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-rw-r--r--sound/core/compress_offload.c67
-rw-r--r--sound/core/control.c2
-rw-r--r--sound/core/pcm.c14
-rw-r--r--sound/core/timer.c2
-rw-r--r--sound/drivers/dummy.c1
-rw-r--r--sound/hda/hdac_regmap.c4
-rw-r--r--sound/pci/au88x0/au88x0_core.c5
-rw-r--r--sound/pci/echoaudio/echoaudio.c4
-rw-r--r--sound/pci/hda/hda_generic.c2
-rw-r--r--sound/pci/hda/hda_intel.c21
-rw-r--r--sound/pci/hda/hda_tegra.c20
-rw-r--r--sound/pci/hda/patch_realtek.c95
-rw-r--r--sound/soc/atmel/Kconfig1
-rw-r--r--sound/soc/atmel/atmel-classd.c5
-rw-r--r--sound/soc/atmel/atmel-pdmic.c5
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c2
-rw-r--r--sound/soc/bcm/Kconfig9
-rw-r--r--sound/soc/bcm/Makefile5
-rw-r--r--sound/soc/bcm/cygnus-pcm.c861
-rw-r--r--sound/soc/bcm/cygnus-ssp.c1529
-rw-r--r--sound/soc/bcm/cygnus-ssp.h139
-rw-r--r--sound/soc/codecs/Kconfig35
-rw-r--r--sound/soc/codecs/Makefile10
-rw-r--r--sound/soc/codecs/adau-utils.c61
-rw-r--r--sound/soc/codecs/adau-utils.h7
-rw-r--r--sound/soc/codecs/adau1373.c38
-rw-r--r--sound/soc/codecs/adau1761-i2c.c2
-rw-r--r--sound/soc/codecs/adau1761-spi.c2
-rw-r--r--sound/soc/codecs/adau1781-i2c.c2
-rw-r--r--sound/soc/codecs/adau1781-spi.c2
-rw-r--r--sound/soc/codecs/adau17x1.c251
-rw-r--r--sound/soc/codecs/adau17x1.h6
-rw-r--r--sound/soc/codecs/adau7002.c80
-rw-r--r--sound/soc/codecs/ak4613.c14
-rw-r--r--sound/soc/codecs/ak4642.c13
-rw-r--r--sound/soc/codecs/arizona.c91
-rw-r--r--sound/soc/codecs/arizona.h22
-rw-r--r--sound/soc/codecs/bt-sco.c52
-rw-r--r--sound/soc/codecs/cs35l33.c1303
-rw-r--r--sound/soc/codecs/cs35l33.h221
-rw-r--r--sound/soc/codecs/cs47l24.c19
-rw-r--r--sound/soc/codecs/cs53l30.c1143
-rw-r--r--sound/soc/codecs/cs53l30.h459
-rw-r--r--sound/soc/codecs/cx20442.c1
-rw-r--r--sound/soc/codecs/da7219-aad.c103
-rw-r--r--sound/soc/codecs/da7219.c34
-rw-r--r--sound/soc/codecs/hdac_hdmi.c27
-rw-r--r--sound/soc/codecs/hdmi-codec.c15
-rw-r--r--sound/soc/codecs/max9860.c753
-rw-r--r--sound/soc/codecs/max9860.h162
-rw-r--r--sound/soc/codecs/pcm1681.c2
-rw-r--r--sound/soc/codecs/pcm179x.c2
-rw-r--r--sound/soc/codecs/pcm5102a.c1
-rw-r--r--sound/soc/codecs/rt286.c7
-rw-r--r--sound/soc/codecs/rt5645.c26
-rw-r--r--sound/soc/codecs/rt5645.h3
-rw-r--r--sound/soc/codecs/rt5670.c2
-rw-r--r--sound/soc/codecs/wm5102.c3
-rw-r--r--sound/soc/codecs/wm5110.c20
-rw-r--r--sound/soc/codecs/wm8940.c1
-rw-r--r--sound/soc/codecs/wm8998.c1
-rw-r--r--sound/soc/codecs/wm_adsp.c34
-rw-r--r--sound/soc/codecs/wm_adsp.h4
-rw-r--r--sound/soc/davinci/davinci-mcasp.c65
-rw-r--r--sound/soc/davinci/davinci-mcasp.h4
-rw-r--r--sound/soc/dwc/Kconfig9
-rw-r--r--sound/soc/dwc/Makefile4
-rw-r--r--sound/soc/dwc/designware_i2s.c231
-rw-r--r--sound/soc/dwc/designware_pcm.c225
-rw-r--r--sound/soc/dwc/local.h128
-rw-r--r--sound/soc/fsl/Kconfig1
-rw-r--r--sound/soc/fsl/fsl_spdif.c2
-rw-r--r--sound/soc/fsl/fsl_ssi.c12
-rw-r--r--sound/soc/intel/Kconfig73
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-compress.c9
-rw-r--r--sound/soc/intel/atom/sst/sst_acpi.c44
-rw-r--r--sound/soc/intel/boards/Makefile2
-rw-r--r--sound/soc/intel/boards/bxt_da7219_max98357a.c460
-rw-r--r--sound/soc/intel/boards/bxt_rt298.c118
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5645.c20
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_max98357a.c51
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_ssm4567.c51
-rw-r--r--sound/soc/intel/boards/skl_rt286.c9
-rw-r--r--sound/soc/intel/common/Makefile4
-rw-r--r--sound/soc/intel/common/sst-acpi.h4
-rw-r--r--sound/soc/intel/common/sst-dsp-priv.h4
-rw-r--r--sound/soc/intel/common/sst-dsp.c69
-rw-r--r--sound/soc/intel/common/sst-dsp.h2
-rw-r--r--sound/soc/intel/common/sst-firmware.c68
-rw-r--r--sound/soc/intel/haswell/sst-haswell-pcm.c1
-rw-r--r--sound/soc/intel/skylake/Makefile2
-rw-r--r--sound/soc/intel/skylake/bxt-sst.c203
-rw-r--r--sound/soc/intel/skylake/skl-messages.c53
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c40
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.h22
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c93
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.c260
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.h102
-rw-r--r--sound/soc/intel/skylake/skl-sst-ipc.c4
-rw-r--r--sound/soc/intel/skylake/skl-sst-ipc.h18
-rw-r--r--sound/soc/intel/skylake/skl-sst-utils.c256
-rw-r--r--sound/soc/intel/skylake/skl-sst.c124
-rw-r--r--sound/soc/intel/skylake/skl-topology.c230
-rw-r--r--sound/soc/intel/skylake/skl-topology.h7
-rw-r--r--sound/soc/intel/skylake/skl.c39
-rw-r--r--sound/soc/intel/skylake/skl.h10
-rw-r--r--sound/soc/sh/rcar/adg.c2
-rw-r--r--sound/soc/soc-compress.c5
-rw-r--r--sound/soc/soc-dapm.c61
-rw-r--r--sound/soc/soc-pcm.c43
-rw-r--r--sound/usb/card.c2
111 files changed, 10134 insertions, 939 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 9b3334be9df2..2c498488af6c 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -67,6 +67,8 @@ struct snd_compr_file {
struct snd_compr_stream stream;
};
+static void error_delayed_work(struct work_struct *work);
+
/*
* a note on stream states used:
* we use following states in the compressed core
@@ -123,6 +125,9 @@ static int snd_compr_open(struct inode *inode, struct file *f)
snd_card_unref(compr->card);
return -ENOMEM;
}
+
+ INIT_DELAYED_WORK(&data->stream.error_work, error_delayed_work);
+
data->stream.ops = compr->ops;
data->stream.direction = dirn;
data->stream.private_data = compr->private_data;
@@ -153,6 +158,8 @@ static int snd_compr_free(struct inode *inode, struct file *f)
struct snd_compr_file *data = f->private_data;
struct snd_compr_runtime *runtime = data->stream.runtime;
+ cancel_delayed_work_sync(&data->stream.error_work);
+
switch (runtime->state) {
case SNDRV_PCM_STATE_RUNNING:
case SNDRV_PCM_STATE_DRAINING:
@@ -237,6 +244,15 @@ snd_compr_ioctl_avail(struct snd_compr_stream *stream, unsigned long arg)
avail = snd_compr_calc_avail(stream, &ioctl_avail);
ioctl_avail.avail = avail;
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ return -EBADFD;
+ case SNDRV_PCM_STATE_XRUN:
+ return -EPIPE;
+ default:
+ break;
+ }
+
if (copy_to_user((__u64 __user *)arg,
&ioctl_avail, sizeof(ioctl_avail)))
return -EFAULT;
@@ -346,11 +362,13 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf,
switch (stream->runtime->state) {
case SNDRV_PCM_STATE_OPEN:
case SNDRV_PCM_STATE_PREPARED:
- case SNDRV_PCM_STATE_XRUN:
case SNDRV_PCM_STATE_SUSPENDED:
case SNDRV_PCM_STATE_DISCONNECTED:
retval = -EBADFD;
goto out;
+ case SNDRV_PCM_STATE_XRUN:
+ retval = -EPIPE;
+ goto out;
}
avail = snd_compr_get_avail(stream);
@@ -399,10 +417,16 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait)
stream = &data->stream;
mutex_lock(&stream->device->lock);
- if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) {
+
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_XRUN:
retval = snd_compr_get_poll(stream) | POLLERR;
goto out;
+ default:
+ break;
}
+
poll_wait(f, &stream->runtime->sleep, wait);
avail = snd_compr_get_avail(stream);
@@ -697,6 +721,45 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
return retval;
}
+static void error_delayed_work(struct work_struct *work)
+{
+ struct snd_compr_stream *stream;
+
+ stream = container_of(work, struct snd_compr_stream, error_work.work);
+
+ mutex_lock(&stream->device->lock);
+
+ stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
+ wake_up(&stream->runtime->sleep);
+
+ mutex_unlock(&stream->device->lock);
+}
+
+/*
+ * snd_compr_stop_error: Report a fatal error on a stream
+ * @stream: pointer to stream
+ * @state: state to transition the stream to
+ *
+ * Stop the stream and set its state.
+ *
+ * Should be called with compressed device lock held.
+ */
+int snd_compr_stop_error(struct snd_compr_stream *stream,
+ snd_pcm_state_t state)
+{
+ if (stream->runtime->state == state)
+ return 0;
+
+ stream->runtime->state = state;
+
+ pr_debug("Changing state to: %d\n", state);
+
+ queue_delayed_work(system_power_efficient_wq, &stream->error_work, 0);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_compr_stop_error);
+
static int snd_compress_wait_for_drain(struct snd_compr_stream *stream)
{
int ret;
diff --git a/sound/core/control.c b/sound/core/control.c
index a85d45595d02..b4fe9b002512 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -160,6 +160,8 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask,
if (snd_BUG_ON(!card || !id))
return;
+ if (card->shutdown)
+ return;
read_lock(&card->ctl_files_rwlock);
#if IS_ENABLED(CONFIG_SND_MIXER_OSS)
card->mixer_oss_change_count++;
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 308c9ecf73db..8e980aa678d0 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -849,6 +849,14 @@ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
}
EXPORT_SYMBOL(snd_pcm_new_internal);
+static void free_chmap(struct snd_pcm_str *pstr)
+{
+ if (pstr->chmap_kctl) {
+ snd_ctl_remove(pstr->pcm->card, pstr->chmap_kctl);
+ pstr->chmap_kctl = NULL;
+ }
+}
+
static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
{
struct snd_pcm_substream *substream, *substream_next;
@@ -871,6 +879,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
kfree(setup);
}
#endif
+ free_chmap(pstr);
if (pstr->substream_count)
put_device(&pstr->dev);
}
@@ -1135,10 +1144,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device)
for (cidx = 0; cidx < 2; cidx++) {
if (!pcm->internal)
snd_unregister_device(&pcm->streams[cidx].dev);
- if (pcm->streams[cidx].chmap_kctl) {
- snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl);
- pcm->streams[cidx].chmap_kctl = NULL;
- }
+ free_chmap(&pcm->streams[cidx]);
}
mutex_unlock(&pcm->open_mutex);
mutex_unlock(&register_mutex);
diff --git a/sound/core/timer.c b/sound/core/timer.c
index e722022d325d..9a6157ea6881 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1955,6 +1955,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer,
qhead = tu->qhead++;
tu->qhead %= tu->queue_size;
+ tu->qused--;
spin_unlock_irq(&tu->qlock);
if (tu->tread) {
@@ -1968,7 +1969,6 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer,
}
spin_lock_irq(&tu->qlock);
- tu->qused--;
if (err < 0)
goto _error;
result += unit;
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index c0f8f613f1f1..172dacd925f5 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -420,6 +420,7 @@ static int dummy_hrtimer_stop(struct snd_pcm_substream *substream)
static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm)
{
+ hrtimer_cancel(&dpcm->timer);
tasklet_kill(&dpcm->tasklet);
}
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index 87041ddd29cb..47a358fab132 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -444,7 +444,7 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg,
err = reg_raw_write(codec, reg, val);
if (err == -EAGAIN) {
err = snd_hdac_power_up_pm(codec);
- if (!err)
+ if (err >= 0)
err = reg_raw_write(codec, reg, val);
snd_hdac_power_down_pm(codec);
}
@@ -470,7 +470,7 @@ static int __snd_hdac_regmap_read_raw(struct hdac_device *codec,
err = reg_raw_read(codec, reg, val, uncached);
if (err == -EAGAIN) {
err = snd_hdac_power_up_pm(codec);
- if (!err)
+ if (err >= 0)
err = reg_raw_read(codec, reg, val, uncached);
snd_hdac_power_down_pm(codec);
}
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 4a054d720112..d3125c169684 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1444,9 +1444,8 @@ static int vortex_wtdma_bufshift(vortex_t * vortex, int wtdma)
int page, p, pp, delta, i;
page =
- (hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) &
- WT_SUBBUF_MASK)
- >> WT_SUBBUF_SHIFT;
+ (hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2))
+ >> WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK;
if (dma->nr_periods >= 4)
delta = (page - dma->period_real) & 3;
else {
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 1cb85aeb0cea..286f5e3686a3 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2200,11 +2200,11 @@ static int snd_echo_resume(struct device *dev)
u32 pipe_alloc_mask;
int err;
- commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL);
+ commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL);
if (commpage_bak == NULL)
return -ENOMEM;
commpage = chip->comm_page;
- memcpy(commpage_bak, commpage, sizeof(struct comm_page));
+ memcpy(commpage_bak, commpage, sizeof(*commpage));
err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
if (err < 0) {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 320445f3bf73..79c7b340acc2 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3977,6 +3977,8 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid,
for (n = 0; n < spec->paths.used; n++) {
path = snd_array_elem(&spec->paths, n);
+ if (!path->depth)
+ continue;
if (path->path[0] == nid ||
path->path[path->depth - 1] == nid) {
bool pin_old = path->pin_enabled;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9a0d1445ca5c..6f8ea13323c1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -365,8 +365,12 @@ enum {
#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
+#define IS_KBL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa171)
+#define IS_KBL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d71)
+#define IS_KBL_H(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa2f0)
#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))
+#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \
+ IS_KBL(pci) || IS_KBL_LP(pci) || IS_KBL_H(pci)
static char *driver_short_names[] = {
[AZX_DRIVER_ICH] = "HDA Intel",
@@ -1214,8 +1218,10 @@ static int azx_free(struct azx *chip)
if (use_vga_switcheroo(hda)) {
if (chip->disabled && hda->probe_continued)
snd_hda_unlock_devices(&chip->bus);
- if (hda->vga_switcheroo_registered)
+ if (hda->vga_switcheroo_registered) {
vga_switcheroo_unregister_client(chip->pci);
+ vga_switcheroo_fini_domain_pm_ops(chip->card->dev);
+ }
}
if (bus->chip_init) {
@@ -2181,6 +2187,15 @@ static const struct pci_device_id azx_ids[] = {
/* Sunrise Point-LP */
{ PCI_DEVICE(0x8086, 0x9d70),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
+ /* Kabylake */
+ { PCI_DEVICE(0x8086, 0xa171),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
+ /* Kabylake-LP */
+ { PCI_DEVICE(0x8086, 0x9d71),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
+ /* Kabylake-H */
+ { PCI_DEVICE(0x8086, 0xa2f0),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
/* Broxton-P(Apollolake) */
{ PCI_DEVICE(0x8086, 0x5a98),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON },
@@ -2254,6 +2269,8 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
{ PCI_DEVICE(0x1002, 0x157a),
.driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+ { PCI_DEVICE(0x1002, 0x15b3),
+ .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
{ PCI_DEVICE(0x1002, 0x793b),
.driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
{ PCI_DEVICE(0x1002, 0x7919),
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 17fd81736d3d..0621920f7617 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -115,20 +115,20 @@ static int substream_free_pages(struct azx *chip,
/*
* Register access ops. Tegra HDA register access is DWORD only.
*/
-static void hda_tegra_writel(u32 value, u32 *addr)
+static void hda_tegra_writel(u32 value, u32 __iomem *addr)
{
writel(value, addr);
}
-static u32 hda_tegra_readl(u32 *addr)
+static u32 hda_tegra_readl(u32 __iomem *addr)
{
return readl(addr);
}
-static void hda_tegra_writew(u16 value, u16 *addr)
+static void hda_tegra_writew(u16 value, u16 __iomem *addr)
{
unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
- void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+ void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
u32 v;
v = readl(dword_addr);
@@ -137,20 +137,20 @@ static void hda_tegra_writew(u16 value, u16 *addr)
writel(v, dword_addr);
}
-static u16 hda_tegra_readw(u16 *addr)
+static u16 hda_tegra_readw(u16 __iomem *addr)
{
unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
- void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+ void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
u32 v;
v = readl(dword_addr);
return (v >> shift) & 0xffff;
}
-static void hda_tegra_writeb(u8 value, u8 *addr)
+static void hda_tegra_writeb(u8 value, u8 __iomem *addr)
{
unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
- void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+ void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
u32 v;
v = readl(dword_addr);
@@ -159,10 +159,10 @@ static void hda_tegra_writeb(u8 value, u8 *addr)
writel(v, dword_addr);
}
-static u8 hda_tegra_readb(u8 *addr)
+static u8 hda_tegra_readb(u8 __iomem *addr)
{
unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
- void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+ void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
u32 v;
v = readl(dword_addr);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d53c25e7a1c1..abcb5a6a1cd9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -346,6 +346,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0234:
case 0x10ec0274:
case 0x10ec0294:
+ case 0x10ec0700:
+ case 0x10ec0701:
+ case 0x10ec0703:
alc_update_coef_idx(codec, 0x10, 1<<15, 0);
break;
case 0x10ec0662:
@@ -2655,6 +2658,7 @@ enum {
ALC269_TYPE_ALC256,
ALC269_TYPE_ALC225,
ALC269_TYPE_ALC294,
+ ALC269_TYPE_ALC700,
};
/*
@@ -2686,6 +2690,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
case ALC269_TYPE_ALC256:
case ALC269_TYPE_ALC225:
case ALC269_TYPE_ALC294:
+ case ALC269_TYPE_ALC700:
ssids = alc269_ssids;
break;
default:
@@ -3618,13 +3623,20 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec,
static void alc_headset_mode_unplugged(struct hda_codec *codec)
{
static struct coef_fw coef0255[] = {
- WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */
WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */
UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */
WRITE_COEFEX(0x57, 0x03, 0x8aa6), /* Direct Drive HP Amp control */
{}
};
+ static struct coef_fw coef0255_1[] = {
+ WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */
+ {}
+ };
+ static struct coef_fw coef0256[] = {
+ WRITE_COEF(0x1b, 0x0c4b), /* LDO and MISC control */
+ {}
+ };
static struct coef_fw coef0233[] = {
WRITE_COEF(0x1b, 0x0c0b),
WRITE_COEF(0x45, 0xc429),
@@ -3677,7 +3689,11 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
switch (codec->core.vendor_id) {
case 0x10ec0255:
+ alc_process_coef_fw(codec, coef0255_1);
+ alc_process_coef_fw(codec, coef0255);
+ break;
case 0x10ec0256:
+ alc_process_coef_fw(codec, coef0256);
alc_process_coef_fw(codec, coef0255);
break;
case 0x10ec0233:
@@ -3896,6 +3912,12 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
WRITE_COEFEX(0x57, 0x03, 0x8ea6),
{}
};
+ static struct coef_fw coef0256[] = {
+ WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */
+ WRITE_COEF(0x1b, 0x0c6b),
+ WRITE_COEFEX(0x57, 0x03, 0x8ea6),
+ {}
+ };
static struct coef_fw coef0233[] = {
WRITE_COEF(0x45, 0xd429),
WRITE_COEF(0x1b, 0x0c2b),
@@ -3936,9 +3958,11 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
switch (codec->core.vendor_id) {
case 0x10ec0255:
- case 0x10ec0256:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0256:
+ alc_process_coef_fw(codec, coef0256);
+ break;
case 0x10ec0233:
case 0x10ec0283:
alc_process_coef_fw(codec, coef0233);
@@ -3978,6 +4002,12 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
WRITE_COEFEX(0x57, 0x03, 0x8ea6),
{}
};
+ static struct coef_fw coef0256[] = {
+ WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */
+ WRITE_COEF(0x1b, 0x0c6b),
+ WRITE_COEFEX(0x57, 0x03, 0x8ea6),
+ {}
+ };
static struct coef_fw coef0233[] = {
WRITE_COEF(0x45, 0xe429),
WRITE_COEF(0x1b, 0x0c2b),
@@ -4018,9 +4048,11 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
switch (codec->core.vendor_id) {
case 0x10ec0255:
- case 0x10ec0256:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0256:
+ alc_process_coef_fw(codec, coef0256);
+ break;
case 0x10ec0233:
case 0x10ec0283:
alc_process_coef_fw(codec, coef0233);
@@ -4266,7 +4298,7 @@ static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec,
static void alc255_set_default_jack_type(struct hda_codec *codec)
{
/* Set to iphone type */
- static struct coef_fw fw[] = {
+ static struct coef_fw alc255fw[] = {
WRITE_COEF(0x1b, 0x880b),
WRITE_COEF(0x45, 0xd089),
WRITE_COEF(0x1b, 0x080b),
@@ -4274,7 +4306,22 @@ static void alc255_set_default_jack_type(struct hda_codec *codec)
WRITE_COEF(0x1b, 0x0c0b),
{}
};
- alc_process_coef_fw(codec, fw);
+ static struct coef_fw alc256fw[] = {
+ WRITE_COEF(0x1b, 0x884b),
+ WRITE_COEF(0x45, 0xd089),
+ WRITE_COEF(0x1b, 0x084b),
+ WRITE_COEF(0x46, 0x0004),
+ WRITE_COEF(0x1b, 0x0c4b),
+ {}
+ };
+ switch (codec->core.vendor_id) {
+ case 0x10ec0255:
+ alc_process_coef_fw(codec, alc255fw);
+ break;
+ case 0x10ec0256:
+ alc_process_coef_fw(codec, alc256fw);
+ break;
+ }
msleep(30);
}
@@ -5587,6 +5634,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x2231, "Thinkpad T560", ALC292_FIXUP_TPT460),
SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460),
SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
@@ -5602,6 +5650,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
+ SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T560p", ALC292_FIXUP_TPT460),
+ SND_PCI_QUIRK(0x17aa, 0x5051, "Thinkpad L460", ALC292_FIXUP_TPT460),
+ SND_PCI_QUIRK(0x17aa, 0x5053, "Thinkpad T460", ALC292_FIXUP_TPT460),
SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -5687,7 +5738,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{}
};
#define ALC225_STANDARD_PINS \
- {0x12, 0xb7a60130}, \
{0x21, 0x04211020}
#define ALC256_STANDARD_PINS \
@@ -5712,10 +5762,24 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC225_STANDARD_PINS,
+ {0x12, 0xb7a60130},
+ {0x14, 0x901701a0}),
+ SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC225_STANDARD_PINS,
+ {0x12, 0xb7a60130},
+ {0x14, 0x901701b0}),
+ SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC225_STANDARD_PINS,
+ {0x12, 0xb7a60150},
{0x14, 0x901701a0}),
SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC225_STANDARD_PINS,
+ {0x12, 0xb7a60150},
{0x14, 0x901701b0}),
+ SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC225_STANDARD_PINS,
+ {0x12, 0xb7a60130},
+ {0x1b, 0x90170110}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
{0x14, 0x90170110},
{0x21, 0x02211020}),
@@ -5775,11 +5839,19 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x12, 0x90a60180},
{0x14, 0x90170130},
{0x21, 0x02211040}),
+ SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell Inspiron 5565", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ {0x12, 0x90a60180},
+ {0x14, 0x90170120},
+ {0x21, 0x02211030}),
SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x12, 0x90a60160},
{0x14, 0x90170120},
{0x21, 0x02211030}),
SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ {0x12, 0x90a60170},
+ {0x14, 0x90170120},
+ {0x21, 0x02211030}),
+ SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC256_STANDARD_PINS),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
@@ -6053,6 +6125,14 @@ static int patch_alc269(struct hda_codec *codec)
case 0x10ec0294:
spec->codec_variant = ALC269_TYPE_ALC294;
break;
+ case 0x10ec0700:
+ case 0x10ec0701:
+ case 0x10ec0703:
+ spec->codec_variant = ALC269_TYPE_ALC700;
+ spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */
+ alc_update_coef_idx(codec, 0x4a, 0, 1 << 15); /* Combo jack auto trigger control */
+ break;
+
}
if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) {
@@ -7008,6 +7088,9 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
HDA_CODEC_ENTRY(0x10ec0670, "ALC670", patch_alc662),
HDA_CODEC_ENTRY(0x10ec0671, "ALC671", patch_alc662),
HDA_CODEC_ENTRY(0x10ec0680, "ALC680", patch_alc680),
+ HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc882),
HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 06e099e802df..22aec9a1e9a4 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -10,6 +10,7 @@ if SND_ATMEL_SOC
config SND_ATMEL_SOC_PDC
tristate
+ depends on HAS_DMA
default m if SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=m
default y if SND_ATMEL_SOC_SSC_PDC=y || (SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=y)
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index 6107de9c538b..6d9b8b44e2da 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -593,11 +593,6 @@ static int atmel_classd_probe(struct platform_device *pdev)
}
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- dev_err(dev, "no memory resource\n");
- return -ENXIO;
- }
-
io_base = devm_ioremap_resource(dev, res);
if (IS_ERR(io_base)) {
ret = PTR_ERR(io_base);
diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c
index aee4787a0b89..5f56da60c92f 100644
--- a/sound/soc/atmel/atmel-pdmic.c
+++ b/sound/soc/atmel/atmel-pdmic.c
@@ -624,11 +624,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev)
}
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- dev_err(dev, "no memory resource\n");
- return -ENXIO;
- }
-
io_base = devm_ioremap_resource(dev, res);
if (IS_ERR(io_base)) {
ret = PTR_ERR(io_base);
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 1267e1af0fae..54c09acd3fed 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -321,7 +321,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
return ret;
}
- dma_params = &ssc_dma_params[dai->id][dir];
+ dma_params = &ssc_dma_params[pdev->id][dir];
dma_params->ssc = ssc_p->ssc;
dma_params->substream = substream;
diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig
index 6a834e109f1d..d528aaceaad9 100644
--- a/sound/soc/bcm/Kconfig
+++ b/sound/soc/bcm/Kconfig
@@ -7,3 +7,12 @@ config SND_BCM2835_SOC_I2S
Say Y or M if you want to add support for codecs attached to
the BCM2835 I2S interface. You will also need
to select the audio interfaces to support below.
+
+config SND_SOC_CYGNUS
+ tristate "SoC platform audio for Broadcom Cygnus chips"
+ depends on ARCH_BCM_CYGNUS || COMPILE_TEST
+ help
+ Say Y if you want to add support for ASoC audio on Broadcom
+ Cygnus chips (bcm958300, bcm958305, bcm911360)
+
+ If you don't know what to do here, say N. \ No newline at end of file
diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile
index bc816b71e5a4..fc739d007884 100644
--- a/sound/soc/bcm/Makefile
+++ b/sound/soc/bcm/Makefile
@@ -3,3 +3,8 @@ snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o
obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o
+# CYGNUS Platform Support
+snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o
+
+obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o
+
diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c
new file mode 100644
index 000000000000..d616e096462e
--- /dev/null
+++ b/sound/soc/bcm/cygnus-pcm.c
@@ -0,0 +1,861 @@
+/*
+ * Copyright (C) 2014-2015 Broadcom Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any
+ * kind, whether express or implied; without even the implied warranty
+ * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <linux/debugfs.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/timer.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "cygnus-ssp.h"
+
+/* Register offset needed for ASoC PCM module */
+
+#define INTH_R5F_STATUS_OFFSET 0x040
+#define INTH_R5F_CLEAR_OFFSET 0x048
+#define INTH_R5F_MASK_SET_OFFSET 0x050
+#define INTH_R5F_MASK_CLEAR_OFFSET 0x054
+
+#define BF_REARM_FREE_MARK_OFFSET 0x344
+#define BF_REARM_FULL_MARK_OFFSET 0x348
+
+/* Ring Buffer Ctrl Regs --- Start */
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_RDADDR_REG_BASE */
+#define SRC_RBUF_0_RDADDR_OFFSET 0x500
+#define SRC_RBUF_1_RDADDR_OFFSET 0x518
+#define SRC_RBUF_2_RDADDR_OFFSET 0x530
+#define SRC_RBUF_3_RDADDR_OFFSET 0x548
+#define SRC_RBUF_4_RDADDR_OFFSET 0x560
+#define SRC_RBUF_5_RDADDR_OFFSET 0x578
+#define SRC_RBUF_6_RDADDR_OFFSET 0x590
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_WRADDR_REG_BASE */
+#define SRC_RBUF_0_WRADDR_OFFSET 0x504
+#define SRC_RBUF_1_WRADDR_OFFSET 0x51c
+#define SRC_RBUF_2_WRADDR_OFFSET 0x534
+#define SRC_RBUF_3_WRADDR_OFFSET 0x54c
+#define SRC_RBUF_4_WRADDR_OFFSET 0x564
+#define SRC_RBUF_5_WRADDR_OFFSET 0x57c
+#define SRC_RBUF_6_WRADDR_OFFSET 0x594
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_BASEADDR_REG_BASE */
+#define SRC_RBUF_0_BASEADDR_OFFSET 0x508
+#define SRC_RBUF_1_BASEADDR_OFFSET 0x520
+#define SRC_RBUF_2_BASEADDR_OFFSET 0x538
+#define SRC_RBUF_3_BASEADDR_OFFSET 0x550
+#define SRC_RBUF_4_BASEADDR_OFFSET 0x568
+#define SRC_RBUF_5_BASEADDR_OFFSET 0x580
+#define SRC_RBUF_6_BASEADDR_OFFSET 0x598
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_ENDADDR_REG_BASE */
+#define SRC_RBUF_0_ENDADDR_OFFSET 0x50c
+#define SRC_RBUF_1_ENDADDR_OFFSET 0x524
+#define SRC_RBUF_2_ENDADDR_OFFSET 0x53c
+#define SRC_RBUF_3_ENDADDR_OFFSET 0x554
+#define SRC_RBUF_4_ENDADDR_OFFSET 0x56c
+#define SRC_RBUF_5_ENDADDR_OFFSET 0x584
+#define SRC_RBUF_6_ENDADDR_OFFSET 0x59c
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_FREE_MARK_REG_BASE */
+#define SRC_RBUF_0_FREE_MARK_OFFSET 0x510
+#define SRC_RBUF_1_FREE_MARK_OFFSET 0x528
+#define SRC_RBUF_2_FREE_MARK_OFFSET 0x540
+#define SRC_RBUF_3_FREE_MARK_OFFSET 0x558
+#define SRC_RBUF_4_FREE_MARK_OFFSET 0x570
+#define SRC_RBUF_5_FREE_MARK_OFFSET 0x588
+#define SRC_RBUF_6_FREE_MARK_OFFSET 0x5a0
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_RDADDR_REG_BASE */
+#define DST_RBUF_0_RDADDR_OFFSET 0x5c0
+#define DST_RBUF_1_RDADDR_OFFSET 0x5d8
+#define DST_RBUF_2_RDADDR_OFFSET 0x5f0
+#define DST_RBUF_3_RDADDR_OFFSET 0x608
+#define DST_RBUF_4_RDADDR_OFFSET 0x620
+#define DST_RBUF_5_RDADDR_OFFSET 0x638
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_WRADDR_REG_BASE */
+#define DST_RBUF_0_WRADDR_OFFSET 0x5c4
+#define DST_RBUF_1_WRADDR_OFFSET 0x5dc
+#define DST_RBUF_2_WRADDR_OFFSET 0x5f4
+#define DST_RBUF_3_WRADDR_OFFSET 0x60c
+#define DST_RBUF_4_WRADDR_OFFSET 0x624
+#define DST_RBUF_5_WRADDR_OFFSET 0x63c
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_BASEADDR_REG_BASE */
+#define DST_RBUF_0_BASEADDR_OFFSET 0x5c8
+#define DST_RBUF_1_BASEADDR_OFFSET 0x5e0
+#define DST_RBUF_2_BASEADDR_OFFSET 0x5f8
+#define DST_RBUF_3_BASEADDR_OFFSET 0x610
+#define DST_RBUF_4_BASEADDR_OFFSET 0x628
+#define DST_RBUF_5_BASEADDR_OFFSET 0x640
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_ENDADDR_REG_BASE */
+#define DST_RBUF_0_ENDADDR_OFFSET 0x5cc
+#define DST_RBUF_1_ENDADDR_OFFSET 0x5e4
+#define DST_RBUF_2_ENDADDR_OFFSET 0x5fc
+#define DST_RBUF_3_ENDADDR_OFFSET 0x614
+#define DST_RBUF_4_ENDADDR_OFFSET 0x62c
+#define DST_RBUF_5_ENDADDR_OFFSET 0x644
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_FULL_MARK_REG_BASE */
+#define DST_RBUF_0_FULL_MARK_OFFSET 0x5d0
+#define DST_RBUF_1_FULL_MARK_OFFSET 0x5e8
+#define DST_RBUF_2_FULL_MARK_OFFSET 0x600
+#define DST_RBUF_3_FULL_MARK_OFFSET 0x618
+#define DST_RBUF_4_FULL_MARK_OFFSET 0x630
+#define DST_RBUF_5_FULL_MARK_OFFSET 0x648
+/* Ring Buffer Ctrl Regs --- End */
+
+/* Error Status Regs --- Start */
+/* AUD_FMM_BF_ESR_ESRX_STATUS_REG_BASE */
+#define ESR0_STATUS_OFFSET 0x900
+#define ESR1_STATUS_OFFSET 0x918
+#define ESR2_STATUS_OFFSET 0x930
+#define ESR3_STATUS_OFFSET 0x948
+#define ESR4_STATUS_OFFSET 0x960
+
+/* AUD_FMM_BF_ESR_ESRX_STATUS_CLEAR_REG_BASE */
+#define ESR0_STATUS_CLR_OFFSET 0x908
+#define ESR1_STATUS_CLR_OFFSET 0x920
+#define ESR2_STATUS_CLR_OFFSET 0x938
+#define ESR3_STATUS_CLR_OFFSET 0x950
+#define ESR4_STATUS_CLR_OFFSET 0x968
+
+/* AUD_FMM_BF_ESR_ESRX_MASK_REG_BASE */
+#define ESR0_MASK_STATUS_OFFSET 0x90c
+#define ESR1_MASK_STATUS_OFFSET 0x924
+#define ESR2_MASK_STATUS_OFFSET 0x93c
+#define ESR3_MASK_STATUS_OFFSET 0x954
+#define ESR4_MASK_STATUS_OFFSET 0x96c
+
+/* AUD_FMM_BF_ESR_ESRX_MASK_SET_REG_BASE */
+#define ESR0_MASK_SET_OFFSET 0x910
+#define ESR1_MASK_SET_OFFSET 0x928
+#define ESR2_MASK_SET_OFFSET 0x940
+#define ESR3_MASK_SET_OFFSET 0x958
+#define ESR4_MASK_SET_OFFSET 0x970
+
+/* AUD_FMM_BF_ESR_ESRX_MASK_CLEAR_REG_BASE */
+#define ESR0_MASK_CLR_OFFSET 0x914
+#define ESR1_MASK_CLR_OFFSET 0x92c
+#define ESR2_MASK_CLR_OFFSET 0x944
+#define ESR3_MASK_CLR_OFFSET 0x95c
+#define ESR4_MASK_CLR_OFFSET 0x974
+/* Error Status Regs --- End */
+
+#define R5F_ESR0_SHIFT 0 /* esr0 = fifo underflow */
+#define R5F_ESR1_SHIFT 1 /* esr1 = ringbuf underflow */
+#define R5F_ESR2_SHIFT 2 /* esr2 = ringbuf overflow */
+#define R5F_ESR3_SHIFT 3 /* esr3 = freemark */
+#define R5F_ESR4_SHIFT 4 /* esr4 = fullmark */
+
+
+/* Mask for R5F register. Set all relevant interrupt for playback handler */
+#define ANY_PLAYBACK_IRQ (BIT(R5F_ESR0_SHIFT) | \
+ BIT(R5F_ESR1_SHIFT) | \
+ BIT(R5F_ESR3_SHIFT))
+
+/* Mask for R5F register. Set all relevant interrupt for capture handler */
+#define ANY_CAPTURE_IRQ (BIT(R5F_ESR2_SHIFT) | BIT(R5F_ESR4_SHIFT))
+
+/*
+ * PERIOD_BYTES_MIN is the number of bytes to at which the interrupt will tick.
