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-rw-r--r--sound/aoa/fabrics/layout.c2
-rw-r--r--sound/core/pcm_lib.c33
-rw-r--r--sound/pci/ac97/ac97_patch.c1
-rw-r--r--sound/pci/azt3328.c11
-rw-r--r--sound/pci/hda/alc268_quirks.c36
-rw-r--r--sound/pci/hda/hda_codec.c6
-rw-r--r--sound/pci/hda/hda_eld.c31
-rw-r--r--sound/pci/hda/patch_cirrus.c10
-rw-r--r--sound/pci/hda/patch_conexant.c57
-rw-r--r--sound/pci/hda/patch_realtek.c39
-rw-r--r--sound/pci/hda/patch_sigmatel.c3
-rw-r--r--sound/soc/blackfin/bf5xx-ad193x.c6
-rw-r--r--sound/soc/codecs/ad193x.c11
-rw-r--r--sound/soc/codecs/ad193x.h5
-rw-r--r--sound/soc/codecs/sta32x.c1
-rw-r--r--sound/soc/codecs/wm8750.c8
-rw-r--r--sound/soc/codecs/wm8903.c5
-rw-r--r--sound/soc/codecs/wm8962.c12
-rw-r--r--sound/soc/codecs/wm8994.c1
-rw-r--r--sound/soc/codecs/wm8996.c28
-rw-r--r--sound/soc/ep93xx/ep93xx-i2s.c5
-rw-r--r--sound/soc/fsl/fsl_dma.c2
-rw-r--r--sound/soc/fsl/mpc5200_dma.c6
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c18
-rw-r--r--sound/soc/fsl/p1022_ds.c4
-rw-r--r--sound/soc/imx/imx-pcm-fiq.c1
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c2
-rw-r--r--sound/soc/omap/ams-delta.c6
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/omap/omap-mcbsp.c4
-rw-r--r--sound/soc/omap/omap-mcbsp.h2
-rw-r--r--sound/soc/omap/omap-pcm.c4
-rw-r--r--sound/soc/omap/omap-pcm.h2
-rw-r--r--sound/soc/omap/rx51.c2
-rw-r--r--sound/soc/samsung/Kconfig1
-rw-r--r--sound/soc/samsung/Makefile2
-rw-r--r--sound/soc/samsung/h1940_uda1380.c1
-rw-r--r--sound/soc/samsung/idma.c453
-rw-r--r--sound/soc/samsung/idma.h26
-rw-r--r--sound/soc/samsung/jive_wm8750.c2
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c1
-rw-r--r--sound/soc/samsung/speyside_wm8962.c8
-rw-r--r--sound/soc/soc-cache.c12
-rw-r--r--sound/soc/soc-core.c6
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/soc/soc-io.c23
-rw-r--r--sound/soc/soc-jack.c4
-rw-r--r--sound/soc/soc-pcm.c3
-rw-r--r--sound/soc/tegra/tegra_pcm.c9
-rw-r--r--sound/soc/tegra/tegra_wm8903.c19
-rw-r--r--sound/usb/caiaq/audio.c37
-rw-r--r--sound/usb/caiaq/device.h1
-rw-r--r--sound/usb/mixer.c3
-rw-r--r--sound/usb/quirks-table.h34
54 files changed, 845 insertions, 170 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 3fd1a7e2492..552b97afbca 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
sdev->pcmid = -1;
list_del(&ldev->list);
layouts_list_items--;
+ kfree(ldev);
outnodev:
of_node_put(sound);
layout_device = NULL;
- kfree(ldev);
return -ENODEV;
}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 86d0caf91b3..62e90b862a0 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
snd_pcm_uframes_t avail = 0;
long wait_time, tout;
+ init_waitqueue_entry(&wait, current);
+ set_current_state(TASK_INTERRUPTIBLE);
+ add_wait_queue(&runtime->tsleep, &wait);
+
if (runtime->no_period_wakeup)
wait_time = MAX_SCHEDULE_TIMEOUT;
else {
@@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
}
wait_time = msecs_to_jiffies(wait_time * 1000);
}
- init_waitqueue_entry(&wait, current);
- add_wait_queue(&runtime->tsleep, &wait);
+
for (;;) {
if (signal_pending(current)) {
err = -ERESTARTSYS;
break;
}
+
+ /*
+ * We need to check if space became available already
+ * (and thus the wakeup happened already) first to close
+ * the race of space already having become available.
+ * This check must happen after been added to the waitqueue
+ * and having current state be INTERRUPTIBLE.
