvc & !use_src)
+ if (use_dvc && !use_src)
dev_err(dev, "DVC is selected without SRC\n");
/* use SSIU or SSI ? */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d4bfd4a9076f..889f4e3d35dc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1325,7 +1325,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
device_initialize(rtd->dev);
rtd->dev->parent = rtd->card->dev;
rtd->dev->release = rtd_release;
- rtd->dev->init_name = name;
+ dev_set_name(rtd->dev, "%s", name);
dev_set_drvdata(rtd->dev, rtd);
mutex_init(&rtd->pcm_mutex);
INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 9577121ce971..ca8037634100 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -21,7 +21,7 @@
*/
#ifndef __TEGRA_ASOC_UTILS_H__
-#define __TEGRA_ASOC_UTILS_H_
+#define __TEGRA_ASOC_UTILS_H__
struct clk;
struct device;
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-26 11:25:38 +0300