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authorTakashi Iwai <tiwai@suse.de>2010-12-13 09:29:52 +0100
committerTakashi Iwai <tiwai@suse.de>2010-12-13 09:29:52 +0100
commit354d14b3f53cc749c5d26d4ba7280d1e931d6bc4 (patch)
tree59b597e44902d8db8bde7deac1e7e707327c6fe6 /sound
parent5144c534d16529bc469396211131e8935589f833 (diff)
parent5b84ba26a9672e615897234fa5efd3eea2d6b295 (diff)
downloadop-kernel-dev-354d14b3f53cc749c5d26d4ba7280d1e931d6bc4.zip
op-kernel-dev-354d14b3f53cc749c5d26d4ba7280d1e931d6bc4.tar.gz
Merge branch 'topic/workq-update' into topic/misc
Diffstat (limited to 'sound')
-rw-r--r--sound/aoa/core/gpio-feature.c7
-rw-r--r--sound/aoa/core/gpio-pmf.c7
-rw-r--r--sound/atmel/abdac.c2
-rw-r--r--sound/core/info.c1
-rw-r--r--sound/core/oss/pcm_oss.c19
-rw-r--r--sound/core/pcm_native.c1
-rw-r--r--sound/core/sound.c1
-rw-r--r--sound/i2c/other/ak4113.c5
-rw-r--r--sound/i2c/other/ak4114.c5
-rw-r--r--sound/pci/ac97/ac97_codec.c6
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c84
-rw-r--r--sound/pci/hda/patch_sigmatel.c6
-rw-r--r--sound/pci/hda/patch_via.c3
-rw-r--r--sound/pci/oxygen/oxygen_lib.c6
-rw-r--r--sound/soc/atmel/Kconfig5
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c9
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c1
-rw-r--r--sound/soc/codecs/max98088.c14
-rw-r--r--sound/soc/codecs/stac9766.c1
-rw-r--r--sound/soc/codecs/tlv320aic3x.c6
-rw-r--r--sound/soc/codecs/tpa6130a2.c4
-rw-r--r--sound/soc/codecs/uda134x.c1
-rw-r--r--sound/soc/codecs/wm8350.c18
-rw-r--r--sound/soc/codecs/wm8523.c1
-rw-r--r--sound/soc/codecs/wm8731.c2
-rw-r--r--sound/soc/codecs/wm8753.c21
-rw-r--r--sound/soc/codecs/wm8776.c1
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/codecs/wm8961.c4
-rw-r--r--sound/soc/codecs/wm8962.c5
-rw-r--r--sound/soc/codecs/wm8994.c6
-rw-r--r--sound/soc/davinci/davinci-evm.c40
-rw-r--r--sound/soc/davinci/davinci-i2s.c15
-rw-r--r--sound/soc/davinci/davinci-mcasp.c13
-rw-r--r--sound/soc/davinci/davinci-sffsdr.c2
-rw-r--r--sound/soc/davinci/davinci-vcif.c16
-rw-r--r--sound/soc/ep93xx/simone.c18
-rw-r--r--sound/soc/fsl/efika-audio-fabric.c1
-rw-r--r--sound/soc/fsl/mpc5200_dma.c1
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c2
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c1
-rw-r--r--sound/soc/fsl/p1022_ds.c1
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c1
-rw-r--r--sound/soc/imx/eukrea-tlv320.c8
-rw-r--r--sound/soc/imx/imx-pcm-dma-mx2.c221
-rw-r--r--sound/soc/imx/imx-ssi.c57
-rw-r--r--sound/soc/imx/imx-ssi.h4
-rw-r--r--sound/soc/imx/phycore-ac97.c33
-rw-r--r--sound/soc/nuc900/nuc900-ac97.c17
-rw-r--r--sound/soc/nuc900/nuc900-audio.h2
-rw-r--r--sound/soc/nuc900/nuc900-pcm.c11
-rw-r--r--sound/soc/omap/Kconfig2
-rw-r--r--sound/soc/omap/omap-mcbsp.c8
-rw-r--r--sound/soc/omap/omap3pandora.c1
-rw-r--r--sound/soc/omap/osk5912.c11
-rw-r--r--sound/soc/pxa/Kconfig1
-rw-r--r--sound/soc/pxa/corgi.c5
-rw-r--r--sound/soc/pxa/magician.c4
-rw-r--r--sound/soc/pxa/poodle.c5
-rw-r--r--sound/soc/pxa/spitz.c5
-rw-r--r--sound/soc/pxa/tosa.c5
-rw-r--r--sound/soc/s3c24xx/Kconfig1
-rw-r--r--sound/soc/s3c24xx/rx1950_uda1380.c20
-rw-r--r--sound/soc/s3c24xx/smdk_spdif.c4
-rw-r--r--sound/soc/s6000/s6000-i2s.c2
-rw-r--r--sound/soc/s6000/s6000-pcm.c2
-rw-r--r--sound/soc/s6000/s6105-ipcam.c2
-rw-r--r--sound/soc/sh/fsi.c25
-rw-r--r--sound/soc/sh/ssi.c2
-rw-r--r--sound/soc/soc-core.c34
-rw-r--r--sound/soc/soc-dapm.c4
-rw-r--r--sound/sound_core.c1
73 files changed, 496 insertions, 367 deletions
diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c
index de8e03a..faa3174 100644
--- a/sound/aoa/core/gpio-feature.c
+++ b/sound/aoa/core/gpio-feature.c
@@ -287,10 +287,9 @@ static void ftr_gpio_exit(struct gpio_runtime *rt)
free_irq(linein_detect_irq, &rt->line_in_notify);
if (rt->line_out_notify.gpio_private)
free_irq(lineout_detect_irq, &rt->line_out_notify);
- cancel_delayed_work(&rt->headphone_notify.work);
- cancel_delayed_work(&rt->line_in_notify.work);
- cancel_delayed_work(&rt->line_out_notify.work);
- flush_scheduled_work();
+ cancel_delayed_work_sync(&rt->headphone_notify.work);
+ cancel_delayed_work_sync(&rt->line_in_notify.work);
+ cancel_delayed_work_sync(&rt->line_out_notify.work);
mutex_destroy(&rt->headphone_notify.mutex);
mutex_destroy(&rt->line_in_notify.mutex);
mutex_destroy(&rt->line_out_notify.mutex);
diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 7e267c9..c8d8a1a 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -107,10 +107,9 @@ static void pmf_gpio_exit(struct gpio_runtime *rt)
/* make sure no work is pending before freeing
* all things */
- cancel_delayed_work(&rt->headphone_notify.work);
- cancel_delayed_work(&rt->line_in_notify.work);
- cancel_delayed_work(&rt->line_out_notify.work);
- flush_scheduled_work();
+ cancel_delayed_work_sync(&rt->headphone_notify.work);
+ cancel_delayed_work_sync(&rt->line_in_notify.work);
+ cancel_delayed_work_sync(&rt->line_out_notify.work);
mutex_destroy(&rt->headphone_notify.mutex);
mutex_destroy(&rt->line_in_notify.mutex);
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 4e47baad..6e24091 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -422,7 +422,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
sample_clk = clk_get(&pdev->dev, "sample_clk");
if (IS_ERR(sample_clk)) {
dev_dbg(&pdev->dev, "no sample clock\n");
- retval = PTR_ERR(pclk);
+ retval = PTR_ERR(sample_clk);
goto out_put_pclk;
}
clk_enable(pclk);
diff --git a/sound/core/info.c b/sound/core/info.c
index b70564e..7077f60 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -23,7 +23,6 @@
#include <linux/time.h>
#include <linux/mm.h>
#include <linux/slab.h>
-#include <linux/smp_lock.h>
#include <linux/string.h>
#include <sound/core.h>
#include <sound/minors.h>
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 5c8c7df..b753ec6 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1510,16 +1510,19 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use
static int snd_pcm_oss_reset(struct snd_pcm_oss_file *pcm_oss_file)
{
struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ int i;
- substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
- if (substream != NULL) {
- snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
- substream->runtime->oss.prepare = 1;
- }
- substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
- if (substream != NULL) {
+ for (i = 0; i < 2; i++) {
+ substream = pcm_oss_file->streams[i];
+ if (!substream)
+ continue;
+ runtime = substream->runtime;
snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
- substream->runtime->oss.prepare = 1;
+ runtime->oss.prepare = 1;
+ runtime->oss.buffer_used = 0;
+ runtime->oss.prev_hw_ptr_period = 0;
+ runtime->oss.period_ptr = 0;
}
return 0;
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index f91a439..0db714e 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -22,7 +22,6 @@
#include <linux/mm.h>
#include <linux/file.h>
#include <linux/slab.h>
-#include <linux/smp_lock.h>
#include <linux/time.h>
#include <linux/pm_qos_params.h>
#include <linux/uio.h>
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 345caea..1c7a3ef 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -21,7 +21,6 @@
#include <linux/init.h>
#include <linux/slab.h>
-#include <linux/smp_lock.h>
#include <linux/time.h>
#include <linux/device.h>
#include <linux/moduleparam.h>
diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index 971a84a..c424d32 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -57,8 +57,7 @@ static void snd_ak4113_free(struct ak4113 *chip)
{
chip->init = 1; /* don't schedule new work */
mb();
- cancel_delayed_work(&chip->work);
- flush_scheduled_work();
+ cancel_delayed_work_sync(&chip->work);
kfree(chip);
}
@@ -141,7 +140,7 @@ void snd_ak4113_reinit(struct ak4113 *chip)
{
chip->init = 1;
mb();
- flush_scheduled_work();
+ flush_delayed_work_sync(&chip->work);
ak4113_init_regs(chip);
/* bring up statistics / event queing */
chip->init = 0;
diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c
index 0341451..d9fb537 100644
--- a/sound/i2c/other/ak4114.c
+++ b/sound/i2c/other/ak4114.c
@@ -67,8 +67,7 @@ static void snd_ak4114_free(struct ak4114 *chip)
{
chip->init = 1; /* don't schedule new work */
mb();
- cancel_delayed_work(&chip->work);
- flush_scheduled_work();
+ cancel_delayed_work_sync(&chip->work);
kfree(chip);
}
@@ -154,7 +153,7 @@ void snd_ak4114_reinit(struct ak4114 *chip)
{
