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-rw-r--r--sound/aoa/soundbus/i2sbus/i2sbus-core.c6
-rw-r--r--sound/arm/pxa2xx-pcm-lib.c2
-rw-r--r--sound/oss/kahlua.c2
-rw-r--r--sound/pci/cs5530.c2
-rw-r--r--sound/pci/hda/patch_realtek.c99
-rw-r--r--sound/pci/hda/patch_sigmatel.c2
-rw-r--r--sound/soc/blackfin/bf5xx-i2s.c34
-rw-r--r--sound/soc/codecs/tlv320aic3x.c16
-rw-r--r--sound/soc/omap/omap-mcbsp.c7
-rw-r--r--sound/sound_core.c2
10 files changed, 133 insertions, 39 deletions
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
index e6beb92c693..b4590df0746 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 1c93eb77cb9..75a0d746fb6 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/oss/kahlua.c b/sound/oss/kahlua.c
index eb9bc365530..c180598f171 100644
--- a/sound/oss/kahlua.c
+++ b/sound/oss/kahlua.c
@@ -1,7 +1,7 @@
/*
* Initialisation code for Cyrix/NatSemi VSA1 softaudio
*
- * (C) Copyright 2003 Red Hat Inc <alan@redhat.com>
+ * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
*
* XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
* The older version (VSA1) provides fairly good soundblaster emulation
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index 4d9378d8120..6dea5b5cc77 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -2,7 +2,7 @@
* cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio
*
* (C) Copyright 2007 Ash Willis <ashwillis@programmer.net>
- * (C) Copyright 2003 Red Hat Inc <alan@redhat.com>
+ * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
*
* This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did
* mess with it a bit. The chip seems to have to have trouble with full duplex
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e72707cb60a..4eceab9bd10 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -307,6 +307,13 @@ struct alc_spec {
/* for PLL fix */
hda_nid_t pll_nid;
unsigned int pll_coef_idx, pll_coef_bit;
+
+#ifdef SND_HDA_NEEDS_RESUME
+#define ALC_MAX_PINS 16
+ unsigned int num_pins;
+ hda_nid_t pin_nids[ALC_MAX_PINS];
+ unsigned int pin_cfgs[ALC_MAX_PINS];
+#endif
};
/*
@@ -2778,6 +2785,64 @@ static void alc_free(struct hda_codec *codec)
codec->spec = NULL; /* to be sure */
}
+#ifdef SND_HDA_NEEDS_RESUME
+static void store_pin_configs(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid, end_nid;
+
+ end_nid = codec->start_nid + codec->num_nodes;
+ for (nid = codec->start_nid; nid < end_nid; nid++) {
+ unsigned int wid_caps = get_wcaps(codec, nid);
+ unsigned int wid_type =
+ (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wid_type != AC_WID_PIN)
+ continue;
+ if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids))
+ break;
+ spec->pin_nids[spec->num_pins] = nid;
+ spec->pin_cfgs[spec->num_pins] =
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ spec->num_pins++;
+ }
+}
+
+static void resume_pin_configs(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_pins; i++) {
+ hda_nid_t pin_nid = spec->pin_nids[i];
+ unsigned int pin_config = spec->pin_cfgs[i];
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+ pin_config & 0x000000ff);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+ (pin_config & 0x0000ff00) >> 8);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+ (pin_config & 0x00ff0000) >> 16);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+ pin_config >> 24);
+ }
+}
+
+static int alc_resume(struct hda_codec *codec)
+{
+ resume_pin_configs(codec);
+ codec->patch_ops.init(codec);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ return 0;
+}
+#else
+#define store_pin_configs(codec)
+#endif
+
/*
*/
static struct hda_codec_ops alc_patch_ops = {
@@ -2786,6 +2851,9 @@ static struct hda_codec_ops alc_patch_ops = {
.init = alc_init,
.free = alc_free,
.unsol_event = alc_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+ .resume = alc_resume,
+#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
.check_power_status = alc_check_power_status,
#endif
@@ -3832,6 +3900,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
+ store_pin_configs(codec);
return 1;
}
@@ -4996,7 +5065,7 @@ 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 +5087,14 @@ 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 +5108,7 @@ 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 +5116,22 @@ 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;
}
@@ -5244,6 +5319,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
}
spec->num_mixers++;
+ store_pin_configs(codec);
return 1;
}
@@ -10307,6 +10383,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ store_pin_configs(codec);
return 1;
}
@@ -11441,6 +11518,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ store_pin_configs(codec);
return 1;
}
@@ -12224,6 +12302,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc269_capture_mixer;
spec->num_mixers++;
+ store_pin_configs(codec);
return 1;
}
@@ -13310,6 +13389,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc861_capture_mixer;
spec->num_mixers++;
+ store_pin_configs(codec);
return 1;
}
@@ -14421,6 +14501,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ store_pin_configs(codec);
return 1;
}
@@ -16252,6 +16333,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc662_capture_mixer;
spec->num_mixers++;
+
+ store_pin_configs(codec);
return 1;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a2ac7205d45..788fdc6f326 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/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 827587f0818..e020c160ee4 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = {
}
};
-static u16 sport_req[][7] = {
- { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
- P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0},
- { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS,
- P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0},
-};
+/*
+ * Setting the TFS pin selector for SPORT 0 based on whether the selected
+ * port id F or G. If the port is F then no conflict should exist for the
+ * TFS. When Port G is selected and EMAC then there is a conflict between
+ * the PHY interrupt line and TFS. Current settings prevent the conflict
+ * by ignoring the TFS pin when Port G is selected. This allows both
+ * ssm2602 using Port G and EMAC concurrently.
+ */
+#ifdef CONFIG_BF527_SPORT0_PORTF
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
+#else
+#define LOCAL_SPORT0_TFS (0)
+#endif
+
+static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+ P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0},
+ {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI,
+ P_SPORT1_RSCLK, P_SPORT1_TFS, 0} };
static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
@@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
ret = -EINVAL;
break;
default:
+ printk(KERN_ERR "%s: Unknown DAI format type\n", __func__);
ret = -EINVAL;
break;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFS:
- ret = -EINVAL;
- break;
- case SND_SOC_DAIFMT_CBM_CFS:
- ret = -EINVAL;
- break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBM_CFS:
case SND_SOC_DAIFMT_CBS_CFM:
ret = -EINVAL;
break;
default:
+ printk(KERN_ERR "%s: Unknown DAI master type\n", __func__);
ret = -EINVAL;
break;
}
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 05336ed7e49..cff276ee261 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 853b33ae343..8485a8a9d0f 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.
diff --git a/sound/sound_core.c b/sound/sound_core.c
index faef87a9bc3..a75b289a5d7 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -57,7 +57,7 @@ module_exit(cleanup_soundcore);
/*
* OSS sound core handling. Breaks out sound functions to submodules
*
- * Author: Alan Cox <alan.cox@linux.org>
+ * Author: Alan Cox <alan@lxorguk.ukuu.org.uk>
*
* Fixes:
*
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-30 05:31:44 +0000