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Message-ID: <s5hk5bz42m6.wl%tiwai@suse.de>
Date: Thu, 23 Oct 2008 19:56:17 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Andrew Morton <akpm@...ux-foundation.org>, perex@...ex.cz,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] ALSA fixes
Hi Linus,
please pull ALSA updates for 2.6.28 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
This contains only the following small bug fixes.
Thanks!
Takashi
==
Harvey Harrison (1):
ALSA: hda - correct bracketing in spdif test in patch_sigmatel.c
Jarkko Nikula (2):
ALSA: ASoC: OMAP: Continue fixing DSP DAI format in McBSP DAI driver
ALSA: ASoC: tlv320aic3x: Fix DSP DAI format and signal polarities matching
Johannes Berg (1):
ALSA: aoa i2sbus: don't overwrite module parameter
Mark Brown (1):
ALSA: Ensure PXA runtime data is initialised
Takashi Iwai (1):
ALSA: hda - Fix conflicting volume controls on ALC260
sound/aoa/soundbus/i2sbus/i2sbus-core.c | 6 +++---
sound/arm/pxa2xx-pcm-lib.c | 2 +-
sound/pci/hda/patch_realtek.c | 22 ++++++++++++++--------
sound/pci/hda/patch_sigmatel.c | 2 +-
sound/soc/codecs/tlv320aic3x.c | 16 ++++++++++------
sound/soc/omap/omap-mcbsp.c | 7 ++-----
6 files changed, 31 insertions(+), 24 deletions(-)
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
index e6beb92..b4590df 100644
--- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
@@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
struct i2sbus_dev *dev;
struct device_node *child = NULL, *sound = NULL;
struct resource *r;
- int i, layout = 0, rlen;
+ int i, layout = 0, rlen, ok = force;
static const char *rnames[] = { "i2sbus: %s (control)",
"i2sbus: %s (tx)",
"i2sbus: %s (rx)" };
@@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
layout = *layout_id;
snprintf(dev->sound.modalias, 32,
"sound-layout-%d", layout);
- force = 1;
+ ok = 1;
}
}
/* for the time being, until we can handle non-layout-id
@@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
* When there are two i2s busses and only one has a layout-id,
* then this depends on the order, but that isn't important
* either as the second one in that case is just a modem. */
- if (!force) {
+ if (!ok) {
kfree(dev);
return -ENODEV;
}
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 1c93eb7..75a0d74 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
goto out;
ret = -ENOMEM;
- rtd = kmalloc(sizeof(*rtd), GFP_KERNEL);
+ rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
if (!rtd)
goto out;
rtd->dma_desc_array =
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e72707c..ef4955c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4996,7 +4996,7 @@ static struct hda_verb alc260_test_init_verbs[] = {
*/
static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
- const char *pfx)
+ const char *pfx, int *vol_bits)
{
hda_nid_t nid_vol;
unsigned long vol_val, sw_val;
@@ -5018,10 +5018,14 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
} else
return 0; /* N/A */
- snprintf(name, sizeof(name), "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
- if (err < 0)
- return err;
+ if (!(*vol_bits & (1 << nid_vol))) {
+ /* first control for the volume widget */
+ snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
+ if (err < 0)
+ return err;
+ *vol_bits |= (1 << nid_vol);
+ }
snprintf(name, sizeof(name), "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
if (err < 0)
@@ -5035,6 +5039,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
{
hda_nid_t nid;
int err;
+ int vols = 0;
spec->multiout.num_dacs = 1;
spec->multiout.dac_nids = spec->private_dac_nids;
@@ -5042,21 +5047,22 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
nid = cfg->line_out_pins[0];
if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Front");
+ err = alc260_add_playback_controls(spec, nid, "Front", &vols);
if (err < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Speaker");
+ err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
if (err < 0)
return err;
}
nid = cfg->hp_pins[0];
if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Headphone");
+ err = alc260_add_playback_controls(spec, nid, "Headphone",
+ &vols);
if (err < 0)
return err;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a2ac720..788fdc6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1282,7 +1282,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
return err;
spec->multiout.share_spdif = 1;
}
- if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) {
+ if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
if (err < 0)
return err;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 05336ed..cff276e 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- /* interface format */
- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_I2S:
+ /*
+ * match both interface format and signal polarities since they
+ * are fixed
+ */
+ switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+ SND_SOC_DAIFMT_INV_MASK)) {
+ case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
break;
- case SND_SOC_DAIFMT_DSP_A:
+ case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
iface_breg |= (0x01 << 6);
break;
- case SND_SOC_DAIFMT_RIGHT_J:
+ case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x02 << 6);
break;
- case SND_SOC_DAIFMT_LEFT_J:
+ case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x03 << 6);
break;
default:
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 853b33a..8485a8a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
break;
case SND_SOC_DAIFMT_DSP_A:
regs->srgr2 |= FPER(wlen * 2 - 1);
- regs->srgr1 |= FWID(0);
+ regs->srgr1 |= FWID(wlen * 2 - 2);
break;
}
@@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
{
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
- unsigned int temp_fmt = fmt;
if (mcbsp_data->configured)
return 0;
@@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
regs->xcr2 |= XDATDLY(0);
- /* Invert bit clock and FS polarity configuration for DSP_A */
- temp_fmt ^= SND_SOC_DAIFMT_IB_IF;
break;
default:
/* Unsupported data format */
@@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
}
/* Set bit clock (CLKX/CLKR) and FS polarities */
- switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
/*
* Normal BCLK + FS.
--
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