chip->init = 1;
mb();
- flush_scheduled_work();
+ flush_delayed_work_sync(&chip->work);
ak4114_init_regs(chip);
/* bring up statistics / event queing */
chip->init = 0;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index a7630e9e..0fc614c 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1014,8 +1014,7 @@ static int snd_ac97_free(struct snd_ac97 *ac97)
{
if (ac97) {
#ifdef CONFIG_SND_AC97_POWER_SAVE
- cancel_delayed_work(&ac97->power_work);
- flush_scheduled_work();
+ cancel_delayed_work_sync(&ac97->power_work);
#endif
snd_ac97_proc_done(ac97);
if (ac97->bus)
@@ -2456,8 +2455,7 @@ void snd_ac97_suspend(struct snd_ac97 *ac97)
if (ac97->build_ops->suspend)
ac97->build_ops->suspend(ac97);
#ifdef CONFIG_SND_AC97_POWER_SAVE
- cancel_delayed_work(&ac97->power_work);
- flush_scheduled_work();
+ cancel_delayed_work_sync(&ac97->power_work);
#endif
snd_ac97_powerdown(ac97);
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3cfb31e..846d1ea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3111,6 +3111,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b7e2348..f17159d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1614,6 +1614,7 @@ do_sku:
spec->init_amp = ALC_INIT_GPIO3;
break;
case 5:
+ default:
spec->init_amp = ALC_INIT_DEFAULT;
break;
}
@@ -2014,6 +2015,36 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
};
/*
+ *ALC888 Acer Aspire 7730G model
+ */
+
+static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+/* Bias voltage on for external mic port */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Front Mic: set to PIN_IN (empty by default) */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/*Enable internal subwoofer */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x17, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/*
* ALC889 Acer Aspire 8930G model
*/
@@ -2200,6 +2231,16 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[2] = 0x17;
}
+static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x17;
+}
+
static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -9524,13 +9565,6 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
{ }
};
-static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
static void alc888_6st_dell_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -9831,7 +9865,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
- SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
@@ -10328,7 +10361,7 @@ static struct alc_config_preset alc882_presets[] = {
.const_channel_count = 6,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .setup = alc888_acer_aspire_6530g_setup,
+ .setup = alc888_acer_aspire_7730g_setup,
.init_hook = alc_automute_amp,
},
[ALC883_MEDION] = {
@@ -14629,7 +14662,10 @@ static int alc275_setup_dual_adc(struct hda_codec *codec)
/* different alc269-variants */
enum {
ALC269_TYPE_NORMAL,
+ ALC269_TYPE_ALC258,
ALC269_TYPE_ALC259,
+ ALC269_TYPE_ALC269VB,
+ ALC269_TYPE_ALC270,
ALC269_TYPE_ALC271X,
};
@@ -15029,7 +15065,7 @@ static int alc269_fill_coef(struct hda_codec *codec)
static int patch_alc269(struct hda_codec *codec)
{
struct alc_spec *spec;
- int board_config;
+ int board_config, coef;
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -15040,14 +15076,23 @@ static int patch_alc269(struct hda_codec *codec)
alc_auto_parse_customize_define(codec);
- if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){
+ coef = alc_read_coef_idx(codec, 0);
+ if ((coef & 0x00f0) == 0x0010) {
if (codec->bus->pci->subsystem_vendor == 0x1025 &&
spec->cdefine.platform_type == 1) {
alc_codec_rename(codec, "ALC271X");
spec->codec_variant = ALC269_TYPE_ALC271X;
- } else {
+ } else if ((coef & 0xf000) == 0x1000) {
+ spec->codec_variant = ALC269_TYPE_ALC270;
+ } else if ((coef & 0xf000) == 0x2000) {
alc_codec_rename(codec, "ALC259");
spec->codec_variant = ALC269_TYPE_ALC259;
+ } else if ((coef & 0xf000) == 0x3000) {
+ alc_codec_rename(codec, "ALC258");
+ spec->codec_variant = ALC269_TYPE_ALC258;
+ } else {
+ alc_codec_rename(codec, "ALC269VB");
+ spec->codec_variant = ALC269_TYPE_ALC269VB;
}
} else
alc_fix_pll_init(codec, 0x20, 0x04, 15);
@@ -15110,7 +15155,7 @@ static int patch_alc269(struct hda_codec *codec)
spec->stream_digital_capture = &alc269_pcm_digital_capture;
if (!spec->adc_nids) { /* wasn't filled automatically? use default */
- if (spec->codec_variant != ALC269_TYPE_NORMAL) {
+ if (spec->codec_variant == ALC269_TYPE_NORMAL) {
spec->adc_nids = alc269_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
spec->capsrc_nids = alc269_capsrc_nids;
@@ -16904,7 +16949,7 @@ static struct alc_config_preset alc861vd_presets[] = {
static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+ return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x22, 0);
}
@@ -18958,6 +19003,8 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
return 0x02;
else if (nid >= 0x0c && nid <= 0x0e)
return nid - 0x0c + 0x02;
+ else if (nid == 0x26) /* ALC887-VD has this DAC too */
+ return 0x25;
else
return 0;
}
@@ -18966,7 +19013,7 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
hda_nid_t dac)
{
- hda_nid_t mix[4];
+ hda_nid_t mix[5];
int i, num;
num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
@@ -19426,7 +19473,10 @@ static int patch_alc888(struct hda_codec *codec)
{
if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){
kfree(codec->chip_name);
- codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
+ if (codec->vendor_id == 0x10ec0887)
+ codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL);
+ else
+ codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
if (!codec->chip_name) {
alc_free(codec);
return -ENOMEM;
@@ -19916,7 +19966,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
.patch = patch_alc882 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
- { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
+ { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
.patch = patch_alc882 },
{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index cfd73af..efa4225 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1627,6 +1627,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1,
"Alienware M17x", STAC_ALIENWARE_M17X),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
+ "Alienware M17x", STAC_ALIENWARE_M17X),
{} /* terminator */
};
@@ -3491,10 +3493,8 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
return err;
}
- if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) {
+ if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1)
snd_hda_add_imux_item(imux, label, index, NULL);
- spec->num_analog_muxes++;
- }
}
return 0;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index d1c3f8d..7f4852a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -263,8 +263,7 @@ static void vt1708_stop_hp_work(struct via_spec *spec)
return;
snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
!spec->vt1708_jack_detectect);
- cancel_delayed_work(&spec->vt1708_hp_work);
- flush_scheduled_work();
+ cancel_delayed_work_sync(&spec->vt1708_hp_work);
}
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 3078ed6..c44c91e 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -565,7 +565,8 @@ static void oxygen_card_free(struct snd_card *card)
oxygen_shutdown(chip);
if (chip->irq >= 0)
free_irq(chip->irq, chip);
- flush_scheduled_work();
+ flush_work_sync(&chip->spdif_input_bits_work);
+ flush_work_sync(&chip->gpio_work);
chip->model.cleanup(chip);
kfree(chip->model_data);
mutex_destroy(&chip->mutex);
@@ -741,7 +742,8 @@ int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
spin_unlock_irq(&chip->reg_lock);
synchronize_irq(chip->irq);
- flush_scheduled_work();
+ flush_work_sync(&chip->spdif_input_bits_work);
+ flush_work_sync(&chip->gpio_work);
chip->interrupt_mask = saved_interrupt_mask;
pci_disable_device(pci);
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index e720d5e..bee3c94 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -16,7 +16,8 @@ config SND_ATMEL_SOC_SSC
config SND_AT91_SOC_SAM9G20_WM8731
tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
- depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+ depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC && \
+ AT91_PROGRAMMABLE_CLOCKS
select SND_ATMEL_SOC_SSC
select SND_SOC_WM8731
help
@@ -25,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731
config SND_AT32_SOC_PLAYPAQ
tristate "SoC Audio support for PlayPaq with WM8510"
- depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+ depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
select SND_ATMEL_SOC_SSC
select SND_SOC_WM8510
help
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 293569d..e521ada 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -222,9 +222,9 @@ static int __init at91sam9g20ek_init(void)
}
pllb = clk_get(NULL, "pllb");
- if (IS_ERR(mclk)) {
+ if (IS_ERR(pllb)) {
printk(KERN_ERR "ASoC: Failed to get PLLB\n");
- ret = PTR_ERR(mclk);
+ ret = PTR_ERR(pllb);
goto err_mclk;
}
ret = clk_set_parent(mclk, pllb);
@@ -240,6 +240,7 @@ static int __init at91sam9g20ek_init(void)
if (!at91sam9g20ek_snd_device) {
printk(KERN_ERR "ASoC: Platform device allocation failed\n");
ret = -ENOMEM;
+ goto err_mclk;
}
platform_set_drvdata(at91sam9g20ek_snd_device,
@@ -248,11 +249,13 @@ static int __init at91sam9g20ek_init(void)
ret = platform_device_add(at91sam9g20ek_snd_device);
if (ret) {
printk(KERN_ERR "ASoC: Platform device allocation failed\n");
- platform_device_put(at91sam9g20ek_snd_device);
+ goto err_device_add;
}
return ret;
+err_device_add:
+ platform_device_put(at91sam9g20ek_snd_device);
err_mclk:
clk_put(mclk);
mclk = NULL;
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index e3d2835..86e0f85 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -167,7 +167,6 @@ static int __init afeb9260_soc_init(void)
return 0;
err1:
- platform_device_del(afeb9260_snd_device);
platform_device_put(afeb9260_snd_device);
return err;
}
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index bc22ee9..d63e287 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -28,6 +28,11 @@
#include <sound/max98088.h>
#include "max98088.h"
+enum max98088_type {
+ MAX98088,
+ MAX98089,
+};
+
struct max98088_cdata {
unsigned int rate;
unsigned int fmt;
@@ -36,6 +41,7 @@ struct max98088_cdata {
struct max98088_priv {
u8 reg_cache[M98088_REG_CNT];
+ enum max98088_type devtype;
void *control_data;
struct max98088_pdata *pdata;
unsigned int sysclk;
@@ -2013,7 +2019,10 @@ err_access:
static int max98088_remove(struct snd_soc_codec *codec)
{
+ struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
+
max98088_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ kfree(max98088->eq_texts);
return 0;
}
@@ -2040,6 +2049,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
if (max98088 == NULL)
return -ENOMEM;
+ max98088->devtype = id->driver_data;
+
i2c_set_clientdata(i2c, max98088);
max98088->control_data = i2c;
max98088->pdata = i2c->dev.platform_data;
@@ -2059,7 +2070,8 @@ static int __devexit max98088_i2c_remove(struct i2c_client *client)
}
static const struct i2c_device_id max98088_i2c_id[] = {
- { "max98088", 0 },
+ { "max98088", MAX98088 },
+ { "max98089", MAX98089 },
{ }
};
MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 00d67cc..061f9e5 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -383,6 +383,7 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
.reg_cache_size = sizeof(stac9766_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
+ .reg_cache_default = stac9766_reg,
};
static __devinit int stac9766_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index fc68779..77b8f9a 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1176,7 +1176,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_gpio);
int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
{
u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
- u8 val, bit = gpio ? 2: 1;
+ u8 val = 0, bit = gpio ? 2 : 1;
aic3x_read(codec, reg, &val);
return (val >> bit) & 1;
@@ -1204,7 +1204,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
int aic3x_headset_detected(struct snd_soc_codec *codec)
{
- u8 val;
+ u8 val = 0;
aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
return (val >> 4) & 1;
}
@@ -1212,7 +1212,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
int aic3x_button_pressed(struct snd_soc_codec *codec)
{
- u8 val;
+ u8 val = 0;
aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
return (val >> 5) & 1;
}
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index ee4fb20..d2c2430 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -78,8 +78,10 @@ static int tpa6130a2_i2c_write(int reg, u8 value)
if (data->power_state) {
val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value);
- if (val < 0)
+ if (val < 0) {
dev_err(&tpa6130a2_client->dev, "Write failed\n");
+ return val;
+ }
}
/* Either powered on or off, we save the context */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 7540a50..464f0cf 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -597,6 +597,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
.resume = uda134x_soc_resume,
.reg_cache_size = sizeof(uda134x_reg),
.reg_word_size = sizeof(u8),
+ .reg_cache_default = uda134x_reg,
.reg_cache_step = 1,
.read = uda134x_read_reg_cache,
.write = uda134x_write,
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f4f1fba..b3e9fac 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -831,7 +831,7 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
}
/* MCLK direction */
- if (dir == WM8350_MCLK_DIR_OUT)
+ if (dir == SND_SOC_CLOCK_OUT)
wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
WM8350_MCLK_DIR);
else
@@ -1586,6 +1586,13 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec)
wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+ /* Make sure AIF tristating is disabled by default */
+ wm8350_clear_bits(wm8350, WM8350_AI_FORMATING, WM8350_AIF_TRI);
+
+ /* Make sure we've got a sane companding setup too */
+ wm8350_clear_bits(wm8350, WM8350_ADC_DAC_COMP,
+ WM8350_DAC_COMP | WM8350_LOOPBACK);
+
/* Make sure jack detect is disabled to start off with */
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
WM8350_JDL_ENA | WM8350_JDR_ENA);
@@ -1619,7 +1626,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec)
{
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
struct wm8350 *wm8350 = dev_get_platdata(codec->dev);
- int ret;
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
WM8350_JDL_ENA | WM8350_JDR_ENA);
@@ -1634,15 +1640,9 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec)
priv->hpr.jack = NULL;
priv->mic.jack = NULL;
- /* cancel any work waiting to be queued. */
- ret = cancel_delayed_work(&codec->delayed_work);
-
/* if there was any work waiting then we run it now and
* wait for its completion */
- if (ret) {
- schedule_delayed_work(&codec->delayed_work, 0);
- flush_scheduled_work();
- }
+ flush_delayed_work_sync(&codec->delayed_work);
wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 712ef7c..9a433a5 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -146,7 +146,6 @@ static int wm8523_startup(struct snd_pcm_substream *substream,
return -EINVAL;
}
- return 0;
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&wm8523->rate_constraint);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 6313858..e725c09 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -526,7 +526,7 @@ static int wm8731_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0);
/* Disable bypass path by default */
- snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0);
+ snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0);
snd_soc_add_controls(codec, wm8731_snd_controls,
ARRAY_SIZE(wm8731_snd_controls));
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8f679a1..84a2367 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1526,25 +1526,6 @@ static int wm8753_resume(struct snd_soc_codec *codec)
return 0;
}
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
-{
- int ret;
-
- /* cancel any work waiting to be queued. */
- ret = cancel_delayed_work(dwork);
-
- /* if there was any work waiting then we run it now and
- * wait for it's completion */
- if (ret) {
- schedule_delayed_work(dwork, 0);
- flush_scheduled_work();
- }
- return ret;
-}
-
static int wm8753_probe(struct snd_soc_codec *codec)
{
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
@@ -1604,7 +1585,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
/* power down chip */
static int wm8753_remove(struct snd_soc_codec *codec)
{
- run_delayed_work(&codec->delayed_work);
+ flush_delayed_work_sync(&codec->delayed_work);
wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 04182c4..0132a27 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -34,7 +34,6 @@
/* codec private data */
struct wm8776_priv {
enum snd_soc_control_type control_type;
- u16 reg_cache[WM8776_CACHEREGNUM];
int sysclk[2];