+ */
+ if (is_playback)
+ avail = snd_pcm_playback_avail(runtime);
+ else
+ avail = snd_pcm_capture_avail(runtime);
+ if (avail >= runtime->twake)
+ break;
snd_pcm_stream_unlock_irq(substream);
- tout = schedule_timeout_interruptible(wait_time);
+
+ tout = schedule_timeout(wait_time);
+
snd_pcm_stream_lock_irq(substream);
+ set_current_state(TASK_INTERRUPTIBLE);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_SUSPENDED:
err = -ESTRPIPE;
@@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
err = -EIO;
break;
}
- if (is_playback)
- avail = snd_pcm_playback_avail(runtime);
- else
- avail = snd_pcm_capture_avail(runtime);
- if (avail >= runtime->twake)
- break;
}
_endloop:
+ set_current_state(TASK_RUNNING);
remove_wait_queue(&runtime->tsleep, &wait);
*availp = avail;
return err;
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1d48b..a872d0a8297 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
0x103c0944, /* HP nc6220 */
0x103c0934, /* HP nc8220 */
0x103c006d, /* HP nx9105 */
+ 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
0x17340088, /* FSC Scenic-W */
0 /* end */
};
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a270c9..579fc0dce12 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
int err;
snd_azf3328_dbgcallenter();
- if (dev >= SNDRV_CARDS)
- return -ENODEV;
+ if (dev >= SNDRV_CARDS) {
+ err = -ENODEV;
+ goto out;
+ }
if (!enable[dev]) {
dev++;
- return -ENOENT;
+ err = -ENOENT;
+ goto out;
}
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
- return err;
+ goto out;
strcpy(card->driver, "AZF3328");
strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
index be58bf2f3ae..2e5876ce71f 100644
--- a/sound/pci/hda/alc268_quirks.c
+++ b/sound/pci/hda/alc268_quirks.c
@@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
static const struct alc_config_preset alc268_presets[] = {
[ALC267_QUANTA_IL1] = {
- .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc267_quanta_il1_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_3ST] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_base_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
@@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = {
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
- .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_toshiba_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_acer_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER_DMIC] = {
- .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ACER_ASPIRE_ONE] = {
- .mixers = { alc268_acer_aspire_one_mixer,
- alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer},
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_aspire_one_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_DELL] = {
- .mixers = { alc268_dell_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
+ .mixers = { alc268_dell_mixer, alc268_beep_mixer},
+ .cap_mixer = alc268_capture_nosrc_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = {
.init_hook = alc_inithook,
},
[ALC268_ZEPTO] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
+ .mixers = { alc268_base_mixer, alc268_beep_mixer },
+ .cap_mixer = alc268_capture_alt_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = {
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
- .mixers = { alc268_test_mixer, alc268_capture_mixer },
+ .mixers = { alc268_test_mixer },
+ .cap_mixer = alc268_capture_mixer,
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_volume_init_verbs,
alc268_beep_init_verbs },
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c238c..f3aefef3721 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
return -1;
}
recursive++;
- for (i = 0; i < nums; i++)
+ for (i = 0; i < nums; i++) {
+ unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+ if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+ continue;
if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
return i;
+ }
return -1;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d09c3..c34f730f481 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
SNDRV_PCM_RATE_192000, /* 7: 192000Hz */
};
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
int byte_index)
{
unsigned int val;
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_HDMI_ELDD, byte_index);
-
#ifdef BE_PARANOID
printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
#endif
-
- if ((val & AC_ELDD_ELD_VALID) == 0) {
- snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
- byte_index);
- val = 0;
- }
-
- return val & AC_ELDD_ELD_DATA;
+ return val;
}
#define GRAB_BITS(buf, byte, lowbit, bits) \
@@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
if (!buf)
return -ENOMEM;
- for (i = 0; i < size; i++)
- buf[i] = hdmi_get_eld_byte(codec, nid, i);
+ for (i = 0; i < size; i++) {
+ unsigned int val = hdmi_get_eld_data(codec, nid, i);
+ if (!(val & AC_ELDD_ELD_VALID)) {
+ if (!i) {
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data byte %d\n", i);
+ val = 0;
+ } else
+ val &= AC_ELDD_ELD_DATA;
+ buf[i] = val;
+ }
ret = hdmi_update_eld(eld, buf, size);
+error:
kfree(buf);
return ret;
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc9b5b..c45f3e69bcf 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
unsigned int *idxp)
{
- int i;
+ int i, idx;
hda_nid_t nid;
nid = codec->start_nid;
@@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
- *idxp = snd_hda_get_conn_index(codec, nid, pin, false);
- if (*idxp >= 0)
+ idx = snd_hda_get_conn_index(codec, nid, pin, false);
+ if (idx >= 0) {
+ *idxp = idx;
return nid;
+ }
}
return 0;
}
@@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
int index, unsigned int pval, int dir,
struct snd_kcontrol **kctlp)
{
- char tmp[32];
+ char tmp[44];
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc949945..7696d05b935 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
#define MAX_AUTO_DACS 5
+#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */
+
/* fill analog DAC list from the widget tree */
static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
{
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
/* fill pin_dac_pair list from the pin and dac list */
static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
int num_pins, hda_nid_t *dacs, int *rest,
- struct pin_dac_pair *filled, int type)
+ struct pin_dac_pair *filled, int nums,
+ int type)
{
- int i, nums;
+ int i, start = nums;
- nums = 0;
- for (i = 0; i < num_pins; i++) {
+ for (i = 0; i < num_pins; i++, nums++) {
filled[nums].pin = pins[i];
filled[nums].type = type;
filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
- nums++;
+ if (filled[nums].dac)
+ continue;
+ if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+ filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+ filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
}
return nums;
}
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
rest = fill_cx_auto_dacs(codec, dacs);
