Merge branch 'for-linus' into for-next
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index ce55c0a..4da41bf 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -30,6 +30,8 @@
"fsl,imx-audio-sgtl5000"
(compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
+ "fsl,imx-audio-wm8960"
+
Required properties:
- compatible : Contains one of entries in the compatible list.
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index e2b712c..c21c38c 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -343,7 +343,7 @@
void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus);
void snd_hdac_bus_update_rirb(struct hdac_bus *bus);
-void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
+int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
void (*ack)(struct hdac_bus *,
struct hdac_stream *));
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index fadd3eb..9106d8e 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -74,6 +74,18 @@
static DEFINE_RWLOCK(snd_pcm_link_rwlock);
static DECLARE_RWSEM(snd_pcm_link_rwsem);
+/* Writer in rwsem may block readers even during its waiting in queue,
+ * and this may lead to a deadlock when the code path takes read sem
+ * twice (e.g. one in snd_pcm_action_nonatomic() and another in
+ * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to
+ * spin until it gets the lock.
+ */
+static inline void down_write_nonblock(struct rw_semaphore *lock)
+{
+ while (!down_write_trylock(lock))
+ cond_resched();
+}
+
/**
* snd_pcm_stream_lock - Lock the PCM stream
* @substream: PCM substream
@@ -1813,7 +1825,7 @@
res = -ENOMEM;
goto _nolock;
}
- down_write(&snd_pcm_link_rwsem);
+ down_write_nonblock(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
substream->runtime->status->state != substream1->runtime->status->state ||
@@ -1860,7 +1872,7 @@
struct snd_pcm_substream *s;
int res = 0;
- down_write(&snd_pcm_link_rwsem);
+ down_write_nonblock(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (!snd_pcm_stream_linked(substream)) {
res = -EALREADY;
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index 8010766..c850345 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -383,15 +383,20 @@
if (snd_BUG_ON(!pool))
return -EINVAL;
- if (pool->ptr) /* should be atomic? */
- return 0;
- pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size);
- if (!pool->ptr)
+ cellptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size);
+ if (!cellptr)
return -ENOMEM;
/* add new cells to the free cell list */
spin_lock_irqsave(&pool->lock, flags);
+ if (pool->ptr) {
+ spin_unlock_irqrestore(&pool->lock, flags);
+ vfree(cellptr);
+ return 0;
+ }
+
+ pool->ptr = cellptr;
pool->free = NULL;
for (cell = 0; cell < pool->size; cell++) {
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index 921fb2b..fe686ee 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -535,19 +535,22 @@
bool is_src, bool ack)
{
struct snd_seq_port_subs_info *grp;
+ struct list_head *list;
+ bool empty;
grp = is_src ? &port->c_src : &port->c_dest;
+ list = is_src ? &subs->src_list : &subs->dest_list;
down_write(&grp->list_mutex);
write_lock_irq(&grp->list_lock);
- if (is_src)
- list_del(&subs->src_list);
- else
- list_del(&subs->dest_list);
+ empty = list_empty(list);
+ if (!empty)
+ list_del_init(list);
grp->exclusive = 0;
write_unlock_irq(&grp->list_lock);
up_write(&grp->list_mutex);
- unsubscribe_port(client, port, grp, &subs->info, ack);
+ if (!empty)
+ unsubscribe_port(client, port, grp, &subs->info, ack);
}
/* connect two ports */
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index b5a17cb..8c48623 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -426,18 +426,22 @@
* @bus: HD-audio core bus
* @status: INTSTS register value
* @ask: callback to be called for woken streams
+ *
+ * Returns the bits of handled streams, or zero if no stream is handled.