};
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 33be84e..fca60a0 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2498,6 +2498,8 @@ static int wm8904_remove(struct snd_soc_codec *codec)
wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
+ kfree(wm8904->retune_mobile_texts);
+ kfree(wm8904->drc_texts);
return 0;
}
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 4f326f6..8340485 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -711,7 +711,7 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream,
if (fs <= 24000)
reg |= WM8961_DACSLOPE;
else
- reg &= WM8961_DACSLOPE;
+ reg &= ~WM8961_DACSLOPE;
snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg);
return 0;
@@ -736,7 +736,7 @@ static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id,
freq /= 2;
} else {
dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq);
- reg &= WM8961_MCLKDIV;
+ reg &= ~WM8961_MCLKDIV;
}
snd_soc_write(codec, WM8961_CLOCKING1, reg);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 894d0cd..e809274 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3500,8 +3500,11 @@ static ssize_t wm8962_beep_set(struct device *dev,
{
struct wm8962_priv *wm8962 = dev_get_drvdata(dev);
long int time;
+ int ret;
- strict_strtol(buf, 10, &time);
+ ret = strict_strtol(buf, 10, &time);
+ if (ret != 0)
+ return ret;
input_event(wm8962->beep, EV_SND, SND_TONE, time);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0db59c3..4d3e6f1 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3903,6 +3903,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
return -ENOMEM;
snd_soc_codec_set_drvdata(codec, wm8994);
+ codec->reg_cache = &wm8994->reg_cache;
+
wm8994->pdata = dev_get_platdata(codec->dev->parent);
wm8994->codec = codec;
@@ -4059,6 +4061,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec)
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
+ kfree(wm8994->retune_mobile_texts);
+ kfree(wm8994->drc_texts);
kfree(wm8994);
return 0;
@@ -4071,6 +4075,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = {
.resume = wm8994_resume,
.read = wm8994_read,
.write = wm8994_write,
+ .readable_register = wm8994_readable,
+ .volatile_register = wm8994_volatile,
.set_bias_level = wm8994_set_bias_level,
};
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 2b07b17..bc9e6b0b 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -157,12 +157,23 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
}
/* davinci-evm digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link evm_dai = {
+static struct snd_soc_dai_link dm6446_evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
- .cpu_dai_name = "davinci-mcasp.0",
+ .cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "tlv320aic3x-hifi",
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-001b",
+ .platform_name = "davinci-pcm-audio",
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link dm355_evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai_name = "davinci-mcbsp.1",
+ .codec_dai_name = "tlv320aic3x-hifi",
+ .codec_name = "tlv320aic3x-codec.1-001b",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
@@ -172,10 +183,10 @@ static struct snd_soc_dai_link dm365_evm_dai = {
#ifdef CONFIG_SND_DM365_AIC3X_CODEC
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
- .cpu_dai_name = "davinci-i2s",
+ .cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "tlv320aic3x-hifi",
.init = evm_aic3x_init,
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-0018",
.ops = &evm_ops,
#elif defined(CONFIG_SND_DM365_VOICE_CODEC)
.name = "Voice Codec - CQ93VC",
@@ -219,10 +230,17 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
.ops = &evm_ops,
};
-/* davinci dm6446, dm355 evm audio machine driver */
-static struct snd_soc_card snd_soc_card_evm = {
- .name = "DaVinci EVM",
- .dai_link = &evm_dai,
+/* davinci dm6446 evm audio machine driver */
+static struct snd_soc_card dm6446_snd_soc_card_evm = {
+ .name = "DaVinci DM6446 EVM",
+ .dai_link = &dm6446_evm_dai,
+ .num_links = 1,
+};
+
+/* davinci dm355 evm audio machine driver */
+static struct snd_soc_card dm355_snd_soc_card_evm = {
+ .name = "DaVinci DM355 EVM",
+ .dai_link = &dm355_evm_dai,
.num_links = 1,
};
@@ -261,10 +279,10 @@ static int __init evm_init(void)
int ret;
if (machine_is_davinci_evm()) {
- evm_snd_dev_data = &snd_soc_card_evm;
+ evm_snd_dev_data = &dm6446_snd_soc_card_evm;
index = 0;
} else if (machine_is_davinci_dm355_evm()) {
- evm_snd_dev_data = &snd_soc_card_evm;
+ evm_snd_dev_data = &dm355_snd_soc_card_evm;
index = 1;
} else if (machine_is_davinci_dm365_evm()) {
evm_snd_dev_data = &dm365_snd_soc_card_evm;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index d46b545..9e0e565 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -426,9 +426,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
snd_pcm_format_t fmt;
unsigned element_cnt = 1;
- dai->capture_dma_data = dev->dma_params;
- dai->playback_dma_data = dev->dma_params;
-
/* general line settings */
spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -601,6 +598,15 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
return ret;
}
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+ return 0;
+}
+
static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -612,6 +618,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+ .startup = davinci_i2s_startup,
.shutdown = davinci_i2s_shutdown,
.prepare = davinci_i2s_prepare,
.trigger = davinci_i2s_trigger,
@@ -749,7 +756,7 @@ static struct platform_driver davinci_mcbsp_driver = {
.probe = davinci_i2s_probe,
.remove = davinci_i2s_remove,
.driver = {
- .name = "davinci-i2s",
+ .name = "davinci-mcbsp",
.owner = THIS_MODULE,
},
};
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 86918ee..fb55d2c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -715,9 +715,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
int word_length;
u8 fifo_level;
- cpu_dai->capture_dma_data = dev->dma_params;
- cpu_dai->playback_dma_data = dev->dma_params;
-
davinci_hw_common_param(dev, substream->stream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
fifo_level = dev->txnumevt;
@@ -799,7 +796,17 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
return ret;
}
+static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+ return 0;
+}
+
static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+ .startup = davinci_mcasp_startup,
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 009b652..6c6666a 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -84,7 +84,7 @@ static struct snd_soc_ops sffsdr_ops = {
static struct snd_soc_dai_link sffsdr_dai = {
.name = "PCM3008", /* Codec name */
.stream_name = "PCM3008 HiFi",
- .cpu_dai_name = "davinci-asp.0",
+ .cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "pcm3008-hifi",
.codec_name = "pcm3008-codec",
.platform_name = "davinci-pcm-audio",
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index ea232f6..9d2afcc 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -97,9 +97,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream,
&davinci_vcif_dev->dma_params[substream->stream];
u32 w;
- dai->capture_dma_data = davinci_vcif_dev->dma_params;
- dai->playback_dma_data = davinci_vcif_dev->dma_params;
-
/* Restart the codec before setup */
davinci_vcif_stop(substream);
davinci_vcif_start(substream);
@@ -174,9 +171,19 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd,
return ret;
}
+static int davinci_vcif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+ return 0;
+}
+
#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000
static struct snd_soc_dai_ops davinci_vcif_dai_ops = {
+ .startup = davinci_vcif_startup,
.trigger = davinci_vcif_trigger,
.hw_params = davinci_vcif_hw_params,
};
@@ -240,7 +247,10 @@ fail:
static int davinci_vcif_remove(struct platform_device *pdev)
{
+ struct davinci_vcif_dev *davinci_vcif_dev = dev_get_drvdata(&pdev->dev);
+
snd_soc_unregister_dai(&pdev->dev);
+ kfree(davinci_vcif_dev);
return 0;
}
diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c
index 4b0d199..2868179 100644
--- a/sound/soc/ep93xx/simone.c
+++ b/sound/soc/ep93xx/simone.c
@@ -54,24 +54,26 @@ static int __init simone_init(void)
ret = platform_device_add(simone_snd_ac97_device);
if (ret)
- goto fail;
+ goto fail1;
simone_snd_device = platform_device_alloc("soc-audio", -1);
if (!simone_snd_device) {
ret = -ENOMEM;
- goto fail;
+ goto fail2;
}