+ * This number should be a multiple of 256. Minimum value is 256
+ */
+#define PERIOD_BYTES_MIN 0x100
+
+static const struct snd_pcm_hardware cygnus_pcm_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+
+ /* A period is basically an interrupt */
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = 0x10000,
+
+ /* period_min/max gives range of approx interrupts per buffer */
+ .periods_min = 2,
+ .periods_max = 8,
+
+ /*
+ * maximum buffer size in bytes = period_bytes_max * periods_max
+ * We allocate this amount of data for each enabled channel
+ */
+ .buffer_bytes_max = 4 * 0x8000,
+};
+
+static u64 cygnus_dma_dmamask = DMA_BIT_MASK(32);
+
+static struct cygnus_aio_port *cygnus_dai_get_dma_data(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+
+ return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream);
+}
+
+static void ringbuf_set_initial(void __iomem *audio_io,
+ struct ringbuf_regs *p_rbuf,
+ bool is_playback,
+ u32 start,
+ u32 periodsize,
+ u32 bufsize)
+{
+ u32 initial_rd;
+ u32 initial_wr;
+ u32 end;
+ u32 fmark_val; /* free or full mark */
+
+ p_rbuf->period_bytes = periodsize;
+ p_rbuf->buf_size = bufsize;
+
+ if (is_playback) {
+ /* Set the pointers to indicate full (flip uppermost bit) */
+ initial_rd = start;
+ initial_wr = initial_rd ^ BIT(31);
+ } else {
+ /* Set the pointers to indicate empty */
+ initial_wr = start;
+ initial_rd = initial_wr;
+ }
+
+ end = start + bufsize - 1;
+
+ /*
+ * The interrupt will fire when free/full mark is *exceeded*
+ * The fmark value must be multiple of PERIOD_BYTES_MIN so set fmark
+ * to be PERIOD_BYTES_MIN less than the period size.
+ */
+ fmark_val = periodsize - PERIOD_BYTES_MIN;
+
+ writel(start, audio_io + p_rbuf->baseaddr);
+ writel(end, audio_io + p_rbuf->endaddr);
+ writel(fmark_val, audio_io + p_rbuf->fmark);
+ writel(initial_rd, audio_io + p_rbuf->rdaddr);
+ writel(initial_wr, audio_io + p_rbuf->wraddr);
+}
+
+static int configure_ringbuf_regs(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ struct ringbuf_regs *p_rbuf;
+ int status = 0;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ /* Map the ssp portnum to a set of ring buffers. */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ p_rbuf = &aio->play_rb_regs;
+
+ switch (aio->portnum) {
+ case 0:
+ *p_rbuf = RINGBUF_REG_PLAYBACK(0);
+ break;
+ case 1:
+ *p_rbuf = RINGBUF_REG_PLAYBACK(2);
+ break;
+ case 2:
+ *p_rbuf = RINGBUF_REG_PLAYBACK(4);
+ break;
+ case 3: /* SPDIF */
+ *p_rbuf = RINGBUF_REG_PLAYBACK(6);
+ break;
+ default:
+ status = -EINVAL;
+ }
+ } else {
+ p_rbuf = &aio->capture_rb_regs;
+
+ switch (aio->portnum) {
+ case 0:
+ *p_rbuf = RINGBUF_REG_CAPTURE(0);
+ break;
+ case 1:
+ *p_rbuf = RINGBUF_REG_CAPTURE(2);
+ break;
+ case 2:
+ *p_rbuf = RINGBUF_REG_CAPTURE(4);
+ break;
+ default:
+ status = -EINVAL;
+ }
+ }
+
+ return status;
+}
+
+static struct ringbuf_regs *get_ringbuf(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ struct ringbuf_regs *p_rbuf = NULL;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ p_rbuf = &aio->play_rb_regs;
+ else
+ p_rbuf = &aio->capture_rb_regs;
+
+ return p_rbuf;
+}
+
+static void enable_intr(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ u32 clear_mask;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ /* The port number maps to the bit position to be cleared */
+ clear_mask = BIT(aio->portnum);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Clear interrupt status before enabling them */
+ writel(clear_mask, aio->cygaud->audio + ESR0_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->cygaud->audio + ESR1_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->cygaud->audio + ESR3_STATUS_CLR_OFFSET);
+ /* Unmask the interrupts of the given port*/
+ writel(clear_mask, aio->cygaud->audio + ESR0_MASK_CLR_OFFSET);
+ writel(clear_mask, aio->cygaud->audio + ESR1_MASK_CLR_OFFSET);
+ writel(clear_mask, aio->cygaud->audio + ESR3_MASK_CLR_OFFSET);
+
+ writel(ANY_PLAYBACK_IRQ,
+ aio->cygaud->audio + INTH_R5F_MASK_CLEAR_OFFSET);
+ } else {
+ writel(clear_mask, aio->cygaud->audio + ESR2_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->cygaud->audio + ESR4_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->cygaud->audio + ESR2_MASK_CLR_OFFSET);
+ writel(clear_mask, aio->cygaud->audio + ESR4_MASK_CLR_OFFSET);
+
+ writel(ANY_CAPTURE_IRQ,
+ aio->cygaud->audio + INTH_R5F_MASK_CLEAR_OFFSET);
+ }
+
+}
+
+static void disable_intr(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct cygnus_aio_port *aio;
+ u32 set_mask;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum);
+
+ /* The port number maps to the bit position to be set */
+ set_mask = BIT(aio->portnum);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Mask the interrupts of the given port*/
+ writel(set_mask, aio->cygaud->audio + ESR0_MASK_SET_OFFSET);
+ writel(set_mask, aio->cygaud->audio + ESR1_MASK_SET_OFFSET);
+ writel(set_mask, aio->cygaud->audio + ESR3_MASK_SET_OFFSET);
+ } else {
+ writel(set_mask, aio->cygaud->audio + ESR2_MASK_SET_OFFSET);
+ writel(set_mask, aio->cygaud->audio + ESR4_MASK_SET_OFFSET);
+ }
+
+}
+
+static int cygnus_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ enable_intr(substream);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ disable_intr(substream);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static void cygnus_pcm_period_elapsed(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ struct ringbuf_regs *p_rbuf = NULL;
+ u32 regval;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ p_rbuf = get_ringbuf(substream);
+
+ /*
+ * If free/full mark interrupt occurs, provide timestamp
+ * to ALSA and update appropriate idx by period_bytes
+ */
+ snd_pcm_period_elapsed(substream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Set the ring buffer to full */
+ regval = readl(aio->cygaud->audio + p_rbuf->rdaddr);
+ regval = regval ^ BIT(31);
+ writel(regval, aio->cygaud->audio + p_rbuf->wraddr);
+ } else {
+ /* Set the ring buffer to empty */
+ regval = readl(aio->cygaud->audio + p_rbuf->wraddr);
+ writel(regval, aio->cygaud->audio + p_rbuf->rdaddr);
+ }
+}
+
+/*
+ * ESR0/1/3 status Description
+ * 0x1 I2S0_out port caused interrupt
+ * 0x2 I2S1_out port caused interrupt
+ * 0x4 I2S2_out port caused interrupt
+ * 0x8 SPDIF_out port caused interrupt
+ */
+static void handle_playback_irq(struct cygnus_audio *cygaud)
+{
+ void __iomem *audio_io;
+ u32 port;
+ u32 esr_status0, esr_status1, esr_status3;
+
+ audio_io = cygaud->audio;
+
+ /*
+ * ESR status gets updates with/without interrupts enabled.
+ * So, check the ESR mask, which provides interrupt enable/
+ * disable status and use it to determine which ESR status
+ * should be serviced.
+ */
+ esr_status0 = readl(audio_io + ESR0_STATUS_OFFSET);
+ esr_status0 &= ~readl(audio_io + ESR0_MASK_STATUS_OFFSET);
+ esr_status1 = readl(audio_io + ESR1_STATUS_OFFSET);
+ esr_status1 &= ~readl(audio_io + ESR1_MASK_STATUS_OFFSET);
+ esr_status3 = readl(audio_io + ESR3_STATUS_OFFSET);
+ esr_status3 &= ~readl(audio_io + ESR3_MASK_STATUS_OFFSET);
+
+ for (port = 0; port < CYGNUS_MAX_PLAYBACK_PORTS; port++) {
+ u32 esrmask = BIT(port);
+
+ /*
+ * Ringbuffer or FIFO underflow
+ * If we get this interrupt then, it is also true that we have
+ * not yet responded to the freemark interrupt.
+ * Log a debug message. The freemark handler below will
+ * handle getting everything going again.
+ */
+ if ((esrmask & esr_status1) || (esrmask & esr_status0)) {
+ dev_dbg(cygaud->dev,
+ "Underrun: esr0=0x%x, esr1=0x%x esr3=0x%x\n",
+ esr_status0, esr_status1, esr_status3);
+ }
+
+ /*
+ * Freemark is hit. This is the normal interrupt.
+ * In typical operation the read and write regs will be equal
+ */
+ if (esrmask & esr_status3) {
+ struct snd_pcm_substream *playstr;
+
+ playstr = cygaud->portinfo[port].play_stream;
+ cygnus_pcm_period_elapsed(playstr);
+ }
+ }
+
+ /* Clear ESR interrupt */
+ writel(esr_status0, audio_io + ESR0_STATUS_CLR_OFFSET);
+ writel(esr_status1, audio_io + ESR1_STATUS_CLR_OFFSET);
+ writel(esr_status3, audio_io + ESR3_STATUS_CLR_OFFSET);
+ /* Rearm freemark logic by writing 1 to the correct bit */
+ writel(esr_status3, audio_io + BF_REARM_FREE_MARK_OFFSET);
+}
+
+/*
+ * ESR2/4 status Description
+ * 0x1 I2S0_in port caused interrupt
+ * 0x2 I2S1_in port caused interrupt
+ * 0x4 I2S2_in port caused interrupt
+ */
+static void handle_capture_irq(struct cygnus_audio *cygaud)
+{
+ void __iomem *audio_io;
+ u32 port;
+ u32 esr_status2, esr_status4;
+
+ audio_io = cygaud->audio;
+
+ /*
+ * ESR status gets updates with/without interrupts enabled.
+ * So, check the ESR mask, which provides interrupt enable/
+ * disable status and use it to determine which ESR status
+ * should be serviced.
+ */
+ esr_status2 = readl(audio_io + ESR2_STATUS_OFFSET);
+ esr_status2 &= ~readl(audio_io + ESR2_MASK_STATUS_OFFSET);
+ esr_status4 = readl(audio_io + ESR4_STATUS_OFFSET);
+ esr_status4 &= ~readl(audio_io + ESR4_MASK_STATUS_OFFSET);
+
+ for (port = 0; port < CYGNUS_MAX_CAPTURE_PORTS; port++) {
+ u32 esrmask = BIT(port);
+
+ /*
+ * Ringbuffer or FIFO overflow
+ * If we get this interrupt then, it is also true that we have
+ * not yet responded to the fullmark interrupt.
+ * Log a debug message. The fullmark handler below will
+ * handle getting everything going again.
+ */
+ if (esrmask & esr_status2)
+ dev_dbg(cygaud->dev,
+ "Overflow: esr2=0x%x\n", esr_status2);
+
+ if (esrmask & esr_status4) {
+ struct snd_pcm_substream *capstr;
+
+ capstr = cygaud->portinfo[port].capture_stream;
+ cygnus_pcm_period_elapsed(capstr);
+ }
+ }
+
+ writel(esr_status2, audio_io + ESR2_STATUS_CLR_OFFSET);
+ writel(esr_status4, audio_io + ESR4_STATUS_CLR_OFFSET);
+ /* Rearm fullmark logic by writing 1 to the correct bit */
+ writel(esr_status4, audio_io + BF_REARM_FULL_MARK_OFFSET);
+}
+
+static irqreturn_t cygnus_dma_irq(int irq, void *data)
+{
+ u32 r5_status;
+ struct cygnus_audio *cygaud = data;
+
+ /*
+ * R5 status bits Description
+ * 0 ESR0 (playback FIFO interrupt)
+ * 1 ESR1 (playback rbuf interrupt)
+ * 2 ESR2 (capture rbuf interrupt)
+ * 3 ESR3 (Freemark play. interrupt)
+ * 4 ESR4 (Fullmark capt. interrupt)
+ */
+ r5_status = readl(cygaud->audio + INTH_R5F_STATUS_OFFSET);
+
+ if (!(r5_status & (ANY_PLAYBACK_IRQ | ANY_CAPTURE_IRQ)))
+ return IRQ_NONE;
+
+ /* If playback interrupt happened */
+ if (ANY_PLAYBACK_IRQ & r5_status) {
+ handle_playback_irq(cygaud);
+ writel(ANY_PLAYBACK_IRQ & r5_status,
+ cygaud->audio + INTH_R5F_CLEAR_OFFSET);
+ }
+
+ /* If capture interrupt happened */
+ if (ANY_CAPTURE_IRQ & r5_status) {
+ handle_capture_irq(cygaud);
+ writel(ANY_CAPTURE_IRQ & r5_status,
+ cygaud->audio + INTH_R5F_CLEAR_OFFSET);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static int cygnus_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cygnus_aio_port *aio;
+ int ret;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ if (!aio)
+ return -ENODEV;
+
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw);
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, PERIOD_BYTES_MIN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, PERIOD_BYTES_MIN);
+ if (ret < 0)
+ return ret;
+ /*
+ * Keep track of which substream belongs to which port.
+ * This info is needed by snd_pcm_period_elapsed() in irq_handler
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aio->play_stream = substream;
+ else
+ aio->capture_stream = substream;
+
+ return 0;
+}
+
+static int cygnus_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct cygnus_aio_port *aio;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aio->play_stream = NULL;
+ else
+ aio->capture_stream = NULL;
+
+ if (!aio->play_stream && !aio->capture_stream)
+ dev_dbg(rtd->cpu_dai->dev, "freed port %d\n", aio->portnum);
+
+ return 0;
+}
+
+static int cygnus_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 cygnus_aio_port *aio;
+ int ret = 0;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ return ret;
+}
+
+static int cygnus_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct cygnus_aio_port *aio;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+static int cygnus_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cygnus_aio_port *aio;
+ unsigned long bufsize, periodsize;
+ int ret = 0;
+ bool is_play;
+ u32 start;
+ struct ringbuf_regs *p_rbuf = NULL;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ bufsize = snd_pcm_lib_buffer_bytes(substream);
+ periodsize = snd_pcm_lib_period_bytes(substream);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n",
+ __func__, bufsize, periodsize);
+
+ configure_ringbuf_regs(substream);
+
+ p_rbuf = get_ringbuf(substream);
+
+ start = runtime->dma_addr;
+
+ is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0;
+
+ ringbuf_set_initial(aio->cygaud->audio, p_rbuf, is_play, start,
+ periodsize, bufsize);
+
+ return ret;
+}
+
+static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ unsigned int res = 0, cur = 0, base = 0;
+ struct ringbuf_regs *p_rbuf = NULL;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ /*
+ * Get the offset of the current read (for playack) or write
+ * index (for capture). Report this value back to the asoc framework.
+ */
+ p_rbuf = get_ringbuf(substream);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cur = readl(aio->cygaud->audio + p_rbuf->rdaddr);
+ else
+ cur = readl(aio->cygaud->audio + p_rbuf->wraddr);
+
+ base = readl(aio->cygaud->audio + p_rbuf->baseaddr);
+
+ /*
+ * Mask off the MSB of the rdaddr,wraddr and baseaddr
+ * since MSB is not part of the address
+ */
+ res = (cur & 0x7fffffff) - (base & 0x7fffffff);
+
+ return bytes_to_frames(substream->runtime, res);
+}
+
+static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size;
+
+ size = cygnus_pcm_hw.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%zx @ %pK\n",
+ __func__, size, buf->area);
+
+ if (!buf->area) {
+ dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__);
+ return -ENOMEM;
+ }
+ buf->bytes = size;
+
+ return 0;
+}
+
+
+static const struct snd_pcm_ops cygnus_pcm_ops = {
+ .open = cygnus_pcm_open,
+ .close = cygnus_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = cygnus_pcm_hw_params,
+ .hw_free = cygnus_pcm_hw_free,
+ .prepare = cygnus_pcm_prepare,
+ .trigger = cygnus_pcm_trigger,
+ .pointer = cygnus_pcm_pointer,
+};
+
+static void cygnus_dma_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (substream) {
+ buf = &substream->dma_buffer;
+ if (buf->area) {
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+ }
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ if (substream) {
+ buf = &substream->dma_buffer;
+ if (buf->area) {
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+ }
+}
+
+static int cygnus_dma_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &cygnus_dma_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = cygnus_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = cygnus_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret) {
+ cygnus_dma_free_dma_buffers(pcm);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver cygnus_soc_platform = {
+ .ops = &cygnus_pcm_ops,
+ .pcm_new = cygnus_dma_new,
+ .pcm_free = cygnus_dma_free_dma_buffers,
+};
+
+int cygnus_soc_platform_register(struct device *dev,
+ struct cygnus_audio *cygaud)
+{
+ int rc = 0;
+
+ dev_dbg(dev, "%s Enter\n", __func__);
+
+ rc = devm_request_irq(dev, cygaud->irq_num, cygnus_dma_irq,
+ IRQF_SHARED, "cygnus-audio", cygaud);
+ if (rc) {
+ dev_err(dev, "%s request_irq error %d\n", __func__, rc);
+ return rc;
+ }
+
+ rc = snd_soc_register_platform(dev, &cygnus_soc_platform);
+ if (rc) {
+ dev_err(dev, "%s failed\n", __func__);
+ return rc;
+ }
+
+ return 0;
+}
+
+int cygnus_soc_platform_unregister(struct device *dev)
+{
+ snd_soc_unregister_platform(dev);
+
+ return 0;
+}
+
+MODULE_LICENSE("GPL v2");
+MODULE_AUTHOR("Broadcom");
+MODULE_DESCRIPTION("Cygnus ASoC PCM module");
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
new file mode 100644
index 000000000000..e710bb0c5637
--- /dev/null
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -0,0 +1,1529 @@
+/*
+ * Copyright (C) 2014-2015 Broadcom Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any
+ * kind, whether express or implied; without even the implied warranty
+ * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "cygnus-ssp.h"
+
+#define DEFAULT_VCO 1354750204
+
+#define CYGNUS_TDM_RATE \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define CAPTURE_FCI_ID_BASE 0x180
+#define CYGNUS_SSP_TRISTATE_MASK 0x001fff
+#define CYGNUS_PLLCLKSEL_MASK 0xf
+
+/* Used with stream_on field to indicate which streams are active */
+#define PLAYBACK_STREAM_MASK BIT(0)
+#define CAPTURE_STREAM_MASK BIT(1)
+
+#define I2S_STREAM_CFG_MASK 0xff003ff
+#define I2S_CAP_STREAM_CFG_MASK 0xf0
+#define SPDIF_STREAM_CFG_MASK 0x3ff
+#define CH_GRP_STEREO 0x1
+
+/* Begin register offset defines */
+#define AUD_MISC_SEROUT_OE_REG_BASE 0x01c
+#define AUD_MISC_SEROUT_SPDIF_OE 12
+#define AUD_MISC_SEROUT_MCLK_OE 3
+#define AUD_MISC_SEROUT_LRCK_OE 2
+#define AUD_MISC_SEROUT_SCLK_OE 1
+#define AUD_MISC_SEROUT_SDAT_OE 0
+
+/* AUD_FMM_BF_CTRL_xxx regs */
+#define BF_DST_CFG0_OFFSET 0x100
+#define BF_DST_CFG1_OFFSET 0x104
+#define BF_DST_CFG2_OFFSET 0x108
+
+#define BF_DST_CTRL0_OFFSET 0x130
+#define BF_DST_CTRL1_OFFSET 0x134
+#define BF_DST_CTRL2_OFFSET 0x138
+
+#define BF_SRC_CFG0_OFFSET 0x148
+#define BF_SRC_CFG1_OFFSET 0x14c
+#define BF_SRC_CFG2_OFFSET 0x150
+#define BF_SRC_CFG3_OFFSET 0x154
+
+#define BF_SRC_CTRL0_OFFSET 0x1c0
+#define BF_SRC_CTRL1_OFFSET 0x1c4
+#define BF_SRC_CTRL2_OFFSET 0x1c8
+#define BF_SRC_CTRL3_OFFSET 0x1cc
+
+#define BF_SRC_GRP0_OFFSET 0x1fc
+#define BF_SRC_GRP1_OFFSET 0x200
+#define BF_SRC_GRP2_OFFSET 0x204
+#define BF_SRC_GRP3_OFFSET 0x208
+
+#define BF_SRC_GRP_EN_OFFSET 0x320
+#define BF_SRC_GRP_FLOWON_OFFSET 0x324
+#define BF_SRC_GRP_SYNC_DIS_OFFSET 0x328
+
+/* AUD_FMM_IOP_OUT_I2S_xxx regs */
+#define OUT_I2S_0_STREAM_CFG_OFFSET 0xa00
+#define OUT_I2S_0_CFG_OFFSET 0xa04
+#define OUT_I2S_0_MCLK_CFG_OFFSET 0xa0c
+
+#define OUT_I2S_1_STREAM_CFG_OFFSET 0xa40
+#define OUT_I2S_1_CFG_OFFSET 0xa44
+#define OUT_I2S_1_MCLK_CFG_OFFSET 0xa4c
+
+#define OUT_I2S_2_STREAM_CFG_OFFSET 0xa80
+#define OUT_I2S_2_CFG_OFFSET 0xa84
+#define OUT_I2S_2_MCLK_CFG_OFFSET 0xa8c
+
+/* AUD_FMM_IOP_OUT_SPDIF_xxx regs */
+#define SPDIF_STREAM_CFG_OFFSET 0xac0
+#define SPDIF_CTRL_OFFSET 0xac4
+#define SPDIF_FORMAT_CFG_OFFSET 0xad8
+#define SPDIF_MCLK_CFG_OFFSET 0xadc
+
+/* AUD_FMM_IOP_PLL_0_xxx regs */
+#define IOP_PLL_0_MACRO_OFFSET 0xb00
+#define IOP_PLL_0_MDIV_Ch0_OFFSET 0xb14
+#define IOP_PLL_0_MDIV_Ch1_OFFSET 0xb18
+#define IOP_PLL_0_MDIV_Ch2_OFFSET 0xb1c
+
+#define IOP_PLL_0_ACTIVE_MDIV_Ch0_OFFSET 0xb30
+#define IOP_PLL_0_ACTIVE_MDIV_Ch1_OFFSET 0xb34
+#define IOP_PLL_0_ACTIVE_MDIV_Ch2_OFFSET 0xb38
+
+/* AUD_FMM_IOP_xxx regs */
+#define IOP_PLL_0_CONTROL_OFFSET 0xb04
+#define IOP_PLL_0_USER_NDIV_OFFSET 0xb08
+#define IOP_PLL_0_ACTIVE_NDIV_OFFSET 0xb20
+#define IOP_PLL_0_RESET_OFFSET 0xb5c
+
+/* AUD_FMM_IOP_IN_I2S_xxx regs */
+#define IN_I2S_0_STREAM_CFG_OFFSET 0x00
+#define IN_I2S_0_CFG_OFFSET 0x04
+#define IN_I2S_1_STREAM_CFG_OFFSET 0x40
+#define IN_I2S_1_CFG_OFFSET 0x44
+#define IN_I2S_2_STREAM_CFG_OFFSET 0x80
+#define IN_I2S_2_CFG_OFFSET 0x84
+
+/* AUD_FMM_IOP_MISC_xxx regs */
+#define IOP_SW_INIT_LOGIC 0x1c0
+
+/* End register offset defines */
+
+
+/* AUD_FMM_IOP_OUT_I2S_x_MCLK_CFG_0_REG */
+#define I2S_OUT_MCLKRATE_SHIFT 16
+
+/* AUD_FMM_IOP_OUT_I2S_x_MCLK_CFG_REG */
+#define I2S_OUT_PLLCLKSEL_SHIFT 0
+
+/* AUD_FMM_IOP_OUT_I2S_x_STREAM_CFG */
+#define I2S_OUT_STREAM_ENA 31
+#define I2S_OUT_STREAM_CFG_GROUP_ID 20
+#define I2S_OUT_STREAM_CFG_CHANNEL_GROUPING 24
+
+/* AUD_FMM_IOP_IN_I2S_x_CAP */
+#define I2S_IN_STREAM_CFG_CAP_ENA 31
+#define I2S_IN_STREAM_CFG_0_GROUP_ID 4
+
+/* AUD_FMM_IOP_OUT_I2S_x_I2S_CFG_REG */
+#define I2S_OUT_CFGX_CLK_ENA 0
+#define I2S_OUT_CFGX_DATA_ENABLE 1
+#define I2S_OUT_CFGX_DATA_ALIGNMENT 6
+#define I2S_OUT_CFGX_BITS_PER_SLOT 13
+#define I2S_OUT_CFGX_VALID_SLOT 14
+#define I2S_OUT_CFGX_FSYNC_WIDTH 18
+#define I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32 26
+#define I2S_OUT_CFGX_SLAVE_MODE 30
+#define I2S_OUT_CFGX_TDM_MODE 31
+
+/* AUD_FMM_BF_CTRL_SOURCECH_CFGx_REG */
+#define BF_SRC_CFGX_SFIFO_ENA 0
+#define BF_SRC_CFGX_BUFFER_PAIR_ENABLE 1
+#define BF_SRC_CFGX_SAMPLE_CH_MODE 2
+#define BF_SRC_CFGX_SFIFO_SZ_DOUBLE 5
+#define BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY 10
+#define BF_SRC_CFGX_BIT_RES 20
+#define BF_SRC_CFGX_PROCESS_SEQ_ID_VALID 31
+
+/* AUD_FMM_BF_CTRL_DESTCH_CFGx_REG */
+#define BF_DST_CFGX_CAP_ENA 0
+#define BF_DST_CFGX_BUFFER_PAIR_ENABLE 1
+#define BF_DST_CFGX_DFIFO_SZ_DOUBLE 2
+#define BF_DST_CFGX_NOT_PAUSE_WHEN_FULL 11
+#define BF_DST_CFGX_FCI_ID 12
+#define BF_DST_CFGX_CAP_MODE 24
+#define BF_DST_CFGX_PROC_SEQ_ID_VALID 31
+
+/* AUD_FMM_IOP_OUT_SPDIF_xxx */
+#define SPDIF_0_OUT_DITHER_ENA 3
+#define SPDIF_0_OUT_STREAM_ENA 31
+
+/* AUD_FMM_IOP_PLL_0_USER */
+#define IOP_PLL_0_USER_NDIV_FRAC 10
+
+/* AUD_FMM_IOP_PLL_0_ACTIVE */
+#define IOP_PLL_0_ACTIVE_NDIV_FRAC 10
+
+
+#define INIT_SSP_REGS(num) (struct cygnus_ssp_regs){ \
+ .i2s_stream_cfg = OUT_I2S_ ##num## _STREAM_CFG_OFFSET, \
+ .i2s_cap_stream_cfg = IN_I2S_ ##num## _STREAM_CFG_OFFSET, \
+ .i2s_cfg = OUT_I2S_ ##num## _CFG_OFFSET, \
+ .i2s_cap_cfg = IN_I2S_ ##num## _CFG_OFFSET, \
+ .i2s_mclk_cfg = OUT_I2S_ ##num## _MCLK_CFG_OFFSET, \
+ .bf_destch_ctrl = BF_DST_CTRL ##num## _OFFSET, \
+ .bf_destch_cfg = BF_DST_CFG ##num## _OFFSET, \
+ .bf_sourcech_ctrl = BF_SRC_CTRL ##num## _OFFSET, \
+ .bf_sourcech_cfg = BF_SRC_CFG ##num## _OFFSET, \
+ .bf_sourcech_grp = BF_SRC_GRP ##num## _OFFSET \
+}
+
+struct pll_macro_entry {
+ u32 mclk;
+ u32 pll_ch_num;
+};
+
+/*
+ * PLL has 3 output channels (1x, 2x, and 4x). Below are
+ * the common MCLK frequencies used by audio driver
+ */
+static const struct pll_macro_entry pll_predef_mclk[] = {
+ { 4096000, 0},
+ { 8192000, 1},
+ {16384000, 2},
+
+ { 5644800, 0},
+ {11289600, 1},
+ {22579200, 2},
+
+ { 6144000, 0},
+ {12288000, 1},
+ {24576000, 2},
+
+ {12288000, 0},
+ {24576000, 1},
+ {49152000, 2},
+
+ {22579200, 0},
+ {45158400, 1},
+ {90316800, 2},
+
+ {24576000, 0},
+ {49152000, 1},
+ {98304000, 2},
+};
+
+/* List of valid frame sizes for tdm mode */
+static const int ssp_valid_tdm_framesize[] = {32, 64, 128, 256, 512};
+
+/*
+ * Use this relationship to derive the sampling rate (lrclk)
+ * lrclk = (mclk) / ((2*mclk_to_sclk_ratio) * (32 * SCLK))).
+ *
+ * Use mclk and pll_ch from the table above
+ *
+ * Valid SCLK = 0/1/2/4/8/12
+ *
+ * mclk_to_sclk_ratio = number of MCLK per SCLK. Division is twice the
+ * value programmed in this field.