*/
-void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
+int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
void (*ack)(struct hdac_bus *,
struct hdac_stream *))
{
struct hdac_stream *azx_dev;
u8 sd_status;
+ int handled = 0;
list_for_each_entry(azx_dev, &bus->stream_list, list) {
if (status & azx_dev->sd_int_sta_mask) {
sd_status = snd_hdac_stream_readb(azx_dev, SD_STS);
snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK);
+ handled |= 1 << azx_dev->index;
if (!azx_dev->substream || !azx_dev->running ||
!(sd_status & SD_INT_COMPLETE))
continue;
@@ -445,6 +449,7 @@
ack(bus, azx_dev);
}
}
+ return handled;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_handle_stream_irq);
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 37cf9ce..27de801 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -930,6 +930,8 @@
struct azx *chip = dev_id;
struct hdac_bus *bus = azx_bus(chip);
u32 status;
+ bool active, handled = false;
+ int repeat = 0; /* count for avoiding endless loop */
#ifdef CONFIG_PM
if (azx_has_pm_runtime(chip))
@@ -939,33 +941,36 @@
spin_lock(&bus->reg_lock);
- if (chip->disabled) {
- spin_unlock(&bus->reg_lock);
- return IRQ_NONE;
- }
+ if (chip->disabled)
+ goto unlock;
- status = azx_readl(chip, INTSTS);
- if (status == 0 || status == 0xffffffff) {
- spin_unlock(&bus->reg_lock);
- return IRQ_NONE;
- }
+ do {
+ status = azx_readl(chip, INTSTS);
+ if (status == 0 || status == 0xffffffff)
+ break;
- snd_hdac_bus_handle_stream_irq(bus, status, stream_update);
+ handled = true;
+ active = false;
+ if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update))
+ active = true;
- /* clear rirb int */
- status = azx_readb(chip, RIRBSTS);
- if (status & RIRB_INT_MASK) {
- if (status & RIRB_INT_RESPONSE) {
- if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
- udelay(80);
- snd_hdac_bus_update_rirb(bus);
+ /* clear rirb int */
+ status = azx_readb(chip, RIRBSTS);
+ if (status & RIRB_INT_MASK) {
+ active = true;
+ if (status & RIRB_INT_RESPONSE) {
+ if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
+ udelay(80);
+ snd_hdac_bus_update_rirb(bus);
+ }
+ azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
}
- azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
- }
+ } while (active && ++repeat < 10);
+ unlock:
spin_unlock(&bus->reg_lock);
- return IRQ_HANDLED;
+ return IRQ_RETVAL(handled);
}
EXPORT_SYMBOL_GPL(azx_interrupt);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4179710..2624cfe 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -363,7 +363,10 @@
((pci)->device == 0x0d0c) || \
((pci)->device == 0x160c))
-#define IS_BROXTON(pci) ((pci)->device == 0x5a98)
+#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
+#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
+#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
+#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci))
static char *driver_short_names[] = {
[AZX_DRIVER_ICH] = "HDA Intel",
@@ -540,13 +543,13 @@
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
snd_hdac_set_codec_wakeup(bus, true);
- if (IS_BROXTON(pci)) {
+ if (IS_SKL_PLUS(pci)) {
pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
val = val & ~INTEL_HDA_CGCTL_MISCBDCGE;
pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
}
azx_init_chip(chip, full_reset);
- if (IS_BROXTON(pci)) {
+ if (IS_SKL_PLUS(pci)) {
pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
val = val | INTEL_HDA_CGCTL_MISCBDCGE;
pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
@@ -555,7 +558,7 @@
snd_hdac_set_codec_wakeup(bus, false);
/* reduce dma latency to avoid noise */
- if (IS_BROXTON(pci))
+ if (IS_BXT(pci))
bxt_reduce_dma_latency(chip);
}
@@ -977,11 +980,6 @@
/* put codec down to D3 at hibernation for Intel SKL+;
* otherwise BIOS may still access the codec and screw up the driver
*/
-#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
-#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
-#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
-#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci))
-
static int azx_freeze_noirq(struct device *dev)
{
struct pci_dev *pci = to_pci_dev(dev);
@@ -2168,10 +2166,10 @@
struct hda_intel *hda;
if (card) {
- /* flush the pending probing work */
+ /* cancel the pending probing work */
chip = card->private_data;
hda = container_of(chip, struct hda_intel, chip);
- flush_work(&hda->probe_work);
+ cancel_work_sync(&hda->probe_work);
snd_card_free(card);
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index efd4980..1f357cd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3801,6 +3801,10 @@
static void alc_headset_mode_default(struct hda_codec *codec)
{
+ static struct coef_fw coef0225[] = {
+ UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10),
+ {}
+ };
static struct coef_fw coef0255[] = {
WRITE_COEF(0x45, 0xc089),
WRITE_COEF(0x45, 0xc489),
@@ -3842,6 +3846,9 @@
};
switch (codec->core.vendor_id) {
+ case 0x10ec0225:
+ alc_process_coef_fw(codec, coef0225);
+ break;
case 0x10ec0255:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0255);
@@ -4749,6 +4756,9 @@
ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
ALC293_FIXUP_LENOVO_SPK_NOISE,
ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
+ ALC255_FIXUP_DELL_SPK_NOISE,
+ ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC280_FIXUP_HP_HEADSET_MIC,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5368,6 +5378,29 @@
.type = HDA_FIXUP_FUNC,
.v.func = alc233_fixup_lenovo_line2_mic_hotkey,
},
+ [ALC255_FIXUP_DELL_SPK_NOISE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_disable_aamix,
+ .chained = true,
+ .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
+ },
+ [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Disable pass-through path for FRONT 14h */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 },
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+ },
+ [ALC280_FIXUP_HP_HEADSET_MIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_disable_aamix,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5410,6 +5443,7 @@
SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
+ SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE),
SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5470,6 +5504,7 @@
SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -5638,10 +5673,10 @@
{0x21, 0x03211020}
static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
- SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+ SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC225_STANDARD_PINS,
{0x14, 0x901701a0}),
- SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+ SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC225_STANDARD_PINS,
{0x14, 0x901701b0}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 3191e0a..d1fb035 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -635,6 +635,7 @@
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0) {
dev_err(prtd->platform->dev, "set integer constraint failed\n");
+ kfree(adata);
return ret;
}
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 33143fe..9178531 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1929,6 +1929,25 @@
{ 1000000, 13500000, 0, 1 },
};
+static const unsigned int pseudo_fref_max[ARIZONA_FLL_MAX_FRATIO] = {
+ 13500000,
+ 6144000,
+ 6144000,
+ 3072000,
+ 3072000,
+ 2822400,
+ 2822400,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 768000,
+};
+
static struct {
unsigned int min;
unsigned int max;
@@ -2042,16 +2061,32 @@
/* Adjust FRATIO/refdiv to avoid integer mode if possible */
refdiv = cfg->refdiv;
+ arizona_fll_dbg(fll, "pseudo: initial ratio=%u fref=%u refdiv=%u\n",
+ init_ratio, Fref, refdiv);
+
while (div <= ARIZONA_FLL_MAX_REFDIV) {
for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
ratio++) {
if ((ARIZONA_FLL_VCO_CORNER / 2) /
- (fll->vco_mult * ratio) < Fref)
+ (fll->vco_mult * ratio) < Fref) {
+ arizona_fll_dbg(fll, "pseudo: hit VCO corner\n");
break;
+ }
+
+ if (Fref > pseudo_fref_max[ratio - 1]) {
+ arizona_fll_dbg(fll,
+ "pseudo: exceeded max fref(%u) for ratio=%u\n",
+ pseudo_fref_max[ratio - 1],
+ ratio);
+ break;
+ }
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
+ arizona_fll_dbg(fll,
+ "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+ Fref, refdiv, div, ratio);
return ratio;
}
}
@@ -2060,6 +2095,9 @@
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
+ arizona_fll_dbg(fll,
+ "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+ Fref, refdiv, div, ratio);
return ratio;
}
}
@@ -2068,6 +2106,9 @@
Fref /= 2;
refdiv++;
init_ratio = arizona_find_fratio(Fref, NULL);
+ arizona_fll_dbg(fll,
+ "pseudo: change fref=%u refdiv=%d(%d) ratio=%u\n",
+ Fref, refdiv, div, init_ratio);
}
arizona_fll_warn(fll, "Falling back to integer mode operation\n");
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index bc08f0c..1bd3164 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -266,6 +266,8 @@
} else {
*mic = false;
regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20);
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0x0400, 0x0000);
}
} else {
regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
@@ -470,24 +472,6 @@
return 0;
}
-static int rt286_vref_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-
- switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
- snd_soc_update_bits(codec,
- RT286_CBJ_CTRL1, 0x0400, 0x0000);
- mdelay(50);
- break;
- default:
- return 0;
- }
-
- return 0;
-}
-
static int rt286_ldo2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -536,7 +520,7 @@
SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1,
12, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1,
- 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU),
+ 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2,
2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1,
@@ -911,8 +895,6 @@
case SND_SOC_BIAS_ON:
mdelay(10);
snd_soc_update_bits(codec,
- RT286_CBJ_CTRL1, 0x0400, 0x0400);
- snd_soc_update_bits(codec,
RT286_DC_GAIN, 0x200, 0x0);
break;
@@ -920,8 +902,6 @@
case SND_SOC_BIAS_STANDBY:
snd_soc_write(codec,
RT286_SET_AUDIO_POWER, AC_PWRST_D3);
- snd_soc_update_bits(codec,
- RT286_CBJ_CTRL1, 0x0400, 0x0000);
break;
default:
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index c61d38b..93e8c90 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -776,7 +776,7 @@
/* IN1/IN2 Control */