platform_set_drvdata(simone_snd_device, &snd_soc_simone);
ret = platform_device_add(simone_snd_device);
- if (ret) {
- platform_device_put(simone_snd_device);
- goto fail;
- }
+ if (ret)
+ goto fail3;
- return ret;
+ return 0;
-fail:
+fail3:
+ platform_device_put(simone_snd_device);
+fail2:
+ platform_device_del(simone_snd_ac97_device);
+fail1:
platform_device_put(simone_snd_ac97_device);
return ret;
}
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 53251e6..108b5d8 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -76,6 +76,7 @@ static __init int efika_fabric_init(void)
rc = platform_device_add(pdev);
if (rc) {
pr_err("efika_fabric_init: platform_device_add() failed\n");
+ platform_device_put(pdev);
return -ENODEV;
}
return 0;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index dce6b55..f92dca0 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -9,7 +9,6 @@
#include <linux/module.h>
#include <linux/of_device.h>
#include <linux/slab.h>
-#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <sound/soc.h>
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 74ffed4..9018fa5 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -160,7 +160,7 @@ static int __devinit psc_i2s_of_probe(struct platform_device *op,
rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
if (rc != 0) {
pr_err("Failed to register DAI\n");
- return 0;
+ return rc;
}
psc_dma = dev_get_drvdata(&op->dev);
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 0d7dcf1..7d7847a 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -498,6 +498,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "platform device add failed\n");
goto error;
}
+ dev_set_drvdata(&pdev->dev, sound_device);
of_node_put(codec_np);
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 63b9eaa..026b756 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -498,6 +498,7 @@ static int p1022_ds_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "platform device add failed\n");
goto error;
}
+ dev_set_drvdata(&pdev->dev, sound_device);
of_node_put(codec_np);
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 25f27ec..ba4d85e 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -76,6 +76,7 @@ static __init int pcm030_fabric_init(void)
rc = platform_device_add(pdev);
if (rc) {
pr_err("pcm030_fabric_init: platform_device_add() failed\n");
+ platform_device_put(pdev);
return -ENODEV;
}
return 0;
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index b596752..dd4fffd 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -34,8 +34,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
@@ -79,10 +79,10 @@ static struct snd_soc_ops eukrea_tlv320_snd_ops = {
static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
- .codec_dai = "tlv320aic23-hifi",
+ .codec_dai_name = "tlv320aic23-hifi",
.platform_name = "imx-pcm-audio.0",
.codec_name = "tlv320aic23-codec.0-001a",
- .cpu_dai = "imx-ssi.0",
+ .cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
};
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index fd493ee..671ef8d 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -20,6 +20,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
+#include <linux/dmaengine.h>
#include <sound/core.h>
#include <sound/initval.h>
@@ -27,165 +28,146 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <mach/dma-mx1-mx2.h>
+#include <mach/dma.h>
#include "imx-ssi.h"
struct imx_pcm_runtime_data {
- int sg_count;
- struct scatterlist *sg_list;
- int period;
+ int period_bytes;
int periods;
- unsigned long dma_addr;
int dma;
- struct snd_pcm_substream *substream;
unsigned long offset;
unsigned long size;
- unsigned long period_cnt;
void *buf;
int period_time;
+ struct dma_async_tx_descriptor *desc;
+ struct dma_chan *dma_chan;
+ struct imx_dma_data dma_data;
};
-/* Called by the DMA framework when a period has elapsed */
-static void imx_ssi_dma_progression(int channel, void *data,
- struct scatterlist *sg)
+static void audio_dma_irq(void *data)
{
- struct snd_pcm_substream *substream = data;
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- if (!sg)
- return;
-
- runtime = iprtd->substream->runtime;
+ iprtd->offset += iprtd->period_bytes;
+ iprtd->offset %= iprtd->period_bytes * iprtd->periods;
- iprtd->offset = sg->dma_address - runtime->dma_addr;
-
- snd_pcm_period_elapsed(iprtd->substream);
+ snd_pcm_period_elapsed(substream);
}
-static void imx_ssi_dma_callback(int channel, void *data)
+static bool filter(struct dma_chan *chan, void *param)
{
- pr_err("%s shouldn't be called\n", __func__);
-}
+ struct imx_pcm_runtime_data *iprtd = param;
-static void snd_imx_dma_err_callback(int channel, void *data, int err)
-{
- struct snd_pcm_substream *substream = data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct imx_pcm_dma_params *dma_params =
- snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- int ret;
+ if (!imx_dma_is_general_purpose(chan))
+ return false;
- pr_err("DMA timeout on channel %d -%s%s%s%s\n",
- channel,
- err & IMX_DMA_ERR_BURST ? " burst" : "",
- err & IMX_DMA_ERR_REQUEST ? " request" : "",
- err & IMX_DMA_ERR_TRANSFER ? " transfer" : "",
- err & IMX_DMA_ERR_BUFFER ? " buffer" : "");
+ chan->private = &iprtd->dma_data;
- imx_dma_disable(iprtd->dma);
- ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
- IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
- substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- DMA_MODE_WRITE : DMA_MODE_READ);
- if (!ret)
- imx_dma_enable(iprtd->dma);
+ return true;
}
-static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
+static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct imx_pcm_dma_params *dma_params;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+ struct dma_slave_config slave_config;
+ dma_cap_mask_t mask;
+ enum dma_slave_buswidth buswidth;
int ret;
dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
- if (iprtd->dma < 0) {
- pr_err("Failed to claim the audio DMA\n");
- return -ENODEV;
- }
+ iprtd->dma_data.peripheral_type = IMX_DMATYPE_SSI;
+ iprtd->dma_data.priority = DMA_PRIO_HIGH;
+ iprtd->dma_data.dma_request = dma_params->dma;
- ret = imx_dma_setup_handlers(iprtd->dma,
- imx_ssi_dma_callback,
- snd_imx_dma_err_callback, substream);
- if (ret)
- goto out;
+ /* Try to grab a DMA channel */
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+ if (!iprtd->dma_chan)
+ return -EINVAL;
- ret = imx_dma_setup_progression_handler(iprtd->dma,
- imx_ssi_dma_progression);
- if (ret) {
- pr_err("Failed to setup the DMA handler\n");
- goto out;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ case SNDRV_PCM_FORMAT_S24_LE:
+ buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ break;
+ default:
+ return 0;
}
- ret = imx_dma_config_channel(iprtd->dma,
- IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
- IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
- dma_params->dma, 1);
- if (ret < 0) {
- pr_err("Cannot configure DMA channel: %d\n", ret);
- goto out;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ slave_config.direction = DMA_TO_DEVICE;
+ slave_config.dst_addr = dma_params->dma_addr;
+ slave_config.dst_addr_width = buswidth;
+ slave_config.dst_maxburst = dma_params->burstsize;
+ } else {
+ slave_config.direction = DMA_FROM_DEVICE;
+ slave_config.src_addr = dma_params->dma_addr;
+ slave_config.src_addr_width = buswidth;
+ slave_config.src_maxburst = dma_params->burstsize;
}
- imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2);
+ ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config);
+ if (ret)
+ return ret;
return 0;
-out:
- imx_dma_free(iprtd->dma);
- return ret;
}
static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- int i;
unsigned long dma_addr;
+ struct dma_chan *chan;
+ struct imx_pcm_dma_params *dma_params;
+ int ret;
- imx_ssi_dma_alloc(substream);
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ ret = imx_ssi_dma_alloc(substream, params);
+ if (ret)
+ return ret;
+ chan = iprtd->dma_chan;
iprtd->size = params_buffer_bytes(params);
iprtd->periods = params_periods(params);
- iprtd->period = params_period_bytes(params);
+ iprtd->period_bytes = params_period_bytes(params);
iprtd->offset = 0;
iprtd->period_time = HZ / (params_rate(params) /
params_period_size(params));
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- if (iprtd->sg_count != iprtd->periods) {
- kfree(iprtd->sg_list);
-
- iprtd->sg_list = kcalloc(iprtd->periods + 1,
- sizeof(struct scatterlist), GFP_KERNEL);