/* parse all analog output pins */
nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
- dacs, &rest, spec->dac_info,
- AUTO_PIN_LINE_OUT);
- nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_HP_OUT);
- nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_SPEAKER_OUT);
+ dacs, &rest, spec->dac_info, 0,
+ AUTO_PIN_LINE_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_HP_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_SPEAKER_OUT);
spec->dac_info_filled = nums;
/* fill multiout struct */
for (i = 0; i < nums; i++) {
hda_nid_t dac = spec->dac_info[i].dac;
- if (!dac)
+ if (!dac || (dac & DAC_SLAVE_FLAG))
continue;
switch (spec->dac_info[i].type) {
case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
}
if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
cx_auto_check_auto_mic(codec);
- if (imux->num_items > 1 && !spec->auto_mic) {
+ if (imux->num_items > 1) {
for (i = 1; i < imux->num_items; i++) {
if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
nid = spec->dac_info[i].dac;
if (!nid)
nid = spec->multiout.dac_nids[0];
+ else if (nid & DAC_SLAVE_FLAG)
+ nid &= ~DAC_SLAVE_FLAG;
select_connection(codec, spec->dac_info[i].pin, nid);
}
if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
hda_nid_t pin, const char *name, int idx)
{
unsigned int caps;
- caps = query_amp_caps(codec, dac, HDA_OUTPUT);
- if (caps & AC_AMPCAP_NUM_STEPS)
- return cx_auto_add_pb_volume(codec, dac, name, idx);
+ if (dac && !(dac & DAC_SLAVE_FLAG)) {
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ if (caps & AC_AMPCAP_NUM_STEPS)
+ return cx_auto_add_pb_volume(codec, dac, name, idx);
+ }
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (caps & AC_AMPCAP_NUM_STEPS)
return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
- if (!spec->dac_info[i].dac)
- continue;
+ hda_nid_t dac = spec->dac_info[i].dac;
type = spec->dac_info[i].type;
if (type == AUTO_PIN_LINE_OUT)
type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
idx = num_spk++;
break;
}
- err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+ err = try_add_pb_volume(codec, dac,
spec->dac_info[i].pin,
label, idx);
if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a1aa09f47f..0503c999e7d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -168,7 +168,7 @@ struct alc_spec {
unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */
unsigned int automute:1; /* HP automute enabled */
unsigned int detect_line:1; /* Line-out detection enabled */
- unsigned int automute_lines:1; /* automute line-out as well */
+ unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */
unsigned int automute_hp_lo:1; /* both HP and LO available */
/* other flags */
@@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec)
if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
return;
- if (!spec->automute_lines || !spec->automute)
+ if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
on = 0;
else
on = spec->jack_present;
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute)
- return;
spec->jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins);
+ if (!spec->automute)
+ return;
update_speakers(codec);
}
@@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute || !spec->detect_line)
- return;
spec->line_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins);
+ if (!spec->automute || !spec->detect_line)
+ return;
update_speakers(codec);
}
@@ -803,7 +803,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
unsigned int val;
if (!spec->automute)
val = 0;
- else if (!spec->automute_lines)
+ else if (!spec->automute_hp_lo || !spec->automute_lines)
val = 1;
else
val = 2;
@@ -824,7 +824,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
spec->automute = 0;
break;
case 1:
- if (spec->automute && !spec->automute_lines)
+ if (spec->automute &&
+ (!spec->automute_hp_lo || !spec->automute_lines))
return 0;
spec->automute = 1;
spec->automute_lines = 0;
@@ -1784,6 +1785,7 @@ static const char * const alc_slave_vols[] = {
"Speaker Playback Volume",
"Mono Playback Volume",
"Line-Out Playback Volume",
+ "PCM Playback Volume",
NULL,
};
@@ -1798,6 +1800,7 @@ static const char * const alc_slave_sws[] = {
"Mono Playback Switch",
"IEC958 Playback Switch",
"Line-Out Playback Switch",
+ "PCM Playback Switch",
NULL,
};
@@ -3081,16 +3084,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
+ hda_nid_t pin, dac;
pin = spec->autocfg.hp_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
- spec->multiout.hp_nid);
+ if (pin) {
+ dac = spec->multiout.hp_nid;
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+ }
pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
- spec->multiout.extra_out_nid[0]);
+ if (pin) {
+ dac = spec->multiout.extra_out_nid[0];
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+ }
}
/*
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b59c00..1b7c11432aa 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac_vrefout_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
{
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
return 1;
}
+#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@@ -6571,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c033cb..5956584ea3a 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
switch (params_rate(params)) {
case 48000:
- clk = 12288000;
+ clk = 24576000;
break;
}
@@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
.cpu_dai_name = "bfin-tdm.0",
.codec_dai_name ="ad193x-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad193x.5",
+ .codec_name = "spi0.5",
.ops = &bf5xx_ad193x_ops,
},
{
@@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
.cpu_dai_name = "bfin-tdm.1",
.codec_dai_name ="ad193x-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad193x.5",
+ .codec_name = "spi0.5",
.ops = &bf5xx_ad193x_ops,
},
};
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5ffe6..eedb6f5e582 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -27,11 +27,6 @@ struct ad193x_priv {
int sysclk;
};
-/* ad193x register cache & default register settings */
-static const u8 ad193x_reg[AD193X_NUM_REGS] = {
- 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
/*
* AD193X volume/mute/de-emphasis etc. controls
*/
@@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
- reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+ reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+ | (word_len << AD193X_DAC_WORD_LEN_SHFT);
snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
@@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
.probe = ad193x_probe,
- .reg_cache_default = ad193x_reg,
- .reg_cache_size = AD193X_NUM_REGS,
- .reg_word_size = sizeof(u16),
};
#if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b549787..cccc2e8e5fb 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
#define AD193X_DAC_LEFT_HIGH (1 << 3)
#define AD193X_DAC_BCLK_INV (1 << 7)
#define AD193X_DAC_CTRL2 0x804
-#define AD193X_DAC_WORD_LEN_MASK 0xC