+ * Valid mclk_to_sclk_ratio = 1 through to 15
+ *
+ * eg: To set lrclk = 48khz, set mclk = 12288000, mclk_to_sclk_ratio = 2,
+ * SCLK = 64
+ */
+struct _ssp_clk_coeff {
+ u32 mclk;
+ u32 sclk_rate;
+ u32 rate;
+ u32 mclk_rate;
+};
+
+static const struct _ssp_clk_coeff ssp_clk_coeff[] = {
+ { 4096000, 32, 16000, 4},
+ { 4096000, 32, 32000, 2},
+ { 4096000, 64, 8000, 4},
+ { 4096000, 64, 16000, 2},
+ { 4096000, 64, 32000, 1},
+ { 4096000, 128, 8000, 2},
+ { 4096000, 128, 16000, 1},
+ { 4096000, 256, 8000, 1},
+
+ { 6144000, 32, 16000, 6},
+ { 6144000, 32, 32000, 3},
+ { 6144000, 32, 48000, 2},
+ { 6144000, 32, 96000, 1},
+ { 6144000, 64, 8000, 6},
+ { 6144000, 64, 16000, 3},
+ { 6144000, 64, 48000, 1},
+ { 6144000, 128, 8000, 3},
+
+ { 8192000, 32, 32000, 4},
+ { 8192000, 64, 16000, 4},
+ { 8192000, 64, 32000, 2},
+ { 8192000, 128, 8000, 4},
+ { 8192000, 128, 16000, 2},
+ { 8192000, 128, 32000, 1},
+ { 8192000, 256, 8000, 2},
+ { 8192000, 256, 16000, 1},
+ { 8192000, 512, 8000, 1},
+
+ {12288000, 32, 32000, 6},
+ {12288000, 32, 48000, 4},
+ {12288000, 32, 96000, 2},
+ {12288000, 32, 192000, 1},
+ {12288000, 64, 16000, 6},
+ {12288000, 64, 32000, 3},
+ {12288000, 64, 48000, 2},
+ {12288000, 64, 96000, 1},
+ {12288000, 128, 8000, 6},
+ {12288000, 128, 16000, 3},
+ {12288000, 128, 48000, 1},
+ {12288000, 256, 8000, 3},
+
+ {16384000, 64, 32000, 4},
+ {16384000, 128, 16000, 4},
+ {16384000, 128, 32000, 2},
+ {16384000, 256, 8000, 4},
+ {16384000, 256, 16000, 2},
+ {16384000, 256, 32000, 1},
+ {16384000, 512, 8000, 2},
+ {16384000, 512, 16000, 1},
+
+ {24576000, 32, 96000, 4},
+ {24576000, 32, 192000, 2},
+ {24576000, 64, 32000, 6},
+ {24576000, 64, 48000, 4},
+ {24576000, 64, 96000, 2},
+ {24576000, 64, 192000, 1},
+ {24576000, 128, 16000, 6},
+ {24576000, 128, 32000, 3},
+ {24576000, 128, 48000, 2},
+ {24576000, 256, 8000, 6},
+ {24576000, 256, 16000, 3},
+ {24576000, 256, 48000, 1},
+ {24576000, 512, 8000, 3},
+
+ {49152000, 32, 192000, 4},
+ {49152000, 64, 96000, 4},
+ {49152000, 64, 192000, 2},
+ {49152000, 128, 32000, 6},
+ {49152000, 128, 48000, 4},
+ {49152000, 128, 96000, 2},
+ {49152000, 128, 192000, 1},
+ {49152000, 256, 16000, 6},
+ {49152000, 256, 32000, 3},
+ {49152000, 256, 48000, 2},
+ {49152000, 256, 96000, 1},
+ {49152000, 512, 8000, 6},
+ {49152000, 512, 16000, 3},
+ {49152000, 512, 48000, 1},
+
+ { 5644800, 32, 22050, 4},
+ { 5644800, 32, 44100, 2},
+ { 5644800, 32, 88200, 1},
+ { 5644800, 64, 11025, 4},
+ { 5644800, 64, 22050, 2},
+ { 5644800, 64, 44100, 1},
+
+ {11289600, 32, 44100, 4},
+ {11289600, 32, 88200, 2},
+ {11289600, 32, 176400, 1},
+ {11289600, 64, 22050, 4},
+ {11289600, 64, 44100, 2},
+ {11289600, 64, 88200, 1},
+ {11289600, 128, 11025, 4},
+ {11289600, 128, 22050, 2},
+ {11289600, 128, 44100, 1},
+
+ {22579200, 32, 88200, 4},
+ {22579200, 32, 176400, 2},
+ {22579200, 64, 44100, 4},
+ {22579200, 64, 88200, 2},
+ {22579200, 64, 176400, 1},
+ {22579200, 128, 22050, 4},
+ {22579200, 128, 44100, 2},
+ {22579200, 128, 88200, 1},
+ {22579200, 256, 11025, 4},
+ {22579200, 256, 22050, 2},
+ {22579200, 256, 44100, 1},
+
+ {45158400, 32, 176400, 4},
+ {45158400, 64, 88200, 4},
+ {45158400, 64, 176400, 2},
+ {45158400, 128, 44100, 4},
+ {45158400, 128, 88200, 2},
+ {45158400, 128, 176400, 1},
+ {45158400, 256, 22050, 4},
+ {45158400, 256, 44100, 2},
+ {45158400, 256, 88200, 1},
+ {45158400, 512, 11025, 4},
+ {45158400, 512, 22050, 2},
+ {45158400, 512, 44100, 1},
+};
+
+static struct cygnus_aio_port *cygnus_dai_get_portinfo(struct snd_soc_dai *dai)
+{
+ struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai);
+
+ return &cygaud->portinfo[dai->id];
+}
+
+static int audio_ssp_init_portregs(struct cygnus_aio_port *aio)
+{
+ u32 value, fci_id;
+ int status = 0;
+
+ switch (aio->port_type) {
+ case PORT_TDM:
+ value = readl(aio->cygaud->audio + aio->regs.i2s_stream_cfg);
+ value &= ~I2S_STREAM_CFG_MASK;
+
+ /* Set Group ID */
+ writel(aio->portnum,
+ aio->cygaud->audio + aio->regs.bf_sourcech_grp);
+
+ /* Configure the AUD_FMM_IOP_OUT_I2S_x_STREAM_CFG reg */
+ value |= aio->portnum << I2S_OUT_STREAM_CFG_GROUP_ID;
+ value |= aio->portnum; /* FCI ID is the port num */
+ value |= CH_GRP_STEREO << I2S_OUT_STREAM_CFG_CHANNEL_GROUPING;
+ writel(value, aio->cygaud->audio + aio->regs.i2s_stream_cfg);
+
+ /* Configure the AUD_FMM_BF_CTRL_SOURCECH_CFGX reg */
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY);
+ value |= BIT(BF_SRC_CFGX_SFIFO_SZ_DOUBLE);
+ value |= BIT(BF_SRC_CFGX_PROCESS_SEQ_ID_VALID);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+
+ /* Configure the AUD_FMM_IOP_IN_I2S_x_CAP_STREAM_CFG_0 reg */
+ value = readl(aio->cygaud->i2s_in +
+ aio->regs.i2s_cap_stream_cfg);
+ value &= ~I2S_CAP_STREAM_CFG_MASK;
+ value |= aio->portnum << I2S_IN_STREAM_CFG_0_GROUP_ID;
+ writel(value, aio->cygaud->i2s_in +
+ aio->regs.i2s_cap_stream_cfg);
+
+ /* Configure the AUD_FMM_BF_CTRL_DESTCH_CFGX_REG_BASE reg */
+ fci_id = CAPTURE_FCI_ID_BASE + aio->portnum;
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_destch_cfg);
+ value |= BIT(BF_DST_CFGX_DFIFO_SZ_DOUBLE);
+ value &= ~BIT(BF_DST_CFGX_NOT_PAUSE_WHEN_FULL);
+ value |= (fci_id << BF_DST_CFGX_FCI_ID);
+ value |= BIT(BF_DST_CFGX_PROC_SEQ_ID_VALID);
+ writel(value, aio->cygaud->audio + aio->regs.bf_destch_cfg);
+
+ /* Enable the transmit pin for this port */
+ value = readl(aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+ value &= ~BIT((aio->portnum * 4) + AUD_MISC_SEROUT_SDAT_OE);
+ writel(value, aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+ break;
+ case PORT_SPDIF:
+ writel(aio->portnum, aio->cygaud->audio + BF_SRC_GRP3_OFFSET);
+
+ value = readl(aio->cygaud->audio + SPDIF_CTRL_OFFSET);
+ value |= BIT(SPDIF_0_OUT_DITHER_ENA);
+ writel(value, aio->cygaud->audio + SPDIF_CTRL_OFFSET);
+
+ /* Enable and set the FCI ID for the SPDIF channel */
+ value = readl(aio->cygaud->audio + SPDIF_STREAM_CFG_OFFSET);
+ value &= ~SPDIF_STREAM_CFG_MASK;
+ value |= aio->portnum; /* FCI ID is the port num */
+ value |= BIT(SPDIF_0_OUT_STREAM_ENA);
+ writel(value, aio->cygaud->audio + SPDIF_STREAM_CFG_OFFSET);
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY);
+ value |= BIT(BF_SRC_CFGX_SFIFO_SZ_DOUBLE);
+ value |= BIT(BF_SRC_CFGX_PROCESS_SEQ_ID_VALID);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+
+ /* Enable the spdif output pin */
+ value = readl(aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+ value &= ~BIT(AUD_MISC_SEROUT_SPDIF_OE);
+ writel(value, aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+ break;
+ default:
+ dev_err(aio->cygaud->dev, "Port not supported\n");
+ status = -EINVAL;
+ }
+
+ return status;
+}
+
+static void audio_ssp_in_enable(struct cygnus_aio_port *aio)
+{
+ u32 value;
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_destch_cfg);
+ value |= BIT(BF_DST_CFGX_CAP_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.bf_destch_cfg);
+
+ writel(0x1, aio->cygaud->audio + aio->regs.bf_destch_ctrl);
+
+ value = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
+ value |= BIT(I2S_OUT_CFGX_CLK_ENA);
+ value |= BIT(I2S_OUT_CFGX_DATA_ENABLE);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_cfg);
+
+ value = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg);
+ value |= BIT(I2S_IN_STREAM_CFG_CAP_ENA);
+ writel(value, aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg);
+
+ aio->streams_on |= CAPTURE_STREAM_MASK;
+}
+
+static void audio_ssp_in_disable(struct cygnus_aio_port *aio)
+{
+ u32 value;
+
+ value = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg);
+ value &= ~BIT(I2S_IN_STREAM_CFG_CAP_ENA);
+ writel(value, aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg);
+
+ aio->streams_on &= ~CAPTURE_STREAM_MASK;
+
+ /* If both playback and capture are off */
+ if (!aio->streams_on) {
+ value = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
+ value &= ~BIT(I2S_OUT_CFGX_CLK_ENA);
+ value &= ~BIT(I2S_OUT_CFGX_DATA_ENABLE);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_cfg);
+ }
+
+ writel(0x0, aio->cygaud->audio + aio->regs.bf_destch_ctrl);
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_destch_cfg);
+ value &= ~BIT(BF_DST_CFGX_CAP_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.bf_destch_cfg);
+}
+
+static int audio_ssp_out_enable(struct cygnus_aio_port *aio)
+{
+ u32 value;
+ int status = 0;
+
+ switch (aio->port_type) {
+ case PORT_TDM:
+ value = readl(aio->cygaud->audio + aio->regs.i2s_stream_cfg);
+ value |= BIT(I2S_OUT_STREAM_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_stream_cfg);
+
+ writel(1, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
+ value |= BIT(I2S_OUT_CFGX_CLK_ENA);
+ value |= BIT(I2S_OUT_CFGX_DATA_ENABLE);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_cfg);
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value |= BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+
+ aio->streams_on |= PLAYBACK_STREAM_MASK;
+ break;
+ case PORT_SPDIF:
+ value = readl(aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET);
+ value |= 0x3;
+ writel(value, aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET);
+
+ writel(1, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value |= BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ break;
+ default:
+ dev_err(aio->cygaud->dev,
+ "Port not supported %d\n", aio->portnum);
+ status = -EINVAL;
+ }
+
+ return status;
+}
+
+static int audio_ssp_out_disable(struct cygnus_aio_port *aio)
+{
+ u32 value;
+ int status = 0;
+
+ switch (aio->port_type) {
+ case PORT_TDM:
+ aio->streams_on &= ~PLAYBACK_STREAM_MASK;
+
+ /* If both playback and capture are off */
+ if (!aio->streams_on) {
+ value = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
+ value &= ~BIT(I2S_OUT_CFGX_CLK_ENA);
+ value &= ~BIT(I2S_OUT_CFGX_DATA_ENABLE);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_cfg);
+ }
+
+ /* set group_sync_dis = 1 */
+ value = readl(aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET);
+ value |= BIT(aio->portnum);
+ writel(value, aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET);
+
+ writel(0, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+
+ /* set group_sync_dis = 0 */
+ value = readl(aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET);
+ value &= ~BIT(aio->portnum);
+ writel(value, aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET);
+
+ value = readl(aio->cygaud->audio + aio->regs.i2s_stream_cfg);
+ value &= ~BIT(I2S_OUT_STREAM_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_stream_cfg);
+
+ /* IOP SW INIT on OUT_I2S_x */
+ value = readl(aio->cygaud->i2s_in + IOP_SW_INIT_LOGIC);
+ value |= BIT(aio->portnum);
+ writel(value, aio->cygaud->i2s_in + IOP_SW_INIT_LOGIC);
+ value &= ~BIT(aio->portnum);
+ writel(value, aio->cygaud->i2s_in + IOP_SW_INIT_LOGIC);
+ break;
+ case PORT_SPDIF:
+ value = readl(aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET);
+ value &= ~0x3;
+ writel(value, aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET);
+ writel(0, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ break;
+ default:
+ dev_err(aio->cygaud->dev,
+ "Port not supported %d\n", aio->portnum);
+ status = -EINVAL;
+ }
+
+ return status;
+}
+
+static int pll_configure_mclk(struct cygnus_audio *cygaud, u32 mclk,
+ struct cygnus_aio_port *aio)
+{
+ int i = 0, error;
+ bool found = false;
+ const struct pll_macro_entry *p_entry;
+ struct clk *ch_clk;
+
+ for (i = 0; i < ARRAY_SIZE(pll_predef_mclk); i++) {
+ p_entry = &pll_predef_mclk[i];
+ if (p_entry->mclk == mclk) {
+ found = true;
+ break;
+ }
+ }
+ if (!found) {
+ dev_err(cygaud->dev,
+ "%s No valid mclk freq (%u) found!\n", __func__, mclk);
+ return -EINVAL;
+ }
+
+ ch_clk = cygaud->audio_clk[p_entry->pll_ch_num];
+
+ if ((aio->clk_trace.cap_en) && (!aio->clk_trace.cap_clk_en)) {
+ error = clk_prepare_enable(ch_clk);
+ if (error) {
+ dev_err(cygaud->dev, "%s clk_prepare_enable failed %d\n",
+ __func__, error);
+ return error;
+ }
+ aio->clk_trace.cap_clk_en = true;
+ }
+
+ if ((aio->clk_trace.play_en) && (!aio->clk_trace.play_clk_en)) {
+ error = clk_prepare_enable(ch_clk);
+ if (error) {
+ dev_err(cygaud->dev, "%s clk_prepare_enable failed %d\n",
+ __func__, error);
+ return error;
+ }
+ aio->clk_trace.play_clk_en = true;
+ }
+
+ error = clk_set_rate(ch_clk, mclk);
+ if (error) {
+ dev_err(cygaud->dev, "%s Set MCLK rate failed: %d\n",
+ __func__, error);
+ return error;
+ }
+
+ return p_entry->pll_ch_num;
+}
+
+static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio,
+ struct cygnus_audio *cygaud)
+{
+ u32 value, i = 0;
+ u32 mask = 0xf;
+ u32 sclk;
+ bool found = false;
+ const struct _ssp_clk_coeff *p_entry = NULL;
+
+ for (i = 0; i < ARRAY_SIZE(ssp_clk_coeff); i++) {
+ p_entry = &ssp_clk_coeff[i];
+ if ((p_entry->rate == aio->lrclk) &&
+ (p_entry->sclk_rate == aio->bit_per_frame) &&
+ (p_entry->mclk == aio->mclk)) {
+ found = true;
+ break;
+ }
+ }
+ if (!found) {
+ dev_err(aio->cygaud->dev,
+ "No valid match found in ssp_clk_coeff array\n");
+ dev_err(aio->cygaud->dev, "lrclk = %u, bits/frame = %u, mclk = %u\n",
+ aio->lrclk, aio->bit_per_frame, aio->mclk);
+ return -EINVAL;
+ }
+
+ sclk = aio->bit_per_frame;
+ if (sclk == 512)
+ sclk = 0;
+ /* sclks_per_1fs_div = sclk cycles/32 */
+ sclk /= 32;
+ /* Set sclk rate */
+ switch (aio->port_type) {
+ case PORT_TDM:
+ /* Set number of bitclks per frame */
+ value = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
+ value &= ~(mask << I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32);
+ value |= sclk << I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32;
+ writel(value, aio->cygaud->audio + aio->regs.i2s_cfg);
+ dev_dbg(aio->cygaud->dev,
+ "SCLKS_PER_1FS_DIV32 = 0x%x\n", value);
+ break;
+ case PORT_SPDIF:
+ break;
+ default:
+ dev_err(aio->cygaud->dev, "Unknown port type\n");
+ return -EINVAL;
+ }
+
+ /* Set MCLK_RATE ssp port (spdif and ssp are the same) */
+ value = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
+ value &= ~(0xf << I2S_OUT_MCLKRATE_SHIFT);
+ value |= (p_entry->mclk_rate << I2S_OUT_MCLKRATE_SHIFT);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
+
+ dev_dbg(aio->cygaud->dev, "mclk cfg reg = 0x%x\n", value);
+ dev_dbg(aio->cygaud->dev, "bits per frame = %u, mclk = %u Hz, lrclk = %u Hz\n",
+ aio->bit_per_frame, aio->mclk, aio->lrclk);
+ return 0;
+}
+
+static int cygnus_ssp_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+ struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai);
+ int rate, bitres;
+ u32 value;
+ u32 mask = 0x1f;
+ int ret = 0;
+
+ dev_dbg(aio->cygaud->dev, "%s port = %d\n", __func__, aio->portnum);
+ dev_dbg(aio->cygaud->dev, "params_channels %d\n",
+ params_channels(params));
+ dev_dbg(aio->cygaud->dev, "rate %d\n", params_rate(params));
+ dev_dbg(aio->cygaud->dev, "format %d\n", params_format(params));
+
+ rate = params_rate(params);
+
+ switch (aio->mode) {
+ case CYGNUS_SSPMODE_TDM:
+ if ((rate == 192000) && (params_channels(params) > 4)) {
+ dev_err(aio->cygaud->dev, "Cannot run %d channels at %dHz\n",
+ params_channels(params), rate);
+ return -EINVAL;
+ }
+ break;
+ case CYGNUS_SSPMODE_I2S:
+ aio->bit_per_frame = 64; /* I2S must be 64 bit per frame */
+ break;
+ default:
+ dev_err(aio->cygaud->dev,
+ "%s port running in unknown mode\n", __func__);
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE);
+ /* Configure channels as mono or stereo/TDM */
+ if (params_channels(params) == 1)
+ value |= BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
+ else
+ value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ if (aio->port_type == PORT_SPDIF) {
+ dev_err(aio->cygaud->dev,
+ "SPDIF does not support 8bit format\n");
+ return -EINVAL;
+ }
+ bitres = 8;
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bitres = 16;
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+ /* 32 bit mode is coded as 0 */
+ bitres = 0;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~(mask << BF_SRC_CFGX_BIT_RES);
+ value |= (bitres << BF_SRC_CFGX_BIT_RES);
+ writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg);
+
+ } else {
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ value = readl(aio->cygaud->audio +
+ aio->regs.bf_destch_cfg);
+ value |= BIT(BF_DST_CFGX_CAP_MODE);
+ writel(value, aio->cygaud->audio +
+ aio->regs.bf_destch_cfg);
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+ value = readl(aio->cygaud->audio +
+ aio->regs.bf_destch_cfg);
+ value &= ~BIT(BF_DST_CFGX_CAP_MODE);
+ writel(value, aio->cygaud->audio +
+ aio->regs.bf_destch_cfg);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ }
+
+ aio->lrclk = rate;
+
+ if (!aio->is_slave)
+ ret = cygnus_ssp_set_clocks(aio, cygaud);
+
+ return ret;
+}
+
+/*
+ * This function sets the mclk frequency for pll clock
+ */
+static int cygnus_ssp_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ int sel;
+ u32 value;
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+ struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(aio->cygaud->dev,
+ "%s Enter port = %d\n", __func__, aio->portnum);
+ sel = pll_configure_mclk(cygaud, freq, aio);
+ if (sel < 0) {
+ dev_err(aio->cygaud->dev,
+ "%s Setting mclk failed.\n", __func__);
+ return -EINVAL;
+ }
+
+ aio->mclk = freq;
+
+ dev_dbg(aio->cygaud->dev, "%s Setting MCLKSEL to %d\n", __func__, sel);
+ value = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
+ value &= ~(0xf << I2S_OUT_PLLCLKSEL_SHIFT);
+ value |= (sel << I2S_OUT_PLLCLKSEL_SHIFT);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
+
+ return 0;
+}
+
+static int cygnus_ssp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+
+ snd_soc_dai_set_dma_data(dai, substream, aio);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aio->clk_trace.play_en = true;
+ else
+ aio->clk_trace.cap_en = true;
+
+ return 0;
+}
+
+static void cygnus_ssp_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aio->clk_trace.play_en = false;
+ else
+ aio->clk_trace.cap_en = false;
+
+ if (!aio->is_slave) {
+ u32 val;
+
+ val = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
+ val &= CYGNUS_PLLCLKSEL_MASK;
+ if (val >= ARRAY_SIZE(aio->cygaud->audio_clk)) {
+ dev_err(aio->cygaud->dev, "Clk index %u is out of bounds\n",
+ val);
+ return;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (aio->clk_trace.play_clk_en) {
+ clk_disable_unprepare(aio->cygaud->
+ audio_clk[val]);
+ aio->clk_trace.play_clk_en = false;
+ }
+ } else {
+ if (aio->clk_trace.cap_clk_en) {
+ clk_disable_unprepare(aio->cygaud->
+ audio_clk[val]);
+ aio->clk_trace.cap_clk_en = false;
+ }
+ }
+ }
+}
+
+/*
+ * Bit Update Notes
+ * 31 Yes TDM Mode (1 = TDM, 0 = i2s)
+ * 30 Yes Slave Mode (1 = Slave, 0 = Master)
+ * 29:26 No Sclks per frame
+ * 25:18 Yes FS Width
+ * 17:14 No Valid Slots
+ * 13 No Bits (1 = 16 bits, 0 = 32 bits)
+ * 12:08 No Bits per samp
+ * 07 Yes Justifcation (1 = LSB, 0 = MSB)
+ * 06 Yes Alignment (1 = Delay 1 clk, 0 = no delay
+ * 05 Yes SCLK polarity (1 = Rising, 0 = Falling)
+ * 04 Yes LRCLK Polarity (1 = High for left, 0 = Low for left)
+ * 03:02 Yes Reserved - write as zero
+ * 01 No Data Enable
+ * 00 No CLK Enable
+ */
+#define I2S_OUT_CFG_REG_UPDATE_MASK 0x3C03FF03
+
+/* Input cfg is same as output, but the FS width is not a valid field */
+#define I2S_IN_CFG_REG_UPDATE_MASK (I2S_OUT_CFG_REG_UPDATE_MASK | 0x03FC0000)
+
+int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai, int len)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+
+ if ((len > 0) && (len < 256)) {
+ aio->fsync_width = len;
+ return 0;
+ } else {
+ return -EINVAL;
+ }
+}
+
+static int cygnus_ssp_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+ u32 ssp_curcfg;
+ u32 ssp_newcfg;
+ u32 ssp_outcfg;
+ u32 ssp_incfg;
+ u32 val;
+ u32 mask;
+
+ dev_dbg(aio->cygaud->dev, "%s Enter fmt: %x\n", __func__, fmt);
+
+ if (aio->port_type == PORT_SPDIF)
+ return -EINVAL;
+
+ ssp_newcfg = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_SLAVE_MODE);
+ aio->is_slave = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ssp_newcfg &= ~BIT(I2S_OUT_CFGX_SLAVE_MODE);
+ aio->is_slave = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT);
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_FSYNC_WIDTH);
+ aio->mode = CYGNUS_SSPMODE_I2S;
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_TDM_MODE);
+
+ /* DSP_A = data after FS, DSP_B = data during FS */
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_A)
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT);
+
+ if ((aio->fsync_width > 0) && (aio->fsync_width < 256))
+ ssp_newcfg |=
+ (aio->fsync_width << I2S_OUT_CFGX_FSYNC_WIDTH);
+ else
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_FSYNC_WIDTH);
+
+ aio->mode = CYGNUS_SSPMODE_TDM;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ /*
+ * SSP out cfg.
+ * Retain bits we do not want to update, then OR in new bits
+ */
+ ssp_curcfg = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
+ ssp_outcfg = (ssp_curcfg & I2S_OUT_CFG_REG_UPDATE_MASK) | ssp_newcfg;
+ writel(ssp_outcfg, aio->cygaud->audio + aio->regs.i2s_cfg);
+
+ /*
+ * SSP in cfg.
+ * Retain bits we do not want to update, then OR in new bits
+ */
+ ssp_curcfg = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg);
+ ssp_incfg = (ssp_curcfg & I2S_IN_CFG_REG_UPDATE_MASK) | ssp_newcfg;
+ writel(ssp_incfg, aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg);
+
+ val = readl(aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+
+ /*
+ * Configure the word clk and bit clk as output or tristate
+ * Each port has 4 bits for controlling its pins.
+ * Shift the mask based upon port number.
