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-rw-r--r--sound/aoa/fabrics/layout.c8
-rw-r--r--sound/aoa/soundbus/i2sbus/core.c3
-rw-r--r--sound/oss/Kconfig2
-rw-r--r--sound/pci/hda/hda_generic.c9
-rw-r--r--sound/pci/hda/patch_realtek.c1
-rw-r--r--sound/soc/codecs/ab8500-codec.h36
-rw-r--r--sound/soc/codecs/cs42l52.c8
-rw-r--r--sound/soc/codecs/cs42l52.h2
-rw-r--r--sound/soc/codecs/da7213.c8
-rw-r--r--sound/soc/codecs/wm0010.c1
-rw-r--r--sound/soc/codecs/wm5110.c4
-rw-r--r--sound/soc/davinci/davinci-mcasp.c7
-rw-r--r--sound/soc/fsl/imx-ssi.c6
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c5
-rw-r--r--sound/soc/soc-compress.c8
-rw-r--r--sound/usb/proc.c22
16 files changed, 71 insertions, 59 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 552b97afbca..61ab640e195 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -113,6 +113,7 @@ MODULE_ALIAS("sound-layout-100");
MODULE_ALIAS("aoa-device-id-14");
MODULE_ALIAS("aoa-device-id-22");
MODULE_ALIAS("aoa-device-id-35");
+MODULE_ALIAS("aoa-device-id-44");
/* onyx with all but microphone connected */
static struct codec_connection onyx_connections_nomic[] = {
@@ -361,6 +362,13 @@ static struct layout layouts[] = {
.connections = tas_connections_nolineout,
},
},
+ /* PowerBook6,5 */
+ { .device_id = 44,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_all,
+ },
+ },
/* PowerBook6,7 */
{ .layout_id = 80,
.codecs[0] = {
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 01065833588..15e76131b50 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -200,7 +200,8 @@ static int i2sbus_add_dev(struct macio_dev *macio,
* We probably cannot handle all device-id machines,
* so restrict to those we do handle for now.
*/
- if (id && (*id == 22 || *id == 14 || *id == 35)) {
+ if (id && (*id == 22 || *id == 14 || *id == 35 ||
+ *id == 44)) {
snprintf(dev->sound.modalias, 32,
"aoa-device-id-%d", *id);
ok = 1;
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 51c4ba95a32..1a964025443 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -250,7 +250,7 @@ config MSND_FIFOSIZE
menuconfig SOUND_OSS
tristate "OSS sound modules"
depends on ISA_DMA_API && VIRT_TO_BUS
- depends on !ISA_DMA_SUPPORT_BROKEN
+ depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN
help
OSS is the Open Sound System suite of sound card drivers. They make
sound programming easier since they provide a common API. Say Y or
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ac079f93c53..ae85bbd2e6f 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -606,6 +606,10 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid,
return false;
}
+/* check whether the NID is referred by any active paths */
+#define is_active_nid_for_any(codec, nid) \
+ is_active_nid(codec, nid, HDA_OUTPUT, 0)
+
/* get the default amp value for the target state */
static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
int dir, unsigned int caps, bool enable)
@@ -759,7 +763,8 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
for (i = 0; i < path->depth; i++) {
hda_nid_t nid = path->path[i];
- if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3)) {
+ if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) &&
+ !is_active_nid_for_any(codec, nid)) {
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_POWER_STATE,
AC_PWRST_D3);
@@ -4157,7 +4162,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
return power_state;
if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER)
return power_state;
- if (is_active_nid(codec, nid, HDA_OUTPUT, 0))
+ if (is_active_nid_for_any(codec, nid))
return power_state;
return AC_PWRST_D3;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6bf47f7326a..59d2e91a9ab 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3482,6 +3482,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h
index 114f69a0c62..306d0bc8455 100644
--- a/sound/soc/codecs/ab8500-codec.h
+++ b/sound/soc/codecs/ab8500-codec.h
@@ -348,25 +348,25 @@
/* AB8500_ADSLOTSELX */
#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00
-#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01
-#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02
-#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03
-#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04
-#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05
-#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06
-#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07
-#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08
-#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x10
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x20
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x30
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x40
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x50
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x60
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x70
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x80
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0xF0
#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00
-#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10
-#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20
-#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30
-#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40
-#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50
-#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60
-#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70
-#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80
-#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x01
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x02
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x03
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x04
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x05
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x06
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x07
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x08
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0x0F
#define AB8500_ADSLOTSELX_EVEN_SHIFT 0
#define AB8500_ADSLOTSELX_ODD_SHIFT 4
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec0..030f53c96ec 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
{ CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */
{ CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */
{ CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */
- { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */
+ { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */
{ CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */
{ CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */
{ CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */
@@ -225,7 +225,7 @@ static const char * const mic_bias_level_text[] = {
};
static const struct soc_enum mic_bias_level_enum =
- SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+ SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -413,7 +413,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
- CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+ CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +441,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
- 6, 0x7f, 0x19, hl_tlv),
+ 0, 0x7f, 0x19, hl_tlv),
SOC_DOUBLE_R("PCM Mixer Switch",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c05907..4277012c471 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
#define CS42L52_PB_CTL1_INV_PCMA (1 << 2)
#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1)
#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK 0x03
#define CS42L52_PB_CTL1_MUTE 3
#define CS42L52_PB_CTL1_UNMUTE 0
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 41230ad1c3e..4a6f1daf911 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1488,17 +1488,17 @@ static int da7213_probe(struct snd_soc_codec *codec)
DA7213_DMIC_DATA_SEL_SHIFT);
break;
}
- switch (pdata->dmic_data_sel) {
+ switch (pdata->dmic_samplephase) {
case DA7213_DMIC_SAMPLE_ON_CLKEDGE:
case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE:
- dmic_cfg |= (pdata->dmic_data_sel <<
+ dmic_cfg |= (pdata->dmic_samplephase <<
DA7213_DMIC_SAMPLEPHASE_SHIFT);
break;
}
- switch (pdata->dmic_data_sel) {
+ switch (pdata->dmic_clk_rate) {
case DA7213_DMIC_CLK_3_0MHZ:
case DA7213_DMIC_CLK_1_5MHZ:
- dmic_cfg |= (pdata->dmic_data_sel <<
+ dmic_cfg |= (pdata->dmic_clk_rate <<
DA7213_DMIC_CLK_RATE_SHIFT);
break;
}
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 8df2b6e1a1a..370af0cbcc9 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -667,6 +667,7 @@ static int wm0010_boot(struct snd_soc_codec *codec)
/* On wm0010 only the CLKCTRL1 value is used */
pll_rec.clkctrl1 = wm0010->pll_clkctrl1;
+ ret = -ENOMEM;
len = pll_rec.length + 8;
out = kzalloc(len, GFP_KERNEL);
if (!out) {
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e0477..ba38f067966 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -976,6 +976,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
if (ret != 0)
return ret;
+ arizona_init_spk(codec);
+
snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
priv->core.arizona->dapm = &codec->dapm;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e..81490febac6 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
int word_length)
{
u32 fmt;
- u32 rotate = (word_length / 4) & 0x7;
+ u32 tx_rotate = (word_length / 4) & 0x7;
+ u32 rx_rotate = (32 - word_length) / 4;
u32 mask = (1ULL << word_length) - 1;
/*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
TXSSZ(fmt), TXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
- TXROT(rotate), TXROT(7));
+ TXROT(tx_rotate), TXROT(7));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
- RXROT(rotate), RXROT(7));
+ RXROT(rx_rotate), RXROT(7));
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
mask);
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 902fab02b85..c6fa03e2114 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -540,11 +540,6 @@ static int imx_ssi_probe(struct platform_device *pdev)
clk_prepare_enable(ssi->clk);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- ret = -ENODEV;
- goto failed_get_resource;
- }
-
ssi->base = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(ssi->base)) {
ret = PTR_ERR(ssi->base);
@@ -633,7 +628,6 @@ failed_pdev_fiq_alloc:
snd_soc_unregister_component(&pdev->dev);
failed_register:
release_mem_region(res->start, resource_size(res));
-failed_get_resource:
clk_disable_unprepare(ssi->clk);
failed_clk:
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index befe68f5928..4c9dad3263c 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -471,11 +471,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
dev_set_drvdata(&pdev->dev, priv);
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!mem) {
- dev_err(&pdev->dev, "platform_get_resource failed\n");
- return -ENXIO;
- }
-
priv->io = devm_ioremap_resource(&pdev->dev, mem);
if (IS_ERR(priv->io))
return PTR_ERR(priv->io);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f2..06a8000aa07 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
goto err;
}
- snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
- SND_SOC_DAPM_STREAM_START);
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+ SND_SOC_DAPM_STREAM_START);
/* cancel any delayed stream shutdown that is pending */
rtd->pop_wait = 0;
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index 135c7687106..5f761ab34c0 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -116,21 +116,22 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
}
static void proc_dump_ep_status(struct snd_usb_substream *subs,
- struct snd_usb_endpoint *ep,
+ struct snd_usb_endpoint *data_ep,
+ struct snd_usb_endpoint *sync_ep,
struct snd_info_buffer *buffer)
{
- if (!ep)
+ if (!data_ep)
return;
- snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize);
+ snd_iprintf(buffer, " Packet Size = %d\n", data_ep->curpacksize);
snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n",
subs->speed == USB_SPEED_FULL
- ? get_full_speed_hz(ep->freqm)
- : get_high_speed_hz(ep->freqm),
- ep->freqm >> 16, ep->freqm & 0xffff);
- if (ep->freqshift != INT_MIN) {
- int res = 16 - ep->freqshift;
+ ? get_full_speed_hz(data_ep->freqm)
+ : get_high_speed_hz(data_ep->freqm),
+ data_ep->freqm >> 16, data_ep->freqm & 0xffff);
+ if (sync_ep && data_ep->freqshift != INT_MIN) {
+ int res = 16 - data_ep->freqshift;
snd_iprintf(buffer, " Feedback Format = %d.%d\n",
- (ep->syncmaxsize > 3 ? 32 : 24) - res, res);
+ (sync_ep->syncmaxsize > 3 ? 32 : 24) - res, res);
}
}
@@ -140,8 +141,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn
snd_iprintf(buffer, " Status: Running\n");
snd_iprintf(buffer, " Interface = %d\n", subs->interface);
snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx);
- proc_dump_ep_status(subs, subs->data_endpoint, buffer);
- proc_dump_ep_status(subs, subs->sync_endpoint, buffer);
+ proc_dump_ep_status(subs, subs->data_endpoint, subs->sync_endpoint, buffer);
} else {
snd_iprintf(buffer, " Status: Stop\n");
}
generated by cgit v1.2.3-18-g5258 (git 2.32.0) at 2024-11-26 06:50:14 +0000