+#define AD193X_DAC_WORD_LEN_SHFT 3
+#define AD193X_DAC_WORD_LEN_MASK 0x18
#define AD193X_DAC_MASTER_MUTE 1
#define AD193X_DAC_CHNL_MUTE 0x805
#define AD193X_DACL1_MUTE 0
@@ -63,7 +64,7 @@
#define AD193X_ADC_CTRL1 0x80f
#define AD193X_ADC_SERFMT_MASK 0x60
#define AD193X_ADC_SERFMT_STEREO (0 << 5)
-#define AD193X_ADC_SERFMT_TDM (1 << 2)
+#define AD193X_ADC_SERFMT_TDM (1 << 5)
#define AD193X_ADC_SERFMT_AUX (2 << 5)
#define AD193X_ADC_WORD_LEN_MASK 0x3
#define AD193X_ADC_CTRL2 0x810
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 409d89d1f34..fbd7eb9e61c 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+ kfree(sta32x);
return ret;
}
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 38f38fddd19..d0003cc3bcd 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -778,11 +778,19 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi)
return 0;
}
+static const struct spi_device_id wm8750_spi_ids[] = {
+ { "wm8750", 0 },
+ { "wm8987", 0 },
+ { },
+};
+MODULE_DEVICE_TABLE(spi, wm8750_spi_ids);
+
static struct spi_driver wm8750_spi_driver = {
.driver = {
.name = "wm8750-codec",
.owner = THIS_MODULE,
},
+ .id_table = wm8750_spi_ids,
.probe = wm8750_spi_probe,
.remove = __devexit_p(wm8750_spi_remove),
};
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 43e3d760766..4ad8ebd290e 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2046,8 +2046,13 @@ static int wm8903_probe(struct snd_soc_codec *codec)
/* power down chip */
static int wm8903_remove(struct snd_soc_codec *codec)
{
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+
wm8903_free_gpio(codec);
wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ if (wm8903->irq)
+ free_irq(wm8903->irq, codec);
+
return 0;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 60d740ebeb5..1725550c293 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (fll) {
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_ENA, WM8962_FLL_ENA);
if (wm8962->irq) {
@@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
WM8962_BIAS_ENA | 0x180);
msleep(5);
-
- snd_soc_update_bits(codec, WM8962_CLOCKING2,
- WM8962_CLKREG_OVD,
- WM8962_CLKREG_OVD);
}
/* VMID 2*250k */
@@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
WM8962_FLL_ENA, fll1);
@@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec)
*/
snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
+ /* Ensure we have soft control over all registers */
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
if (pdata) {
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 09e680ae88b..b393f9fac97 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2981,6 +2981,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
wm8994->hubs.dcs_readback_mode = 1;
break;
}
+ break;
case WM8958:
wm8994->hubs.dcs_readback_mode = 1;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index ab8e9d1aaff..0cdb9d10567 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = {
};
static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text);
+ SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
static const char *hpf_mode_text[] = {
"HiFi", "Custom", "Voice"
@@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
-SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-
-SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux),
+SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux),
SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0),
@@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
{ "AIF2RX0", NULL, "AIFCLK" },
{ "AIF2RX1", NULL, "AIFCLK" },
+ { "AIF1TX0", NULL, "AIFCLK" },
+ { "AIF1TX1", NULL, "AIFCLK" },
+ { "AIF1TX2", NULL, "AIFCLK" },
+ { "AIF1TX3", NULL, "AIFCLK" },
+ { "AIF1TX4", NULL, "AIFCLK" },
+ { "AIF1TX5", NULL, "AIFCLK" },
+
+ { "AIF2TX0", NULL, "AIFCLK" },
+ { "AIF2TX1", NULL, "AIFCLK" },
+
{ "DSP1RXL", NULL, "SYSDSPCLK" },
{ "DSP1RXR", NULL, "SYSDSPCLK" },
{ "DSP2RXL", NULL, "SYSDSPCLK" },
@@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
+ /* Clear any pending completions (eg, from failed startups) */
+ try_wait_for_completion(&wm8996->fll_lock);
+
snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
WM8996_FLL_ENA, WM8996_FLL_ENA);
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 56efa0c1c9a..099614e1665 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
err = -ENODEV;
- goto fail;
+ goto fail_free_info;
}
info->mem = request_mem_region(res->start, resource_size(res),
pdev->name);
if (!info->mem) {
err = -EBUSY;
- goto fail;
+ goto fail_free_info;
}
info->regs = ioremap(info->mem->start, resource_size(info->mem));
@@ -435,6 +435,7 @@ fail_unmap_mem:
iounmap(info->regs);
fail_release_mem:
release_mem_region(info->mem->start, resource_size(info->mem));
+fail_free_info:
kfree(info);
fail:
return err;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 732208c8c0b..cb50598338e 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
* assume that device_node pointers are a valid comparison.
*/
np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
}
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index fd0dc46afc3..5c6c2457386 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
.pcm_free = &psc_dma_free,
};
-static int mpc5200_hpcd_probe(struct of_device *op)
+static int mpc5200_hpcd_probe(struct platform_device *op)
{
phys_addr_t fifo;
struct psc_dma *psc_dma;
@@ -487,7 +487,7 @@ out_unmap:
return ret;
}
-static int mpc5200_hpcd_remove(struct of_device *op)
+static int mpc5200_hpcd_remove(struct platform_device *op)
{
struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
@@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match);
static struct platform_driver mpc5200_hpcd_of_driver = {
.probe = mpc5200_hpcd_probe,
.remove = mpc5200_hpcd_remove,
- .dev = {
+ .driver = {
.owner = THIS_MODULE,
.name = "mpc5200-pcm-audio",
.of_match_table = mpc5200_hpcd_match,
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a1929795958..358f0baaf71 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
- if (!machine_data)
- return -ENOMEM;
+ if (!machine_data) {
+ ret = -ENOMEM;
+ goto error_alloc;
+ }
machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
machine_data->dai[0].ops = &mpc8610_hpcd_ops;
@@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
ret = platform_device_add(sound_device);
if (ret) {
dev_err(&pdev->dev, "platform device add failed\n");
- goto error;
+ goto error_sound;
}
dev_set_drvdata(&pdev->dev, sound_device);
@@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
return 0;
+error_sound:
+ platform_device_unregister(sound_device);
error:
- of_node_put(codec_np);
-
- if (sound_device)
- platform_device_unregister(sound_device);
-
kfree(machine_data);
-
+error_alloc:
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 8fa4d5f8eda..fcb862eb0c7 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np,
* dai->platform name should already point to an allocated buffer.