- if (!iprtd->sg_list)
- return -ENOMEM;
- iprtd->sg_count = iprtd->periods + 1;
- }
-
- sg_init_table(iprtd->sg_list, iprtd->sg_count);
dma_addr = runtime->dma_addr;
- for (i = 0; i < iprtd->periods; i++) {
- iprtd->sg_list[i].page_link = 0;
- iprtd->sg_list[i].offset = 0;
- iprtd->sg_list[i].dma_address = dma_addr;
- iprtd->sg_list[i].length = iprtd->period;
- dma_addr += iprtd->period;
+ iprtd->buf = (unsigned int *)substream->dma_buffer.area;
+
+ iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+ iprtd->period_bytes * iprtd->periods,
+ iprtd->period_bytes,
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ if (!iprtd->desc) {
+ dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+ return -EINVAL;
}
- /* close the loop */
- iprtd->sg_list[iprtd->sg_count - 1].offset = 0;
- iprtd->sg_list[iprtd->sg_count - 1].length = 0;
- iprtd->sg_list[iprtd->sg_count - 1].page_link =
- ((unsigned long) iprtd->sg_list | 0x01) & ~0x02;
+ iprtd->desc->callback = audio_dma_irq;
+ iprtd->desc->callback_param = substream;
+
return 0;
}
@@ -194,41 +176,21 @@ static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- if (iprtd->dma >= 0) {
- imx_dma_free(iprtd->dma);
- iprtd->dma = -EINVAL;
+ if (iprtd->dma_chan) {
+ dma_release_channel(iprtd->dma_chan);
+ iprtd->dma_chan = NULL;
}
- kfree(iprtd->sg_list);
- iprtd->sg_list = NULL;
-
return 0;
}
static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct imx_pcm_dma_params *dma_params;
- struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- int err;
dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- iprtd->substream = substream;
- iprtd->buf = (unsigned int *)substream->dma_buffer.area;
- iprtd->period_cnt = 0;
-
- pr_debug("%s: buf: %p period: %d periods: %d\n",
- __func__, iprtd->buf, iprtd->period, iprtd->periods);
-
- err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
- IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
- substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- DMA_MODE_WRITE : DMA_MODE_READ);
- if (err)
- return err;
-
return 0;
}
@@ -241,14 +203,14 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- imx_dma_enable(iprtd->dma);
+ dmaengine_submit(iprtd->desc);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- imx_dma_disable(iprtd->dma);
+ dmaengine_terminate_all(iprtd->dma_chan);
break;
default:
@@ -263,6 +225,9 @@ static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+ pr_debug("%s: %ld %ld\n", __func__, iprtd->offset,
+ bytes_to_frames(substream->runtime, iprtd->offset));
+
return bytes_to_frames(substream->runtime, iprtd->offset);
}
@@ -279,7 +244,7 @@ static struct snd_pcm_hardware snd_imx_hardware = {
.channels_max = 2,
.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
.period_bytes_min = 128,
- .period_bytes_max = 16 * 1024,
+ .period_bytes_max = 65535, /* Limited by SDMA engine */
.periods_min = 2,
.periods_max = 255,
.fifo_size = 0,
@@ -304,11 +269,23 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
}
snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
+
+ return 0;
+}
+
+static int snd_imx_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+ kfree(iprtd);
+
return 0;
}
static struct snd_pcm_ops imx_pcm_ops = {
.open = snd_imx_open,
+ .close = snd_imx_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_imx_pcm_hw_params,
.hw_free = snd_imx_pcm_hw_free,
@@ -340,7 +317,6 @@ static struct platform_driver imx_pcm_driver = {
.name = "imx-pcm-audio",
.owner = THIS_MODULE,
},
-
.probe = imx_soc_platform_probe,
.remove = __devexit_p(imx_soc_platform_remove),
};
@@ -356,4 +332,3 @@ static void __exit snd_imx_pcm_exit(void)
platform_driver_unregister(&imx_pcm_driver);
}
module_exit(snd_imx_pcm_exit);
-
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index d4bd345..390b6ff 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -439,7 +439,22 @@ void imx_pcm_free(struct snd_pcm *pcm)
}
EXPORT_SYMBOL_GPL(imx_pcm_free);
+static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
+ uint32_t val;
+
+ snd_soc_dai_set_drvdata(dai, ssi);
+
+ val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
+ SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
+ writel(val, ssi->base + SSI_SFCSR);
+
+ return 0;
+}
+
static struct snd_soc_dai_driver imx_ssi_dai = {
+ .probe = imx_ssi_dai_probe,
.playback = {
.channels_min = 2,
.channels_max = 2,
@@ -455,20 +470,6 @@ static struct snd_soc_dai_driver imx_ssi_dai = {
.ops = &imx_ssi_pcm_dai_ops,
};
-static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
-{
- struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
- uint32_t val;
-
- snd_soc_dai_set_drvdata(dai, ssi);
-
- val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
- SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
- writel(val, ssi->base + SSI_SFCSR);
-
- return 0;
-}
-
static struct snd_soc_dai_driver imx_ac97_dai = {
.probe = imx_ssi_dai_probe,
.ac97_control = 1,
@@ -677,9 +678,25 @@ static int imx_ssi_probe(struct platform_device *pdev)
goto failed_register;
}
- ssi->soc_platform_pdev = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
- if (!ssi->soc_platform_pdev)
+ ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
+ if (!ssi->soc_platform_pdev_fiq) {
+ ret = -ENOMEM;
+ goto failed_pdev_fiq_alloc;
+ }
+
+ platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi);
+ ret = platform_device_add(ssi->soc_platform_pdev_fiq);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to add platform device\n");
+ goto failed_pdev_fiq_add;
+ }
+
+ ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id);
+ if (!ssi->soc_platform_pdev) {
+ ret = -ENOMEM;
goto failed_pdev_alloc;
+ }
+
platform_set_drvdata(ssi->soc_platform_pdev, ssi);
ret = platform_device_add(ssi->soc_platform_pdev);
if (ret) {
@@ -692,6 +709,10 @@ static int imx_ssi_probe(struct platform_device *pdev)
failed_pdev_add:
platform_device_put(ssi->soc_platform_pdev);
failed_pdev_alloc:
+ platform_device_del(ssi->soc_platform_pdev_fiq);
+failed_pdev_fiq_add:
+ platform_device_put(ssi->soc_platform_pdev_fiq);
+failed_pdev_fiq_alloc:
snd_soc_unregister_dai(&pdev->dev);
failed_register:
failed_ac97:
@@ -712,8 +733,8 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev)
struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
struct imx_ssi *ssi = platform_get_drvdata(pdev);
- platform_device_del(ssi->soc_platform_pdev);
- platform_device_put(ssi->soc_platform_pdev);
+ platform_device_unregister(ssi->soc_platform_pdev);
+ platform_device_unregister(ssi->soc_platform_pdev_fiq);
snd_soc_unregister_dai(&pdev->dev);
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index 53b780d..a4406a1 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -185,6 +185,9 @@
#define DRV_NAME "imx-ssi"
+#include <linux/dmaengine.h>
+#include <mach/dma.h>
+
struct imx_pcm_dma_params {
int dma;
unsigned long dma_addr;
@@ -212,6 +215,7 @@ struct imx_ssi {
int enabled;
struct platform_device *soc_platform_pdev;
+ struct platform_device *soc_platform_pdev_fiq;
};
struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev,
diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c
index 6a65dd7..9eabc28 100644
--- a/sound/soc/imx/phycore-ac97.c
+++ b/sound/soc/imx/phycore-ac97.c
@@ -20,9 +20,6 @@
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include "../codecs/wm9712.h"
-#include "imx-ssi.h"
-
static struct snd_soc_card imx_phycore;
static struct snd_soc_ops imx_phycore_hifi_ops = {
@@ -41,11 +38,12 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = {
};
static struct snd_soc_card imx_phycore = {
- .name = "PhyCORE-audio",
+ .name = "PhyCORE-ac97-audio",
.dai_link = imx_phycore_dai_ac97,
.num_links = ARRAY_SIZE(imx_phycore_dai_ac97),
};
+static struct platform_device *imx_phycore_snd_ac97_device;
static struct platform_device *imx_phycore_snd_device;
static int __init imx_phycore_init(void)
@@ -56,29 +54,42 @@ static int __init imx_phycore_init(void)
/* return happy. We might run on a totally different machine */
return 0;
- imx_phycore_snd_device = platform_device_alloc("soc-audio", -1);
- if (!imx_phycore_snd_device)
+ imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+ if (!imx_phycore_snd_ac97_device)
return -ENOMEM;
- platform_set_drvdata(imx_phycore_snd_device, &imx_phycore);
- ret = platform_device_add(imx_phycore_snd_device);
+ platform_set_drvdata(imx_phycore_snd_ac97_device, &imx_phycore);
+ ret = platform_device_add(imx_phycore_snd_ac97_device);