SOC_SINGLE_TLV("IN1 Boost", RT5645_IN1_CTRL1,
- RT5645_BST_SFT1, 8, 0, bst_tlv),
+ RT5645_BST_SFT1, 12, 0, bst_tlv),
SOC_SINGLE_TLV("IN2 Boost", RT5645_IN2_CTRL,
RT5645_BST_SFT2, 8, 0, bst_tlv),
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 820d8fa..fb8ea05 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3985,7 +3985,6 @@
if (rt5659 == NULL)
return -ENOMEM;
- rt5659->i2c = i2c;
i2c_set_clientdata(i2c, rt5659);
if (pdata)
@@ -4157,24 +4156,17 @@
INIT_DELAYED_WORK(&rt5659->jack_detect_work, rt5659_jack_detect_work);
- if (rt5659->i2c->irq) {
- ret = request_threaded_irq(rt5659->i2c->irq, NULL, rt5659_irq,
- IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+ if (i2c->irq) {
+ ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
+ rt5659_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
| IRQF_ONESHOT, "rt5659", rt5659);
if (ret)
dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
}
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659,
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659,
rt5659_dai, ARRAY_SIZE(rt5659_dai));
-
- if (ret) {
- if (rt5659->i2c->irq)
- free_irq(rt5659->i2c->irq, rt5659);
- }
-
- return 0;
}
static int rt5659_i2c_remove(struct i2c_client *i2c)
@@ -4191,24 +4183,29 @@
regmap_write(rt5659->regmap, RT5659_RESET, 0);
}
+#ifdef CONFIG_OF
static const struct of_device_id rt5659_of_match[] = {
{ .compatible = "realtek,rt5658", },
{ .compatible = "realtek,rt5659", },
- {},
+ { },
};
+MODULE_DEVICE_TABLE(of, rt5659_of_match);
+#endif
+#ifdef CONFIG_ACPI
static struct acpi_device_id rt5659_acpi_match[] = {
- { "10EC5658", 0},
- { "10EC5659", 0},
- { },
+ { "10EC5658", 0, },
+ { "10EC5659", 0, },
+ { },
};
MODULE_DEVICE_TABLE(acpi, rt5659_acpi_match);
+#endif
struct i2c_driver rt5659_i2c_driver = {
.driver = {
.name = "rt5659",
.owner = THIS_MODULE,
- .of_match_table = rt5659_of_match,
+ .of_match_table = of_match_ptr(rt5659_of_match),
.acpi_match_table = ACPI_PTR(rt5659_acpi_match),
},
.probe = rt5659_i2c_probe,
diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h
index 8f07ee9..d31c9e5 100644
--- a/sound/soc/codecs/rt5659.h
+++ b/sound/soc/codecs/rt5659.h
@@ -1792,7 +1792,6 @@
struct snd_soc_codec *codec;
struct rt5659_platform_data pdata;
struct regmap *regmap;
- struct i2c_client *i2c;
struct gpio_desc *gpiod_ldo1_en;
struct gpio_desc *gpiod_reset;
struct snd_soc_jack *hs_jack;
diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c
index 21ca3a5..d374c18 100644
--- a/sound/soc/codecs/sigmadsp-i2c.c
+++ b/sound/soc/codecs/sigmadsp-i2c.c
@@ -31,7 +31,10 @@
kfree(buf);
- return ret;
+ if (ret < 0)
+ return ret;
+
+ return 0;
}
static int sigmadsp_read_i2c(void *control_data,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 6088d30..97c0f1e 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2382,6 +2382,7 @@
static int wm5110_remove(struct platform_device *pdev)
{
+ snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index ff23772..d7f444f 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -240,13 +240,13 @@
SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
7, 1, 1),
-SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
- WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
-SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
- WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
- WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
+ WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
+ WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
+ WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
WM8960_RINPATH, 4, 3, 0, micboost_tlv),
@@ -643,29 +643,31 @@
return -EINVAL;
}
- /* check if the sysclk frequency is available. */
- for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
- if (sysclk_divs[i] == -1)
- continue;
- sysclk = freq_out / sysclk_divs[i];
- for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
- if (sysclk == dac_divs[j] * lrclk) {
+ if (wm8960->clk_id != WM8960_SYSCLK_PLL) {
+ /* check if the sysclk frequency is available. */
+ for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
+ if (sysclk_divs[i] == -1)
+ continue;
+ sysclk = freq_out / sysclk_divs[i];
+ for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
+ if (sysclk != dac_divs[j] * lrclk)
+ continue;
for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k)
if (sysclk == bclk * bclk_divs[k] / 10)
break;
if (k != ARRAY_SIZE(bclk_divs))
break;
}
+ if (j != ARRAY_SIZE(dac_divs))
+ break;
}
- if (j != ARRAY_SIZE(dac_divs))
- break;
- }
- if (i != ARRAY_SIZE(sysclk_divs)) {
- goto configure_clock;
- } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) {
- dev_err(codec->dev, "failed to configure clock\n");
- return -EINVAL;
+ if (i != ARRAY_SIZE(sysclk_divs)) {
+ goto configure_clock;
+ } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) {
+ dev_err(codec->dev, "failed to configure clock\n");
+ return -EINVAL;
+ }
}
/* get a available pll out frequency and set pll */
for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index ce664c2..bff258d 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -645,6 +645,8 @@
dev->dev = &pdev->dev;
+ dev->i2s_reg_comp1 = I2S_COMP_PARAM_1;
+ dev->i2s_reg_comp2 = I2S_COMP_PARAM_2;