*/
ret = of_address_to_resource(dma_channel_np, 0, &res);
- if (ret)
+ if (ret) {
+ of_node_put(dma_channel_np);
return ret;
+ }
snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
(unsigned long long) res.start, dma_channel_np->name);
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 309c59e6fb6..7945625e0e0 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -240,7 +240,6 @@ static int ssi_irq = 0;
static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_card *card = rtd->card->snd_card;
struct snd_soc_dai *dai = rtd->cpu_dai;
struct snd_pcm *pcm = rtd->pcm;
int ret;
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index a33fc51f363..d0bcf3fcea0 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
if (!priv->mem) {
dev_err(&pdev->dev, "request_mem_region failed\n");
err = -EBUSY;
- goto error;
+ goto err_alloc;
}
priv->io = ioremap(priv->mem->start, SZ_16K);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 30fe0d0efe1..0aa475f92ef 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
}
/* Set codec bias level */
- ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
/* Add hook switch - can be used to control the codec from userspace
* even if line discipline fails */
@@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void)
ams_delta_hook_switch_gpios);
/* Keep modem power on */
- ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(&ams_delta_audio_card,
+ &ams_delta_audio_card.rtd[0].codec->dapm,
+ SND_SOC_BIAS_STANDBY);
platform_device_unregister(cx20442_platform_device);
platform_device_unregister(ams_delta_audio_platform_device);
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 83d213bfd3d..62e292f4931 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -3,7 +3,7 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -402,6 +402,6 @@ static void __exit n810_soc_exit(void)
module_init(n810_soc_init);
module_exit(n810_soc_exit);
-MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
MODULE_DESCRIPTION("ALSA SoC Nokia N810");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 07b77235724..ebcc2d4d2b1 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -3,7 +3,7 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
@@ -780,6 +780,6 @@ static void __exit snd_omap_mcbsp_exit(void)
}
module_exit(snd_omap_mcbsp_exit);
-MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
MODULE_DESCRIPTION("OMAP I2S SoC Interface");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 9a7dedd6f5a..65cde9d3807 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -3,7 +3,7 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index b2f5751edae..9b5c88ac35b 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -3,7 +3,7 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
@@ -436,6 +436,6 @@ static void __exit snd_omap_pcm_exit(void)
}
module_exit(snd_omap_pcm_exit);
-MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index a0ed1dbb52d..f95fe306417 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -3,7 +3,7 @@
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
* Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 0aae998b654..893300a53ba 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -5,7 +5,7 @@
*
* Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
* Eduardo Valentin <eduardo.valentin@nokia.com>
- * Jarkko Nikula <jhnikula@gmail.com>
+ * Jarkko Nikula <jarkko.nikula@bitmer.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index b99091fc34e..65f980ef287 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
+ select SND_SOC_WM1250_EV1
config SND_SOC_SPEYSIDE_WM8962
tristate "Audio support for Wolfson Speyside with WM8962"
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 9eb3b12eb72..8509d3c4366 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -1,5 +1,6 @@
# S3c24XX Platform Support
snd-soc-s3c24xx-objs := dma.o
+snd-soc-idma-objs := idma.o
snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
snd-soc-ac97-objs := ac97.o
@@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o
obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o
obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o
+obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o
# S3C24XX Machine Support
snd-soc-jive-wm8750-objs := jive_wm8750.o
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 241f55d0066..c6c65892294 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,6 +13,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
new file mode 100644
index 00000000000..ebde0740ab1
--- /dev/null
+++ b/sound/soc/samsung/idma.c
@@ -0,0 +1,453 @@
+/*
+ * sound/soc/samsung/idma.c
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd.
+ * http://www.samsung.com
+ *
+ * I2S0's Internal DMA driver
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "i2s.h"
+#include "idma.h"
+#include "dma.h"
+#include "i2s-regs.h"
+
+#define ST_RUNNING (1<<0)
+#define ST_OPENED (1<<1)
+
+static const struct snd_pcm_hardware idma_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_U24_LE |
+ SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S8,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = MAX_IDMA_BUFFER,
+ .period_bytes_min = 128,
+ .period_bytes_max = MAX_IDMA_PERIOD,
+ .periods_min = 1,
+ .periods_max = 2,
+};
+
+struct idma_ctrl {
+ spinlock_t lock;
+ int state;
+ dma_addr_t start;
+ dma_addr_t pos;
+ dma_addr_t end;
+ dma_addr_t period;
+ dma_addr_t periodsz;
+ void *token;
+ void (*cb)(void *dt, int bytes_xfer);
+};
+
+static struct idma_info {
+ spinlock_t lock;
+ void __iomem *regs;
+ dma_addr_t lp_tx_addr;
+} idma;
+
+static void idma_getpos(dma_addr_t *src)
+{
+ *src = idma.lp_tx_addr +
+ (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4;
+}
+
+static int idma_enqueue(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct idma_ctrl *prtd = substream->runtime->private_data;
+ u32 val;
+
+ spin_lock(&prtd->lock);
+ prtd->token = (void *) substream;
+ spin_unlock(&prtd->lock);
+
+ /* Internal DMA Level0 Interrupt Address */
+ val = idma.lp_tx_addr + prtd->periodsz;
+ writel(val, idma.regs + I2SLVL0ADDR);
+
+ /* Start address0 of I2S internal DMA operation. */
+ val = idma.lp_tx_addr;
+ writel(val, idma.regs + I2SSTR0);
+
+ /*
+ * Transfer block size for I2S internal DMA.
+ * Should decide transfer size before start dma operation
+ */
+ val = readl(idma.regs + I2SSIZE);
+ val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT);
+ val |= (((runtime->dma_bytes >> 2) &
+ I2SSIZE_TRNMSK) << I2SSIZE_SHIFT);
+ writel(val, idma.regs + I2SSIZE);
+