+ if (ret)
+ goto fail1;
imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1);
- if (!imx_phycore_snd_device)
- return -ENOMEM;
+ if (!imx_phycore_snd_device) {
+ ret = -ENOMEM;
+ goto fail2;
+ }
ret = platform_device_add(imx_phycore_snd_device);
if (ret) {
printk(KERN_ERR "ASoC: Platform device allocation failed\n");
- platform_device_put(imx_phycore_snd_device);
+ goto fail3;
}
+ return 0;
+
+fail3:
+ platform_device_put(imx_phycore_snd_device);
+fail2:
+ platform_device_del(imx_phycore_snd_ac97_device);
+fail1:
+ platform_device_put(imx_phycore_snd_ac97_device);
return ret;
}
static void __exit imx_phycore_exit(void)
{
platform_device_unregister(imx_phycore_snd_device);
+ platform_device_unregister(imx_phycore_snd_ac97_device);
}
late_initcall(imx_phycore_init);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 293dc74..dac6732 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -49,7 +49,7 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97,
mutex_lock(&ac97_mutex);
val = nuc900_checkready();
- if (!!val) {
+ if (val) {
dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
goto out;
}
@@ -102,7 +102,7 @@ static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
mutex_lock(&ac97_mutex);
tmp = nuc900_checkready();
- if (!!tmp)
+ if (tmp)
dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
/* clear the R_WB bit and write register index */
@@ -149,7 +149,7 @@ static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97)
udelay(100);
val = nuc900_checkready();
- if (!!val)
+ if (val)
dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
mutex_unlock(&ac97_mutex);
@@ -263,8 +263,7 @@ static int nuc900_ac97_trigger(struct snd_pcm_substream *substream,
return ret;
}
-static int nuc900_ac97_probe(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int nuc900_ac97_probe(struct snd_soc_dai *dai)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
unsigned long val;
@@ -284,12 +283,12 @@ static int nuc900_ac97_probe(struct platform_device *pdev,
return 0;
}
-static void nuc900_ac97_remove(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int nuc900_ac97_remove(struct snd_soc_dai *dai)
{
struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
clk_disable(nuc900_audio->clk);
+ return 0;
}
static struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
@@ -313,7 +312,7 @@ static struct snd_soc_dai_driver nuc900_ac97_dai = {
.channels_max = 2,
},
.ops = &nuc900_ac97_dai_ops,
-}
+};
static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
{
@@ -384,7 +383,6 @@ out0:
static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
{
-
snd_soc_unregister_dai(&pdev->dev);
clk_put(nuc900_ac97_data->clk);
@@ -392,6 +390,7 @@ static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
release_mem_region(nuc900_ac97_data->res->start,
resource_size(nuc900_ac97_data->res));
+ kfree(nuc900_ac97_data);
nuc900_ac97_data = NULL;
return 0;
diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h
index aeed8ea..59f7e8e 100644
--- a/sound/soc/nuc900/nuc900-audio.h
+++ b/sound/soc/nuc900/nuc900-audio.h
@@ -110,4 +110,6 @@ struct nuc900_audio {
};
+extern struct nuc900_audio *nuc900_ac97_data;
+
#endif /*end _NUC900_AUDIO_H */
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 195d1ac..8263f56 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -50,12 +50,12 @@ static int nuc900_dma_hw_params(struct snd_pcm_substream *substream,
unsigned long flags;
int ret = 0;
- spin_lock_irqsave(&nuc900_audio->lock, flags);
-
ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
if (ret < 0)
return ret;
+ spin_lock_irqsave(&nuc900_audio->lock, flags);
+
nuc900_audio->substream = substream;
nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr;
nuc900_audio->buffersize[substream->stream] =
@@ -169,6 +169,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
unsigned long flags, val;
+ int ret = 0;
spin_lock_irqsave(&nuc900_audio->lock, flags);
@@ -197,10 +198,10 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
}
spin_unlock_irqrestore(&nuc900_audio->lock, flags);
- return 0;
+ return ret;
}
static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -332,7 +333,7 @@ static struct snd_soc_platform_driver nuc900_soc_platform = {
.ops = &nuc900_dma_ops,
.pcm_new = nuc900_dma_new,
.pcm_free = nuc900_dma_free_dma_buffers,
-}
+};
static int __devinit nuc900_soc_platform_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index d542ea2..a088db6 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -12,8 +12,8 @@ config SND_OMAP_SOC_MCPDM
config SND_OMAP_SOC_N810
tristate "SoC Audio support for Nokia N810"
depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
+ depends on OMAP_MUX
select SND_OMAP_SOC_MCBSP
- select OMAP_MUX
select SND_SOC_TLV320AIC3X
help
Say Y if you want to add support for SoC audio on Nokia N810.
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d211c9f..7e84f24 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -644,15 +644,23 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case OMAP_MCBSP_CLKR_SRC_CLKR:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR);
break;
case OMAP_MCBSP_CLKR_SRC_CLKX:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX);
break;
case OMAP_MCBSP_FSR_SRC_FSR:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR);
break;
case OMAP_MCBSP_FSR_SRC_FSX:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX);
break;
default:
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index dbd9d96..4ee33ce 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -306,6 +306,7 @@ static int __init omap3pandora_soc_init(void)
pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n",
dev_name(&omap3pandora_snd_device->dev),
PTR_ERR(omap3pandora_dac_reg));
+ ret = PTR_ERR(omap3pandora_dac_reg);
goto fail3;
}
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index f0e6625..65ae00e 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -177,7 +177,8 @@ static int __init osk_soc_init(void)
tlv320aic23_mclk = clk_get(dev, "mclk");
if (IS_ERR(tlv320aic23_mclk)) {
printk(KERN_ERR "Could not get mclk clock\n");
- return -ENODEV;
+ err = PTR_ERR(tlv320aic23_mclk);
+ goto err2;
}
/*
@@ -188,7 +189,7 @@ static int __init osk_soc_init(void)
if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
err = -ECANCELED;
- goto err1;
+ goto err3;
}
}
@@ -196,9 +197,12 @@ static int __init osk_soc_init(void)
(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
return 0;
-err1:
+
+err3:
clk_put(tlv320aic23_mclk);
+err2:
platform_device_del(osk_snd_device);
+err1:
platform_device_put(osk_snd_device);
return err;
@@ -207,6 +211,7 @@ err1:
static void __exit osk_soc_exit(void)
{
+ clk_put(tlv320aic23_mclk);
platform_device_unregister(osk_snd_device);
}
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 37f191b..580f485 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,6 +1,7 @@
config SND_PXA2XX_SOC
tristate "SoC Audio for the Intel PXA2xx chip"
depends on ARCH_PXA
+ select SND_ARM
select SND_PXA2XX_LIB
help
Say Y or M if you want to add support for codecs attached to
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 97e9423..f451acd 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -100,8 +100,13 @@ static int corgi_startup(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
corgi_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index b8207ce..5ef0526 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -72,9 +72,13 @@ static int magician_startup(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
magician_ext_control(codec);
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index af84ee9..84edd03 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -77,8 +77,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
poodle_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index f470f36..0b30d7d 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -108,8 +108,13 @@ static int spitz_startup(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
spitz_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 73d0edd..7b983f9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -81,8 +81,13 @@ static int tosa_startup(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
tosa_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 8a6b53c..d85bf8a 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -2,6 +2,7 @@ config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3CXXXX chips"
depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210
select S3C64XX_DMA if ARCH_S3C64XX
+ select S3C2410_DMA if ARCH_S3C2410
help
Say Y or M if you want to add support for codecs attached to
the S3C24XX AC97 or I2S interfaces. You will also need to
diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c
index ffd5cf2..468cc11 100644
--- a/sound/soc/s3c24xx/rx1950_uda1380.c
+++ b/sound/soc/s3c24xx/rx1950_uda1380.c
@@ -50,7 +50,6 @@ static unsigned int rates[] = {
16000,
44100,
48000,
- 88200,
};
static struct snd_pcm_hw_constraint_list hw_rates = {
@@ -130,7 +129,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
};
static struct platform_device *s3c24xx_snd_device;
-static struct clk *xtal;
static int rx1950_startup(struct snd_pcm_substream *substream)
{
@@ -179,10 +177,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
case 44100:
case 88200:
clk_source = S3C24XX_CLKSRC_MPLL;
- fs_mode = S3C2410_IISMOD_256FS;
- div = clk_get_rate(xtal) / (256 * rate);
- if (clk_get_rate(xtal) % (256 * rate) > (128 * rate))
- div++;
+ fs_mode = S3C2410_IISMOD_384FS;
+ div = 1;
break;
default:
printk(KERN_ERR "%s: rate %d is not supported\n",
@@ -210,7 +206,7 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
/* set MCLK division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
- S3C2410_IISMOD_384FS);
+ fs_mode);
if (ret < 0)
return ret;
@@ -295,17 +291,8 @@ static int __init rx1950_init(void)
goto err_plat_add;
}
- xtal = clk_get(&s3c24xx_snd_device->dev, "xtal");
-
- if (IS_ERR(xtal)) {
- ret = PTR_ERR(xtal);
- platform_device_unregister(s3c24xx_snd_device);
- goto err_clk;
- }
-
return 0;
-err_clk:
err_plat_add:
err_plat_alloc:
err_gpio_conf:
@@ -320,7 +307,6 @@ static void __exit rx1950_exit(void)
platform_device_unregister(s3c24xx_snd_device);
snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
hp_jack_gpios);
- clk_put(xtal);
gpio_free(S3C2410_GPA(1));
}
diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c
index f31d22a..c8bd904 100644
--- a/sound/soc/s3c24xx/smdk_spdif.c
+++ b/sound/soc/s3c24xx/smdk_spdif.c
@@ -38,7 +38,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev)
}
mout_epll = clk_get(NULL, "mout_epll");
- if (IS_ERR(fout_epll)) {
+ if (IS_ERR(mout_epll)) {
printk(KERN_WARNING "%s: Cannot find mout_epll.\n",
__func__);
ret = -EINVAL;
@@ -54,7 +54,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev)
}
sclk_spdif = clk_get(NULL, "sclk_spdif");
- if (IS_ERR(fout_epll)) {
+ if (IS_ERR(sclk_spdif)) {
printk(KERN_WARNING "%s: Cannot find sclk_spdif.\n",
__func__);
ret = -EINVAL;
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 8778faa..3052f64 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -434,7 +434,7 @@ static struct snd_soc_dai_driver s6000_i2s_dai = {
.rate_max = 1562500,
},
.ops = &s6000_i2s_dai_ops,
-}
+};
static int __devinit s6000_i2s_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 271fd22..ab3ccae 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -473,7 +473,7 @@ static int s6000_pcm_new(struct snd_card *card,
}
res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED,
- s6000_soc_platform.name, pcm);
+ "s6000-audio", pcm);
if (res) {
printk(KERN_ERR "s6000-pcm couldn't get IRQ\n");
return res;
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 96c05e1..c1244c5 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -167,7 +167,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_sync(codec);
- snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec));
+ snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec));
return 0;
}
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 507e709..4c2404b 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -132,6 +132,8 @@ struct fsi_priv {
struct fsi_stream playback;
struct fsi_stream capture;
+ long rate;
+
u32 mst_ctrl;
};
@@ -854,10 +856,17 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
{
struct fsi_priv *fsi = fsi_get_priv(substream);
int is_play = fsi_is_play(substream);
+ struct fsi_master *master = fsi_get_master(fsi);
+ int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
fsi_irq_disable(fsi, is_play);
fsi_clk_ctrl(fsi, 0);
+ set_rate = master->info->set_rate;
+ if (set_rate && fsi->rate)
+ set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
+ fsi->rate = 0;
+
pm_runtime_put_sync(dai->dev);
}
@@ -891,20 +900,20 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_master *master = fsi_get_master(fsi);
- int (*set_rate)(int is_porta, int rate) = master->info->set_rate;
+ int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
int fsi_ver = master->core->ver;
- int is_play = fsi_is_play(substream);
+ long rate = params_rate(params);
int ret;
- /* if slave mode, set_rate is not needed */
- if (!fsi_is_master_mode(fsi, is_play))
+ set_rate = master->info->set_rate;
+ if (!set_rate)
return 0;
- /* it is error if no set_rate */
- if (!set_rate)
- return -EIO;
+ ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
+ if (ret < 0) /* error */
+ return ret;
- ret = set_rate(fsi_is_port_a(fsi), params_rate(params));
+ fsi->rate = rate;
if (ret > 0) {
u32 data = 0;
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 40bbdf1..05192d9 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -387,7 +387,7 @@ static int __devinit sh4_soc_dai_probe(struct platform_device *pdev)
static int __devexit sh4_soc_dai_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai));
+ snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai));
return 0;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 614a8b3..b54ea9a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -67,25 +67,6 @@ static int pmdown_time = 5000;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
-{
- int ret;
-
- /* cancel any work waiting to be queued. */
- ret = cancel_delayed_work(dwork);
-
- /* if there was any work waiting then we run it now and
- * wait for it's completion */
- if (ret) {
- schedule_delayed_work(dwork, 0);
- flush_scheduled_work();
- }
- return ret;
-}
-
/* codec register dump */
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
{
@@ -1016,7 +997,7 @@ static int soc_suspend(struct device *dev)
/* close any waiting streams and save state */
for (i = 0; i < card->num_rtd; i++) {
- run_delayed_work(&card->rtd[i].delayed_work);
+ flush_delayed_work_sync(&card->rtd[i].delayed_work);
card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level;
}
@@ -1687,7 +1668,7 @@ static int soc_remove(struct platform_device *pdev)
/* make sure any delayed work runs */
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
- run_delayed_work(&rtd->delayed_work);
+ flush_delayed_work_sync(&rtd->delayed_work);
}
/* remove and free each DAI */
@@ -1718,7 +1699,7 @@ static int soc_poweroff(struct device *dev)
* now, we're shutting down so no imminent restart. */
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
- run_delayed_work(&rtd->delayed_work);
+ flush_delayed_work_sync(&rtd->delayed_work);
}
snd_soc_dapm_shutdown(card);
@@ -3043,8 +3024,10 @@ int snd_soc_register_dais(struct device *dev,
for (i = 0; i < count; i++) {
dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL);
- if (dai == NULL)
- return -ENOMEM;
+ if (dai == NULL) {
+ ret = -ENOMEM;
+ goto err;
+ }
/* create DAI component name */
dai->name = fmt_multiple_name(dev, &dai_drv[i]);
@@ -3263,9 +3246,6 @@ int snd_soc_register_codec(struct device *dev,
return 0;
error:
- for (i--; i >= 0; i--)
- snd_soc_unregister_dai(dev);
-
if (codec->reg_cache)
kfree(codec->reg_cache);
kfree(codec->name);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7d85c64..75ed649 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -683,12 +683,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
struct snd_soc_dapm_widget *b,
int sort[])
{
- if (a->codec != b->codec)
- return (unsigned long)a - (unsigned long)b;
if (sort[a->id] != sort[b->id])
return sort[a->id] - sort[b->id];
if (a->reg != b->reg)
return a->reg - b->reg;
+ if (a->codec != b->codec)
+ return (unsigned long)a->codec - (unsigned long)b->codec;
return 0;
}
diff --git a/sound/sound_core.c b/sound/sound_core.c
index c03bbae..5580ace 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -104,7 +104,6 @@ module_exit(cleanup_soundcore);
#include <linux/init.h>
#include <linux/slab.h>
-#include <linux/smp_lock.h>
#include <linux/types.h>
#include <linux/kernel.h>
#include <linux/sound.h>
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