+ */
+ mask = BIT(AUD_MISC_SEROUT_LRCK_OE)
+ | BIT(AUD_MISC_SEROUT_SCLK_OE)
+ | BIT(AUD_MISC_SEROUT_MCLK_OE);
+ mask = mask << (aio->portnum * 4);
+ if (aio->is_slave)
+ /* Set bit for tri-state */
+ val |= mask;
+ else
+ /* Clear bit for drive */
+ val &= ~mask;
+
+ dev_dbg(aio->cygaud->dev, "%s Set OE bits 0x%x\n", __func__, val);
+ writel(val, aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+
+ return 0;
+}
+
+static int cygnus_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+ struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(aio->cygaud->dev,
+ "%s cmd %d at port = %d\n", __func__, cmd, aio->portnum);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ audio_ssp_out_enable(aio);
+ else
+ audio_ssp_in_enable(aio);
+ cygaud->active_ports++;
+
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ audio_ssp_out_disable(aio);
+ else
+ audio_ssp_in_disable(aio);
+ cygaud->active_ports--;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+ u32 value;
+ int bits_per_slot = 0; /* default to 32-bits per slot */
+ int frame_bits;
+ unsigned int active_slots;
+ bool found = false;
+ int i;
+
+ if (tx_mask != rx_mask) {
+ dev_err(aio->cygaud->dev,
+ "%s tx_mask must equal rx_mask\n", __func__);
+ return -EINVAL;
+ }
+
+ active_slots = hweight32(tx_mask);
+
+ if ((active_slots < 0) || (active_slots > 16))
+ return -EINVAL;
+
+ /* Slot value must be even */
+ if (active_slots % 2)
+ return -EINVAL;
+
+ /* We encode 16 slots as 0 in the reg */
+ if (active_slots == 16)
+ active_slots = 0;
+
+ /* Slot Width is either 16 or 32 */
+ switch (slot_width) {
+ case 16:
+ bits_per_slot = 1;
+ break;
+ case 32:
+ bits_per_slot = 0;
+ break;
+ default:
+ bits_per_slot = 0;
+ dev_warn(aio->cygaud->dev,
+ "%s Defaulting Slot Width to 32\n", __func__);
+ }
+
+ frame_bits = slots * slot_width;
+
+ for (i = 0; i < ARRAY_SIZE(ssp_valid_tdm_framesize); i++) {
+ if (ssp_valid_tdm_framesize[i] == frame_bits) {
+ found = true;
+ break;
+ }
+ }
+
+ if (!found) {
+ dev_err(aio->cygaud->dev,
+ "%s In TDM mode, frame bits INVALID (%d)\n",
+ __func__, frame_bits);
+ return -EINVAL;
+ }
+
+ aio->bit_per_frame = frame_bits;
+
+ dev_dbg(aio->cygaud->dev, "%s active_slots %u, bits per frame %d\n",
+ __func__, active_slots, frame_bits);
+
+ /* Set capture side of ssp port */
+ value = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg);
+ value &= ~(0xf << I2S_OUT_CFGX_VALID_SLOT);
+ value |= (active_slots << I2S_OUT_CFGX_VALID_SLOT);
+ value &= ~BIT(I2S_OUT_CFGX_BITS_PER_SLOT);
+ value |= (bits_per_slot << I2S_OUT_CFGX_BITS_PER_SLOT);
+ writel(value, aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg);
+
+ /* Set playback side of ssp port */
+ value = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
+ value &= ~(0xf << I2S_OUT_CFGX_VALID_SLOT);
+ value |= (active_slots << I2S_OUT_CFGX_VALID_SLOT);
+ value &= ~BIT(I2S_OUT_CFGX_BITS_PER_SLOT);
+ value |= (bits_per_slot << I2S_OUT_CFGX_BITS_PER_SLOT);
+ writel(value, aio->cygaud->audio + aio->regs.i2s_cfg);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int cygnus_ssp_suspend(struct snd_soc_dai *cpu_dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+
+ if (!aio->is_slave) {
+ u32 val;
+
+ val = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
+ val &= CYGNUS_PLLCLKSEL_MASK;
+ if (val >= ARRAY_SIZE(aio->cygaud->audio_clk)) {
+ dev_err(aio->cygaud->dev, "Clk index %u is out of bounds\n",
+ val);
+ return -EINVAL;
+ }
+
+ if (aio->clk_trace.cap_clk_en)
+ clk_disable_unprepare(aio->cygaud->audio_clk[val]);
+ if (aio->clk_trace.play_clk_en)
+ clk_disable_unprepare(aio->cygaud->audio_clk[val]);
+
+ aio->pll_clk_num = val;
+ }
+
+ return 0;
+}
+
+static int cygnus_ssp_resume(struct snd_soc_dai *cpu_dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+ int error;
+
+ if (!aio->is_slave) {
+ if (aio->clk_trace.cap_clk_en) {
+ error = clk_prepare_enable(aio->cygaud->
+ audio_clk[aio->pll_clk_num]);
+ if (error) {
+ dev_err(aio->cygaud->dev, "%s clk_prepare_enable failed\n",
+ __func__);
+ return -EINVAL;
+ }
+ }
+ if (aio->clk_trace.play_clk_en) {
+ error = clk_prepare_enable(aio->cygaud->
+ audio_clk[aio->pll_clk_num]);
+ if (error) {
+ if (aio->clk_trace.cap_clk_en)
+ clk_disable_unprepare(aio->cygaud->
+ audio_clk[aio->pll_clk_num]);
+ dev_err(aio->cygaud->dev, "%s clk_prepare_enable failed\n",
+ __func__);
+ return -EINVAL;
+ }
+ }
+ }
+
+ return 0;
+}
+#else
+#define cygnus_ssp_suspend NULL
+#define cygnus_ssp_resume NULL
+#endif
+
+static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = {
+ .startup = cygnus_ssp_startup,
+ .shutdown = cygnus_ssp_shutdown,
+ .trigger = cygnus_ssp_trigger,
+ .hw_params = cygnus_ssp_hw_params,
+ .set_fmt = cygnus_ssp_set_fmt,
+ .set_sysclk = cygnus_ssp_set_sysclk,
+ .set_tdm_slot = cygnus_set_dai_tdm_slot,
+};
+
+
+#define INIT_CPU_DAI(num) { \
+ .name = "cygnus-ssp" #num, \
+ .playback = { \
+ .channels_min = 1, \
+ .channels_max = 16, \
+ .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
+ SNDRV_PCM_RATE_192000, \
+ .formats = SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE, \
+ }, \
+ .capture = { \
+ .channels_min = 2, \
+ .channels_max = 16, \
+ .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
+ SNDRV_PCM_RATE_192000, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE, \
+ }, \
+ .ops = &cygnus_ssp_dai_ops, \
+ .suspend = cygnus_ssp_suspend, \
+ .resume = cygnus_ssp_resume, \
+}
+
+static const struct snd_soc_dai_driver cygnus_ssp_dai_info[] = {
+ INIT_CPU_DAI(0),
+ INIT_CPU_DAI(1),
+ INIT_CPU_DAI(2),
+};
+
+static struct snd_soc_dai_driver cygnus_spdif_dai_info = {
+ .name = "cygnus-spdif",
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &cygnus_ssp_dai_ops,
+ .suspend = cygnus_ssp_suspend,
+ .resume = cygnus_ssp_resume,
+};
+
+static struct snd_soc_dai_driver cygnus_ssp_dai[CYGNUS_MAX_PORTS];
+
+static const struct snd_soc_component_driver cygnus_ssp_component = {
+ .name = "cygnus-audio",
+};
+
+/*
+ * Return < 0 if error
+ * Return 0 if disabled
+ * Return 1 if enabled and node is parsed successfully
+ */
+static int parse_ssp_child_node(struct platform_device *pdev,
+ struct device_node *dn,
+ struct cygnus_audio *cygaud,
+ struct snd_soc_dai_driver *p_dai)
+{
+ struct cygnus_aio_port *aio;
+ struct cygnus_ssp_regs ssp_regs[3];
+ u32 rawval;
+ int portnum = -1;
+ enum cygnus_audio_port_type port_type;
+
+ if (of_property_read_u32(dn, "reg", &rawval)) {
+ dev_err(&pdev->dev, "Missing reg property\n");
+ return -EINVAL;
+ }
+
+ portnum = rawval;
+ switch (rawval) {
+ case 0:
+ ssp_regs[0] = INIT_SSP_REGS(0);
+ port_type = PORT_TDM;
+ break;
+ case 1:
+ ssp_regs[1] = INIT_SSP_REGS(1);
+ port_type = PORT_TDM;
+ break;
+ case 2:
+ ssp_regs[2] = INIT_SSP_REGS(2);
+ port_type = PORT_TDM;
+ break;
+ case 3:
+ port_type = PORT_SPDIF;
+ break;
+ default:
+ dev_err(&pdev->dev, "Bad value for reg %u\n", rawval);
+ return -EINVAL;
+ }
+
+ aio = &cygaud->portinfo[portnum];
+ aio->cygaud = cygaud;
+ aio->portnum = portnum;
+ aio->port_type = port_type;
+ aio->fsync_width = -1;
+
+ switch (port_type) {
+ case PORT_TDM:
+ aio->regs = ssp_regs[portnum];
+ *p_dai = cygnus_ssp_dai_info[portnum];
+ aio->mode = CYGNUS_SSPMODE_UNKNOWN;
+ break;
+
+ case PORT_SPDIF:
+ aio->regs.bf_sourcech_cfg = BF_SRC_CFG3_OFFSET;
+ aio->regs.bf_sourcech_ctrl = BF_SRC_CTRL3_OFFSET;
+ aio->regs.i2s_mclk_cfg = SPDIF_MCLK_CFG_OFFSET;
+ aio->regs.i2s_stream_cfg = SPDIF_STREAM_CFG_OFFSET;
+ *p_dai = cygnus_spdif_dai_info;
+
+ /* For the purposes of this code SPDIF can be I2S mode */
+ aio->mode = CYGNUS_SSPMODE_I2S;
+ break;
+ default:
+ dev_err(&pdev->dev, "Bad value for port_type %d\n", port_type);
+ return -EINVAL;
+ }
+
+ dev_dbg(&pdev->dev, "%s portnum = %d\n", __func__, aio->portnum);
+ aio->streams_on = 0;
+ aio->cygaud->dev = &pdev->dev;
+ aio->clk_trace.play_en = false;
+ aio->clk_trace.cap_en = false;
+
+ audio_ssp_init_portregs(aio);
+ return 0;
+}
+
+static int audio_clk_init(struct platform_device *pdev,
+ struct cygnus_audio *cygaud)
+{
+ int i;
+ char clk_name[PROP_LEN_MAX];
+
+ for (i = 0; i < ARRAY_SIZE(cygaud->audio_clk); i++) {
+ snprintf(clk_name, PROP_LEN_MAX, "ch%d_audio", i);
+
+ cygaud->audio_clk[i] = devm_clk_get(&pdev->dev, clk_name);
+ if (IS_ERR(cygaud->audio_clk[i]))
+ return PTR_ERR(cygaud->audio_clk[i]);
+ }
+
+ return 0;
+}
+
+static int cygnus_ssp_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct device_node *child_node;
+ struct resource *res = pdev->resource;
+ struct cygnus_audio *cygaud;
+ int err = -EINVAL;
+ int node_count;
+ int active_port_count;
+
+ cygaud = devm_kzalloc(dev, sizeof(struct cygnus_audio), GFP_KERNEL);
+ if (!cygaud)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, cygaud);
+
+ res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "aud");
+ cygaud->audio = devm_ioremap_resource(dev, res);
+ if (IS_ERR(cygaud->audio))
+ return PTR_ERR(cygaud->audio);
+
+ res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "i2s_in");
+ cygaud->i2s_in = devm_ioremap_resource(dev, res);
+ if (IS_ERR(cygaud->i2s_in))
+ return PTR_ERR(cygaud->i2s_in);
+
+ /* Tri-state all controlable pins until we know that we need them */
+ writel(CYGNUS_SSP_TRISTATE_MASK,
+ cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+
+ node_count = of_get_child_count(pdev->dev.of_node);
+ if ((node_count < 1) || (node_count > CYGNUS_MAX_PORTS)) {
+ dev_err(dev, "child nodes is %d. Must be between 1 and %d\n",
+ node_count, CYGNUS_MAX_PORTS);
+ return -EINVAL;
+ }
+
+ active_port_count = 0;
+
+ for_each_available_child_of_node(pdev->dev.of_node, child_node) {
+ err = parse_ssp_child_node(pdev, child_node, cygaud,
+ &cygnus_ssp_dai[active_port_count]);
+
+ /* negative is err, 0 is active and good, 1 is disabled */
+ if (err < 0)
+ return err;
+ else if (!err) {
+ dev_dbg(dev, "Activating DAI: %s\n",
+ cygnus_ssp_dai[active_port_count].name);
+ active_port_count++;
+ }
+ }
+
+ cygaud->dev = dev;
+ cygaud->active_ports = 0;
+
+ dev_dbg(dev, "Registering %d DAIs\n", active_port_count);
+ err = snd_soc_register_component(dev, &cygnus_ssp_component,
+ cygnus_ssp_dai, active_port_count);
+ if (err) {
+ dev_err(dev, "snd_soc_register_dai failed\n");
+ return err;
+ }
+
+ cygaud->irq_num = platform_get_irq(pdev, 0);
+ if (cygaud->irq_num <= 0) {
+ dev_err(dev, "platform_get_irq failed\n");
+ err = cygaud->irq_num;
+ goto err_irq;
+ }
+
+ err = audio_clk_init(pdev, cygaud);
+ if (err) {
+ dev_err(dev, "audio clock initialization failed\n");
+ goto err_irq;
+ }
+
+ err = cygnus_soc_platform_register(dev, cygaud);
+ if (err) {
+ dev_err(dev, "platform reg error %d\n", err);
+ goto err_irq;
+ }
+
+ return 0;
+
+err_irq:
+ snd_soc_unregister_component(dev);
+ return err;
+}
+
+static int cygnus_ssp_remove(struct platform_device *pdev)
+{
+ cygnus_soc_platform_unregister(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+
+ return 0;
+}
+
+static const struct of_device_id cygnus_ssp_of_match[] = {
+ { .compatible = "brcm,cygnus-audio" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, cygnus_ssp_of_match);
+
+static struct platform_driver cygnus_ssp_driver = {
+ .probe = cygnus_ssp_probe,
+ .remove = cygnus_ssp_remove,
+ .driver = {
+ .name = "cygnus-ssp",
+ .of_match_table = cygnus_ssp_of_match,
+ },
+};
+
+module_platform_driver(cygnus_ssp_driver);
+
+MODULE_ALIAS("platform:cygnus-ssp");
+MODULE_LICENSE("GPL v2");
+MODULE_AUTHOR("Broadcom");
+MODULE_DESCRIPTION("Cygnus ASoC SSP Interface");
diff --git a/sound/soc/bcm/cygnus-ssp.h b/sound/soc/bcm/cygnus-ssp.h
new file mode 100644
index 000000000000..33dd34305928
--- /dev/null
+++ b/sound/soc/bcm/cygnus-ssp.h
@@ -0,0 +1,139 @@
+/*
+ * Copyright (C) 2014-2015 Broadcom Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any
+ * kind, whether express or implied; without even the implied warranty
+ * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#ifndef __CYGNUS_SSP_H__
+#define __CYGNUS_SSP_H__
+
+#define CYGNUS_TDM_DAI_MAX_SLOTS 16
+
+#define CYGNUS_MAX_PLAYBACK_PORTS 4
+#define CYGNUS_MAX_CAPTURE_PORTS 3
+#define CYGNUS_MAX_I2S_PORTS 3
+#define CYGNUS_MAX_PORTS CYGNUS_MAX_PLAYBACK_PORTS
+#define CYGNUS_AUIDO_MAX_NUM_CLKS 3
+
+#define CYGNUS_SSP_FRAMEBITS_DIV 1
+
+#define CYGNUS_SSPMODE_I2S 0
+#define CYGNUS_SSPMODE_TDM 1
+#define CYGNUS_SSPMODE_UNKNOWN -1
+
+#define CYGNUS_SSP_CLKSRC_PLL 0
+
+/* Max string length of our dt property names */
+#define PROP_LEN_MAX 40
+
+struct ringbuf_regs {
+ unsigned rdaddr;
+ unsigned wraddr;
+ unsigned baseaddr;
+ unsigned endaddr;
+ unsigned fmark; /* freemark for play, fullmark for caputure */
+ unsigned period_bytes;
+ unsigned buf_size;
+};
+
+#define RINGBUF_REG_PLAYBACK(num) ((struct ringbuf_regs) { \
+ .rdaddr = SRC_RBUF_ ##num## _RDADDR_OFFSET, \
+ .wraddr = SRC_RBUF_ ##num## _WRADDR_OFFSET, \
+ .baseaddr = SRC_RBUF_ ##num## _BASEADDR_OFFSET, \
+ .endaddr = SRC_RBUF_ ##num## _ENDADDR_OFFSET, \
+ .fmark = SRC_RBUF_ ##num## _FREE_MARK_OFFSET, \
+ .period_bytes = 0, \
+ .buf_size = 0, \
+})
+
+#define RINGBUF_REG_CAPTURE(num) ((struct ringbuf_regs) { \
+ .rdaddr = DST_RBUF_ ##num## _RDADDR_OFFSET, \
+ .wraddr = DST_RBUF_ ##num## _WRADDR_OFFSET, \
+ .baseaddr = DST_RBUF_ ##num## _BASEADDR_OFFSET, \
+ .endaddr = DST_RBUF_ ##num## _ENDADDR_OFFSET, \
+ .fmark = DST_RBUF_ ##num## _FULL_MARK_OFFSET, \
+ .period_bytes = 0, \
+ .buf_size = 0, \
+})
+
+enum cygnus_audio_port_type {
+ PORT_TDM,
+ PORT_SPDIF,
+};
+
+struct cygnus_ssp_regs {
+ u32 i2s_stream_cfg;
+ u32 i2s_cfg;
+ u32 i2s_cap_stream_cfg;
+ u32 i2s_cap_cfg;
+ u32 i2s_mclk_cfg;
+
+ u32 bf_destch_ctrl;
+ u32 bf_destch_cfg;
+ u32 bf_sourcech_ctrl;
+ u32 bf_sourcech_cfg;
+ u32 bf_sourcech_grp;
+};
+
+struct cygnus_track_clk {
+ bool cap_en;
+ bool play_en;
+ bool cap_clk_en;
+ bool play_clk_en;
+};
+
+struct cygnus_aio_port {
+ int portnum;
+ int mode;
+ bool is_slave;
+ int streams_on; /* will be 0 if both capture and play are off */
+ int fsync_width;
+ int port_type;
+
+ u32 mclk;
+ u32 lrclk;
+ u32 bit_per_frame;
+ u32 pll_clk_num;
+
+ struct cygnus_audio *cygaud;
+ struct cygnus_ssp_regs regs;
+
+ struct ringbuf_regs play_rb_regs;
+ struct ringbuf_regs capture_rb_regs;
+
+ struct snd_pcm_substream *play_stream;
+ struct snd_pcm_substream *capture_stream;
+
+ struct cygnus_track_clk clk_trace;
+};
+
+
+struct cygnus_audio {
+ struct cygnus_aio_port portinfo[CYGNUS_MAX_PORTS];
+
+ int irq_num;
+ void __iomem *audio;
+ struct device *dev;
+ void __iomem *i2s_in;
+
+ struct clk *audio_clk[CYGNUS_AUIDO_MAX_NUM_CLKS];
+ int active_ports;
+ unsigned long vco_rate;
+};
+
+extern int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai);
+extern int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd);
+extern int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai,
+ int len);
+extern int cygnus_soc_platform_register(struct device *dev,
+ struct cygnus_audio *cygaud);
+extern int cygnus_soc_platform_unregister(struct device *dev);
+extern int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai,
+ int len);
+#endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4d82a58ff6b0..7f516b6ee150 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_ADAU1977_SPI if SPI_MASTER
select SND_SOC_ADAU1977_I2C if I2C
select SND_SOC_ADAU1701 if I2C
+ select SND_SOC_ADAU7002
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
@@ -46,6 +47,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_BT_SCO
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS35L32 if I2C
+ select SND_SOC_CS35L33 if I2C
select SND_SOC_CS42L51_I2C if I2C
select SND_SOC_CS42L52 if I2C && INPUT
select SND_SOC_CS42L56 if I2C && INPUT
@@ -57,6 +59,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CS42XX8_I2C if I2C
select SND_SOC_CS4349 if I2C
select SND_SOC_CS47L24 if MFD_CS47L24
+ select SND_SOC_CS53L30 if I2C
select SND_SOC_CX20442 if TTY
select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
select SND_SOC_DA7213 if I2C
@@ -84,6 +87,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98925 if I2C
select SND_SOC_MAX98926 if I2C
select SND_SOC_MAX9850 if I2C
+ select SND_SOC_MAX9860 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_MC13783 if MFD_MC13XXX
@@ -269,8 +273,12 @@ config SND_SOC_AD1980
config SND_SOC_AD73311
tristate
+config SND_SOC_ADAU_UTILS
+ tristate
+
config SND_SOC_ADAU1373
tristate
+ select SND_SOC_ADAU_UTILS
config SND_SOC_ADAU1701
tristate "Analog Devices ADAU1701 CODEC"
@@ -280,6 +288,7 @@ config SND_SOC_ADAU1701
config SND_SOC_ADAU17X1
tristate
select SND_SOC_SIGMADSP_REGMAP
+ select SND_SOC_ADAU_UTILS
config SND_SOC_ADAU1761
tristate
@@ -322,6 +331,9 @@ config SND_SOC_ADAU1977_I2C
select SND_SOC_ADAU1977
select REGMAP_I2C
+config SND_SOC_ADAU7002
+ tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter"
+
config SND_SOC_ADAV80X
tristate
@@ -371,7 +383,7 @@ config SND_SOC_ALC5632
tristate
config SND_SOC_BT_SCO
- tristate
+ tristate "Dummy BT SCO codec driver"
config SND_SOC_CQ0093VC
tristate
@@ -380,6 +392,10 @@ config SND_SOC_CS35L32
tristate "Cirrus Logic CS35L32 CODEC"
depends on I2C
+config SND_SOC_CS35L33
+ tristate "Cirrus Logic CS35L33 CODEC"
+ depends on I2C
+
config SND_SOC_CS42L51
tristate
@@ -450,6 +466,11 @@ config SND_SOC_CS4349
config SND_SOC_CS47L24
tristate
+# Cirrus Logic Quad-Channel ADC
+config SND_SOC_CS53L30
+ tristate "Cirrus Logic CS53L30 CODEC"
+ depends on I2C
+
config SND_SOC_CX20442
tristate
depends on TTY
@@ -483,9 +504,10 @@ config SND_SOC_DMIC
tristate
config SND_SOC_HDMI_CODEC
- tristate
- select SND_PCM_ELD
- select SND_PCM_IEC958
+ tristate
+ select SND_PCM_ELD
+ select SND_PCM_IEC958
+ select HDMI
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
@@ -547,6 +569,11 @@ config SND_SOC_MAX98926
config SND_SOC_MAX9850
tristate
+config SND_SOC_MAX9860
+ tristate "Maxim MAX9860 Mono Audio Voice Codec"
+ depends on I2C
+ select REGMAP_I2C
+
config SND_SOC_PCM1681
tristate "Texas Instruments PCM1681 CODEC"
depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 0f548fd34ca3..ff97d7392e8b 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -7,6 +7,7 @@ snd-soc-ad193x-spi-objs := ad193x-spi.o
snd-soc-ad193x-i2c-objs := ad193x-i2c.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
+snd-soc-adau-utils-objs := adau-utils.o
snd-soc-adau1373-objs := adau1373.o
snd-soc-adau1701-objs := adau1701.o
snd-soc-adau17x1-objs := adau17x1.o
@@ -19,6 +20,7 @@ snd-soc-adau1781-spi-objs := adau1781-spi.o
snd-soc-adau1977-objs := adau1977.o
snd-soc-adau1977-spi-objs := adau1977-spi.o
snd-soc-adau1977-i2c-objs := adau1977-i2c.o
+snd-soc-adau7002-objs := adau7002.o
snd-soc-adav80x-objs := adav80x.o
snd-soc-adav801-objs := adav801.o
snd-soc-adav803-objs := adav803.o
@@ -35,6 +37,7 @@ snd-soc-arizona-objs := arizona.o
snd-soc-bt-sco-objs := bt-sco.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs35l32-objs := cs35l32.o
+snd-soc-cs35l33-objs := cs35l33.o
snd-soc-cs42l51-objs := cs42l51.o
snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o
snd-soc-cs42l52-objs := cs42l52.o
@@ -49,6 +52,7 @@ snd-soc-cs42xx8-objs := cs42xx8.o
snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
snd-soc-cs4349-objs := cs4349.o
snd-soc-cs47l24-objs := cs47l24.o
+snd-soc-cs53l30-objs := cs53l30.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
snd-soc-da7213-objs := da7213.o
@@ -79,6 +83,7 @@ snd-soc-max9867-objs := max9867.o
snd-soc-max98925-objs := max98925.o
snd-soc-max98926-objs := max98926.o
snd-soc-max9850-objs := max9850.o
+snd-soc-max9860-objs := max9860.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
snd-soc-nau8825-objs := nau8825.o
@@ -220,6 +225,7 @@ obj-$(CONFIG_SND_SOC_AD193X_SPI) += snd-soc-ad193x-spi.o
obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU_UTILS) += snd-soc-adau-utils.o
obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o
obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o
obj-$(CONFIG_SND_SOC_ADAU17X1) += snd-soc-adau17x1.o
@@ -232,6 +238,7 @@ obj-$(CONFIG_SND_SOC_ADAU1781_SPI) += snd-soc-adau1781-spi.o
obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o
obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o
obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o
+obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o
obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o
obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o
obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o
@@ -250,6 +257,7 @@ obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o
obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o
+obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o
obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o
obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o
@@ -264,6 +272,7 @@ obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o
obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o
obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o
obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o
+obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o
@@ -293,6 +302,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o
obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o
obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
+obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o
diff --git a/sound/soc/codecs/adau-utils.c b/sound/soc/codecs/adau-utils.c
new file mode 100644
index 000000000000..19d6a6f41b12
--- /dev/null
+++ b/sound/soc/codecs/adau-utils.c
@@ -0,0 +1,61 @@
+/*
+ * Shared helper functions for devices from the ADAU family
+ *
+ * Copyright 2011-2016 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/gcd.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+
+#include "adau-utils.h"
+
+int adau_calc_pll_cfg(unsigned int freq_in, unsigned int freq_out,
+ uint8_t regs[5])
+{
+ unsigned int r, n, m, i, j;
+ unsigned int div;
+
+ if (!freq_out) {
+ r = 0;
+ n = 0;
+ m = 0;
+ div = 0;
+ } else {
+ if (freq_out % freq_in != 0) {
+ div = DIV_ROUND_UP(freq_in, 13500000);
+ freq_in /= div;
+ r = freq_out / freq_in;
+ i = freq_out % freq_in;
+ j = gcd(i, freq_in);
+ n = i / j;
+ m = freq_in / j;
+ div--;
+ } else {
+ r = freq_out / freq_in;
+ n = 0;
+ m = 0;
+ div = 0;
+ }
+ if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2)
+ return -EINVAL;
+ }
+
+ regs[0] = m >> 8;
+ regs[1] = m & 0xff;
+ regs[2] = n >> 8;
+ regs[3] = n & 0xff;
+ regs[4] = (r << 3) | (div << 1);
+ if (m != 0)
+ regs[4] |= 1; /* Fractional mode */
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(adau_calc_pll_cfg);
+
+MODULE_DESCRIPTION("ASoC ADAU audio CODECs shared helper functions");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/adau-utils.h b/sound/soc/codecs/adau-utils.h
new file mode 100644
index 000000000000..939b5f37762f
--- /dev/null
+++ b/sound/soc/codecs/adau-utils.h
@@ -0,0 +1,7 @@
+#ifndef SOUND_SOC_CODECS_ADAU_PLL_H
+#define SOUND_SOC_CODECS_ADAU_PLL_H
+
+int adau_calc_pll_cfg(unsigned int freq_in, unsigned int freq_out,
+ uint8_t regs[5]);
+
+#endif
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index fe1353a797b9..1556b360fa15 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -23,6 +23,7 @@
#include <sound/adau1373.h>
#include "adau1373.h"
+#include "adau-utils.h"
struct adau1373_dai {
unsigned int clk_src;
@@ -1254,7 +1255,8 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
{
struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
unsigned int dpll_div = 0;
- unsigned int x, r, n, m, i, j, mode;
+ uint8_t pll_regs[5];
+ int ret;
switch (pll_id) {
case ADAU1373_PLL1:
@@ -1295,27 +1297,8 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
dpll_div++;
}
- if (freq_out % freq_in != 0) {
- /* fout = fin * (r + (n/m)) / x */
- x = DIV_ROUND_UP(freq_in, 13500000);
- freq_in /= x;
- r = freq_out / freq_in;
- i = freq_out % freq_in;
- j = gcd(i, freq_in);
- n = i / j;
- m = freq_in / j;
- x--;
- mode = 1;
- } else {
- /* fout = fin / r */
- r = freq_out / freq_in;
- n = 0;
- m = 0;
- x = 0;
- mode = 0;
- }
-
- if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff)
+ ret = adau_calc_pll_cfg(freq_in, freq_out, pll_regs);
+ if (ret)
return -EINVAL;
if (dpll_div) {
@@ -1330,12 +1313,11 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id),
(source << 4) | dpll_div);
- regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff);
- regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff);
- regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff);
- regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff);
- regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id),
- (r << 3) | (x << 1) | mode);
+ regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), pll_regs[0]);
+ regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), pll_regs[1]);
+ regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), pll_regs[2]);
+ regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), pll_regs[3]);
+ regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), pll_regs[4]);
/* Set sysclk to pll_rate / 4 */
regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09);
diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c
index 8de010f758cd..9e7f257f17f8 100644
--- a/sound/soc/codecs/adau1761-i2c.c
+++ b/sound/soc/codecs/adau1761-i2c.c
@@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client,
static int adau1761_i2c_remove(struct i2c_client *client)
{
- snd_soc_unregister_codec(&client->dev);
+ adau17x1_remove(&client->dev);
return 0;
}
diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c
index d9171245bd9f..a0b214be759a 100644
--- a/sound/soc/codecs/adau1761-spi.c
+++ b/sound/soc/codecs/adau1761-spi.c
@@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi)
static int adau1761_spi_remove(struct spi_device *spi)
{
- snd_soc_unregister_codec(&spi->dev);
+ adau17x1_remove(&spi->dev);
return 0;
}
diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c
index 06cbca84cf02..7b9d1802d159 100644
--- a/sound/soc/codecs/adau1781-i2c.c
+++ b/sound/soc/codecs/adau1781-i2c.c
@@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client,
static int adau1781_i2c_remove(struct i2c_client *client)
{
- snd_soc_unregister_codec(&client->dev);
+ adau17x1_remove(&client->dev);
return 0;
}
diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c
index 3d965a01b99c..9b233544d2e8 100644
--- a/sound/soc/codecs/adau1781-spi.c
+++ b/sound/soc/codecs/adau1781-spi.c
@@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi)
static int adau1781_spi_remove(struct spi_device *spi)
{
- snd_soc_unregister_codec(&spi->dev);
+ adau17x1_remove(&spi->dev);
return 0;
}
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index fcf05b254ecd..439aa3ff1f99 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -9,6 +9,7 @@
#include <linux/module.h>
#include <linux/init.h>
+#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <sound/core.h>
@@ -23,6 +24,7 @@
#include "sigmadsp.h"
#include "adau17x1.h"
+#include "adau-utils.h"
static const char * const adau17x1_capture_mixer_boost_text[] = {
"Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3",
@@ -302,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau)
}
EXPORT_SYMBOL_GPL(adau17x1_has_dsp);
+static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adau *adau = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ if (freq_in < 8000000 || freq_in > 27000000)
+ return -EINVAL;
+
+ ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs);
+ if (ret < 0)
+ return ret;
+
+ /* The PLL register is 6 bytes long and can only be written at once. */
+ ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL,
+ adau->pll_regs, ARRAY_SIZE(adau->pll_regs));
+ if (ret)
+ return ret;
+
+ adau->pll_freq = freq_out;
+
+ return 0;
+}
+
+static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec);
+ struct adau *adau = snd_soc_codec_get_drvdata(dai->codec);
+ bool is_pll;
+ bool was_pll;
+
+ switch (clk_id) {
+ case ADAU17X1_CLK_SRC_MCLK:
+ is_pll = false;
+ break;
+ case ADAU17X1_CLK_SRC_PLL_AUTO:
+ if (!adau->mclk)
+ return -EINVAL;
+ /* Fall-through */
+ case ADAU17X1_CLK_SRC_PLL:
+ is_pll = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (adau->clk_src) {
+ case ADAU17X1_CLK_SRC_MCLK:
+ was_pll = false;
+ break;
+ case ADAU17X1_CLK_SRC_PLL:
+ case ADAU17X1_CLK_SRC_PLL_AUTO:
+ was_pll = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adau->sysclk = freq;
+
+ if (is_pll != was_pll) {
+ if (is_pll) {
+ snd_soc_dapm_add_routes(dapm,
+ &adau17x1_dapm_pll_route, 1);
+ } else {
+ snd_soc_dapm_del_routes(dapm,
+ &adau17x1_dapm_pll_route, 1);
+ }
+ }
+
+ adau->clk_src = clk_id;
+
+ return 0;
+}
+
+static int adau17x1_auto_pll(struct snd_soc_dai *dai,
+ struct snd_pcm_hw_params *params)
+{
+ struct adau *adau = snd_soc_dai_get_drvdata(dai);
+ unsigned int pll_rate;
+
+ switch (params_rate(params)) {
+ case 48000:
+ case 8000:
+ case 12000:
+ case 16000:
+ case 24000:
+ case 32000:
+ case 96000:
+ pll_rate = 48000 * 1024;
+ break;
+ case 44100:
+ case 7350:
+ case 11025:
+ case 14700:
+ case 22050:
+ case 29400:
+ case 88200:
+ pll_rate = 44100 * 1024;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK,
+ clk_get_rate(adau->mclk), pll_rate);
+}
+
static int adau17x1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
@@ -311,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
unsigned int freq;
int ret;
- if (adau->clk_src == ADAU17X1_CLK_SRC_PLL)
+ switch (adau->clk_src) {
+ case ADAU17X1_CLK_SRC_PLL_AUTO:
+ ret = adau17x1_auto_pll(dai, params);
+ if (ret)
+ return ret;
+ /* Fall-through */
+ case ADAU17X1_CLK_SRC_PLL:
freq = adau->pll_freq;
- else
+ break;
+ default:
freq = adau->sysclk;
+ break;
+ }
if (freq % params_rate(params) != 0)
return -EINVAL;
@@ -386,93 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
ADAU17X1_SERIAL_PORT1_DELAY_MASK, val);
}
-static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
- int source, unsigned int freq_in, unsigned int freq_out)
-{
- struct snd_soc_codec *codec = dai->codec;
- struct adau *adau = snd_soc_codec_get_drvdata(codec);
- unsigned int r, n, m, i, j;
- unsigned int div;
- int ret;
-
- if (freq_in < 8000000 || freq_in > 27000000)
- return -EINVAL;
-
- if (!freq_out) {
- r = 0;
- n = 0;
- m = 0;
- div = 0;
- } else {
- if (freq_out % freq_in != 0) {
- div = DIV_ROUND_UP(freq_in, 13500000);
- freq_in /= div;
- r = freq_out / freq_in;
- i = freq_out % freq_in;
- j = gcd(i, freq_in);
- n = i / j;
- m = freq_in / j;
- div--;
- } else {
- r = freq_out / freq_in;
- n = 0;
- m = 0;
- div = 0;
- }
- if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2)
- return -EINVAL;
- }
-
- adau->pll_regs[0] = m >> 8;
- adau->pll_regs[1] = m & 0xff;
- adau->pll_regs[2] = n >> 8;
- adau->pll_regs[3] = n & 0xff;
- adau->pll_regs[4] = (r << 3) | (div << 1);
- if (m != 0)
- adau->pll_regs[4] |= 1; /* Fractional mode */
-
- /* The PLL register is 6 bytes long and can only be written at once. */
- ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL,
- adau->pll_regs, ARRAY_SIZE(adau->pll_regs));
- if (ret)
- return ret;
-
- adau->pll_freq = freq_out;
-
- return 0;
-}
-
-static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai,
- int clk_id, unsigned int freq, int dir)
-{
- struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec);
- struct adau *adau = snd_soc_codec_get_drvdata(dai->codec);
-
- switch (clk_id) {
- case ADAU17X1_CLK_SRC_MCLK:
- case ADAU17X1_CLK_SRC_PLL:
- break;
- default:
- return -EINVAL;
- }
-
- adau->sysclk = freq;
-
- if (adau->clk_src != clk_id) {
- if (clk_id == ADAU17X1_CLK_SRC_PLL) {
- snd_soc_dapm_add_routes(dapm,
- &adau17x1_dapm_pll_route, 1);
- } else {
- snd_soc_dapm_del_routes(dapm,
- &adau17x1_dapm_pll_route, 1);
- }
- }
-
- adau->clk_src = clk_id;
-
- return 0;
-}
-
static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai,
unsigned int fmt)
{
@@ -857,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec)
ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes,
ARRAY_SIZE(adau17x1_no_dsp_dapm_routes));
}
+
+ if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK)
+ snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1);
+
return ret;
}
EXPORT_SYMBOL_GPL(adau17x1_add_routes);
@@ -879,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
const char *firmware_name)
{
struct adau *adau;
+ int ret;
if (IS_ERR(regmap))
return PTR_ERR(regmap);
@@ -887,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
if (!adau)
return -ENOMEM;
+ adau->mclk = devm_clk_get(dev, "mclk");
+ if (IS_ERR(adau->mclk)) {
+ if (PTR_ERR(adau->mclk) != -ENOENT)
+ return PTR_ERR(adau->mclk);
+ /* Clock is optional (for the driver) */
+ adau->mclk = NULL;
+ } else if (adau->mclk) {
+ adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO;
+
+ /*
+ * Any valid PLL output rate will work at this point, use one
+ * that is likely to be chosen later as well. The register will
+ * be written when the PLL is powered up for the first time.