+ val = readl(idma.regs + I2SAHB);
+ val |= AHB_INTENLVL0;
+ writel(val, idma.regs + I2SAHB);
+
+ return 0;
+}
+
+static void idma_setcallbk(struct snd_pcm_substream *substream,
+ void (*cb)(void *, int))
+{
+ struct idma_ctrl *prtd = substream->runtime->private_data;
+
+ spin_lock(&prtd->lock);
+ prtd->cb = cb;
+ spin_unlock(&prtd->lock);
+}
+
+static void idma_control(int op)
+{
+ u32 val = readl(idma.regs + I2SAHB);
+
+ spin_lock(&idma.lock);
+
+ switch (op) {
+ case LPAM_DMA_START:
+ val |= (AHB_INTENLVL0 | AHB_DMAEN);
+ break;
+ case LPAM_DMA_STOP:
+ val &= ~(AHB_INTENLVL0 | AHB_DMAEN);
+ break;
+ default:
+ spin_unlock(&idma.lock);
+ return;
+ }
+
+ writel(val, idma.regs + I2SAHB);
+ spin_unlock(&idma.lock);
+}
+
+static void idma_done(void *id, int bytes_xfer)
+{
+ struct snd_pcm_substream *substream = id;
+ struct idma_ctrl *prtd = substream->runtime->private_data;
+
+ if (prtd && (prtd->state & ST_RUNNING))
+ snd_pcm_period_elapsed(substream);
+}
+
+static int idma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct idma_ctrl *prtd = substream->runtime->private_data;
+ u32 mod = readl(idma.regs + I2SMOD);
+ u32 ahb = readl(idma.regs + I2SAHB);
+
+ ahb |= (AHB_DMARLD | AHB_INTMASK);
+ mod |= MOD_TXS_IDMA;
+ writel(ahb, idma.regs + I2SAHB);
+ writel(mod, idma.regs + I2SMOD);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ prtd->start = prtd->pos = runtime->dma_addr;
+ prtd->period = params_periods(params);
+ prtd->periodsz = params_period_bytes(params);
+ prtd->end = runtime->dma_addr + runtime->dma_bytes;
+
+ idma_setcallbk(substream, idma_done);
+
+ return 0;
+}
+
+static int idma_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ return 0;
+}
+
+static int idma_prepare(struct snd_pcm_substream *substream)
+{
+ struct idma_ctrl *prtd = substream->runtime->private_data;
+
+ prtd->pos = prtd->start;
+
+ /* flush the DMA channel */
+ idma_control(LPAM_DMA_STOP);
+ idma_enqueue(substream);
+
+ return 0;
+}
+
+static int idma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct idma_ctrl *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ spin_lock(&prtd->lock);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ prtd->state |= ST_RUNNING;
+ idma_control(LPAM_DMA_START);
+ break;
+
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ prtd->state &= ~ST_RUNNING;
+ idma_control(LPAM_DMA_STOP);
+ break;
+
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ spin_unlock(&prtd->lock);
+
+ return ret;
+}
+
+static snd_pcm_uframes_t
+ idma_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct idma_ctrl *prtd = runtime->private_data;
+ dma_addr_t src;
+ unsigned long res;
+
+ spin_lock(&prtd->lock);
+
+ idma_getpos(&src);
+ res = src - prtd->start;
+
+ spin_unlock(&prtd->lock);
+
+ return bytes_to_frames(substream->runtime, res);
+}
+
+static int idma_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long size, offset;
+ int ret;
+
+ /* From snd_pcm_lib_mmap_iomem */
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ vma->vm_flags |= VM_IO;
+ size = vma->vm_end - vma->vm_start;
+ offset = vma->vm_pgoff << PAGE_SHIFT;
+ ret = io_remap_pfn_range(vma, vma->vm_start,
+ (runtime->dma_addr + offset) >> PAGE_SHIFT,
+ size, vma->vm_page_prot);
+
+ return ret;
+}
+
+static irqreturn_t iis_irq(int irqno, void *dev_id)
+{
+ struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id;
+ u32 iiscon, iisahb, val, addr;
+
+ iisahb = readl(idma.regs + I2SAHB);
+ iiscon = readl(idma.regs + I2SCON);
+
+ val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0;
+
+ if (val) {
+ iisahb |= val;
+ writel(iisahb, idma.regs + I2SAHB);
+
+ addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr;
+ addr += prtd->periodsz;
+ addr %= (prtd->end - prtd->start);
+ addr += idma.lp_tx_addr;
+
+ writel(addr, idma.regs + I2SLVL0ADDR);
+
+ if (prtd->cb)
+ prtd->cb(prtd->token, prtd->period);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static int idma_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct idma_ctrl *prtd;
+ int ret;
+
+ snd_soc_set_runtime_hwparams(substream, &idma_hardware);
+
+ prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd);
+ if (ret < 0) {
+ pr_err("fail to claim i2s irq , ret = %d\n", ret);
+ kfree(prtd);
+ return ret;
+ }
+
+ spin_lock_init(&prtd->lock);
+
+ runtime->private_data = prtd;
+
+ return 0;
+}
+
+static int idma_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct idma_ctrl *prtd = runtime->private_data;
+
+ free_irq(IRQ_I2S0, prtd);
+
+ if (!prtd)
+ pr_err("idma_close called with prtd == NULL\n");
+
+ kfree(prtd);
+
+ return 0;
+}
+
+static struct snd_pcm_ops idma_ops = {
+ .open = idma_open,
+ .close = idma_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .trigger = idma_trigger,
+ .pointer = idma_pointer,
+ .mmap = idma_mmap,
+ .hw_params = idma_hw_params,
+ .hw_free = idma_hw_free,
+ .prepare = idma_prepare,
+};
+
+static void idma_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (!substream)
+ return;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ return;
+
+ iounmap(buf->area);
+
+ buf->area = NULL;
+ buf->addr = 0;
+}
+
+static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+
+ /* Assign PCM buffer pointers */
+ buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS;
+ buf->addr = idma.lp_tx_addr;
+ buf->bytes = idma_hardware.buffer_bytes_max;
+ buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes);
+
+ return 0;
+}
+
+static u64 idma_mask = DMA_BIT_MASK(32);
+
+static int idma_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &idma_mask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (dai->driver->playback.channels_min)
+ ret = preallocate_idma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+
+ return ret;
+}
+
+void idma_reg_addr_init(void *regs, dma_addr_t addr)
+{
+ spin_lock_init(&idma.lock);
+ idma.regs = regs;
+ idma.lp_tx_addr = addr;
+}
+
+struct snd_soc_platform_driver asoc_idma_platform = {
+ .ops = &idma_ops,
+ .pcm_new = idma_new,
+ .pcm_free = idma_free,
+};
+
+static int __devinit asoc_idma_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform);
+}
+
+static int __devexit asoc_idma_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_idma_driver = {
+ .driver = {
+ .name = "samsung-idma",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = asoc_idma_platform_probe,
+ .remove = __devexit_p(asoc_idma_platform_remove),
+};
+
+static int __init asoc_idma_init(void)
+{
+ return platform_driver_register(&asoc_idma_driver);
+}
+module_init(asoc_idma_init);
+