if (pdata) {
dev->capability = pdata->cap;
clk_id = NULL;
@@ -652,9 +654,6 @@
if (dev->quirks & DW_I2S_QUIRK_COMP_REG_OFFSET) {
dev->i2s_reg_comp1 = pdata->i2s_reg_comp1;
dev->i2s_reg_comp2 = pdata->i2s_reg_comp2;
- } else {
- dev->i2s_reg_comp1 = I2S_COMP_PARAM_1;
- dev->i2s_reg_comp2 = I2S_COMP_PARAM_2;
}
ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
} else {
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 40dfd8a..ed8de10 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -112,20 +112,6 @@
struct fsl_ssi_reg_val tx;
};
-static const struct reg_default fsl_ssi_reg_defaults[] = {
- {CCSR_SSI_SCR, 0x00000000},
- {CCSR_SSI_SIER, 0x00003003},
- {CCSR_SSI_STCR, 0x00000200},
- {CCSR_SSI_SRCR, 0x00000200},
- {CCSR_SSI_STCCR, 0x00040000},
- {CCSR_SSI_SRCCR, 0x00040000},
- {CCSR_SSI_SACNT, 0x00000000},
- {CCSR_SSI_STMSK, 0x00000000},
- {CCSR_SSI_SRMSK, 0x00000000},
- {CCSR_SSI_SACCEN, 0x00000000},
- {CCSR_SSI_SACCDIS, 0x00000000},
-};
-
static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -190,8 +176,7 @@
.val_bits = 32,
.reg_stride = 4,
.val_format_endian = REGMAP_ENDIAN_NATIVE,
- .reg_defaults = fsl_ssi_reg_defaults,
- .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults),
+ .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1,
.readable_reg = fsl_ssi_readable_reg,
.volatile_reg = fsl_ssi_volatile_reg,
.precious_reg = fsl_ssi_precious_reg,
@@ -201,6 +186,7 @@
struct fsl_ssi_soc_data {
bool imx;
+ bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */
bool offline_config;
u32 sisr_write_mask;
};
@@ -303,6 +289,7 @@
static struct fsl_ssi_soc_data fsl_ssi_imx21 = {
.imx = true,
+ .imx21regs = true,
.offline_config = true,
.sisr_write_mask = 0,
};
@@ -586,8 +573,12 @@
*/
regmap_write(regs, CCSR_SSI_SACNT,
CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV);
- regmap_write(regs, CCSR_SSI_SACCDIS, 0xff);
- regmap_write(regs, CCSR_SSI_SACCEN, 0x300);
+
+ /* no SACC{ST,EN,DIS} regs on imx21-class SSI */
+ if (!ssi_private->soc->imx21regs) {
+ regmap_write(regs, CCSR_SSI_SACCDIS, 0xff);
+ regmap_write(regs, CCSR_SSI_SACCEN, 0x300);
+ }
/*
* Enable SSI, Transmit and Receive. AC97 has to communicate with the
@@ -1397,6 +1388,7 @@
struct resource *res;
void __iomem *iomem;
char name[64];
+ struct regmap_config regconfig = fsl_ssi_regconfig;
of_id = of_match_device(fsl_ssi_ids, &pdev->dev);
if (!of_id || !of_id->data)
@@ -1444,15 +1436,25 @@
return PTR_ERR(iomem);
ssi_private->ssi_phys = res->start;
+ if (ssi_private->soc->imx21regs) {
+ /*
+ * According to datasheet imx21-class SSI
+ * don't have SACC{ST,EN,DIS} regs.
+ */
+ regconfig.max_register = CCSR_SSI_SRMSK;
+ regconfig.num_reg_defaults_raw =
+ CCSR_SSI_SRMSK / sizeof(uint32_t) + 1;
+ }
+
ret = of_property_match_string(np, "clock-names", "ipg");
if (ret < 0) {
ssi_private->has_ipg_clk_name = false;
ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
- &fsl_ssi_regconfig);
+ ®config);
} else {
ssi_private->has_ipg_clk_name = true;
ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev,
- "ipg", iomem, &fsl_ssi_regconfig);
+ "ipg", iomem, ®config);
}
if (IS_ERR(ssi_private->regs)) {
dev_err(&pdev->dev, "Failed to init register map\n");
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index a407e83..fb896b2 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -72,8 +72,6 @@
goto end;
}
- platform_set_drvdata(pdev, data);
-
end:
of_node_put(spdif_np);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 1ded881..2389ab4 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -99,7 +99,7 @@
if (ret && ret != -ENOTSUPP)
goto err;
}
-
+ return 0;
err:
return ret;
}
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 803f95e..7d7c872 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -30,11 +30,15 @@
config SND_SOC_INTEL_SST
tristate
select SND_SOC_INTEL_SST_ACPI if ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
depends on (X86 || COMPILE_TEST)
config SND_SOC_INTEL_SST_ACPI
tristate
+config SND_SOC_INTEL_SST_MATCH
+ tristate
+
config SND_SOC_INTEL_HASWELL
tristate
@@ -57,7 +61,7 @@
config SND_SOC_INTEL_BYT_RT5640_MACH
tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
depends on X86_INTEL_LPSS && I2C
- depends on DW_DMAC_CORE=y && (SND_SOC_INTEL_BYTCR_RT5640_MACH = n)
+ depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n)
select SND_SOC_INTEL_SST
select SND_SOC_INTEL_BAYTRAIL
select SND_SOC_RT5640
@@ -69,7 +73,7 @@
config SND_SOC_INTEL_BYT_MAX98090_MACH
tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec"
depends on X86_INTEL_LPSS && I2C
- depends on DW_DMAC_CORE=y
+ depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n)
select SND_SOC_INTEL_SST
select SND_SOC_INTEL_BAYTRAIL
select SND_SOC_MAX98090
@@ -97,6 +101,7 @@
select SND_SOC_RT5640
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5640 audio codec.
@@ -109,6 +114,7 @@
select SND_SOC_RT5651
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5651 audio codec.
@@ -121,6 +127,7 @@
select SND_SOC_RT5670
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5672 audio codec.
@@ -133,6 +140,7 @@
select SND_SOC_RT5645
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5645/5650 audio codec.
