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Message-ID: <s5hk3mx6997.wl%tiwai@suse.de>
Date: Fri, 17 May 2013 15:12:04 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nel.org>,
Liam Girdwood <lgirdwood@...il.com>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for v3.10-rc2
Linus,
please pull sound fixes for v3.10-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git sound-3.10
The topmost commit is e6135fe960bba1d1a3a6b6818839e6d8d3f21293
----------------------------------------------------------------
sound fixes for v3.10-rc2
A fairly calm update at this time, as seen in the short log, only one
fix per person: including,
- a few ASoC fixes (da7213 dmic, ux500 AD slot, wm0010 error path)
- a copule of HD-audio fixes
- a few other misc fixes (MIPS allmodconfig, proc output in usb,
old PowerBook support)
----------------------------------------------------------------
Axel Lin (1):
ASoC: da7213: Fix setting dmic_samplephase and dmic_clk_rate
David Henningsson (1):
ALSA: hda - Add headset mic support for another Dell machine
Fabio Baltieri (1):
ASoC: ux500: Swap even/odd AD slot definitions
Michael Ellerman (1):
ALSA: snd-aoa: Add a layout entry for PowerBook6,5
Paul Bolle (1):
sound: Fix make allmodconfig on MIPS correctly
Takashi Iwai (1):
ALSA: hda - Check the activity of the NID to be powered down
Torstein Hegge (1):
ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format
Wei Yongjun (1):
ASoC: wm0010: fix error return code in wm0010_boot()
---
sound/aoa/fabrics/layout.c | 8 ++++++++
sound/aoa/soundbus/i2sbus/core.c | 3 ++-
sound/oss/Kconfig | 2 +-
sound/pci/hda/hda_generic.c | 9 +++++++--
sound/pci/hda/patch_realtek.c | 1 +
sound/soc/codecs/ab8500-codec.h | 36 ++++++++++++++++++------------------
sound/soc/codecs/da7213.c | 8 ++++----
sound/soc/codecs/wm0010.c | 1 +
sound/usb/proc.c | 22 +++++++++++-----------
9 files changed, 53 insertions(+), 37 deletions(-)
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 552b97a..61ab640 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 0106583..15e7613 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 51c4ba9..1a96402 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 ac079f9..ae85bbd2 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 6bf47f7..59d2e91 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 114f69a..306d0bc 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/da7213.c b/sound/soc/codecs/da7213.c
index 41230ad..4a6f1da 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 8df2b6e..370af0c 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/usb/proc.c b/sound/usb/proc.c
index 135c768..5f761ab 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");
}
--
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