+static void __exit asoc_idma_exit(void)
+{
+ platform_driver_unregister(&asoc_idma_driver);
+}
+module_exit(asoc_idma_exit);
+
+MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar@gmail.com>");
+MODULE_DESCRIPTION("Samsung ASoC IDMA Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h
new file mode 100644
index 00000000000..48273216166
--- /dev/null
+++ b/sound/soc/samsung/idma.h
@@ -0,0 +1,26 @@
+/*
+ * sound/soc/samsung/idma.h
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd
+ * http://www.samsung.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __SND_SOC_SAMSUNG_IDMA_H_
+#define __SND_SOC_SAMSUNG_IDMA_H_
+
+extern void idma_reg_addr_init(void *regs, dma_addr_t addr);
+
+/* dma_state */
+#define LPAM_DMA_STOP 0
+#define LPAM_DMA_START 1
+
+#define MAX_IDMA_PERIOD (128 * 1024)
+#define MAX_IDMA_BUFFER (160 * 1024)
+
+#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 3b53ad54bc3..14eb6ea69e7 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = {
.cpu_dai_name = "s3c2412-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "samsung-audio",
- .codec_name = "wm8750-codec.0-0x1a",
+ .codec_name = "wm8750-codec.0-001a",
.init = jive_wm8750_init,
.ops = &jive_ops,
};
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 1e574a5d440..bc8c1676459 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,6 +17,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 8ac42bf8209..72535f2daaf 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
@@ -37,7 +40,7 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
44100 * 256,
SND_SOC_CLOCK_IN);
if (ret < 0) {
- pr_err("Failed to set SYSCLK: %d\n");
+ pr_err("Failed to set SYSCLK: %d\n", ret);
return ret;
}
}
@@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_STANDBY:
ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index d9f8aded51f..20b7f3b003a 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -203,14 +203,14 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec)
rbnode = rb_entry(node, struct snd_soc_rbtree_node, node);
for (i = 0; i < rbnode->blklen; ++i) {
regtmp = rbnode->base_reg + i;
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, regtmp));
val = snd_soc_rbtree_get_register(rbnode, i);
def = snd_soc_get_cache_val(codec->reg_def_copy, i,
rbnode->word_size);
if (val == def)
continue;
+ WARN_ON(!snd_soc_codec_writable_register(codec, regtmp));
+
codec->cache_bypass = 1;
ret = snd_soc_write(codec, regtmp, val);
codec->cache_bypass = 0;
@@ -563,8 +563,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
lzo_blocks = codec->reg_cache;
for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) {
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, i));
+ WARN_ON(!snd_soc_codec_writable_register(codec, i));
ret = snd_soc_cache_read(codec, i, &val);
if (ret)
return ret;
@@ -823,8 +822,6 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
codec_drv = codec->driver;
for (i = 0; i < codec_drv->reg_cache_size; ++i) {
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, i));
ret = snd_soc_cache_read(codec, i, &val);
if (ret)
return ret;
@@ -832,6 +829,9 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
if (snd_soc_get_cache_val(codec->reg_def_copy,
i, codec_drv->reg_word_size) == val)
continue;
+
+ WARN_ON(!snd_soc_codec_writable_register(codec, i));
+
ret = snd_soc_write(codec, i, val);
if (ret)
return ret;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca2749..d2ef014af21 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1633,7 +1633,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec,
if (codec->readable_register)
return codec->readable_register(codec, reg);
else
- return 0;
+ return 1;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register);
@@ -1651,7 +1651,7 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec,
if (codec->writable_register)
return codec->writable_register(codec, reg);
else
- return 0;
+ return 1;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register);
@@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
if (prefix) {
name_len = strlen(long_name) + strlen(prefix) + 2;
- name = kmalloc(name_len, GFP_ATOMIC);
+ name = kmalloc(name_len, GFP_KERNEL);
if (!name)
return NULL;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7e15914b363..d67c637557a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2763,7 +2763,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
/**
* snd_soc_dapm_free - free dapm resources
- * @card: SoC device
+ * @dapm: DAPM context
*
* Free all dapm widgets and resources.
*/
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c8058..a62f7dd4ba9 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
#define snd_soc_16_8_read_i2c NULL
#endif
+#if defined(CONFIG_SPI_MASTER)
+static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ struct spi_device *spi = codec->control_data;
+
+ const u16 reg = cpu_to_be16(r | 0x100);
+ u8 data;
+ int ret;
+
+ ret = spi_write_then_read(spi, &reg, 2, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_16_8_read_spi NULL
+#endif
+
static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
@@ -295,6 +314,7 @@ static struct {
int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+ unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
} io_types[] = {
{
.addr_bits = 4, .data_bits = 12,
@@ -318,6 +338,7 @@ static struct {
.addr_bits = 16, .data_bits = 8,
.write = snd_soc_16_8_write,
.i2c_read = snd_soc_16_8_read_i2c,
+ .spi_read = snd_soc_16_8_read_spi,
},
{
.addr_bits = 16, .data_bits = 16,
@@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
#ifdef CONFIG_SPI_MASTER
codec->hw_write = do_spi_write;
#endif
+ if (io_types[i].spi_read)
+ codec->hw_read = io_types[i].spi_read;
codec->control_data = container_of(codec->dev,
struct spi_device,
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98d584..fa31d9c2abd 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
snd_soc_dapm_sync(dapm);
- snd_jack_report(jack->jack, status);
+ snd_jack_report(jack->jack, jack->status);
out:
mutex_unlock(&codec->mutex);
@@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
IRQF_TRIGGER_FALLING,
gpios[i].name,
&gpios[i]);
- if (ret)
+ if (ret < 0)
goto err;
if (gpios[i].wake) {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b5759397afa..2879c883eeb 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
codec_dai->active--;
codec->active--;
+ if (!cpu_dai->active && !codec_dai->active)
+ rtd->rate = 0;
+
/* Muting the DAC suppresses artifacts caused during digital
* shutdown, for example from stopping clocks.