@@ -145,6 +153,7 @@
select SND_SOC_TS3A227E
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with MAX98090 audio codec it also can support TI jack chip as aux device.
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 55c33dc..52ed434 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -528,6 +528,7 @@
.ops = &sst_compr_dai_ops,
.playback = {
.stream_name = "Compress Playback",
+ .channels_min = 1,
},
},
/* BE CPU Dais */
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 7396ddb..2cbcbe4 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -212,7 +212,10 @@
{
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
- channels->min = channels->max = 4;
+ if (params_channels(params) == 2)
+ channels->min = channels->max = 2;
+ else
+ channels->min = channels->max = 4;
return 0;
}
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 668fdee..fbbb25c 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -1,13 +1,10 @@
snd-soc-sst-dsp-objs := sst-dsp.o
-ifneq ($(CONFIG_SND_SST_IPC_ACPI),)
-snd-soc-sst-acpi-objs := sst-match-acpi.o
-else
-snd-soc-sst-acpi-objs := sst-acpi.o sst-match-acpi.o
-endif
-
+snd-soc-sst-acpi-objs := sst-acpi.o
+snd-soc-sst-match-objs := sst-match-acpi.o
snd-soc-sst-ipc-objs := sst-ipc.o
snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
+obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o
diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c
index 7a85c57..2c5eda1 100644
--- a/sound/soc/intel/common/sst-acpi.c
+++ b/sound/soc/intel/common/sst-acpi.c
@@ -215,6 +215,7 @@
.dma_size = SST_LPT_DSP_DMA_SIZE,
};
+#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI)
static struct sst_acpi_mach baytrail_machines[] = {
{ "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL },
{ "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL },
@@ -231,11 +232,14 @@
.sst_id = SST_DEV_ID_BYT,
.resindex_dma_base = -1,
};
+#endif
static const struct acpi_device_id sst_acpi_match[] = {
{ "INT33C8", (unsigned long)&sst_acpi_haswell_desc },
{ "INT3438", (unsigned long)&sst_acpi_broadwell_desc },
+#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI)
{ "80860F28", (unsigned long)&sst_acpi_baytrail_desc },
+#endif
{ }
};
MODULE_DEVICE_TABLE(acpi, sst_acpi_match);
diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/intel/common/sst-match-acpi.c
index dd077e1..3b4539d 100644
--- a/sound/soc/intel/common/sst-match-acpi.c
+++ b/sound/soc/intel/common/sst-match-acpi.c
@@ -41,3 +41,6 @@
return NULL;
}
EXPORT_SYMBOL_GPL(sst_acpi_find_machine);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index de6dac4..4629372 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -688,14 +688,14 @@
/* get src queue index */
src_index = skl_get_queue_index(src_mcfg->m_out_pin, dst_id, out_max);
if (src_index < 0)
- return -EINVAL;
+ return 0;
msg.src_queue = src_index;
/* get dst queue index */
dst_index = skl_get_queue_index(dst_mcfg->m_in_pin, src_id, in_max);
if (dst_index < 0)
- return -EINVAL;
+ return 0;
msg.dst_queue = dst_index;
@@ -747,7 +747,7 @@
skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
- if (src_mcfg->m_state < SKL_MODULE_INIT_DONE &&
+ if (src_mcfg->m_state < SKL_MODULE_INIT_DONE ||
dst_mcfg->m_state < SKL_MODULE_INIT_DONE)
return 0;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index f355325..b6e6b61 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -863,6 +863,7 @@
else
delay += hstream->bufsize;
}
+ delay = (hstream->bufsize == delay) ? 0 : delay;
if (delay >= hstream->period_bytes) {
dev_info(bus->dev,
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 4624556..a294fee 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -54,12 +54,9 @@
/*
* Each pipelines needs memory to be allocated. Check if we have free memory
- * from available pool. Then only add this to pool
- * This is freed when pipe is deleted
- * Note: DSP does actual memory management we only keep track for complete
- * pool
+ * from available pool.
*/
-static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
+static bool skl_is_pipe_mem_avail(struct skl *skl,
struct skl_module_cfg *mconfig)
{
struct skl_sst *ctx = skl->skl_sst;
@@ -74,10 +71,20 @@
"exceeds ppl memory available %d mem %d\n",
skl->resource.max_mem, skl->resource.mem);
return false;
+ } else {
+ return true;
}
+}
+/*
+ * Add the mem to the mem pool. This is freed when pipe is deleted.
+ * Note: DSP does actual memory management we only keep track for complete
+ * pool
+ */
+static void skl_tplg_alloc_pipe_mem(struct skl *skl,
+ struct skl_module_cfg *mconfig)
+{
skl->resource.mem += mconfig->pipe->memory_pages;
- return true;
}
/*
@@ -85,10 +92,10 @@
* quantified in MCPS (Million Clocks Per Second) required for module/pipe
*
* Each pipelines needs mcps to be allocated. Check if we have mcps for this
- * pipe. This adds the mcps to driver counter
- * This is removed on pipeline delete
+ * pipe.