+ */
+ ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024,
+ adau->pll_regs);
+ if (ret < 0)
+ return ret;
+
+ ret = clk_prepare_enable(adau->mclk);
+ if (ret)
+ return ret;
+ }
+
adau->regmap = regmap;
adau->switch_mode = switch_mode;
adau->type = type;
@@ -910,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
}
EXPORT_SYMBOL_GPL(adau17x1_probe);
+void adau17x1_remove(struct device *dev)
+{
+ struct adau *adau = dev_get_drvdata(dev);
+
+ snd_soc_unregister_codec(dev);
+ if (adau->mclk)
+ clk_disable_unprepare(adau->mclk);
+}
+EXPORT_SYMBOL_GPL(adau17x1_remove);
+
MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code");
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index 5ae87a084d97..bf04b7efee40 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -22,13 +22,18 @@ enum adau17x1_pll_src {
};
enum adau17x1_clk_src {
+ /* Automatically configure PLL based on the sample rate */
+ ADAU17X1_CLK_SRC_PLL_AUTO,
ADAU17X1_CLK_SRC_MCLK,
ADAU17X1_CLK_SRC_PLL,
};
+struct clk;
+
struct adau {
unsigned int sysclk;
unsigned int pll_freq;
+ struct clk *mclk;
enum adau17x1_clk_src clk_src;
enum adau17x1_type type;
@@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec);
int adau17x1_probe(struct device *dev, struct regmap *regmap,
enum adau17x1_type type, void (*switch_mode)(struct device *dev),
const char *firmware_name);
+void adau17x1_remove(struct device *dev);
int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec,
enum adau17x1_micbias_voltage micbias);
bool adau17x1_readable_register(struct device *dev, unsigned int reg);
diff --git a/sound/soc/codecs/adau7002.c b/sound/soc/codecs/adau7002.c
new file mode 100644
index 000000000000..9df72c6adcca
--- /dev/null
+++ b/sound/soc/codecs/adau7002.c
@@ -0,0 +1,80 @@
+/*
+ * ADAU7002 Stereo PDM-to-I2S/TDM converter driver
+ *
+ * Copyright 2014-2016 Analog Devices
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static const struct snd_soc_dapm_widget adau7002_widgets[] = {
+ SND_SOC_DAPM_INPUT("PDM_DAT"),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("IOVDD", 0, 0),
+};
+
+static const struct snd_soc_dapm_route adau7002_routes[] = {
+ { "Capture", NULL, "PDM_DAT" },
+ { "Capture", NULL, "IOVDD" },
+};
+
+static struct snd_soc_dai_driver adau7002_dai = {
+ .name = "adau7002-hifi",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 20,
+ },
+};
+
+static const struct snd_soc_codec_driver adau7002_codec_driver = {
+ .dapm_widgets = adau7002_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adau7002_widgets),
+ .dapm_routes = adau7002_routes,
+ .num_dapm_routes = ARRAY_SIZE(adau7002_routes),
+};
+
+static int adau7002_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &adau7002_codec_driver,
+ &adau7002_dai, 1);
+}
+
+static int adau7002_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id adau7002_dt_ids[] = {
+ { .compatible = "adi,adau7002", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, adau7002_dt_ids);
+#endif
+
+static struct platform_driver adau7002_driver = {
+ .driver = {
+ .name = "adau7002",
+ .of_match_table = of_match_ptr(adau7002_dt_ids),
+ },
+ .probe = adau7002_probe,
+ .remove = adau7002_remove,
+};
+module_platform_driver(adau7002_driver);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ADAU7002 Stereo PDM-to-I2S/TDM Converter driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 647f69de6baa..97798d250f08 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -146,6 +146,7 @@ static const struct regmap_config ak4613_regmap_cfg = {
.max_register = 0x16,
.reg_defaults = ak4613_reg,
.num_reg_defaults = ARRAY_SIZE(ak4613_reg),
+ .cache_type = REGCACHE_RBTREE,
};
static const struct of_device_id ak4613_of_match[] = {
@@ -436,15 +437,25 @@ static struct snd_soc_dai_driver ak4613_dai = {
.symmetric_rates = 1,
};
-static int ak4613_resume(struct snd_soc_codec *codec)
+static int ak4613_suspend(struct snd_soc_codec *codec)
{
struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+ regcache_cache_only(regmap, true);
regcache_mark_dirty(regmap);
+ return 0;
+}
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+ struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+ regcache_cache_only(regmap, false);
return regcache_sync(regmap);
}
static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+ .suspend = ak4613_suspend,
.resume = ak4613_resume,
.set_bias_level = ak4613_set_bias_level,
.controls = ak4613_snd_controls,
@@ -530,7 +541,6 @@ static int ak4613_i2c_remove(struct i2c_client *client)
static struct i2c_driver ak4613_i2c_driver = {
.driver = {
.name = "ak4613-codec",
- .owner = THIS_MODULE,
.of_match_table = ak4613_of_match,
},
.probe = ak4613_i2c_probe,
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4d8b9e49e8d6..cc941d66ec3d 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -523,15 +523,23 @@ static struct snd_soc_dai_driver ak4642_dai = {
.symmetric_rates = 1,
};
-static int ak4642_resume(struct snd_soc_codec *codec)
+static int ak4642_suspend(struct snd_soc_codec *codec)
{
struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+ regcache_cache_only(regmap, true);
regcache_mark_dirty(regmap);
- regcache_sync(regmap);
return 0;
}
+static int ak4642_resume(struct snd_soc_codec *codec)
+{
+ struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+ regcache_cache_only(regmap, false);
+ regcache_sync(regmap);
+ return 0;
+}
static int ak4642_probe(struct snd_soc_codec *codec)
{
struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
@@ -544,6 +552,7 @@ static int ak4642_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
.probe = ak4642_probe,
+ .suspend = ak4642_suspend,
.resume = ak4642_resume,
.set_bias_level = ak4642_set_bias_level,
.controls = ak4642_snd_controls,
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 664a8c044ffb..ecfdbfcae366 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -85,30 +85,9 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
- bool manual_ena = false;
int val;
- switch (arizona->type) {
- case WM5102:
- switch (arizona->rev) {
- case 0:
- break;
- default:
- manual_ena = true;
- break;
- }
- default:
- break;
- }
-
switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
- if (!priv->spk_ena && manual_ena) {
- regmap_write_async(arizona->regmap, 0x4f5, 0x25a);
- priv->spk_ena_pending = true;
- }
- break;
case SND_SOC_DAPM_POST_PMU:
val = snd_soc_read(codec, ARIZONA_INTERRUPT_RAW_STATUS_3);
if (val & ARIZONA_SPK_OVERHEAT_STS) {
@@ -120,33 +99,12 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
regmap_update_bits_async(arizona->regmap,
ARIZONA_OUTPUT_ENABLES_1,
1 << w->shift, 1 << w->shift);
-
- if (priv->spk_ena_pending) {
- msleep(75);
- regmap_write_async(arizona->regmap, 0x4f5, 0xda);
- priv->spk_ena_pending = false;
- priv->spk_ena++;
- }
break;
case SND_SOC_DAPM_PRE_PMD:
- if (manual_ena) {
- priv->spk_ena--;
- if (!priv->spk_ena)
- regmap_write_async(arizona->regmap,
- 0x4f5, 0x25a);
- }
-
regmap_update_bits_async(arizona->regmap,
ARIZONA_OUTPUT_ENABLES_1,
1 << w->shift, 0);
break;
- case SND_SOC_DAPM_POST_PMD:
- if (manual_ena) {
- if (!priv->spk_ena)
- regmap_write_async(arizona->regmap,
- 0x4f5, 0x0da);
- }
- break;
default:
break;
}
@@ -324,6 +282,17 @@ int arizona_init_gpio(struct snd_soc_codec *codec)
}
EXPORT_SYMBOL_GPL(arizona_init_gpio);
+int arizona_init_notifiers(struct snd_soc_codec *codec)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+
+ BLOCKING_INIT_NOTIFIER_HEAD(&arizona->notifier);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_notifiers);
+
const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
"None",
"Tone Generator 1",
@@ -619,7 +588,7 @@ const struct soc_enum arizona_asrc_rate1 =
arizona_rate_text, arizona_rate_val);
EXPORT_SYMBOL_GPL(arizona_asrc_rate1);
-static const char *arizona_vol_ramp_text[] = {
+static const char * const arizona_vol_ramp_text[] = {
"0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB",
"15ms/6dB", "30ms/6dB",
};
@@ -648,7 +617,7 @@ SOC_ENUM_SINGLE_DECL(arizona_out_vi_ramp,
arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_out_vi_ramp);
-static const char *arizona_lhpf_mode_text[] = {
+static const char * const arizona_lhpf_mode_text[] = {
"Low-pass", "High-pass"
};
@@ -676,7 +645,7 @@ SOC_ENUM_SINGLE_DECL(arizona_lhpf4_mode,
arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf4_mode);
-static const char *arizona_ng_hold_text[] = {
+static const char * const arizona_ng_hold_text[] = {
"30ms", "120ms", "250ms", "500ms",
};
@@ -810,6 +779,14 @@ const struct soc_enum arizona_output_anc_src[] = {
};
EXPORT_SYMBOL_GPL(arizona_output_anc_src);
+const struct snd_kcontrol_new arizona_voice_trigger_switch[] = {
+ SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0),
+ SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 1, 1, 0),
+ SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 2, 1, 0),
+ SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 3, 1, 0),
+};
+EXPORT_SYMBOL_GPL(arizona_voice_trigger_switch);
+
static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
{
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
@@ -2573,6 +2550,30 @@ int arizona_lhpf_coeff_put(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_GPL(arizona_lhpf_coeff_put);
+int arizona_register_notifier(struct snd_soc_codec *codec,
+ struct notifier_block *nb,
+ int (*notify)(struct notifier_block *nb,
+ unsigned long action, void *data))
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+
+ nb->notifier_call = notify;
+
+ return blocking_notifier_chain_register(&arizona->notifier, nb);
+}
+EXPORT_SYMBOL_GPL(arizona_register_notifier);
+
+int arizona_unregister_notifier(struct snd_soc_codec *codec,
+ struct notifier_block *nb)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+
+ return blocking_notifier_chain_unregister(&arizona->notifier, nb);
+}
+EXPORT_SYMBOL_GPL(arizona_unregister_notifier);
+
MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ce0531b8c632..69da1ef3a17c 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -63,6 +63,9 @@
#define ARIZONA_DVFS_SR1_RQ 0x001
#define ARIZONA_DVFS_ADSP1_RQ 0x100
+/* Notifier events */
+#define ARIZONA_NOTIFY_VOICE_TRIGGER 0x1
+
struct arizona;
struct wm_adsp;
@@ -87,14 +90,15 @@ struct arizona_priv {
unsigned int out_down_pending;
unsigned int out_down_delay;
- unsigned int spk_ena:2;
- unsigned int spk_ena_pending:1;
-
unsigned int dvfs_reqs;
struct mutex dvfs_lock;
bool dvfs_cached;
};
+struct arizona_voice_trigger_info {
+ int core;
+};
+
#define ARIZONA_NUM_MIXER_INPUTS 104
extern const unsigned int arizona_mixer_tlv[];
@@ -248,6 +252,8 @@ extern const struct soc_enum arizona_anc_input_src[];
extern const struct soc_enum arizona_anc_ng_enum;
extern const struct soc_enum arizona_output_anc_src[];
+extern const struct snd_kcontrol_new arizona_voice_trigger_switch[];
+
extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event);
@@ -306,6 +312,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source,
extern int arizona_init_spk(struct snd_soc_codec *codec);
extern int arizona_init_gpio(struct snd_soc_codec *codec);
extern int arizona_init_mono(struct snd_soc_codec *codec);
+extern int arizona_init_notifiers(struct snd_soc_codec *codec);
extern int arizona_free_spk(struct snd_soc_codec *codec);
@@ -317,4 +324,13 @@ int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
extern const char *arizona_sample_rate_val_to_name(unsigned int rate_val);
+
+extern int arizona_register_notifier(struct snd_soc_codec *codec,
+ struct notifier_block *nb,
+ int (*notify)(struct notifier_block *nb,
+ unsigned long action,
+ void *data));
+extern int arizona_unregister_notifier(struct snd_soc_codec *codec,
+ struct notifier_block *nb);
+
#endif
diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c
index b084ad113e96..2a8d0ee141d4 100644
--- a/sound/soc/codecs/bt-sco.c
+++ b/sound/soc/codecs/bt-sco.c
@@ -25,22 +25,41 @@ static const struct snd_soc_dapm_route bt_sco_routes[] = {
{ "TX", NULL, "Playback" },
};
-static struct snd_soc_dai_driver bt_sco_dai = {
- .name = "bt-sco-pcm",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 1,
- .channels_max = 1,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
- .capture = {
- .stream_name = "Capture",
- .channels_min = 1,
- .channels_max = 1,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+static struct snd_soc_dai_driver bt_sco_dai[] = {
+ {
+ .name = "bt-sco-pcm",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
},
+ {
+ .name = "bt-sco-pcm-wb",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ }
};
static struct snd_soc_codec_driver soc_codec_dev_bt_sco = {
@@ -53,7 +72,7 @@ static struct snd_soc_codec_driver soc_codec_dev_bt_sco = {
static int bt_sco_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco,
- &bt_sco_dai, 1);
+ bt_sco_dai, ARRAY_SIZE(bt_sco_dai));
}
static int bt_sco_remove(struct platform_device *pdev)
@@ -77,6 +96,7 @@ MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids);
#if defined(CONFIG_OF)
static const struct of_device_id bt_sco_codec_of_match[] = {
{ .compatible = "delta,dfbmcs320", },
+ { .compatible = "linux,bt-sco", },
{},
};
MODULE_DEVICE_TABLE(of, bt_sco_codec_of_match);
diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c
new file mode 100644
index 000000000000..6f9c1addcd7f
--- /dev/null
+++ b/sound/soc/codecs/cs35l33.c
@@ -0,0 +1,1303 @@
+/*
+ * cs35l33.c -- CS35L33 ALSA SoC audio driver
+ *
+ * Copyright 2016 Cirrus Logic, Inc.
+ *
+ * Author: Paul Handrigan <paul.handrigan@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <sound/cs35l33.h>
+#include <linux/pm_runtime.h>
+#include <linux/regulator/consumer.h>
+#include <linux/regulator/machine.h>
+#include <linux/of_gpio.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
+
+#include "cs35l33.h"
+
+#define CS35L33_BOOT_DELAY 50
+
+struct cs35l33_private {
+ struct snd_soc_codec *codec;
+ struct cs35l33_pdata pdata;
+ struct regmap *regmap;
+ struct gpio_desc *reset_gpio;
+ bool amp_cal;
+ int mclk_int;
+ struct regulator_bulk_data core_supplies[2];
+ int num_core_supplies;
+ bool is_tdm_mode;
+ bool enable_soft_ramp;
+};
+
+static const struct reg_default cs35l33_reg[] = {
+ {CS35L33_PWRCTL1, 0x85},
+ {CS35L33_PWRCTL2, 0xFE},
+ {CS35L33_CLK_CTL, 0x0C},
+ {CS35L33_BST_PEAK_CTL, 0x90},
+ {CS35L33_PROTECT_CTL, 0x55},
+ {CS35L33_BST_CTL1, 0x00},
+ {CS35L33_BST_CTL2, 0x01},
+ {CS35L33_ADSP_CTL, 0x00},
+ {CS35L33_ADC_CTL, 0xC8},
+ {CS35L33_DAC_CTL, 0x14},
+ {CS35L33_DIG_VOL_CTL, 0x00},
+ {CS35L33_CLASSD_CTL, 0x04},
+ {CS35L33_AMP_CTL, 0x90},
+ {CS35L33_INT_MASK_1, 0xFF},
+ {CS35L33_INT_MASK_2, 0xFF},
+ {CS35L33_DIAG_LOCK, 0x00},
+ {CS35L33_DIAG_CTRL_1, 0x40},
+ {CS35L33_DIAG_CTRL_2, 0x00},
+ {CS35L33_HG_MEMLDO_CTL, 0x62},
+ {CS35L33_HG_REL_RATE, 0x03},
+ {CS35L33_LDO_DEL, 0x12},
+ {CS35L33_HG_HEAD, 0x0A},
+ {CS35L33_HG_EN, 0x05},
+ {CS35L33_TX_VMON, 0x00},
+ {CS35L33_TX_IMON, 0x03},
+ {CS35L33_TX_VPMON, 0x02},
+ {CS35L33_TX_VBSTMON, 0x05},
+ {CS35L33_TX_FLAG, 0x06},
+ {CS35L33_TX_EN1, 0x00},
+ {CS35L33_TX_EN2, 0x00},
+ {CS35L33_TX_EN3, 0x00},
+ {CS35L33_TX_EN4, 0x00},
+ {CS35L33_RX_AUD, 0x40},
+ {CS35L33_RX_SPLY, 0x03},
+ {CS35L33_RX_ALIVE, 0x04},
+ {CS35L33_BST_CTL4, 0x63},
+};
+
+static const struct reg_sequence cs35l33_patch[] = {
+ { 0x00, 0x99, 0 },
+ { 0x59, 0x02, 0 },
+ { 0x52, 0x30, 0 },
+ { 0x39, 0x45, 0 },
+ { 0x57, 0x30, 0 },
+ { 0x2C, 0x68, 0 },
+ { 0x00, 0x00, 0 },
+};
+
+static bool cs35l33_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS35L33_DEVID_AB:
+ case CS35L33_DEVID_CD:
+ case CS35L33_DEVID_E:
+ case CS35L33_REV_ID:
+ case CS35L33_INT_STATUS_1:
+ case CS35L33_INT_STATUS_2:
+ case CS35L33_HG_STATUS:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool cs35l33_writeable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ /* these are read only registers */
+ case CS35L33_DEVID_AB:
+ case CS35L33_DEVID_CD:
+ case CS35L33_DEVID_E:
+ case CS35L33_REV_ID:
+ case CS35L33_INT_STATUS_1:
+ case CS35L33_INT_STATUS_2:
+ case CS35L33_HG_STATUS:
+ return false;
+ default:
+ return true;
+ }
+}
+
+static bool cs35l33_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS35L33_DEVID_AB:
+ case CS35L33_DEVID_CD:
+ case CS35L33_DEVID_E:
+ case CS35L33_REV_ID:
+ case CS35L33_PWRCTL1:
+ case CS35L33_PWRCTL2:
+ case CS35L33_CLK_CTL:
+ case CS35L33_BST_PEAK_CTL:
+ case CS35L33_PROTECT_CTL:
+ case CS35L33_BST_CTL1:
+ case CS35L33_BST_CTL2:
+ case CS35L33_ADSP_CTL:
+ case CS35L33_ADC_CTL:
+ case CS35L33_DAC_CTL:
+ case CS35L33_DIG_VOL_CTL:
+ case CS35L33_CLASSD_CTL:
+ case CS35L33_AMP_CTL:
+ case CS35L33_INT_MASK_1:
+ case CS35L33_INT_MASK_2:
+ case CS35L33_INT_STATUS_1:
+ case CS35L33_INT_STATUS_2:
+ case CS35L33_DIAG_LOCK:
+ case CS35L33_DIAG_CTRL_1:
+ case CS35L33_DIAG_CTRL_2:
+ case CS35L33_HG_MEMLDO_CTL:
+ case CS35L33_HG_REL_RATE:
+ case CS35L33_LDO_DEL:
+ case CS35L33_HG_HEAD:
+ case CS35L33_HG_EN:
+ case CS35L33_TX_VMON:
+ case CS35L33_TX_IMON:
+ case CS35L33_TX_VPMON:
+ case CS35L33_TX_VBSTMON:
+ case CS35L33_TX_FLAG:
+ case CS35L33_TX_EN1:
+ case CS35L33_TX_EN2:
+ case CS35L33_TX_EN3:
+ case CS35L33_TX_EN4:
+ case CS35L33_RX_AUD:
+ case CS35L33_RX_SPLY:
+ case CS35L33_RX_ALIVE:
+ case CS35L33_BST_CTL4:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0);
+static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0);
+
+static const struct snd_kcontrol_new cs35l33_snd_controls[] = {
+
+ SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL,
+ 4, 0x09, 0, classd_ctl_tlv),
+ SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL,
+ 0, 0x34, 0xE4, dac_tlv),
+};
+
+static int cs35l33_spkrdrv_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);
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (!priv->amp_cal) {
+ usleep_range(8000, 9000);
+ priv->amp_cal = true;
+ regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL,
+ CS35L33_AMP_CAL, 0);
+ dev_dbg(codec->dev, "Amp calibration done\n");
+ }
+ dev_dbg(codec->dev, "Amp turned on\n");
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ dev_dbg(codec->dev, "Amp turned off\n");
+ break;
+ default:
+ dev_err(codec->dev, "Invalid event = 0x%x\n", event);
+ break;
+ }
+
+ return 0;
+}
+
+static int cs35l33_sdin_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);
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL1,
+ CS35L33_PDN_BST, 0);
+ val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM;
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL2,
+ CS35L33_PDN_TDM, val);
+ dev_dbg(codec->dev, "BST turned on\n");
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ dev_dbg(codec->dev, "SDIN turned on\n");
+ if (!priv->amp_cal) {
+ regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL,
+ CS35L33_AMP_CAL, CS35L33_AMP_CAL);
+ dev_dbg(codec->dev, "Amp calibration started\n");
+ usleep_range(10000, 11000);
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL2,
+ CS35L33_PDN_TDM, CS35L33_PDN_TDM);
+ usleep_range(4000, 4100);
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL1,
+ CS35L33_PDN_BST, CS35L33_PDN_BST);
+ dev_dbg(codec->dev, "BST and SDIN turned off\n");
+ break;
+ default:
+ dev_err(codec->dev, "Invalid event = 0x%x\n", event);
+
+ }
+
+ return 0;
+}
+
+static int cs35l33_sdout_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);
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM;
+ unsigned int mask2 = CS35L33_SDOUT_3ST_TDM;
+ unsigned int val, val2;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (priv->is_tdm_mode) {
+ /* set sdout_3st_i2s and reset pdn_tdm */
+ val = CS35L33_SDOUT_3ST_I2S;
+ /* reset sdout_3st_tdm */
+ val2 = 0;
+ } else {
+ /* reset sdout_3st_i2s and set pdn_tdm */
+ val = CS35L33_PDN_TDM;
+ /* set sdout_3st_tdm */
+ val2 = CS35L33_SDOUT_3ST_TDM;
+ }
+ dev_dbg(codec->dev, "SDOUT turned on\n");
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM;
+ val2 = CS35L33_SDOUT_3ST_TDM;
+ dev_dbg(codec->dev, "SDOUT turned off\n");
+ break;
+ default:
+ dev_err(codec->dev, "Invalid event = 0x%x\n", event);
+ return 0;
+ }
+
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL2,
+ mask, val);
+ regmap_update_bits(priv->regmap, CS35L33_CLK_CTL,
+ mask2, val2);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = {
+
+ SND_SOC_DAPM_OUTPUT("SPK"),
+ SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0,
+ cs35l33_spkrdrv_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2,
+ 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_INPUT("MON"),
+
+ SND_SOC_DAPM_ADC("VMON", NULL,
+ CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1),
+ SND_SOC_DAPM_ADC("IMON", NULL,
+ CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1),
+ SND_SOC_DAPM_ADC("VPMON", NULL,
+ CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1),
+ SND_SOC_DAPM_ADC("VBSTMON", NULL,
+ CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1),
+
+ SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0,
+ cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_PRE_PMD),
+};
+
+static const struct snd_soc_dapm_route cs35l33_audio_map[] = {
+ {"SDIN", NULL, "CS35L33 Playback"},
+ {"SPKDRV", NULL, "SDIN"},
+ {"SPK", NULL, "SPKDRV"},
+
+ {"VMON", NULL, "MON"},
+ {"IMON", NULL, "MON"},
+
+ {"SDOUT", NULL, "VMON"},
+ {"SDOUT", NULL, "IMON"},
+ {"CS35L33 Capture", NULL, "SDOUT"},
+};
+
+static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = {
+ {"SPKDRV", NULL, "VPMON"},
+ {"VPMON", NULL, "CS35L33 Playback"},
+};
+
+static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = {
+ {"SDOUT", NULL, "VPMON"},
+ {"VPMON", NULL, "MON"},
+ {"SDOUT", NULL, "VBSTMON"},
+ {"VBSTMON", NULL, "MON"},
+};
+
+static int cs35l33_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ unsigned int val;
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL1,
+ CS35L33_PDN_ALL, 0);
+ regmap_update_bits(priv->regmap, CS35L33_CLK_CTL,
+ CS35L33_MCLKDIS, 0);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL1,
+ CS35L33_PDN_ALL, CS35L33_PDN_ALL);
+ regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val);
+ usleep_range(1000, 1100);
+ if (val & CS35L33_PDN_DONE)
+ regmap_update_bits(priv->regmap, CS35L33_CLK_CTL,
+ CS35L33_MCLKDIS, CS35L33_MCLKDIS);
+ break;
+ case SND_SOC_BIAS_OFF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+struct cs35l33_mclk_div {
+ int mclk;
+ int srate;
+ u8 adsp_rate;
+ u8 int_fs_ratio;
+};
+
+static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = {
+ /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */
+ {5644800, 11025, 0x4, CS35L33_INT_FS_RATE},
+ {5644800, 22050, 0x8, CS35L33_INT_FS_RATE},
+ {5644800, 44100, 0xC, CS35L33_INT_FS_RATE},
+
+ {6000000, 8000, 0x1, 0},
+ {6000000, 11025, 0x2, 0},
+ {6000000, 11029, 0x3, 0},
+ {6000000, 12000, 0x4, 0},
+ {6000000, 16000, 0x5, 0},
+ {6000000, 22050, 0x6, 0},
+ {6000000, 22059, 0x7, 0},
+ {6000000, 24000, 0x8, 0},
+ {6000000, 32000, 0x9, 0},
+ {6000000, 44100, 0xA, 0},
+ {6000000, 44118, 0xB, 0},
+ {6000000, 48000, 0xC, 0},
+
+ {6144000, 8000, 0x1, CS35L33_INT_FS_RATE},
+ {6144000, 12000, 0x4, CS35L33_INT_FS_RATE},
+ {6144000, 16000, 0x5, CS35L33_INT_FS_RATE},
+ {6144000, 24000, 0x8, CS35L33_INT_FS_RATE},
+ {6144000, 32000, 0x9, CS35L33_INT_FS_RATE},
+ {6144000, 48000, 0xC, CS35L33_INT_FS_RATE},
+};
+
+static int cs35l33_get_mclk_coeff(int mclk, int srate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) {
+ if (cs35l33_mclk_coeffs[i].mclk == mclk &&
+ cs35l33_mclk_coeffs[i].srate == srate)
+ return i;
+ }
+ return -EINVAL;
+}
+
+static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL,
+ CS35L33_MS_MASK, CS35L33_MS_MASK);
+ dev_dbg(codec->dev, "Audio port in master mode\n");
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL,
+ CS35L33_MS_MASK, 0);
+ dev_dbg(codec->dev, "Audio port in slave mode\n");
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ /*
+ * tdm mode in cs35l33 resembles dsp-a mode very
+ * closely, it is dsp-a with fsync shifted left by half bclk
+ */
+ priv->is_tdm_mode = true;
+ dev_dbg(codec->dev, "Audio port in TDM mode\n");
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ priv->is_tdm_mode = false;
+ dev_dbg(codec->dev, "Audio port in I2S mode\n");
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+ int sample_size = params_width(params);
+ int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params));
+
+ if (coeff < 0)
+ return coeff;
+
+ regmap_update_bits(priv->regmap, CS35L33_CLK_CTL,
+ CS35L33_ADSP_FS | CS35L33_INT_FS_RATE,
+ cs35l33_mclk_coeffs[coeff].int_fs_ratio
+ | cs35l33_mclk_coeffs[coeff].adsp_rate);
+
+ if (priv->is_tdm_mode) {
+ sample_size = (sample_size / 8) - 1;
+ if (sample_size > 2)
+ sample_size = 2;
+ regmap_update_bits(priv->regmap, CS35L33_RX_AUD,
+ CS35L33_AUDIN_RX_DEPTH,
+ sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT);
+ }
+
+ dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n",
+ params_rate(params), params_width(params));
+
+ return 0;
+}
+
+static const unsigned int cs35l33_src_rates[] = {
+ 8000, 11025, 11029, 12000, 16000, 22050,
+ 22059, 24000, 32000, 44100, 44118, 48000
+};
+
+static const struct snd_pcm_hw_constraint_list cs35l33_constraints = {
+ .count = ARRAY_SIZE(cs35l33_src_rates),
+ .list = cs35l33_src_rates,
+};
+
+static int cs35l33_pcm_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &cs35l33_constraints);
+ return 0;
+}
+
+static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (tristate) {
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL2,
+ CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S);
+ regmap_update_bits(priv->regmap, CS35L33_CLK_CTL,
+ CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM);
+ } else {
+ regmap_update_bits(priv->regmap, CS35L33_PWRCTL2,
+ CS35L33_SDOUT_3ST_I2S, 0);
+ regmap_update_bits(priv->regmap, CS35L33_CLK_CTL,
+ CS35L33_SDOUT_3ST_TDM, 0);
+ }
+
+ return 0;
+}
+
+static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int reg, bit_pos, i;
+ int slot, slot_num;
+
+ if (slot_width != 8)
+ return -EINVAL;
+
+ /* scan rx_mask for aud slot */
+ slot = ffs(rx_mask) - 1;
+ if (slot >= 0) {
+ regmap_update_bits(priv->regmap, CS35L33_RX_AUD,
+ CS35L33_X_LOC, slot);
+ dev_dbg(codec->dev, "Audio starts from slots %d", slot);
+ }
+
+ /*
+ * scan tx_mask: vmon(2 slots); imon (2 slots);
+ * vpmon (1 slot) vbstmon (1 slot)
+ */
+ slot = ffs(tx_mask) - 1;
+ slot_num = 0;
+
+ for (i = 0; i < 2 ; i++) {
+ /* disable vpmon/vbstmon: enable later if set in tx_mask */
+ regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i,
+ CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE
+ | CS35L33_X_LOC);
+ }
+
+ /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/
+ snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route,
+ ARRAY_SIZE(cs35l33_vp_vbst_mon_route));
+
+ while (slot >= 0) {
+ /* configure VMON_TX_LOC */
+ if (slot_num == 0) {
+ regmap_update_bits(priv->regmap, CS35L33_TX_VMON,
+ CS35L33_X_STATE | CS35L33_X_LOC, slot);
+ dev_dbg(codec->dev, "VMON enabled in slots %d-%d",
+ slot, slot + 1);
+ }
+
+ /* configure IMON_TX_LOC */
+ if (slot_num == 3) {
+ regmap_update_bits(priv->regmap, CS35L33_TX_IMON,
+ CS35L33_X_STATE | CS35L33_X_LOC, slot);
+ dev_dbg(codec->dev, "IMON enabled in slots %d-%d",
+ slot, slot + 1);
+ }
+
+ /* configure VPMON_TX_LOC */
+ if (slot_num == 4) {
+ regmap_update_bits(priv->regmap, CS35L33_TX_VPMON,
+ CS35L33_X_STATE | CS35L33_X_LOC, slot);
+ snd_soc_dapm_add_routes(dapm,
+ &cs35l33_vp_vbst_mon_route[0], 2);
+ dev_dbg(codec->dev, "VPMON enabled in slots %d", slot);
+ }
+
+ /* configure VBSTMON_TX_LOC */
+ if (slot_num == 5) {
+ regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON,
+ CS35L33_X_STATE | CS35L33_X_LOC, slot);
+ snd_soc_dapm_add_routes(dapm,
+ &cs35l33_vp_vbst_mon_route[2], 2);
+ dev_dbg(codec->dev,
+ "VBSTMON enabled in slots %d", slot);
+ }
+
+ /* Enable the relevant tx slot */
+ reg = CS35L33_TX_EN4 - (slot/8);
+ bit_pos = slot - ((slot / 8) * (8));
+ regmap_update_bits(priv->regmap, reg,
+ 1 << bit_pos, 1 << bit_pos);
+
+ tx_mask &= ~(1 << slot);
+ slot = ffs(tx_mask) - 1;
+ slot_num++;
+ }
+
+ return 0;
+}
+
+static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec,
+ int clk_id, int source, unsigned int freq, int dir)
+{
+ struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec);
+
+ switch (freq) {
+ case CS35L33_MCLK_5644:
+ case CS35L33_MCLK_6:
+ case CS35L33_MCLK_6144:
+ regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL,
+ CS35L33_MCLKDIV2, 0);
+ cs35l33->mclk_int = freq;
+ break;
+ case CS35L33_MCLK_11289:
+ case CS35L33_MCLK_12:
+ case CS35L33_MCLK_12288:
+ regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL,
+ CS35L33_MCLKDIV2, CS35L33_MCLKDIV2);
+ cs35l33->mclk_int = freq/2;
+ break;
+ default:
+ cs35l33->mclk_int = 0;
+ return -EINVAL;
+ }
+
+ dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n",
+ freq, cs35l33->mclk_int);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops cs35l33_ops = {
+ .startup = cs35l33_pcm_startup,
+ .set_tristate = cs35l33_set_tristate,
+ .set_fmt = cs35l33_set_dai_fmt,
+ .hw_params = cs35l33_pcm_hw_params,
+ .set_tdm_slot = cs35l33_set_tdm_slot,
+};
+
+static struct snd_soc_dai_driver cs35l33_dai = {
+ .name = "cs35l33-dai",
+ .id = 0,
+ .playback = {
+ .stream_name = "CS35L33 Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = CS35L33_RATES,
+ .formats = CS35L33_FORMATS,
+ },
+ .capture = {
+ .stream_name = "CS35L33 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = CS35L33_RATES,
+ .formats = CS35L33_FORMATS,
+ },
+ .ops = &cs35l33_ops,
+ .symmetric_rates = 1,
+};
+
+static int cs35l33_set_hg_data(struct snd_soc_codec *codec,
+ struct cs35l33_pdata *pdata)
+{
+ struct cs35l33_hg *hg_config = &pdata->hg_config;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (hg_config->enable_hg_algo) {
+ regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL,
+ CS35L33_MEM_DEPTH_MASK,
+ hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT);
+ regmap_write(priv->regmap, CS35L33_HG_REL_RATE,
+ hg_config->release_rate);
+ regmap_update_bits(priv->regmap, CS35L33_HG_HEAD,
+ CS35L33_HD_RM_MASK,
+ hg_config->hd_rm << CS35L33_HD_RM_SHIFT);
+ regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL,
+ CS35L33_LDO_THLD_MASK,
+ hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT);
+ regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL,
+ CS35L33_LDO_DISABLE_MASK,
+ hg_config->ldo_path_disable <<
+ CS35L33_LDO_DISABLE_SHIFT);
+ regmap_update_bits(priv->regmap, CS35L33_LDO_DEL,
+ CS35L33_LDO_ENTRY_DELAY_MASK,
+ hg_config->ldo_entry_delay <<
+ CS35L33_LDO_ENTRY_DELAY_SHIFT);
+ if (hg_config->vp_hg_auto) {
+ regmap_update_bits(priv->regmap, CS35L33_HG_EN,
+ CS35L33_VP_HG_AUTO_MASK,
+ CS35L33_VP_HG_AUTO_MASK);
+ snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route,
+ ARRAY_SIZE(cs35l33_vphg_auto_route));
+ }
+ regmap_update_bits(priv->regmap, CS35L33_HG_EN,
+ CS35L33_VP_HG_MASK,
+ hg_config->vp_hg << CS35L33_VP_HG_SHIFT);
+ regmap_update_bits(priv->regmap, CS35L33_LDO_DEL,
+ CS35L33_VP_HG_RATE_MASK,
+ hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT);
+ regmap_update_bits(priv->regmap, CS35L33_LDO_DEL,
+ CS35L33_VP_HG_VA_MASK,
+ hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT);
+ regmap_update_bits(priv->regmap, CS35L33_HG_EN,
+ CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK);
+ }
+ return 0;
+}
+
+static int cs35l33_set_bst_ipk(struct snd_soc_codec *codec, unsigned int bst)
+{
+ struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0, steps = 0;
+
+ /* Boost current in uA */
+ if (bst > 3600000 || bst < 1850000) {
+ dev_err(codec->dev, "Invalid boost current %d\n", bst);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ if (bst % 15625) {
+ dev_err(codec->dev, "Current not a multiple of 15625uA (%d)\n",
+ bst);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ while (bst > 1850000) {
+ bst -= 15625;
+ steps++;
+ }
+
+ regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL,
+ steps+0x70);
+
+err:
+ return ret;
+}
+
+static int cs35l33_probe(struct snd_soc_codec *codec)
+{
+ struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec);
+
+ cs35l33->codec = codec;
+ pm_runtime_get_sync(codec->dev);
+
+ regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL,
+ CS35L33_ALIVE_WD_DIS, 0x8);
+ regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2,
+ CS35L33_ALIVE_WD_DIS2,
+ CS35L33_ALIVE_WD_DIS2);
+
+ /* Set Platform Data */
+ regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1,
+ CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl);
+ regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL,
+ CS35L33_AMP_DRV_SEL_MASK,
+ cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT);
+
+ if (cs35l33->pdata.boost_ipk)
+ cs35l33_set_bst_ipk(codec, cs35l33->pdata.boost_ipk);
+
+ if (cs35l33->enable_soft_ramp) {
+ snd_soc_update_bits(codec, CS35L33_DAC_CTL,
+ CS35L33_DIGSFT, CS35L33_DIGSFT);
+ snd_soc_update_bits(codec, CS35L33_DAC_CTL,
+ CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate);
+ } else {
+ snd_soc_update_bits(codec, CS35L33_DAC_CTL,
+ CS35L33_DIGSFT, 0);
+ }
+
+ /* update IMON scaling rate if different from default of 0x8 */
+ if (cs35l33->pdata.imon_adc_scale != 0x8)
+ snd_soc_update_bits(codec, CS35L33_ADC_CTL,
+ CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale);
+
+ cs35l33_set_hg_data(codec, &(cs35l33->pdata));
+
+ /*
+ * unmask important interrupts that causes the chip to enter
+ * speaker safe mode and hence deserves user attention
+ */
+ regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1,
+ CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT |
+ CS35L33_M_CAL_ERR, 0);
+
+ pm_runtime_put_sync(codec->dev);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = {
+ .probe = cs35l33_probe,
+
+ .set_bias_level = cs35l33_set_bias_level,
+ .set_sysclk = cs35l33_codec_set_sysclk,
+
+ .dapm_widgets = cs35l33_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets),
+ .dapm_routes = cs35l33_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map),
+ .controls = cs35l33_snd_controls,
+ .num_controls = ARRAY_SIZE(cs35l33_snd_controls),
+
+ .idle_bias_off = true,
+};
+
+static const struct regmap_config cs35l33_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS35L33_MAX_REGISTER,
+ .reg_defaults = cs35l33_reg,
+ .num_reg_defaults = ARRAY_SIZE(cs35l33_reg),
+ .volatile_reg = cs35l33_volatile_register,
+ .readable_reg = cs35l33_readable_register,
+ .writeable_reg = cs35l33_writeable_register,
+ .cache_type = REGCACHE_RBTREE,
+ .use_single_rw = true,
+};
+
+static int __maybe_unused cs35l33_runtime_resume(struct device *dev)
+{
+ struct cs35l33_private *cs35l33 = dev_get_drvdata(dev);
+ int ret;
+
+ dev_dbg(dev, "%s\n", __func__);
+
+ if (cs35l33->reset_gpio)
+ gpiod_set_value_cansleep(cs35l33->reset_gpio, 0);
+
+ ret = regulator_bulk_enable(cs35l33->num_core_supplies,
+ cs35l33->core_supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable core supplies: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(cs35l33->regmap, false);
+
+ if (cs35l33->reset_gpio)
+ gpiod_set_value_cansleep(cs35l33->reset_gpio, 1);
+
+ msleep(CS35L33_BOOT_DELAY);
+
+ ret = regcache_sync(cs35l33->regmap);
+ if (ret != 0) {
+ dev_err(dev, "Failed to restore register cache\n");
+ goto err;
+ }
+
+ return 0;
+
+err:
+ regcache_cache_only(cs35l33->regmap, true);
+ regulator_bulk_disable(cs35l33->num_core_supplies,
+ cs35l33->core_supplies);
+
+ return ret;
+}
+
+static int __maybe_unused cs35l33_runtime_suspend(struct device *dev)
+{
+ struct cs35l33_private *cs35l33 = dev_get_drvdata(dev);
+
+ dev_dbg(dev, "%s\n", __func__);
+
+ /* redo the calibration in next power up */
+ cs35l33->amp_cal = false;
+
+ regcache_cache_only(cs35l33->regmap, true);
+ regcache_mark_dirty(cs35l33->regmap);
+ regulator_bulk_disable(cs35l33->num_core_supplies,
+ cs35l33->core_supplies);
+
+ return 0;
+}
+
+static const struct dev_pm_ops cs35l33_pm_ops = {
+ SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend,
+ cs35l33_runtime_resume,
+ NULL)
+};
+
+static int cs35l33_get_hg_data(const struct device_node *np,
+ struct cs35l33_pdata *pdata)
+{
+ struct device_node *hg;
+ struct cs35l33_hg *hg_config = &pdata->hg_config;
+ u32 val32;
+
+ hg = of_get_child_by_name(np, "cirrus,hg-algo");
+ hg_config->enable_hg_algo = hg ? true : false;
+
+ if (hg_config->enable_hg_algo) {
+ if (of_property_read_u32(hg, "cirrus,mem-depth", &val32) >= 0)
+ hg_config->mem_depth = val32;
+ if (of_property_read_u32(hg, "cirrus,release-rate",
+ &val32) >= 0)
+ hg_config->release_rate = val32;
+ if (of_property_read_u32(hg, "cirrus,ldo-thld", &val32) >= 0)
+ hg_config->ldo_thld = val32;
+ if (of_property_read_u32(hg, "cirrus,ldo-path-disable",
+ &val32) >= 0)
+ hg_config->ldo_path_disable = val32;
+ if (of_property_read_u32(hg, "cirrus,ldo-entry-delay",
+ &val32) >= 0)
+ hg_config->ldo_entry_delay = val32;
+
+ hg_config->vp_hg_auto = of_property_read_bool(hg,
+ "cirrus,vp-hg-auto");
+
+ if (of_property_read_u32(hg, "cirrus,vp-hg", &val32) >= 0)
+ hg_config->vp_hg = val32;
+ if (of_property_read_u32(hg, "cirrus,vp-hg-rate", &val32) >= 0)
+ hg_config->vp_hg_rate = val32;
+ if (of_property_read_u32(hg, "cirrus,vp-hg-va", &val32) >= 0)
+ hg_config->vp_hg_va = val32;
+ }
+
+ of_node_put(hg);
+
+ return 0;
+}
+
+static irqreturn_t cs35l33_irq_thread(int irq, void *data)
+{
+ struct cs35l33_private *cs35l33 = data;
+ struct snd_soc_codec *codec = cs35l33->codec;
+ unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2;
+
+ regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2,
+ &sticky_val2);
+ regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1,
+ &sticky_val1);
+ regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2);
+ regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1);
+
+ /* Check to see if the unmasked bits are active,
+ * if not then exit.