*/
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index ff86e5e3db6..c7cfd96e991 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -309,9 +309,14 @@ static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
{
- struct snd_pcm_substream *substream = pcm->streams[stream].substream;
- struct snd_dma_buffer *buf = &substream->dma_buffer;
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ return;
+ buf = &substream->dma_buffer;
if (!buf->area)
return;
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index a42e9ac30f2..be27f1d229a 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -56,6 +56,7 @@
#define GPIO_HP_MUTE BIT(1)
#define GPIO_INT_MIC_EN BIT(2)
#define GPIO_EXT_MIC_EN BIT(3)
+#define GPIO_HP_DET BIT(4)
struct tegra_wm8903 {
struct tegra_asoc_utils_data util_data;
@@ -304,6 +305,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack,
1,
&tegra_wm8903_hp_jack_gpio);
+ machine->gpio_requested |= GPIO_HP_DET;
}
snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
@@ -317,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
/* FIXME: Calculate automatically based on DAPM routes? */
- if (!machine_is_harmony() && !machine_is_ventana())
+ if (!machine_is_harmony())
snd_soc_dapm_nc_pin(dapm, "IN1L");
if (!machine_is_seaboard() && !machine_is_aebl())
snd_soc_dapm_nc_pin(dapm, "IN1R");
@@ -393,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, card);
snd_soc_card_set_drvdata(card, machine);
- if (machine_is_harmony() || machine_is_ventana()) {
+ if (machine_is_harmony()) {
card->dapm_routes = harmony_audio_map;
card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
} else if (machine_is_seaboard()) {
@@ -429,10 +431,10 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
struct tegra_wm8903_platform_data *pdata = machine->pdata;
- snd_soc_unregister_card(card);
-
- tegra_asoc_utils_fini(&machine->util_data);
-
+ if (machine->gpio_requested & GPIO_HP_DET)
+ snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
+ 1,
+ &tegra_wm8903_hp_jack_gpio);
if (machine->gpio_requested & GPIO_EXT_MIC_EN)
gpio_free(pdata->gpio_ext_mic_en);
if (machine->gpio_requested & GPIO_INT_MIC_EN)
@@ -441,6 +443,11 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
gpio_free(pdata->gpio_hp_mute);
if (machine->gpio_requested & GPIO_SPKR_EN)
gpio_free(pdata->gpio_spkr_en);
+ machine->gpio_requested = 0;
+
+ snd_soc_unregister_card(card);
+
+ tegra_asoc_utils_fini(&machine->util_data);
kfree(machine);
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index d0d493ca28a..2cf87f5afed 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev)
for (i = 0; i < N_URBS; i++) {
usb_kill_urb(dev->data_urbs_in[i]);
- usb_kill_urb(dev->data_urbs_out[i]);
+
+ if (test_bit(i, &dev->outurb_active_mask))
+ usb_kill_urb(dev->data_urbs_out[i]);
}
+
+ dev->outurb_active_mask = 0;
}
static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
@@ -612,8 +616,9 @@ static void read_completed(struct urb *urb)
{
struct snd_usb_caiaq_cb_info *info = urb->context;
struct snd_usb_caiaqdev *dev;
- struct urb *out;
- int frame, len, send_it = 0, outframe = 0;
+ struct urb *out = NULL;
+ int i, frame, len, send_it = 0, outframe = 0;
+ size_t offset = 0;
if (urb->status || !info)
return;
@@ -623,7 +628,17 @@ static void read_completed(struct urb *urb)
if (!dev->streaming)
return;
- out = dev->data_urbs_out[info->index];
+ /* find an unused output urb that is unused */
+ for (i = 0; i < N_URBS; i++)
+ if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) {
+ out = dev->data_urbs_out[i];
+ break;
+ }
+
+ if (!out) {
+ log("Unable to find an output urb to use\n");
+ goto requeue;
+ }
/* read the recently received packet and send back one which has
* the same layout */
@@ -634,7 +649,8 @@ static void read_completed(struct urb *urb)
len = urb->iso_frame_desc[outframe].actual_length;
out->iso_frame_desc[outframe].length = len;
out->iso_frame_desc[outframe].actual_length = 0;
- out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+ out->iso_frame_desc[outframe].offset = offset;
+ offset += len;
if (len > 0) {
spin_lock(&dev->spinlock);
@@ -650,11 +666,15 @@ static void read_completed(struct urb *urb)
}
if (send_it) {
- out->number_of_packets = FRAMES_PER_URB;
+ out->number_of_packets = outframe;
out->transfer_flags = URB_ISO_ASAP;
usb_submit_urb(out, GFP_ATOMIC);
+ } else {
+ struct snd_usb_caiaq_cb_info *oinfo = out->context;
+ clear_bit(oinfo->index, &dev->outurb_active_mask);
}
+requeue:
/* re-submit inbound urb */
for (frame = 0; frame < FRAMES_PER_URB; frame++) {
urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
@@ -676,6 +696,8 @@ static void write_completed(struct urb *urb)
dev->output_running = 1;
wake_up(&dev->prepare_wait_queue);
}
+
+ clear_bit(info->index, &dev->outurb_active_mask);
}
static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
@@ -827,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
if (!dev->data_cb_info)
return -ENOMEM;
+ dev->outurb_active_mask = 0;
+ BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8));
+
for (i = 0; i < N_URBS; i++) {
dev->data_cb_info[i].dev = dev;
dev->data_cb_info[i].index = i;
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index b2b310194ff..3f9c6339ae9 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -96,6 +96,7 @@ struct snd_usb_caiaqdev {
int input_panic, output_panic, warned;
char *audio_in_buf, *audio_out_buf;
unsigned int samplerates, bpp;
+ unsigned long outurb_active_mask;
struct snd_pcm_substream *sub_playback[MAX_STREAMS];
struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c04d7c71ac8..cdd19d7fe50 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p,
if (p && p->dB) {
cval->dBmin = p->dB->min;
cval->dBmax = p->dB->max;
+ cval->initialized = 1;
}
}
@@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
" Switch" : " Volume");
if (control == UAC_FU_VOLUME) {
check_mapped_dB(map, cval);
- if (cval->dBmin < cval->dBmax) {
+ if (cval->dBmin < cval->dBmax || !cval->initialized) {
kctl->tlv.c = mixer_vol_tlv;
kctl->vd[0].access |=
SNDRV_CTL_ELEM_ACCESS_TLV_READ |
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 4d4f86552a2..a42e3ef3832 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
+{
+ USB_DEVICE(0x0582, 0x0130),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "BOSS", */
+ /* .product_name = "MICRO BR-80", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
/* Guillemot devices */
{