*/
-static bool skl_tplg_alloc_pipe_mcps(struct skl *skl,
+
+static bool skl_is_pipe_mcps_avail(struct skl *skl,
struct skl_module_cfg *mconfig)
{
struct skl_sst *ctx = skl->skl_sst;
@@ -98,13 +105,18 @@
"%s: module_id %d instance %d\n", __func__,
mconfig->id.module_id, mconfig->id.instance_id);
dev_err(ctx->dev,
- "exceeds ppl memory available %d > mem %d\n",
+ "exceeds ppl mcps available %d > mem %d\n",
skl->resource.max_mcps, skl->resource.mcps);
return false;
+ } else {
+ return true;
}
+}
+static void skl_tplg_alloc_pipe_mcps(struct skl *skl,
+ struct skl_module_cfg *mconfig)
+{
skl->resource.mcps += mconfig->mcps;
- return true;
}
/*
@@ -411,7 +423,7 @@
mconfig = w->priv;
/* check resource available */
- if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+ if (!skl_is_pipe_mcps_avail(skl, mconfig))
return -ENOMEM;
if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) {
@@ -435,6 +447,7 @@
ret = skl_tplg_set_module_params(w, ctx);
if (ret < 0)
return ret;
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
}
return 0;
@@ -477,10 +490,10 @@
struct skl_sst *ctx = skl->skl_sst;
/* check resource available */
- if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+ if (!skl_is_pipe_mcps_avail(skl, mconfig))
return -EBUSY;
- if (!skl_tplg_alloc_pipe_mem(skl, mconfig))
+ if (!skl_is_pipe_mem_avail(skl, mconfig))
return -ENOMEM;
/*
@@ -526,11 +539,15 @@
src_module = dst_module;
}
+ skl_tplg_alloc_pipe_mem(skl, mconfig);
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
+
return 0;
}
static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
struct skl *skl,
+ struct snd_soc_dapm_widget *src_w,
struct skl_module_cfg *src_mconfig)
{
struct snd_soc_dapm_path *p;
@@ -547,6 +564,10 @@
dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name);
next_sink = p->sink;
+
+ if (!is_skl_dsp_widget_type(p->sink))
+ return skl_tplg_bind_sinks(p->sink, skl, src_w, src_mconfig);
+
/*
* here we will check widgets in sink pipelines, so that
* can be any widgets type and we are only interested if
@@ -576,7 +597,7 @@
}
if (!sink)
- return skl_tplg_bind_sinks(next_sink, skl, src_mconfig);
+ return skl_tplg_bind_sinks(next_sink, skl, src_w, src_mconfig);
return 0;
}
@@ -605,7 +626,7 @@
* if sink is not started, start sink pipe first, then start
* this pipe
*/
- ret = skl_tplg_bind_sinks(w, skl, src_mconfig);
+ ret = skl_tplg_bind_sinks(w, skl, w, src_mconfig);
if (ret)
return ret;
@@ -773,10 +794,7 @@
continue;
}
- ret = skl_unbind_modules(ctx, src_module, dst_module);
- if (ret < 0)
- return ret;
-
+ skl_unbind_modules(ctx, src_module, dst_module);
src_module = dst_module;
}
@@ -814,9 +832,6 @@
* This is a connecter and if path is found that means
* unbind between source and sink has not happened yet
*/
- ret = skl_stop_pipe(ctx, sink_mconfig->pipe);
- if (ret < 0)
- return ret;
ret = skl_unbind_modules(ctx, src_mconfig,
sink_mconfig);
}
@@ -842,6 +857,12 @@
case SND_SOC_DAPM_PRE_PMU:
return skl_tplg_mixer_dapm_pre_pmu_event(w, skl);
+ case SND_SOC_DAPM_POST_PMU:
+ return skl_tplg_mixer_dapm_post_pmu_event(w, skl);
+
+ case SND_SOC_DAPM_PRE_PMD:
+ return skl_tplg_mixer_dapm_pre_pmd_event(w, skl);
+
case SND_SOC_DAPM_POST_PMD:
return skl_tplg_mixer_dapm_post_pmd_event(w, skl);
}
@@ -916,6 +937,13 @@
skl_get_module_params(skl->skl_sst, (u32 *)bc->params,
bc->max, bc->param_id, mconfig);
+ /* decrement size for TLV header */
+ size -= 2 * sizeof(u32);
+
+ /* check size as we don't want to send kernel data */
+ if (size > bc->max)
+ size = bc->max;
+
if (bc->params) {
if (copy_to_user(data, &bc->param_id, sizeof(u32)))
return -EFAULT;
@@ -1510,6 +1538,7 @@
&skl_tplg_ops, fw, 0);
if (ret < 0) {
dev_err(bus->dev, "tplg component load failed%d\n", ret);
+ release_firmware(fw);
return -EINVAL;
}
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 443a15d..092705e 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -614,8 +614,6 @@
goto out_unregister;
/*configure PM */
- pm_runtime_set_autosuspend_delay(bus->dev, SKL_SUSPEND_DELAY);
- pm_runtime_use_autosuspend(bus->dev);
pm_runtime_put_noidle(bus->dev);
pm_runtime_allow(bus->dev);
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 15c04e2..9769676 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -9,7 +9,7 @@
config SND_SOC_MT8173_MAX98090
tristate "ASoC Audio driver for MT8173 with MAX98090 codec"
- depends on SND_SOC_MEDIATEK
+ depends on SND_SOC_MEDIATEK && I2C
select SND_SOC_MAX98090
help
This adds ASoC driver for Mediatek MT8173 boards
@@ -19,7 +19,7 @@
config SND_SOC_MT8173_RT5650_RT5676
tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs"
- depends on SND_SOC_MEDIATEK
+ depends on SND_SOC_MEDIATEK && I2C
select SND_SOC_RT5645
select SND_SOC_RT5677
help
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index c866ade..a6c7b8d 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -381,9 +381,19 @@
__raw_writel(BM_SAIF_CTRL_CLKGATE,
saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+ clk_prepare(saif->clk);
+
return 0;
}
+static void mxs_saif_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ clk_unprepare(saif->clk);
+}
+
/*
* Should only be called when port is inactive.