+ */
+ if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2))
+ return IRQ_NONE;
+
+ regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1,
+ &current_val);
+
+ /* handle the interrupts */
+
+ if (sticky_val1 & CS35L33_AMP_SHORT) {
+ dev_crit(codec->dev, "Amp short error\n");
+ if (!(current_val & CS35L33_AMP_SHORT)) {
+ dev_dbg(codec->dev,
+ "Amp short error release\n");
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL,
+ CS35L33_AMP_SHORT_RLS, 0);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL,
+ CS35L33_AMP_SHORT_RLS,
+ CS35L33_AMP_SHORT_RLS);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS,
+ 0);
+ }
+ }
+
+ if (sticky_val1 & CS35L33_CAL_ERR) {
+ dev_err(codec->dev, "Cal error\n");
+
+ /* redo the calibration in next power up */
+ cs35l33->amp_cal = false;
+
+ if (!(current_val & CS35L33_CAL_ERR)) {
+ dev_dbg(codec->dev, "Cal error release\n");
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS,
+ 0);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS,
+ CS35L33_CAL_ERR_RLS);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS,
+ 0);
+ }
+ }
+
+ if (sticky_val1 & CS35L33_OTE) {
+ dev_crit(codec->dev, "Over temperature error\n");
+ if (!(current_val & CS35L33_OTE)) {
+ dev_dbg(codec->dev,
+ "Over temperature error release\n");
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_OTE_RLS,
+ CS35L33_OTE_RLS);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0);
+ }
+ }
+
+ if (sticky_val1 & CS35L33_OTW) {
+ dev_err(codec->dev, "Over temperature warning\n");
+ if (!(current_val & CS35L33_OTW)) {
+ dev_dbg(codec->dev,
+ "Over temperature warning release\n");
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_OTW_RLS,
+ CS35L33_OTW_RLS);
+ regmap_update_bits(cs35l33->regmap,
+ CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0);
+ }
+ }
+ if (CS35L33_ALIVE_ERR & sticky_val1)
+ dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n");
+
+ if (CS35L33_MCLK_ERR & sticky_val1)
+ dev_err(codec->dev, "ERROR: MCLK Interrupt\n");
+
+ if (CS35L33_VMON_OVFL & sticky_val2)
+ dev_err(codec->dev,
+ "ERROR: VMON Overflow Interrupt\n");
+
+ if (CS35L33_IMON_OVFL & sticky_val2)
+ dev_err(codec->dev,
+ "ERROR: IMON Overflow Interrupt\n");
+
+ if (CS35L33_VPMON_OVFL & sticky_val2)
+ dev_err(codec->dev,
+ "ERROR: VPMON Overflow Interrupt\n");
+
+ return IRQ_HANDLED;
+}
+
+static const char * const cs35l33_core_supplies[] = {
+ "VA",
+ "VP",
+};
+
+static int cs35l33_of_get_pdata(struct device *dev,
+ struct cs35l33_private *cs35l33)
+{
+ struct device_node *np = dev->of_node;
+ struct cs35l33_pdata *pdata = &cs35l33->pdata;
+ u32 val32;
+
+ if (!np)
+ return 0;
+
+ if (of_property_read_u32(np, "cirrus,boost-ctl", &val32) >= 0) {
+ pdata->boost_ctl = val32;
+ pdata->amp_drv_sel = 1;
+ }
+
+ if (of_property_read_u32(np, "cirrus,ramp-rate", &val32) >= 0) {
+ pdata->ramp_rate = val32;
+ cs35l33->enable_soft_ramp = true;
+ }
+
+ if (of_property_read_u32(np, "cirrus,boost-ipk", &val32) >= 0)
+ pdata->boost_ipk = val32;
+
+ if (of_property_read_u32(np, "cirrus,imon-adc-scale", &val32) >= 0) {
+ if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6))
+ pdata->imon_adc_scale = val32;
+ else
+ /* use default value */
+ pdata->imon_adc_scale = 0x8;
+ } else {
+ /* use default value */
+ pdata->imon_adc_scale = 0x8;
+ }
+
+ cs35l33_get_hg_data(np, pdata);
+
+ return 0;
+}
+
+static int cs35l33_i2c_probe(struct i2c_client *i2c_client,
+ const struct i2c_device_id *id)
+{
+ struct cs35l33_private *cs35l33;
+ struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev);
+ int ret, devid, i;
+ unsigned int reg;
+
+ cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private),
+ GFP_KERNEL);
+ if (!cs35l33)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c_client, cs35l33);
+ cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap);
+ if (IS_ERR(cs35l33->regmap)) {
+ ret = PTR_ERR(cs35l33->regmap);
+ dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(cs35l33->regmap, true);
+
+ for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++)
+ cs35l33->core_supplies[i].supply
+ = cs35l33_core_supplies[i];
+ cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies);
+
+ ret = devm_regulator_bulk_get(&i2c_client->dev,
+ cs35l33->num_core_supplies,
+ cs35l33->core_supplies);
+ if (ret != 0) {
+ dev_err(&i2c_client->dev,
+ "Failed to request core supplies: %d\n",
+ ret);
+ return ret;
+ }
+
+ if (pdata) {
+ cs35l33->pdata = *pdata;
+ } else {
+ cs35l33_of_get_pdata(&i2c_client->dev, cs35l33);
+ pdata = &cs35l33->pdata;
+ }
+
+ ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL,
+ cs35l33_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW,
+ "cs35l33", cs35l33);
+ if (ret != 0)
+ dev_warn(&i2c_client->dev, "Failed to request IRQ: %d\n", ret);
+
+ /* We could issue !RST or skip it based on AMP topology */
+ cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev,
+ "reset-gpios", GPIOD_OUT_HIGH);
+ if (IS_ERR(cs35l33->reset_gpio)) {
+ dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n",
+ __func__);
+ return PTR_ERR(cs35l33->reset_gpio);
+ }
+
+ ret = regulator_bulk_enable(cs35l33->num_core_supplies,
+ cs35l33->core_supplies);
+ if (ret != 0) {
+ dev_err(&i2c_client->dev,
+ "Failed to enable core supplies: %d\n",
+ ret);
+ return ret;
+ }
+
+ if (cs35l33->reset_gpio)
+ gpiod_set_value_cansleep(cs35l33->reset_gpio, 1);
+
+ msleep(CS35L33_BOOT_DELAY);
+ regcache_cache_only(cs35l33->regmap, false);
+
+ /* initialize codec */
+ ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg);
+ devid = (reg & 0xFF) << 12;
+ ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg);
+ devid |= (reg & 0xFF) << 4;
+ ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg);
+ devid |= (reg & 0xF0) >> 4;
+
+ if (devid != CS35L33_CHIP_ID) {
+ dev_err(&i2c_client->dev,
+ "CS35L33 Device ID (%X). Expected ID %X\n",
+ devid, CS35L33_CHIP_ID);
+ goto err_enable;
+ }
+
+ ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "Get Revision ID failed\n");
+ goto err_enable;
+ }
+
+ dev_info(&i2c_client->dev,
+ "Cirrus Logic CS35L33, Revision: %02X\n", reg & 0xFF);
+
+ ret = regmap_register_patch(cs35l33->regmap,
+ cs35l33_patch, ARRAY_SIZE(cs35l33_patch));
+ if (ret < 0) {
+ dev_err(&i2c_client->dev,
+ "Error in applying regmap patch: %d\n", ret);
+ goto err_enable;
+ }
+
+ /* disable mclk and tdm */
+ regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL,
+ CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM,
+ CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM);
+
+ pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100);
+ pm_runtime_use_autosuspend(&i2c_client->dev);
+ pm_runtime_set_active(&i2c_client->dev);
+ pm_runtime_enable(&i2c_client->dev);
+
+ ret = snd_soc_register_codec(&i2c_client->dev,
+ &soc_codec_dev_cs35l33, &cs35l33_dai, 1);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "%s: Register codec failed\n",
+ __func__);
+ goto err_enable;
+ }
+
+ return 0;
+
+err_enable:
+ regulator_bulk_disable(cs35l33->num_core_supplies,
+ cs35l33->core_supplies);
+
+ return ret;
+}
+
+static int cs35l33_i2c_remove(struct i2c_client *client)
+{
+ struct cs35l33_private *cs35l33 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+
+ if (cs35l33->reset_gpio)
+ gpiod_set_value_cansleep(cs35l33->reset_gpio, 0);
+
+ pm_runtime_disable(&client->dev);
+ regulator_bulk_disable(cs35l33->num_core_supplies,
+ cs35l33->core_supplies);
+
+ return 0;
+}
+
+static const struct of_device_id cs35l33_of_match[] = {
+ { .compatible = "cirrus,cs35l33", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, cs35l33_of_match);
+
+static const struct i2c_device_id cs35l33_id[] = {
+ {"cs35l33", 0},
+ {}
+};
+
+MODULE_DEVICE_TABLE(i2c, cs35l33_id);
+
+static struct i2c_driver cs35l33_i2c_driver = {
+ .driver = {
+ .name = "cs35l33",
+ .pm = &cs35l33_pm_ops,
+ .of_match_table = cs35l33_of_match,
+
+ },
+ .id_table = cs35l33_id,
+ .probe = cs35l33_i2c_probe,
+ .remove = cs35l33_i2c_remove,
+
+};
+module_i2c_driver(cs35l33_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS35L33 driver");
+MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan@cirrus.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h
new file mode 100644
index 000000000000..c045737d1a5f
--- /dev/null
+++ b/sound/soc/codecs/cs35l33.h
@@ -0,0 +1,221 @@
+/*
+ * cs35l33.h -- CS35L33 ALSA SoC audio driver
+ *
+ * Copyright 2016 Cirrus Logic, Inc.
+ *
+ * Author: Paul Handrigan <paul.handrigan@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef __CS35L33_H__
+#define __CS35L33_H__
+
+#define CS35L33_CHIP_ID 0x00035A33
+#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */
+#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */
+#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */
+#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */
+#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */
+#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */
+#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */
+#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */
+#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */
+#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */
+#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */
+#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */
+#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */
+#define CS35L33_ADC_CTL 0x0E /* ADC Control */
+#define CS35L33_DAC_CTL 0x0F /* DAC Control */
+#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */
+#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */
+#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */
+#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */
+#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */
+#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */
+#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */
+#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */
+#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */
+#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */
+#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */
+#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */
+#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */
+#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */
+#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */
+#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */
+#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */
+#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */
+#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */
+#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */
+#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */
+#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */
+#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */
+#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */
+#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */
+#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */
+#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */
+#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */
+#define CS35L33_HG_STATUS 0x3F /* H/G Status */
+#define CS35L33_MAX_REGISTER 0x59
+
+#define CS35L33_MCLK_5644 5644800
+#define CS35L33_MCLK_6144 6144000
+#define CS35L33_MCLK_6 6000000
+#define CS35L33_MCLK_11289 11289600
+#define CS35L33_MCLK_12 12000000
+#define CS35L33_MCLK_12288 12288000
+
+/* CS35L33_PWRCTL1 */
+#define CS35L33_PDN_AMP (1 << 7)
+#define CS35L33_PDN_BST (1 << 2)
+#define CS35L33_PDN_ALL 1
+
+/* CS35L33_PWRCTL2 */
+#define CS35L33_PDN_VMON_SHIFT 7
+#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT)
+#define CS35L33_PDN_IMON_SHIFT 6
+#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT)
+#define CS35L33_PDN_VPMON_SHIFT 5
+#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT)
+#define CS35L33_PDN_VBSTMON_SHIFT 4
+#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT)
+#define CS35L33_SDOUT_3ST_I2S_SHIFT 3
+#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT)
+#define CS35L33_PDN_SDIN_SHIFT 2
+#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT)
+#define CS35L33_PDN_TDM_SHIFT 1
+#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT)
+
+/* CS35L33_CLK_CTL */
+#define CS35L33_MCLKDIS (1 << 7)
+#define CS35L33_MCLKDIV2 (1 << 6)
+#define CS35L33_SDOUT_3ST_TDM (1 << 5)
+#define CS35L33_INT_FS_RATE (1 << 4)
+#define CS35L33_ADSP_FS 0xF
+
+/* CS35L33_PROTECT_CTL */
+#define CS35L33_ALIVE_WD_DIS (3 << 2)
+
+/* CS35L33_BST_CTL1 */
+#define CS35L33_BST_CTL_SRC (1 << 6)
+#define CS35L33_BST_CTL_SHIFT (1 << 5)
+#define CS35L33_BST_CTL_MASK 0x3F
+
+/* CS35L33_BST_CTL2 */
+#define CS35L33_TDM_WD_SEL (1 << 4)
+#define CS35L33_ALIVE_WD_DIS2 (1 << 3)
+#define CS35L33_VBST_SR_STEP 0x3
+
+/* CS35L33_ADSP_CTL */
+#define CS35L33_ADSP_DRIVE (1 << 7)
+#define CS35L33_MS_MASK (1 << 6)
+#define CS35L33_SDIN_LOC (3 << 4)
+#define CS35L33_ALIVE_RATE 0x3
+
+/* CS35L33_ADC_CTL */
+#define CS35L33_INV_VMON (1 << 7)
+#define CS35L33_INV_IMON (1 << 6)
+#define CS35L33_ADC_NOTCH_DIS (1 << 5)
+#define CS35L33_IMON_SCALE 0xF
+
+/* CS35L33_DAC_CTL */
+#define CS35L33_INV_DAC (1 << 7)
+#define CS35L33_DAC_NOTCH_DIS (1 << 5)
+#define CS35L33_DIGSFT (1 << 4)
+#define CS35L33_DSR_RATE 0xF
+
+/* CS35L33_CLASSD_CTL */
+#define CS35L33_AMP_SD (1 << 6)
+#define CS35L33_AMP_DRV_SEL_SRC (1 << 5)
+#define CS35L33_AMP_DRV_SEL_MASK 0x10
+#define CS35L33_AMP_DRV_SEL_SHIFT 4
+#define CS35L33_AMP_CAL (1 << 3)
+#define CS35L33_GAIN_CHG_ZC_MASK 0x04
+#define CS35L33_GAIN_CHG_ZC_SHIFT 2
+#define CS35L33_CLASS_D_CTL_MASK 0x3F
+
+/* CS35L33_AMP_CTL */
+#define CS35L33_AMP_GAIN 0xF0
+#define CS35L33_CAL_ERR_RLS (1 << 3)
+#define CS35L33_AMP_SHORT_RLS (1 << 2)
+#define CS35L33_OTW_RLS (1 << 1)
+#define CS35L33_OTE_RLS 1
+
+/* CS35L33_INT_MASK_1 */
+#define CS35L33_M_CAL_ERR_SHIFT 6
+#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT)
+#define CS35L33_M_ALIVE_ERR_SHIFT 5
+#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT)
+#define CS35L33_M_AMP_SHORT_SHIFT 2
+#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT)
+#define CS35L33_M_OTW_SHIFT 1
+#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT)
+#define CS35L33_M_OTE_SHIFT 0
+#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT)
+
+/* CS35L33_INT_STATUS_1 */
+#define CS35L33_CAL_ERR (1 << 6)
+#define CS35L33_ALIVE_ERR (1 << 5)
+#define CS35L33_ADSPCLK_ERR (1 << 4)
+#define CS35L33_MCLK_ERR (1 << 3)
+#define CS35L33_AMP_SHORT (1 << 2)
+#define CS35L33_OTW (1 << 1)
+#define CS35L33_OTE (1 << 0)
+
+/* CS35L33_INT_STATUS_2 */
+#define CS35L33_VMON_OVFL (1 << 7)
+#define CS35L33_IMON_OVFL (1 << 6)
+#define CS35L33_VPMON_OVFL (1 << 5)
+#define CS35L33_VBSTMON_OVFL (1 << 4)
+#define CS35L33_PDN_DONE 1
+
+/* CS35L33_BST_CTL4 */
+#define CS35L33_BST_RGS 0x70
+#define CS35L33_BST_COEFF3 0xF
+
+/* CS35L33_HG_MEMLDO_CTL */
+#define CS35L33_MEM_DEPTH_SHIFT 5
+#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT)
+#define CS35L33_LDO_THLD_SHIFT 1
+#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT)
+#define CS35L33_LDO_DISABLE_SHIFT 0
+#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT)
+
+/* CS35L33_LDO_DEL */
+#define CS35L33_VP_HG_VA_SHIFT 5
+#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT)
+#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2
+#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT)
+#define CS35L33_VP_HG_RATE_SHIFT 0
+#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT)
+
+/* CS35L33_HG_HEAD */
+#define CS35L33_HD_RM_SHIFT 0
+#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT)
+
+/* CS35L33_HG_EN */
+#define CS35L33_CLASS_HG_ENA_SHIFT 7
+#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT)
+#define CS35L33_VP_HG_AUTO_SHIFT 6
+#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6)
+#define CS35L33_VP_HG_SHIFT 0
+#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT)
+
+#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000)
+#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+/* CS35L33_{RX,TX}_X */
+#define CS35L33_X_STATE_SHIFT 7
+#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT)
+#define CS35L33_X_LOC_SHIFT 0
+#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT)
+
+/* CS35L33_RX_AUD */
+#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5
+#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT)
+
+#endif
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 5ec5a682d186..954a4f5d3338 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -359,6 +359,11 @@ SND_SOC_DAPM_INPUT("IN2R"),
SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
+SND_SOC_DAPM_OUTPUT("DSP Voice Trigger"),
+
+SND_SOC_DAPM_SWITCH("DSP3 Voice Trigger", SND_SOC_NOPM, 2, 0,
+ &arizona_voice_trigger_switch[2]),
+
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
@@ -899,10 +904,16 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = {
{ "MICSUPP", NULL, "SYSCLK" },
+ { "DRC1 Signal Activity", NULL, "SYSCLK" },
+ { "DRC2 Signal Activity", NULL, "SYSCLK" },
{ "DRC1 Signal Activity", NULL, "DRC1L" },
{ "DRC1 Signal Activity", NULL, "DRC1R" },
{ "DRC2 Signal Activity", NULL, "DRC2L" },
{ "DRC2 Signal Activity", NULL, "DRC2R" },
+
+ { "DSP Voice Trigger", NULL, "SYSCLK" },
+ { "DSP Voice Trigger", NULL, "DSP3 Voice Trigger" },
+ { "DSP3 Voice Trigger", "Switch", "DSP3" },
};
static int cs47l24_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
@@ -1067,6 +1078,7 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
{
struct cs47l24_priv *priv = data;
struct arizona *arizona = priv->core.arizona;
+ struct arizona_voice_trigger_info info;
int serviced = 0;
int i, ret;
@@ -1074,6 +1086,12 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
if (ret != -ENODEV)
serviced++;
+ if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) {
+ info.core = i;
+ arizona_call_notifiers(arizona,
+ ARIZONA_NOTIFY_VOICE_TRIGGER,
+ &info);
+ }
}
if (!serviced) {
@@ -1096,6 +1114,7 @@ static int cs47l24_codec_probe(struct snd_soc_codec *codec)
arizona_init_spk(codec);
arizona_init_gpio(codec);
arizona_init_mono(codec);
+ arizona_init_notifiers(codec);
ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1,
"ADSP2 Compressed IRQ", cs47l24_adsp2_irq,
diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c
new file mode 100644
index 000000000000..2c0d9c430a8c
--- /dev/null
+++ b/sound/soc/codecs/cs53l30.c
@@ -0,0 +1,1143 @@
+/*
+ * cs53l30.c -- CS53l30 ALSA Soc Audio driver
+ *
+ * Copyright 2015 Cirrus Logic, Inc.
+ *
+ * Authors: Paul Handrigan <Paul.Handrigan@cirrus.com>,
+ * Tim Howe <Tim.Howe@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of_gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/regulator/consumer.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "cs53l30.h"
+
+#define CS53L30_NUM_SUPPLIES 2
+static const char *const cs53l30_supply_names[CS53L30_NUM_SUPPLIES] = {
+ "VA",
+ "VP",
+};
+
+struct cs53l30_private {
+ struct regulator_bulk_data supplies[CS53L30_NUM_SUPPLIES];
+ struct regmap *regmap;
+ struct gpio_desc *reset_gpio;
+ struct gpio_desc *mute_gpio;
+ struct clk *mclk;
+ bool use_sdout2;
+ u32 mclk_rate;
+};
+
+static const struct reg_default cs53l30_reg_defaults[] = {
+ { CS53L30_PWRCTL, CS53L30_PWRCTL_DEFAULT },
+ { CS53L30_MCLKCTL, CS53L30_MCLKCTL_DEFAULT },
+ { CS53L30_INT_SR_CTL, CS53L30_INT_SR_CTL_DEFAULT },
+ { CS53L30_MICBIAS_CTL, CS53L30_MICBIAS_CTL_DEFAULT },
+ { CS53L30_ASPCFG_CTL, CS53L30_ASPCFG_CTL_DEFAULT },
+ { CS53L30_ASP_CTL1, CS53L30_ASP_CTL1_DEFAULT },
+ { CS53L30_ASP_TDMTX_CTL1, CS53L30_ASP_TDMTX_CTLx_DEFAULT },
+ { CS53L30_ASP_TDMTX_CTL2, CS53L30_ASP_TDMTX_CTLx_DEFAULT },
+ { CS53L30_ASP_TDMTX_CTL3, CS53L30_ASP_TDMTX_CTLx_DEFAULT },
+ { CS53L30_ASP_TDMTX_CTL4, CS53L30_ASP_TDMTX_CTLx_DEFAULT },
+ { CS53L30_ASP_TDMTX_EN1, CS53L30_ASP_TDMTX_ENx_DEFAULT },
+ { CS53L30_ASP_TDMTX_EN2, CS53L30_ASP_TDMTX_ENx_DEFAULT },
+ { CS53L30_ASP_TDMTX_EN3, CS53L30_ASP_TDMTX_ENx_DEFAULT },
+ { CS53L30_ASP_TDMTX_EN4, CS53L30_ASP_TDMTX_ENx_DEFAULT },
+ { CS53L30_ASP_TDMTX_EN5, CS53L30_ASP_TDMTX_ENx_DEFAULT },
+ { CS53L30_ASP_TDMTX_EN6, CS53L30_ASP_TDMTX_ENx_DEFAULT },
+ { CS53L30_ASP_CTL2, CS53L30_ASP_CTL2_DEFAULT },
+ { CS53L30_SFT_RAMP, CS53L30_SFT_RMP_DEFAULT },
+ { CS53L30_LRCK_CTL1, CS53L30_LRCK_CTLx_DEFAULT },
+ { CS53L30_LRCK_CTL2, CS53L30_LRCK_CTLx_DEFAULT },
+ { CS53L30_MUTEP_CTL1, CS53L30_MUTEP_CTL1_DEFAULT },
+ { CS53L30_MUTEP_CTL2, CS53L30_MUTEP_CTL2_DEFAULT },
+ { CS53L30_INBIAS_CTL1, CS53L30_INBIAS_CTL1_DEFAULT },
+ { CS53L30_INBIAS_CTL2, CS53L30_INBIAS_CTL2_DEFAULT },
+ { CS53L30_DMIC1_STR_CTL, CS53L30_DMIC1_STR_CTL_DEFAULT },
+ { CS53L30_DMIC2_STR_CTL, CS53L30_DMIC2_STR_CTL_DEFAULT },
+ { CS53L30_ADCDMIC1_CTL1, CS53L30_ADCDMICx_CTL1_DEFAULT },
+ { CS53L30_ADCDMIC1_CTL2, CS53L30_ADCDMIC1_CTL2_DEFAULT },
+ { CS53L30_ADC1_CTL3, CS53L30_ADCx_CTL3_DEFAULT },
+ { CS53L30_ADC1_NG_CTL, CS53L30_ADCx_NG_CTL_DEFAULT },
+ { CS53L30_ADC1A_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT },
+ { CS53L30_ADC1B_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT },
+ { CS53L30_ADC1A_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT },
+ { CS53L30_ADC1B_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT },
+ { CS53L30_ADCDMIC2_CTL1, CS53L30_ADCDMICx_CTL1_DEFAULT },
+ { CS53L30_ADCDMIC2_CTL2, CS53L30_ADCDMIC1_CTL2_DEFAULT },
+ { CS53L30_ADC2_CTL3, CS53L30_ADCx_CTL3_DEFAULT },
+ { CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_CTL_DEFAULT },
+ { CS53L30_ADC2A_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT },
+ { CS53L30_ADC2B_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT },
+ { CS53L30_ADC2A_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT },
+ { CS53L30_ADC2B_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT },
+ { CS53L30_INT_MASK, CS53L30_DEVICE_INT_MASK },
+};
+
+static bool cs53l30_volatile_register(struct device *dev, unsigned int reg)
+{
+ if (reg == CS53L30_IS)
+ return true;
+ else
+ return false;
+}
+
+static bool cs53l30_writeable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS53L30_DEVID_AB:
+ case CS53L30_DEVID_CD:
+ case CS53L30_DEVID_E:
+ case CS53L30_REVID:
+ case CS53L30_IS:
+ return false;
+ default:
+ return true;
+ }
+}
+
+static bool cs53l30_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS53L30_DEVID_AB:
+ case CS53L30_DEVID_CD:
+ case CS53L30_DEVID_E:
+ case CS53L30_REVID:
+ case CS53L30_PWRCTL:
+ case CS53L30_MCLKCTL:
+ case CS53L30_INT_SR_CTL:
+ case CS53L30_MICBIAS_CTL:
+ case CS53L30_ASPCFG_CTL:
+ case CS53L30_ASP_CTL1:
+ case CS53L30_ASP_TDMTX_CTL1:
+ case CS53L30_ASP_TDMTX_CTL2:
+ case CS53L30_ASP_TDMTX_CTL3:
+ case CS53L30_ASP_TDMTX_CTL4:
+ case CS53L30_ASP_TDMTX_EN1:
+ case CS53L30_ASP_TDMTX_EN2:
+ case CS53L30_ASP_TDMTX_EN3:
+ case CS53L30_ASP_TDMTX_EN4:
+ case CS53L30_ASP_TDMTX_EN5:
+ case CS53L30_ASP_TDMTX_EN6:
+ case CS53L30_ASP_CTL2:
+ case CS53L30_SFT_RAMP:
+ case CS53L30_LRCK_CTL1:
+ case CS53L30_LRCK_CTL2:
+ case CS53L30_MUTEP_CTL1:
+ case CS53L30_MUTEP_CTL2:
+ case CS53L30_INBIAS_CTL1:
+ case CS53L30_INBIAS_CTL2:
+ case CS53L30_DMIC1_STR_CTL:
+ case CS53L30_DMIC2_STR_CTL:
+ case CS53L30_ADCDMIC1_CTL1:
+ case CS53L30_ADCDMIC1_CTL2:
+ case CS53L30_ADC1_CTL3:
+ case CS53L30_ADC1_NG_CTL:
+ case CS53L30_ADC1A_AFE_CTL:
+ case CS53L30_ADC1B_AFE_CTL:
+ case CS53L30_ADC1A_DIG_VOL:
+ case CS53L30_ADC1B_DIG_VOL:
+ case CS53L30_ADCDMIC2_CTL1:
+ case CS53L30_ADCDMIC2_CTL2:
+ case CS53L30_ADC2_CTL3:
+ case CS53L30_ADC2_NG_CTL:
+ case CS53L30_ADC2A_AFE_CTL:
+ case CS53L30_ADC2B_AFE_CTL:
+ case CS53L30_ADC2A_DIG_VOL:
+ case CS53L30_ADC2B_DIG_VOL:
+ case CS53L30_INT_MASK:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0);
+static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0);
+static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
+static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1);
+static DECLARE_TLV_DB_SCALE(pga_preamp_tlv, 0, 10000, 0);
+
+static const char * const input1_sel_text[] = {
+ "DMIC1 On AB In",
+ "DMIC1 On A In",
+ "DMIC1 On B In",
+ "ADC1 On AB In",
+ "ADC1 On A In",
+ "ADC1 On B In",
+ "DMIC1 Off ADC1 Off",
+};
+
+static unsigned int const input1_sel_values[] = {
+ CS53L30_CH_TYPE,
+ CS53L30_ADCxB_PDN | CS53L30_CH_TYPE,
+ CS53L30_ADCxA_PDN | CS53L30_CH_TYPE,
+ CS53L30_DMICx_PDN,
+ CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN,
+ CS53L30_ADCxA_PDN | CS53L30_DMICx_PDN,
+ CS53L30_ADCxA_PDN | CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN,
+};
+
+static const char * const input2_sel_text[] = {
+ "DMIC2 On AB In",
+ "DMIC2 On A In",
+ "DMIC2 On B In",
+ "ADC2 On AB In",
+ "ADC2 On A In",
+ "ADC2 On B In",
+ "DMIC2 Off ADC2 Off",
+};
+
+static unsigned int const input2_sel_values[] = {
+ 0x0,
+ CS53L30_ADCxB_PDN,
+ CS53L30_ADCxA_PDN,
+ CS53L30_DMICx_PDN,
+ CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN,
+ CS53L30_ADCxA_PDN | CS53L30_DMICx_PDN,
+ CS53L30_ADCxA_PDN | CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN,
+};
+
+static const char * const input1_route_sel_text[] = {
+ "ADC1_SEL", "DMIC1_SEL",
+};
+
+static const struct soc_enum input1_route_sel_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, CS53L30_CH_TYPE_SHIFT,
+ ARRAY_SIZE(input1_route_sel_text),
+ input1_route_sel_text);
+
+static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0,
+ CS53L30_ADCDMICx_PDN_MASK, input1_sel_text,
+ input1_sel_values);
+
+static const struct snd_kcontrol_new input1_route_sel_mux =
+ SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum);
+
+static const char * const input2_route_sel_text[] = {
+ "ADC2_SEL", "DMIC2_SEL",
+};
+
+/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */
+static const struct soc_enum input2_route_sel_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0,
+ ARRAY_SIZE(input2_route_sel_text),
+ input2_route_sel_text);
+
+static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0,
+ CS53L30_ADCDMICx_PDN_MASK, input2_sel_text,
+ input2_sel_values);
+
+static const struct snd_kcontrol_new input2_route_sel_mux =
+ SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum);
+
+/*
+ * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal)
+ * TB - Time base
+ * NOTE: If MCLK_INT_SCALE = 0, then TB=1
+ */
+static const char * const cs53l30_ng_delay_text[] = {
+ "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms",
+};
+
+static const struct soc_enum adc1_ng_delay_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, CS53L30_ADCx_NG_DELAY_SHIFT,
+ ARRAY_SIZE(cs53l30_ng_delay_text),
+ cs53l30_ng_delay_text);
+
+static const struct soc_enum adc2_ng_delay_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_DELAY_SHIFT,
+ ARRAY_SIZE(cs53l30_ng_delay_text),
+ cs53l30_ng_delay_text);
+
+/* The noise gate threshold selected will depend on NG Boost */
+static const char * const cs53l30_ng_thres_text[] = {
+ "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB",
+ "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB",
+};
+
+static const struct soc_enum adc1_ng_thres_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, CS53L30_ADCx_NG_THRESH_SHIFT,
+ ARRAY_SIZE(cs53l30_ng_thres_text),
+ cs53l30_ng_thres_text);
+
+static const struct soc_enum adc2_ng_thres_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_THRESH_SHIFT,
+ ARRAY_SIZE(cs53l30_ng_thres_text),
+ cs53l30_ng_thres_text);
+
+/* Corner frequencies are with an Fs of 48kHz. */
+static const char * const hpf_corner_freq_text[] = {
+ "1.86Hz", "120Hz", "235Hz", "466Hz",
+};
+
+static const struct soc_enum adc1_hpf_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, CS53L30_ADCx_HPF_CF_SHIFT,
+ ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text);
+
+static const struct soc_enum adc2_hpf_enum =
+ SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, CS53L30_ADCx_HPF_CF_SHIFT,
+ ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text);
+
+static const struct snd_kcontrol_new cs53l30_snd_controls[] = {
+ SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP,
+ CS53L30_DIGSFT_SHIFT, 1, 0),
+ SOC_SINGLE("ADC1 Noise Gate Ganging Switch", CS53L30_ADC1_CTL3,
+ CS53L30_ADCx_NG_ALL_SHIFT, 1, 0),
+ SOC_SINGLE("ADC2 Noise Gate Ganging Switch", CS53L30_ADC2_CTL3,
+ CS53L30_ADCx_NG_ALL_SHIFT, 1, 0),
+ SOC_SINGLE("ADC1A Noise Gate Enable Switch", CS53L30_ADC1_NG_CTL,
+ CS53L30_ADCxA_NG_SHIFT, 1, 0),
+ SOC_SINGLE("ADC1B Noise Gate Enable Switch", CS53L30_ADC1_NG_CTL,
+ CS53L30_ADCxB_NG_SHIFT, 1, 0),
+ SOC_SINGLE("ADC2A Noise Gate Enable Switch", CS53L30_ADC2_NG_CTL,
+ CS53L30_ADCxA_NG_SHIFT, 1, 0),
+ SOC_SINGLE("ADC2B Noise Gate Enable Switch", CS53L30_ADC2_NG_CTL,
+ CS53L30_ADCxB_NG_SHIFT, 1, 0),
+ SOC_SINGLE("ADC1 Notch Filter Switch", CS53L30_ADCDMIC1_CTL2,
+ CS53L30_ADCx_NOTCH_DIS_SHIFT, 1, 1),
+ SOC_SINGLE("ADC2 Notch Filter Switch", CS53L30_ADCDMIC2_CTL2,
+ CS53L30_ADCx_NOTCH_DIS_SHIFT, 1, 1),
+ SOC_SINGLE("ADC1A Invert Switch", CS53L30_ADCDMIC1_CTL2,
+ CS53L30_ADCxA_INV_SHIFT, 1, 0),
+ SOC_SINGLE("ADC1B Invert Switch", CS53L30_ADCDMIC1_CTL2,
+ CS53L30_ADCxB_INV_SHIFT, 1, 0),
+ SOC_SINGLE("ADC2A Invert Switch", CS53L30_ADCDMIC2_CTL2,
+ CS53L30_ADCxA_INV_SHIFT, 1, 0),
+ SOC_SINGLE("ADC2B Invert Switch", CS53L30_ADCDMIC2_CTL2,
+ CS53L30_ADCxB_INV_SHIFT, 1, 0),
+
+ SOC_SINGLE_TLV("ADC1A Digital Boost Volume", CS53L30_ADCDMIC1_CTL2,
+ CS53L30_ADCxA_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv),
+ SOC_SINGLE_TLV("ADC1B Digital Boost Volume", CS53L30_ADCDMIC1_CTL2,
+ CS53L30_ADCxB_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv),
+ SOC_SINGLE_TLV("ADC2A Digital Boost Volume", CS53L30_ADCDMIC2_CTL2,
+ CS53L30_ADCxA_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv),
+ SOC_SINGLE_TLV("ADC2B Digital Boost Volume", CS53L30_ADCDMIC2_CTL2,
+ CS53L30_ADCxB_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv),
+ SOC_SINGLE_TLV("ADC1 NG Boost Volume", CS53L30_ADC1_NG_CTL,
+ CS53L30_ADCx_NG_BOOST_SHIFT, 1, 0, adc_ng_boost_tlv),
+ SOC_SINGLE_TLV("ADC2 NG Boost Volume", CS53L30_ADC2_NG_CTL,
+ CS53L30_ADCx_NG_BOOST_SHIFT, 1, 0, adc_ng_boost_tlv),
+
+ SOC_DOUBLE_R_TLV("ADC1 Preamplifier Volume", CS53L30_ADC1A_AFE_CTL,
+ CS53L30_ADC1B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT,
+ 2, 0, pga_preamp_tlv),
+ SOC_DOUBLE_R_TLV("ADC2 Preamplifier Volume", CS53L30_ADC2A_AFE_CTL,
+ CS53L30_ADC2B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT,
+ 2, 0, pga_preamp_tlv),
+
+ SOC_ENUM("Input 1 Channel Select", input1_sel_enum),
+ SOC_ENUM("Input 2 Channel Select", input2_sel_enum),
+
+ SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum),
+ SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum),
+ SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum),
+ SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum),
+ SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum),
+ SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum),
+
+ SOC_SINGLE_SX_TLV("ADC1A PGA Volume",
+ CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+ SOC_SINGLE_SX_TLV("ADC1B PGA Volume",
+ CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+ SOC_SINGLE_SX_TLV("ADC2A PGA Volume",
+ CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+ SOC_SINGLE_SX_TLV("ADC2B PGA Volume",
+ CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv),
+
+ SOC_SINGLE_SX_TLV("ADC1A Digital Volume",
+ CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+ SOC_SINGLE_SX_TLV("ADC1B Digital Volume",
+ CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+ SOC_SINGLE_SX_TLV("ADC2A Digital Volume",
+ CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+ SOC_SINGLE_SX_TLV("ADC2B Digital Volume",
+ CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv),
+};
+
+static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("IN1_DMIC1"),
+ SND_SOC_DAPM_INPUT("IN2"),
+ SND_SOC_DAPM_INPUT("IN3_DMIC2"),
+ SND_SOC_DAPM_INPUT("IN4"),
+ SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL,
+ CS53L30_MIC1_BIAS_PDN_SHIFT, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL,
+ CS53L30_MIC2_BIAS_PDN_SHIFT, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL,
+ CS53L30_MIC3_BIAS_PDN_SHIFT, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL,
+ CS53L30_MIC4_BIAS_PDN_SHIFT, 1, NULL, 0),
+
+ SND_SOC_DAPM_AIF_OUT("ASP_SDOUT1", NULL, 0, CS53L30_ASP_CTL1,
+ CS53L30_ASP_SDOUTx_PDN_SHIFT, 1),
+ SND_SOC_DAPM_AIF_OUT("ASP_SDOUT2", NULL, 0, CS53L30_ASP_CTL2,
+ CS53L30_ASP_SDOUTx_PDN_SHIFT, 1),
+
+ SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0,
+ &input1_route_sel_mux),
+ SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0,
+ &input2_route_sel_mux),
+
+ SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1,
+ CS53L30_ADCxA_PDN_SHIFT, 1),
+ SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1,
+ CS53L30_ADCxB_PDN_SHIFT, 1),
+ SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1,
+ CS53L30_ADCxA_PDN_SHIFT, 1),
+ SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1,
+ CS53L30_ADCxB_PDN_SHIFT, 1),
+ SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1,
+ CS53L30_DMICx_PDN_SHIFT, 1),
+ SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1,
+ CS53L30_DMICx_PDN_SHIFT, 1),
+};
+
+static const struct snd_soc_dapm_route cs53l30_dapm_routes[] = {
+ /* ADC Input Paths */
+ {"ADC1A", NULL, "IN1_DMIC1"},
+ {"Input Mux 1", "ADC1_SEL", "ADC1A"},
+ {"ADC1B", NULL, "IN2"},
+
+ {"ADC2A", NULL, "IN3_DMIC2"},
+ {"Input Mux 2", "ADC2_SEL", "ADC2A"},
+ {"ADC2B", NULL, "IN4"},
+
+ /* MIC Bias Paths */
+ {"ADC1A", NULL, "MIC1 Bias"},
+ {"ADC1B", NULL, "MIC2 Bias"},
+ {"ADC2A", NULL, "MIC3 Bias"},
+ {"ADC2B", NULL, "MIC4 Bias"},
+
+ /* DMIC Paths */
+ {"DMIC1", NULL, "IN1_DMIC1"},
+ {"Input Mux 1", "DMIC1_SEL", "DMIC1"},
+
+ {"DMIC2", NULL, "IN3_DMIC2"},
+ {"Input Mux 2", "DMIC2_SEL", "DMIC2"},
+};
+
+static const struct snd_soc_dapm_route cs53l30_dapm_routes_sdout1[] = {
+ /* Output Paths when using SDOUT1 only */
+ {"ASP_SDOUT1", NULL, "ADC1A" },
+ {"ASP_SDOUT1", NULL, "Input Mux 1"},
+ {"ASP_SDOUT1", NULL, "ADC1B"},
+
+ {"ASP_SDOUT1", NULL, "ADC2A"},
+ {"ASP_SDOUT1", NULL, "Input Mux 2"},
+ {"ASP_SDOUT1", NULL, "ADC2B"},
+
+ {"Capture", NULL, "ASP_SDOUT1"},
+};
+
+static const struct snd_soc_dapm_route cs53l30_dapm_routes_sdout2[] = {
+ /* Output Paths when using both SDOUT1 and SDOUT2 */
+ {"ASP_SDOUT1", NULL, "ADC1A" },
+ {"ASP_SDOUT1", NULL, "Input Mux 1"},
+ {"ASP_SDOUT1", NULL, "ADC1B"},
+
+ {"ASP_SDOUT2", NULL, "ADC2A"},
+ {"ASP_SDOUT2", NULL, "Input Mux 2"},
+ {"ASP_SDOUT2", NULL, "ADC2B"},
+
+ {"Capture", NULL, "ASP_SDOUT1"},
+ {"Capture", NULL, "ASP_SDOUT2"},
+};
+
+struct cs53l30_mclk_div {
+ u32 mclk_rate;
+ u32 srate;
+ u8 asp_rate;
+ u8 internal_fs_ratio;
+ u8 mclk_int_scale;
+};
+
+static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = {
+ /* NOTE: Enable MCLK_INT_SCALE to save power. */
+
+ /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */
+ {5644800, 11025, 0x4, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {5644800, 22050, 0x8, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {5644800, 44100, 0xC, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+
+ {6000000, 8000, 0x1, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 11025, 0x2, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 12000, 0x4, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 16000, 0x5, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 22050, 0x6, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 24000, 0x8, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 32000, 0x9, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 44100, 0xA, 0, CS53L30_MCLK_INT_SCALE},
+ {6000000, 48000, 0xC, 0, CS53L30_MCLK_INT_SCALE},
+
+ {6144000, 8000, 0x1, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 11025, 0x2, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 12000, 0x4, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 16000, 0x5, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 22050, 0x6, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 24000, 0x8, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 32000, 0x9, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 44100, 0xA, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6144000, 48000, 0xC, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+
+ {6400000, 8000, 0x1, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 11025, 0x2, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 12000, 0x4, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 16000, 0x5, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 22050, 0x6, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 24000, 0x8, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 32000, 0x9, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 44100, 0xA, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+ {6400000, 48000, 0xC, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE},
+};
+
+struct cs53l30_mclkx_div {
+ u32 mclkx;
+ u8 ratio;
+ u8 mclkdiv;
+};
+
+static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = {
+ {5644800, 1, CS53L30_MCLK_DIV_BY_1},
+ {6000000, 1, CS53L30_MCLK_DIV_BY_1},
+ {6144000, 1, CS53L30_MCLK_DIV_BY_1},
+ {11289600, 2, CS53L30_MCLK_DIV_BY_2},
+ {12288000, 2, CS53L30_MCLK_DIV_BY_2},
+ {12000000, 2, CS53L30_MCLK_DIV_BY_2},
+ {19200000, 3, CS53L30_MCLK_DIV_BY_3},
+};
+
+static int cs53l30_get_mclkx_coeff(int mclkx)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) {
+ if (cs53l30_mclkx_coeffs[i].mclkx == mclkx)
+ return i;
+ }
+
+ return -EINVAL;
+}
+
+static int cs53l30_get_mclk_coeff(int mclk_rate, int srate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) {
+ if (cs53l30_mclk_coeffs[i].mclk_rate == mclk_rate &&
+ cs53l30_mclk_coeffs[i].srate == srate)
+ return i;
+ }
+
+ return -EINVAL;
+}
+
+static int cs53l30_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec);
+ int mclkx_coeff;
+ u32 mclk_rate;
+
+ /* MCLKX -> MCLK */
+ mclkx_coeff = cs53l30_get_mclkx_coeff(freq);
+ if (mclkx_coeff < 0)
+ return mclkx_coeff;
+
+ mclk_rate = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx /
+ cs53l30_mclkx_coeffs[mclkx_coeff].ratio;
+
+ regmap_update_bits(priv->regmap, CS53L30_MCLKCTL,
+ CS53L30_MCLK_DIV_MASK,
+ cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv);
+
+ priv->mclk_rate = mclk_rate;
+
+ return 0;
+}
+
+static int cs53l30_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec);
+ u8 aspcfg = 0, aspctl1 = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aspcfg |= CS53L30_ASP_MS;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* Set TDM_PDN to turn off TDM mode -- Reset default */
+ aspctl1 |= CS53L30_ASP_TDM_PDN;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /*
+ * Clear TDM_PDN to turn on TDM mode; Use ASP_SCLK_INV = 0
+ * with SHIFT_LEFT = 1 combination as Figure 4-13 shows in
+ * the CS53L30 datasheet
+ */
+ aspctl1 |= CS53L30_SHIFT_LEFT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Check to see if the SCLK is inverted */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_NF:
+ case SND_SOC_DAIFMT_IB_IF:
+ aspcfg ^= CS53L30_ASP_SCLK_INV;
+ break;
+ default:
+ break;
+ }
+
+ regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL,
+ CS53L30_ASP_MS | CS53L30_ASP_SCLK_INV, aspcfg);
+
+ regmap_update_bits(priv->regmap, CS53L30_ASP_CTL1,
+ CS53L30_ASP_TDM_PDN | CS53L30_SHIFT_LEFT, aspctl1);
+
+ return 0;
+}
+
+static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec);
+ int srate = params_rate(params);
+ int mclk_coeff;
+
+ /* MCLK -> srate */
+ mclk_coeff = cs53l30_get_mclk_coeff(priv->mclk_rate, srate);
+ if (mclk_coeff < 0)
+ return -EINVAL;
+
+ regmap_update_bits(priv->regmap, CS53L30_INT_SR_CTL,
+ CS53L30_INTRNL_FS_RATIO_MASK,
+ cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio);
+
+ regmap_update_bits(priv->regmap, CS53L30_MCLKCTL,
+ CS53L30_MCLK_INT_SCALE_MASK,
+ cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale);
+
+ regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL,
+ CS53L30_ASP_RATE_MASK,
+ cs53l30_mclk_coeffs[mclk_coeff].asp_rate);
+
+ return 0;
+}
+
+static int cs53l30_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int reg;
+ int i, inter_max_check, ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY)
+ regmap_update_bits(priv->regmap, CS53L30_PWRCTL,
+ CS53L30_PDN_LP_MASK, 0);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (dapm->bias_level == SND_SOC_BIAS_OFF) {
+ ret = clk_prepare_enable(priv->mclk);
+ if (ret) {
+ dev_err(codec->dev,
+ "failed to enable MCLK: %d\n", ret);
+ return ret;
+ }
+ regmap_update_bits(priv->regmap, CS53L30_MCLKCTL,
+ CS53L30_MCLK_DIS_MASK, 0);
+ regmap_update_bits(priv->regmap, CS53L30_PWRCTL,
+ CS53L30_PDN_ULP_MASK, 0);
+ msleep(50);
+ } else {
+ regmap_update_bits(priv->regmap, CS53L30_PWRCTL,
+ CS53L30_PDN_ULP_MASK,
+ CS53L30_PDN_ULP);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ regmap_update_bits(priv->regmap, CS53L30_INT_MASK,
+ CS53L30_PDN_DONE, 0);
+ /*
+ * If digital softramp is set, the amount of time required
+ * for power down increases and depends on the digital
+ * volume setting.
+ */
+
+ /* Set the max possible time if digsft is set */
+ regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg);
+ if (reg & CS53L30_DIGSFT_MASK)
+ inter_max_check = CS53L30_PDN_POLL_MAX;
+ else
+ inter_max_check = 10;
+
+ regmap_update_bits(priv->regmap, CS53L30_PWRCTL,
+ CS53L30_PDN_ULP_MASK,
+ CS53L30_PDN_ULP);
+ /* PDN_DONE will take a min of 20ms to be set.*/
+ msleep(20);
+ /* Clr status */
+ regmap_read(priv->regmap, CS53L30_IS, &reg);
+ for (i = 0; i < inter_max_check; i++) {
+ if (inter_max_check < 10) {
+ usleep_range(1000, 1100);
+ regmap_read(priv->regmap, CS53L30_IS, &reg);
+ if (reg & CS53L30_PDN_DONE)
+ break;
+ } else {
+ usleep_range(10000, 10100);
+ regmap_read(priv->regmap, CS53L30_IS, &reg);
+ if (reg & CS53L30_PDN_DONE)
+ break;
+ }
+ }
+ /* PDN_DONE is set. We now can disable the MCLK */
+ regmap_update_bits(priv->regmap, CS53L30_INT_MASK,
+ CS53L30_PDN_DONE, CS53L30_PDN_DONE);
+ regmap_update_bits(priv->regmap, CS53L30_MCLKCTL,
+ CS53L30_MCLK_DIS_MASK,
+ CS53L30_MCLK_DIS);
+ clk_disable_unprepare(priv->mclk);
+ break;
+ }
+
+ return 0;
+}
+
+static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec);
+ u8 val = tristate ? CS53L30_ASP_3ST : 0;
+
+ return regmap_update_bits(priv->regmap, CS53L30_ASP_CTL1,
+ CS53L30_ASP_3ST_MASK, val);
+}
+
+static unsigned int const cs53l30_src_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list src_constraints = {
+ .count = ARRAY_SIZE(cs53l30_src_rates),
+ .list = cs53l30_src_rates,
+};
+
+static int cs53l30_pcm_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &src_constraints);
+
+ return 0;
+}
+
+/*
+ * Note: CS53L30 counts the slot number per byte while ASoC counts the slot
+ * number per slot_width. So there is a difference between the slots of ASoC
+ * and the slots of CS53L30.
+ */
+static int cs53l30_set_dai_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int loc[CS53L30_TDM_SLOT_MAX] = {48, 48, 48, 48};
+ unsigned int slot_next, slot_step;
+ u64 tx_enable = 0;
+ int i;
+
+ if (!rx_mask) {
+ dev_err(dai->dev, "rx masks must not be 0\n");
+ return -EINVAL;
+ }
+
+ /* Assuming slot_width is not supposed to be greater than 64 */
+ if (slots <= 0 || slot_width <= 0 || slot_width > 64) {
+ dev_err(dai->dev, "invalid slot number or slot width\n");
+ return -EINVAL;
+ }
+
+ if (slot_width & 0x7) {
+ dev_err(dai->dev, "slot width must count in byte\n");
+ return -EINVAL;
+ }
+
+ /* How many bytes in each ASoC slot */
+ slot_step = slot_width >> 3;
+
+ for (i = 0; rx_mask && i < CS53L30_TDM_SLOT_MAX; i++) {
+ /* Find the first slot from LSB */
+ slot_next = __ffs(rx_mask);
+ /* Save the slot location by converting to CS53L30 slot */
+ loc[i] = slot_next * slot_step;
+ /* Create the mask of CS53L30 slot */
+ tx_enable |= (u64)((u64)(1 << slot_step) - 1) << (u64)loc[i];
+ /* Clear this slot from rx_mask */
+ rx_mask &= ~(1 << slot_next);
+ }
+
+ /* Error out to avoid slot shift */
+ if (rx_mask && i == CS53L30_TDM_SLOT_MAX) {
+ dev_err(dai->dev, "rx_mask exceeds max slot number: %d\n",
+ CS53L30_TDM_SLOT_MAX);
+ return -EINVAL;
+ }
+
+ /* Validate the last active CS53L30 slot */
+ slot_next = loc[i - 1] + slot_step - 1;
+ if (slot_next > 47) {
+ dev_err(dai->dev, "slot selection out of bounds: %u\n",
+ slot_next);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < CS53L30_TDM_SLOT_MAX && loc[i] != 48; i++) {
+ regmap_update_bits(priv->regmap, CS53L30_ASP_TDMTX_CTL(i),
+ CS53L30_ASP_CHx_TX_LOC_MASK, loc[i]);
+ dev_dbg(dai->dev, "loc[%d]=%x\n", i, loc[i]);
+ }
+
+ for (i = 0; i < CS53L30_ASP_TDMTX_ENx_MAX && tx_enable; i++) {
+ regmap_write(priv->regmap, CS53L30_ASP_TDMTX_ENx(i),
+ tx_enable & 0xff);
+ tx_enable >>= 8;
+ dev_dbg(dai->dev, "en_reg=%x, tx_enable=%llx\n",
+ CS53L30_ASP_TDMTX_ENx(i), tx_enable & 0xff);
+ }
+
+ return 0;
+}
+
+static int cs53l30_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
+{
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec);
+
+ if (priv->mute_gpio)
+ gpiod_set_value_cansleep(priv->mute_gpio, mute);
+
+ return 0;
+}
+
+/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */
+#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
+
+#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops cs53l30_ops = {
+ .startup = cs53l30_pcm_startup,
+ .hw_params = cs53l30_pcm_hw_params,
+ .set_fmt = cs53l30_set_dai_fmt,
+ .set_sysclk = cs53l30_set_sysclk,
+ .set_tristate = cs53l30_set_tristate,
+ .set_tdm_slot = cs53l30_set_dai_tdm_slot,
+ .mute_stream = cs53l30_mute_stream,
+};
+
+static struct snd_soc_dai_driver cs53l30_dai = {
+ .name = "cs53l30",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = CS53L30_RATES,
+ .formats = CS53L30_FORMATS,
+ },
+ .ops = &cs53l30_ops,
+ .symmetric_rates = 1,
+};
+
+static int cs53l30_codec_probe(struct snd_soc_codec *codec)
+{
+ struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+
+ if (priv->use_sdout2)
+ snd_soc_dapm_add_routes(dapm, cs53l30_dapm_routes_sdout2,
+ ARRAY_SIZE(cs53l30_dapm_routes_sdout2));
+ else
+ snd_soc_dapm_add_routes(dapm, cs53l30_dapm_routes_sdout1,
+ ARRAY_SIZE(cs53l30_dapm_routes_sdout1));
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver cs53l30_driver = {
+ .probe = cs53l30_codec_probe,
+ .set_bias_level = cs53l30_set_bias_level,
+ .idle_bias_off = true,
+
+ .dapm_widgets = cs53l30_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets),
+ .dapm_routes = cs53l30_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(cs53l30_dapm_routes),
+
+ .controls = cs53l30_snd_controls,
+ .num_controls = ARRAY_SIZE(cs53l30_snd_controls),
+};
+
+static struct regmap_config cs53l30_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS53L30_MAX_REGISTER,
+ .reg_defaults = cs53l30_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults),
+ .volatile_reg = cs53l30_volatile_register,
+ .writeable_reg = cs53l30_writeable_register,
+ .readable_reg = cs53l30_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int cs53l30_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ const struct device_node *np = client->dev.of_node;
+ struct device *dev = &client->dev;
+ struct cs53l30_private *cs53l30;
+ unsigned int devid = 0;
+ unsigned int reg;
+ int ret = 0, i;
+ u8 val;
+
+ cs53l30 = devm_kzalloc(dev, sizeof(*cs53l30), GFP_KERNEL);
+ if (!cs53l30)
+ return -ENOMEM;
+
+ for (i = 0; i < ARRAY_SIZE(cs53l30->supplies); i++)
+ cs53l30->supplies[i].supply = cs53l30_supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs53l30->supplies),
+ cs53l30->supplies);
+ if (ret) {
+ dev_err(dev, "failed to get supplies: %d\n", ret);
+ return ret;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(cs53l30->supplies),
+ cs53l30->supplies);
+ if (ret) {
+ dev_err(dev, "failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ /* Reset the Device */
+ cs53l30->reset_gpio = devm_gpiod_get_optional(dev, "reset",
+ GPIOD_OUT_LOW);
+ if (IS_ERR(cs53l30->reset_gpio)) {
+ ret = PTR_ERR(cs53l30->reset_gpio);
+ goto error;
+ }
+
+ if (cs53l30->reset_gpio)
+ gpiod_set_value_cansleep(cs53l30->reset_gpio, 1);
+
+ i2c_set_clientdata(client, cs53l30);
+
+ cs53l30->mclk_rate = 0;
+
+ cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap);
+ if (IS_ERR(cs53l30->regmap)) {
+ ret = PTR_ERR(cs53l30->regmap);
+ dev_err(dev, "regmap_init() failed: %d\n", ret);
+ goto error;
+ }
+
+ /* Initialize codec */
+ ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg);
+ devid = reg << 12;
+
+ ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg);
+ devid |= reg << 4;
+
+ ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg);
+ devid |= (reg & 0xF0) >> 4;
+
+ if (devid != CS53L30_DEVID) {
+ ret = -ENODEV;
+ dev_err(dev, "Device ID (%X). Expected %X\n",
+ devid, CS53L30_DEVID);
+ goto error;
+ }
+
+ ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg);
+ if (ret < 0) {
+ dev_err(dev, "failed to get Revision ID: %d\n", ret);
+ goto error;
+ }
+
+ /* Check if MCLK provided */
+ cs53l30->mclk = devm_clk_get(dev, "mclk");
+ if (IS_ERR(cs53l30->mclk)) {
+ if (PTR_ERR(cs53l30->mclk) == -EPROBE_DEFER) {
+ ret = -EPROBE_DEFER;
+ goto error;
+ }
+ /* Otherwise mark the mclk pointer to NULL */
+ cs53l30->mclk = NULL;
+ }
+
+ /* Fetch the MUTE control */
+ cs53l30->mute_gpio = devm_gpiod_get_optional(dev, "mute",
+ GPIOD_OUT_HIGH);
+ if (IS_ERR(cs53l30->mute_gpio)) {
+ ret = PTR_ERR(cs53l30->mute_gpio);
+ goto error;
+ }
+
+ if (cs53l30->mute_gpio) {
+ /* Enable MUTE controls via MUTE pin */
+ regmap_write(cs53l30->regmap, CS53L30_MUTEP_CTL1,
+ CS53L30_MUTEP_CTL1_MUTEALL);
+ /* Flip the polarity of MUTE pin */
+ if (gpiod_is_active_low(cs53l30->mute_gpio))
+ regmap_update_bits(cs53l30->regmap, CS53L30_MUTEP_CTL2,
+ CS53L30_MUTE_PIN_POLARITY, 0);
+ }
+
+ if (!of_property_read_u8(np, "cirrus,micbias-lvl", &val))
+ regmap_update_bits(cs53l30->regmap, CS53L30_MICBIAS_CTL,
+ CS53L30_MIC_BIAS_CTRL_MASK, val);
+
+ if (of_property_read_bool(np, "cirrus,use-sdout2"))
+ cs53l30->use_sdout2 = true;
+
+ dev_info(dev, "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF);
+
+ ret = snd_soc_register_codec(dev, &cs53l30_driver, &cs53l30_dai, 1);
+ if (ret) {
+ dev_err(dev, "failed to register codec: %d\n", ret);
+ goto error;
+ }
+
+ return 0;
+
+error:
+ regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies),
+ cs53l30->supplies);
+ return ret;
+}
+
+static int cs53l30_i2c_remove(struct i2c_client *client)
+{
+ struct cs53l30_private *cs53l30 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+
+ /* Hold down reset */
+ if (cs53l30->reset_gpio)
+ gpiod_set_value_cansleep(cs53l30->reset_gpio, 0);
+
+ regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies),
+ cs53l30->supplies);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int cs53l30_runtime_suspend(struct device *dev)
+{
+ struct cs53l30_private *cs53l30 = dev_get_drvdata(dev);
+
+ regcache_cache_only(cs53l30->regmap, true);
+
+ /* Hold down reset */
+ if (cs53l30->reset_gpio)
+ gpiod_set_value_cansleep(cs53l30->reset_gpio, 0);
+
+ regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies),
+ cs53l30->supplies);
+
+ return 0;
+}
+
+static int cs53l30_runtime_resume(struct device *dev)
+{
+ struct cs53l30_private *cs53l30 = dev_get_drvdata(dev);
+ int ret;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(cs53l30->supplies),
+ cs53l30->supplies);
+ if (ret) {
+ dev_err(dev, "failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ if (cs53l30->reset_gpio)
+ gpiod_set_value_cansleep(cs53l30->reset_gpio, 1);
+
+ regcache_cache_only(cs53l30->regmap, false);
+ ret = regcache_sync(cs53l30->regmap);
+ if (ret) {
+ dev_err(dev, "failed to synchronize regcache: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops cs53l30_runtime_pm = {
+ SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume,
+ NULL)
+};
+
+static const struct of_device_id cs53l30_of_match[] = {
+ { .compatible = "cirrus,cs53l30", },
+ {},
+};
+
+MODULE_DEVICE_TABLE(of, cs53l30_of_match);
+
+static const struct i2c_device_id cs53l30_id[] = {
+ { "cs53l30", 0 },
+ {}
+};
+
+MODULE_DEVICE_TABLE(i2c, cs53l30_id);
+
+static struct i2c_driver cs53l30_i2c_driver = {
+ .driver = {
+ .name = "cs53l30",
+ .pm = &cs53l30_runtime_pm,
+ },
+ .id_table = cs53l30_id,
+ .probe = cs53l30_i2c_probe,
+ .remove = cs53l30_i2c_remove,
+};
+
+module_i2c_driver(cs53l30_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS53L30 driver");
+MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan@cirrus.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h
new file mode 100644
index 000000000000..5e39da568749
--- /dev/null
+++ b/sound/soc/codecs/cs53l30.h
@@ -0,0 +1,459 @@
+/*
+ * ALSA SoC CS53L30 codec driver
+ *
+ * Copyright 2015 Cirrus Logic, Inc.