* although can be called multiple times by upper layers.
@@ -424,8 +434,6 @@
return ret;
}
- /* prepare clk in hw_param, enable in trigger */
- clk_prepare(saif->clk);
if (saif != master_saif) {
/*
* Set an initial clock rate for the saif internal logic to work
@@ -611,6 +619,7 @@
static const struct snd_soc_dai_ops mxs_saif_dai_ops = {
.startup = mxs_saif_startup,
+ .shutdown = mxs_saif_shutdown,
.trigger = mxs_saif_trigger,
.prepare = mxs_saif_prepare,
.hw_params = mxs_saif_hw_params,
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 79688aa..4aeb8e1 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -440,18 +440,18 @@
}
static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream,
- struct snd_soc_pcm_runtime *soc_runtime)
+ struct snd_soc_pcm_runtime *rt)
{
struct snd_dma_buffer *buf = &substream->dma_buffer;
size_t size = lpass_platform_pcm_hardware.buffer_bytes_max;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = soc_runtime->dev;
+ buf->dev.dev = rt->platform->dev;
buf->private_data = NULL;
- buf->area = dma_alloc_coherent(soc_runtime->dev, size, &buf->addr,
+ buf->area = dma_alloc_coherent(rt->platform->dev, size, &buf->addr,
GFP_KERNEL);
if (!buf->area) {
- dev_err(soc_runtime->dev, "%s: Could not allocate DMA buffer\n",
+ dev_err(rt->platform->dev, "%s: Could not allocate DMA buffer\n",
__func__);
return -ENOMEM;
}
@@ -461,12 +461,12 @@
}
static void lpass_platform_free_buffer(struct snd_pcm_substream *substream,
- struct snd_soc_pcm_runtime *soc_runtime)
+ struct snd_soc_pcm_runtime *rt)
{
struct snd_dma_buffer *buf = &substream->dma_buffer;
if (buf->area) {
- dma_free_coherent(soc_runtime->dev, buf->bytes, buf->area,
+ dma_free_coherent(rt->dev, buf->bytes, buf->area,
buf->addr);
}
buf->area = NULL;
@@ -499,9 +499,6 @@
snd_soc_pcm_set_drvdata(soc_runtime, data);
- soc_runtime->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- soc_runtime->dev->dma_mask = &soc_runtime->dev->coherent_dma_mask;
-
ret = lpass_platform_alloc_buffer(substream, soc_runtime);
if (ret)
return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 5a2812f..0d37079 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -310,7 +310,7 @@
};
static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
- struct snd_kcontrol *kcontrol)
+ struct snd_kcontrol *kcontrol, const char *ctrl_name)
{
struct dapm_kcontrol_data *data;
struct soc_mixer_control *mc;
@@ -333,7 +333,7 @@
if (mc->autodisable) {
struct snd_soc_dapm_widget template;
- name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+ name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name,
"Autodisable");
if (!name) {
ret = -ENOMEM;
@@ -371,7 +371,7 @@
if (e->autodisable) {
struct snd_soc_dapm_widget template;
- name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+ name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name,
"Autodisable");
if (!name) {
ret = -ENOMEM;
@@ -871,7 +871,7 @@
kcontrol->private_free = dapm_kcontrol_free;
- ret = dapm_kcontrol_data_alloc(w, kcontrol);
+ ret = dapm_kcontrol_data_alloc(w, kcontrol, name);
if (ret) {
snd_ctl_free_one(kcontrol);
goto exit_free;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e898b42..1af4f23 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1810,7 +1810,8 @@
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
- (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND))
continue;
dev_dbg(be->dev, "ASoC: hw_free BE %s\n",
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index b79875e..47de8af 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -2458,7 +2458,6 @@
else
err = snd_usbmidi_create_endpoints(umidi, endpoints);
if (err < 0) {
- snd_usbmidi_free(umidi);
return err;
}