+ *
+ * Author: Paul Handrigan <Paul.Handrigan@cirrus.com>,
+ * Tim Howe <Tim.Howe@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef __CS53L30_H__
+#define __CS53L30_H__
+
+/* I2C Registers */
+#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */
+#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */
+#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */
+#define CS53L30_REVID 0x05 /* Revision ID [RO]. */
+#define CS53L30_PWRCTL 0x06 /* Power Control. */
+#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */
+#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */
+#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */
+#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */
+#define CS53L30_ASP_CTL1 0x0D /* ASP1 Control. */
+#define CS53L30_ASP_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */
+#define CS53L30_ASP_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */
+#define CS53L30_ASP_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */
+#define CS53L30_ASP_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */
+#define CS53L30_ASP_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */
+#define CS53L30_ASP_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */
+#define CS53L30_ASP_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */
+#define CS53L30_ASP_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */
+#define CS53L30_ASP_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */
+#define CS53L30_ASP_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */
+#define CS53L30_ASP_CTL2 0x18 /* ASP2 Control. */
+#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */
+#define CS53L30_LRCK_CTL1 0x1B /* LRCK Control 1. */
+#define CS53L30_LRCK_CTL2 0x1C /* LRCK Control 2. */
+#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */
+#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */
+#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */
+#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */
+#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */
+#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */
+#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */
+#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */
+#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */
+#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */
+#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */
+#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */
+#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */
+#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */
+#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */
+#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */
+#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */
+#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */
+#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */
+#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */
+#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */
+#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */
+#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */
+#define CS53L30_IS 0x36 /* Interrupt Status. */
+#define CS53L30_MAX_REGISTER 0x36
+
+#define CS53L30_TDM_SLOT_MAX 4
+#define CS53L30_ASP_TDMTX_CTL(x) (CS53L30_ASP_TDMTX_CTL1 + (x))
+/* x : index for registers; n : index for slot; 8 slots per register */
+#define CS53L30_ASP_TDMTX_ENx(x) (CS53L30_ASP_TDMTX_EN6 - (x))
+#define CS53L30_ASP_TDMTX_ENn(n) CS53L30_ASP_TDMTX_ENx((n) >> 3)
+#define CS53L30_ASP_TDMTX_ENx_MAX 6
+
+/* Device ID */
+#define CS53L30_DEVID 0x53A30
+
+/* PDN_DONE Poll Maximum
+ * If soft ramp is set it will take much longer to power down
+ * the system.
+ */
+#define CS53L30_PDN_POLL_MAX 90
+
+/* Bitfield Definitions */
+
+/* R6 (0x06) CS53L30_PWRCTL - Power Control */
+#define CS53L30_PDN_ULP_SHIFT 7
+#define CS53L30_PDN_ULP_MASK (1 << CS53L30_PDN_ULP_SHIFT)
+#define CS53L30_PDN_ULP (1 << CS53L30_PDN_ULP_SHIFT)
+#define CS53L30_PDN_LP_SHIFT 6
+#define CS53L30_PDN_LP_MASK (1 << CS53L30_PDN_LP_SHIFT)
+#define CS53L30_PDN_LP (1 << CS53L30_PDN_LP_SHIFT)
+#define CS53L30_DISCHARGE_FILT_SHIFT 5
+#define CS53L30_DISCHARGE_FILT_MASK (1 << CS53L30_DISCHARGE_FILT_SHIFT)
+#define CS53L30_DISCHARGE_FILT (1 << CS53L30_DISCHARGE_FILT_SHIFT)
+#define CS53L30_THMS_PDN_SHIFT 4
+#define CS53L30_THMS_PDN_MASK (1 << CS53L30_THMS_PDN_SHIFT)
+#define CS53L30_THMS_PDN (1 << CS53L30_THMS_PDN_SHIFT)
+
+#define CS53L30_PWRCTL_DEFAULT (CS53L30_THMS_PDN)
+
+/* R7 (0x07) CS53L30_MCLKCTL - MCLK Control */
+#define CS53L30_MCLK_DIS_SHIFT 7
+#define CS53L30_MCLK_DIS_MASK (1 << CS53L30_MCLK_DIS_SHIFT)
+#define CS53L30_MCLK_DIS (1 << CS53L30_MCLK_DIS_SHIFT)
+#define CS53L30_MCLK_INT_SCALE_SHIFT 6
+#define CS53L30_MCLK_INT_SCALE_MASK (1 << CS53L30_MCLK_INT_SCALE_SHIFT)
+#define CS53L30_MCLK_INT_SCALE (1 << CS53L30_MCLK_INT_SCALE_SHIFT)
+#define CS53L30_DMIC_DRIVE_SHIFT 5
+#define CS53L30_DMIC_DRIVE_MASK (1 << CS53L30_DMIC_DRIVE_SHIFT)
+#define CS53L30_DMIC_DRIVE (1 << CS53L30_DMIC_DRIVE_SHIFT)
+#define CS53L30_MCLK_DIV_SHIFT 2
+#define CS53L30_MCLK_DIV_WIDTH 2
+#define CS53L30_MCLK_DIV_MASK (((1 << CS53L30_MCLK_DIV_WIDTH) - 1) << CS53L30_MCLK_DIV_SHIFT)
+#define CS53L30_MCLK_DIV_BY_1 (0x0 << CS53L30_MCLK_DIV_SHIFT)
+#define CS53L30_MCLK_DIV_BY_2 (0x1 << CS53L30_MCLK_DIV_SHIFT)
+#define CS53L30_MCLK_DIV_BY_3 (0x2 << CS53L30_MCLK_DIV_SHIFT)
+#define CS53L30_SYNC_EN_SHIFT 1
+#define CS53L30_SYNC_EN_MASK (1 << CS53L30_SYNC_EN_SHIFT)
+#define CS53L30_SYNC_EN (1 << CS53L30_SYNC_EN_SHIFT)
+
+#define CS53L30_MCLKCTL_DEFAULT (CS53L30_MCLK_DIV_BY_2)
+
+/* R8 (0x08) CS53L30_INT_SR_CTL - Internal Sample Rate Control */
+#define CS53L30_INTRNL_FS_RATIO_SHIFT 4
+#define CS53L30_INTRNL_FS_RATIO_MASK (1 << CS53L30_INTRNL_FS_RATIO_SHIFT)
+#define CS53L30_INTRNL_FS_RATIO (1 << CS53L30_INTRNL_FS_RATIO_SHIFT)
+#define CS53L30_MCLK_19MHZ_EN_SHIFT 0
+#define CS53L30_MCLK_19MHZ_EN_MASK (1 << CS53L30_MCLK_19MHZ_EN_SHIFT)
+#define CS53L30_MCLK_19MHZ_EN (1 << CS53L30_MCLK_19MHZ_EN_SHIFT)
+
+/* 0x6 << 1 is reserved bits */
+#define CS53L30_INT_SR_CTL_DEFAULT (CS53L30_INTRNL_FS_RATIO | 0x6 << 1)
+
+/* R10 (0x0A) CS53L30_MICBIAS_CTL - Mic Bias Control */
+#define CS53L30_MIC4_BIAS_PDN_SHIFT 7
+#define CS53L30_MIC4_BIAS_PDN_MASK (1 << CS53L30_MIC4_BIAS_PDN_SHIFT)
+#define CS53L30_MIC4_BIAS_PDN (1 << CS53L30_MIC4_BIAS_PDN_SHIFT)
+#define CS53L30_MIC3_BIAS_PDN_SHIFT 6
+#define CS53L30_MIC3_BIAS_PDN_MASK (1 << CS53L30_MIC3_BIAS_PDN_SHIFT)
+#define CS53L30_MIC3_BIAS_PDN (1 << CS53L30_MIC3_BIAS_PDN_SHIFT)
+#define CS53L30_MIC2_BIAS_PDN_SHIFT 5
+#define CS53L30_MIC2_BIAS_PDN_MASK (1 << CS53L30_MIC2_BIAS_PDN_SHIFT)
+#define CS53L30_MIC2_BIAS_PDN (1 << CS53L30_MIC2_BIAS_PDN_SHIFT)
+#define CS53L30_MIC1_BIAS_PDN_SHIFT 4
+#define CS53L30_MIC1_BIAS_PDN_MASK (1 << CS53L30_MIC1_BIAS_PDN_SHIFT)
+#define CS53L30_MIC1_BIAS_PDN (1 << CS53L30_MIC1_BIAS_PDN_SHIFT)
+#define CS53L30_MICx_BIAS_PDN (0xf << CS53L30_MIC1_BIAS_PDN_SHIFT)
+#define CS53L30_VP_MIN_SHIFT 2
+#define CS53L30_VP_MIN_MASK (1 << CS53L30_VP_MIN_SHIFT)
+#define CS53L30_VP_MIN (1 << CS53L30_VP_MIN_SHIFT)
+#define CS53L30_MIC_BIAS_CTRL_SHIFT 0
+#define CS53L30_MIC_BIAS_CTRL_WIDTH 2
+#define CS53L30_MIC_BIAS_CTRL_MASK (((1 << CS53L30_MIC_BIAS_CTRL_WIDTH) - 1) << CS53L30_MIC_BIAS_CTRL_SHIFT)
+#define CS53L30_MIC_BIAS_CTRL_HIZ (0 << CS53L30_MIC_BIAS_CTRL_SHIFT)
+#define CS53L30_MIC_BIAS_CTRL_1V8 (1 << CS53L30_MIC_BIAS_CTRL_SHIFT)
+#define CS53L30_MIC_BIAS_CTRL_2V75 (2 << CS53L30_MIC_BIAS_CTRL_SHIFT)
+
+#define CS53L30_MICBIAS_CTL_DEFAULT (CS53L30_MICx_BIAS_PDN | CS53L30_VP_MIN)
+
+/* R12 (0x0C) CS53L30_ASPCFG_CTL - ASP Configuration Control */
+#define CS53L30_ASP_MS_SHIFT 7
+#define CS53L30_ASP_MS_MASK (1 << CS53L30_ASP_MS_SHIFT)
+#define CS53L30_ASP_MS (1 << CS53L30_ASP_MS_SHIFT)
+#define CS53L30_ASP_SCLK_INV_SHIFT 4
+#define CS53L30_ASP_SCLK_INV_MASK (1 << CS53L30_ASP_SCLK_INV_SHIFT)
+#define CS53L30_ASP_SCLK_INV (1 << CS53L30_ASP_SCLK_INV_SHIFT)
+#define CS53L30_ASP_RATE_SHIFT 0
+#define CS53L30_ASP_RATE_WIDTH 4
+#define CS53L30_ASP_RATE_MASK (((1 << CS53L30_ASP_RATE_WIDTH) - 1) << CS53L30_ASP_RATE_SHIFT)
+#define CS53L30_ASP_RATE_48K (0xc << CS53L30_ASP_RATE_SHIFT)
+
+#define CS53L30_ASPCFG_CTL_DEFAULT (CS53L30_ASP_RATE_48K)
+
+/* R13/R24 (0x0D/0x18) CS53L30_ASP_CTL1 & CS53L30_ASP_CTL2 - ASP Control 1~2 */
+#define CS53L30_ASP_TDM_PDN_SHIFT 7
+#define CS53L30_ASP_TDM_PDN_MASK (1 << CS53L30_ASP_TDM_PDN_SHIFT)
+#define CS53L30_ASP_TDM_PDN (1 << CS53L30_ASP_TDM_PDN_SHIFT)
+#define CS53L30_ASP_SDOUTx_PDN_SHIFT 6
+#define CS53L30_ASP_SDOUTx_PDN_MASK (1 << CS53L30_ASP_SDOUTx_PDN_SHIFT)
+#define CS53L30_ASP_SDOUTx_PDN (1 << CS53L30_ASP_SDOUTx_PDN_SHIFT)
+#define CS53L30_ASP_3ST_SHIFT 5
+#define CS53L30_ASP_3ST_MASK (1 << CS53L30_ASP_3ST_SHIFT)
+#define CS53L30_ASP_3ST (1 << CS53L30_ASP_3ST_SHIFT)
+#define CS53L30_SHIFT_LEFT_SHIFT 4
+#define CS53L30_SHIFT_LEFT_MASK (1 << CS53L30_SHIFT_LEFT_SHIFT)
+#define CS53L30_SHIFT_LEFT (1 << CS53L30_SHIFT_LEFT_SHIFT)
+#define CS53L30_ASP_SDOUTx_DRIVE_SHIFT 0
+#define CS53L30_ASP_SDOUTx_DRIVE_MASK (1 << CS53L30_ASP_SDOUTx_DRIVE_SHIFT)
+#define CS53L30_ASP_SDOUTx_DRIVE (1 << CS53L30_ASP_SDOUTx_DRIVE_SHIFT)
+
+#define CS53L30_ASP_CTL1_DEFAULT (CS53L30_ASP_TDM_PDN)
+#define CS53L30_ASP_CTL2_DEFAULT (0)
+
+/* R14 (0x0E) ~ R17 (0x11) CS53L30_ASP_TDMTX_CTLx - ASP TDM TX Control 1~4 */
+#define CS53L30_ASP_CHx_TX_STATE_SHIFT 7
+#define CS53L30_ASP_CHx_TX_STATE_MASK (1 << CS53L30_ASP_CHx_TX_STATE_SHIFT)
+#define CS53L30_ASP_CHx_TX_STATE (1 << CS53L30_ASP_CHx_TX_STATE_SHIFT)
+#define CS53L30_ASP_CHx_TX_LOC_SHIFT 0
+#define CS53L30_ASP_CHx_TX_LOC_WIDTH 6
+#define CS53L30_ASP_CHx_TX_LOC_MASK (((1 << CS53L30_ASP_CHx_TX_LOC_WIDTH) - 1) << CS53L30_ASP_CHx_TX_LOC_SHIFT)
+#define CS53L30_ASP_CHx_TX_LOC_MAX (47 << CS53L30_ASP_CHx_TX_LOC_SHIFT)
+#define CS53L30_ASP_CHx_TX_LOC(x) ((x) << CS53L30_ASP_CHx_TX_LOC_SHIFT)
+
+#define CS53L30_ASP_TDMTX_CTLx_DEFAULT (CS53L30_ASP_CHx_TX_LOC_MAX)
+
+/* R18 (0x12) ~ R23 (0x17) CS53L30_ASP_TDMTX_ENx - ASP TDM TX Enable 1~6 */
+#define CS53L30_ASP_TDMTX_ENx_DEFAULT (0)
+
+/* R26 (0x1A) CS53L30_SFT_RAMP - Soft Ramp Control */
+#define CS53L30_DIGSFT_SHIFT 5
+#define CS53L30_DIGSFT_MASK (1 << CS53L30_DIGSFT_SHIFT)
+#define CS53L30_DIGSFT (1 << CS53L30_DIGSFT_SHIFT)
+
+#define CS53L30_SFT_RMP_DEFAULT (0)
+
+/* R28 (0x1C) CS53L30_LRCK_CTL2 - LRCK Control 2 */
+#define CS53L30_LRCK_50_NPW_SHIFT 3
+#define CS53L30_LRCK_50_NPW_MASK (1 << CS53L30_LRCK_50_NPW_SHIFT)
+#define CS53L30_LRCK_50_NPW (1 << CS53L30_LRCK_50_NPW_SHIFT)
+#define CS53L30_LRCK_TPWH_SHIFT 0
+#define CS53L30_LRCK_TPWH_WIDTH 3
+#define CS53L30_LRCK_TPWH_MASK (((1 << CS53L30_LRCK_TPWH_WIDTH) - 1) << CS53L30_LRCK_TPWH_SHIFT)
+#define CS53L30_LRCK_TPWH(x) (((x) << CS53L30_LRCK_TPWH_SHIFT) & CS53L30_LRCK_TPWH_MASK)
+
+#define CS53L30_LRCK_CTLx_DEFAULT (0)
+
+/* R31 (0x1F) CS53L30_MUTEP_CTL1 - MUTE Pin Control 1 */
+#define CS53L30_MUTE_PDN_ULP_SHIFT 7
+#define CS53L30_MUTE_PDN_ULP_MASK (1 << CS53L30_MUTE_PDN_ULP_SHIFT)
+#define CS53L30_MUTE_PDN_ULP (1 << CS53L30_MUTE_PDN_ULP_SHIFT)
+#define CS53L30_MUTE_PDN_LP_SHIFT 6
+#define CS53L30_MUTE_PDN_LP_MASK (1 << CS53L30_MUTE_PDN_LP_SHIFT)
+#define CS53L30_MUTE_PDN_LP (1 << CS53L30_MUTE_PDN_LP_SHIFT)
+#define CS53L30_MUTE_M4B_PDN_SHIFT 4
+#define CS53L30_MUTE_M4B_PDN_MASK (1 << CS53L30_MUTE_M4B_PDN_SHIFT)
+#define CS53L30_MUTE_M4B_PDN (1 << CS53L30_MUTE_M4B_PDN_SHIFT)
+#define CS53L30_MUTE_M3B_PDN_SHIFT 3
+#define CS53L30_MUTE_M3B_PDN_MASK (1 << CS53L30_MUTE_M3B_PDN_SHIFT)
+#define CS53L30_MUTE_M3B_PDN (1 << CS53L30_MUTE_M3B_PDN_SHIFT)
+#define CS53L30_MUTE_M2B_PDN_SHIFT 2
+#define CS53L30_MUTE_M2B_PDN_MASK (1 << CS53L30_MUTE_M2B_PDN_SHIFT)
+#define CS53L30_MUTE_M2B_PDN (1 << CS53L30_MUTE_M2B_PDN_SHIFT)
+#define CS53L30_MUTE_M1B_PDN_SHIFT 1
+#define CS53L30_MUTE_M1B_PDN_MASK (1 << CS53L30_MUTE_M1B_PDN_SHIFT)
+#define CS53L30_MUTE_M1B_PDN (1 << CS53L30_MUTE_M1B_PDN_SHIFT)
+/* Note: be careful - x starts from 0 */
+#define CS53L30_MUTE_MxB_PDN_SHIFT(x) (CS53L30_MUTE_M1B_PDN_SHIFT + (x))
+#define CS53L30_MUTE_MxB_PDN_MASK(x) (1 << CS53L30_MUTE_MxB_PDN_SHIFT(x))
+#define CS53L30_MUTE_MxB_PDN(x) (1 << CS53L30_MUTE_MxB_PDN_SHIFT(x))
+#define CS53L30_MUTE_MB_ALL_PDN_SHIFT 0
+#define CS53L30_MUTE_MB_ALL_PDN_MASK (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT)
+#define CS53L30_MUTE_MB_ALL_PDN (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT)
+
+#define CS53L30_MUTEP_CTL1_MUTEALL (0xdf)
+#define CS53L30_MUTEP_CTL1_DEFAULT (0)
+
+/* R32 (0x20) CS53L30_MUTEP_CTL2 - MUTE Pin Control 2 */
+#define CS53L30_MUTE_PIN_POLARITY_SHIFT 7
+#define CS53L30_MUTE_PIN_POLARITY_MASK (1 << CS53L30_MUTE_PIN_POLARITY_SHIFT)
+#define CS53L30_MUTE_PIN_POLARITY (1 << CS53L30_MUTE_PIN_POLARITY_SHIFT)
+#define CS53L30_MUTE_ASP_TDM_PDN_SHIFT 6
+#define CS53L30_MUTE_ASP_TDM_PDN_MASK (1 << CS53L30_MUTE_ASP_TDM_PDN_SHIFT)
+#define CS53L30_MUTE_ASP_TDM_PDN (1 << CS53L30_MUTE_ASP_TDM_PDN_SHIFT)
+#define CS53L30_MUTE_ASP_SDOUT2_PDN_SHIFT 5
+#define CS53L30_MUTE_ASP_SDOUT2_PDN_MASK (1 << CS53L30_MUTE_ASP_SDOUT2_PDN_SHIFT)
+#define CS53L30_MUTE_ASP_SDOUT2_PDN (1 << CS53L30_MUTE_ASP_SDOUT2_PDN_SHIFT)
+#define CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT 4
+#define CS53L30_MUTE_ASP_SDOUT1_PDN_MASK (1 << CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT)
+#define CS53L30_MUTE_ASP_SDOUT1_PDN (1 << CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT)
+/* Note: be careful - x starts from 0 */
+#define CS53L30_MUTE_ASP_SDOUTx_PDN_SHIFT(x) ((x) + CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT)
+#define CS53L30_MUTE_ASP_SDOUTx_PDN_MASK(x) (1 << CS53L30_MUTE_ASP_SDOUTx_PDN_SHIFT(x))
+#define CS53L30_MUTE_ASP_SDOUTx_PDN (1 << CS53L30_MUTE_ASP_SDOUTx_PDN_SHIFT(x))
+#define CS53L30_MUTE_ADC2B_PDN_SHIFT 3
+#define CS53L30_MUTE_ADC2B_PDN_MASK (1 << CS53L30_MUTE_ADC2B_PDN_SHIFT)
+#define CS53L30_MUTE_ADC2B_PDN (1 << CS53L30_MUTE_ADC2B_PDN_SHIFT)
+#define CS53L30_MUTE_ADC2A_PDN_SHIFT 2
+#define CS53L30_MUTE_ADC2A_PDN_MASK (1 << CS53L30_MUTE_ADC2A_PDN_SHIFT)
+#define CS53L30_MUTE_ADC2A_PDN (1 << CS53L30_MUTE_ADC2A_PDN_SHIFT)
+#define CS53L30_MUTE_ADC1B_PDN_SHIFT 1
+#define CS53L30_MUTE_ADC1B_PDN_MASK (1 << CS53L30_MUTE_ADC1B_PDN_SHIFT)
+#define CS53L30_MUTE_ADC1B_PDN (1 << CS53L30_MUTE_ADC1B_PDN_SHIFT)
+#define CS53L30_MUTE_ADC1A_PDN_SHIFT 0
+#define CS53L30_MUTE_ADC1A_PDN_MASK (1 << CS53L30_MUTE_ADC1A_PDN_SHIFT)
+#define CS53L30_MUTE_ADC1A_PDN (1 << CS53L30_MUTE_ADC1A_PDN_SHIFT)
+
+#define CS53L30_MUTEP_CTL2_DEFAULT (CS53L30_MUTE_PIN_POLARITY)
+
+/* R33 (0x21) CS53L30_INBIAS_CTL1 - Input Bias Control 1 */
+#define CS53L30_IN4M_BIAS_SHIFT 6
+#define CS53L30_IN4M_BIAS_WIDTH 2
+#define CS53L30_IN4M_BIAS_MASK (((1 << CS53L30_IN4M_BIAS_WIDTH) - 1) << CS53L30_IN4M_BIAS_SHIFT)
+#define CS53L30_IN4M_BIAS_OPEN (0 << CS53L30_IN4M_BIAS_SHIFT)
+#define CS53L30_IN4M_BIAS_PULL_DOWN (1 << CS53L30_IN4M_BIAS_SHIFT)
+#define CS53L30_IN4M_BIAS_VCM (2 << CS53L30_IN4M_BIAS_SHIFT)
+#define CS53L30_IN4P_BIAS_SHIFT 4
+#define CS53L30_IN4P_BIAS_WIDTH 2
+#define CS53L30_IN4P_BIAS_MASK (((1 << CS53L30_IN4P_BIAS_WIDTH) - 1) << CS53L30_IN4P_BIAS_SHIFT)
+#define CS53L30_IN4P_BIAS_OPEN (0 << CS53L30_IN4P_BIAS_SHIFT)
+#define CS53L30_IN4P_BIAS_PULL_DOWN (1 << CS53L30_IN4P_BIAS_SHIFT)
+#define CS53L30_IN4P_BIAS_VCM (2 << CS53L30_IN4P_BIAS_SHIFT)
+#define CS53L30_IN3M_BIAS_SHIFT 2
+#define CS53L30_IN3M_BIAS_WIDTH 2
+#define CS53L30_IN3M_BIAS_MASK (((1 << CS53L30_IN3M_BIAS_WIDTH) - 1) << CS53L30_IN4M_BIAS_SHIFT)
+#define CS53L30_IN3M_BIAS_OPEN (0 << CS53L30_IN3M_BIAS_SHIFT)
+#define CS53L30_IN3M_BIAS_PULL_DOWN (1 << CS53L30_IN3M_BIAS_SHIFT)
+#define CS53L30_IN3M_BIAS_VCM (2 << CS53L30_IN3M_BIAS_SHIFT)
+#define CS53L30_IN3P_BIAS_SHIFT 0
+#define CS53L30_IN3P_BIAS_WIDTH 2
+#define CS53L30_IN3P_BIAS_MASK (((1 << CS53L30_IN3P_BIAS_WIDTH) - 1) << CS53L30_IN3P_BIAS_SHIFT)
+#define CS53L30_IN3P_BIAS_OPEN (0 << CS53L30_IN3P_BIAS_SHIFT)
+#define CS53L30_IN3P_BIAS_PULL_DOWN (1 << CS53L30_IN3P_BIAS_SHIFT)
+#define CS53L30_IN3P_BIAS_VCM (2 << CS53L30_IN3P_BIAS_SHIFT)
+
+#define CS53L30_INBIAS_CTL1_DEFAULT (CS53L30_IN4M_BIAS_VCM | CS53L30_IN4P_BIAS_VCM |\
+ CS53L30_IN3M_BIAS_VCM | CS53L30_IN3P_BIAS_VCM)
+
+/* R34 (0x22) CS53L30_INBIAS_CTL2 - Input Bias Control 2 */
+#define CS53L30_IN2M_BIAS_SHIFT 6
+#define CS53L30_IN2M_BIAS_WIDTH 2
+#define CS53L30_IN2M_BIAS_MASK (((1 << CS53L30_IN2M_BIAS_WIDTH) - 1) << CS53L30_IN2M_BIAS_SHIFT)
+#define CS53L30_IN2M_BIAS_OPEN (0 << CS53L30_IN2M_BIAS_SHIFT)
+#define CS53L30_IN2M_BIAS_PULL_DOWN (1 << CS53L30_IN2M_BIAS_SHIFT)
+#define CS53L30_IN2M_BIAS_VCM (2 << CS53L30_IN2M_BIAS_SHIFT)
+#define CS53L30_IN2P_BIAS_SHIFT 4
+#define CS53L30_IN2P_BIAS_WIDTH 2
+#define CS53L30_IN2P_BIAS_MASK (((1 << CS53L30_IN2P_BIAS_WIDTH) - 1) << CS53L30_IN2P_BIAS_SHIFT)
+#define CS53L30_IN2P_BIAS_OPEN (0 << CS53L30_IN2P_BIAS_SHIFT)
+#define CS53L30_IN2P_BIAS_PULL_DOWN (1 << CS53L30_IN2P_BIAS_SHIFT)
+#define CS53L30_IN2P_BIAS_VCM (2 << CS53L30_IN2P_BIAS_SHIFT)
+#define CS53L30_IN1M_BIAS_SHIFT 2
+#define CS53L30_IN1M_BIAS_WIDTH 2
+#define CS53L30_IN1M_BIAS_MASK (((1 << CS53L30_IN1M_BIAS_WIDTH) - 1) << CS53L30_IN1M_BIAS_SHIFT)
+#define CS53L30_IN1M_BIAS_OPEN (0 << CS53L30_IN1M_BIAS_SHIFT)
+#define CS53L30_IN1M_BIAS_PULL_DOWN (1 << CS53L30_IN1M_BIAS_SHIFT)
+#define CS53L30_IN1M_BIAS_VCM (2 << CS53L30_IN1M_BIAS_SHIFT)
+#define CS53L30_IN1P_BIAS_SHIFT 0
+#define CS53L30_IN1P_BIAS_WIDTH 2
+#define CS53L30_IN1P_BIAS_MASK (((1 << CS53L30_IN1P_BIAS_WIDTH) - 1) << CS53L30_IN1P_BIAS_SHIFT)
+#define CS53L30_IN1P_BIAS_OPEN (0 << CS53L30_IN1P_BIAS_SHIFT)
+#define CS53L30_IN1P_BIAS_PULL_DOWN (1 << CS53L30_IN1P_BIAS_SHIFT)
+#define CS53L30_IN1P_BIAS_VCM (2 << CS53L30_IN1P_BIAS_SHIFT)
+
+#define CS53L30_INBIAS_CTL2_DEFAULT (CS53L30_IN2M_BIAS_VCM | CS53L30_IN2P_BIAS_VCM |\
+ CS53L30_IN1M_BIAS_VCM | CS53L30_IN1P_BIAS_VCM)
+
+/* R35 (0x23) & R36 (0x24) CS53L30_DMICx_STR_CTL - DMIC1 & DMIC2 Stereo Control */
+#define CS53L30_DMICx_STEREO_ENB_SHIFT 5
+#define CS53L30_DMICx_STEREO_ENB_MASK (1 << CS53L30_DMICx_STEREO_ENB_SHIFT)
+#define CS53L30_DMICx_STEREO_ENB (1 << CS53L30_DMICx_STEREO_ENB_SHIFT)
+
+/* 0x88 and 0xCC are reserved bits */
+#define CS53L30_DMIC1_STR_CTL_DEFAULT (CS53L30_DMICx_STEREO_ENB | 0x88)
+#define CS53L30_DMIC2_STR_CTL_DEFAULT (CS53L30_DMICx_STEREO_ENB | 0xCC)
+
+/* R37/R45 (0x25/0x2D) CS53L30_ADCDMICx_CTL1 - ADC1/DMIC1 & ADC2/DMIC2 Control 1 */
+#define CS53L30_ADCxB_PDN_SHIFT 7
+#define CS53L30_ADCxB_PDN_MASK (1 << CS53L30_ADCxB_PDN_SHIFT)
+#define CS53L30_ADCxB_PDN (1 << CS53L30_ADCxB_PDN_SHIFT)
+#define CS53L30_ADCxA_PDN_SHIFT 6
+#define CS53L30_ADCxA_PDN_MASK (1 << CS53L30_ADCxA_PDN_SHIFT)
+#define CS53L30_ADCxA_PDN (1 << CS53L30_ADCxA_PDN_SHIFT)
+#define CS53L30_DMICx_PDN_SHIFT 2
+#define CS53L30_DMICx_PDN_MASK (1 << CS53L30_DMICx_PDN_SHIFT)
+#define CS53L30_DMICx_PDN (1 << CS53L30_DMICx_PDN_SHIFT)
+#define CS53L30_DMICx_SCLK_DIV_SHIFT 1
+#define CS53L30_DMICx_SCLK_DIV_MASK (1 << CS53L30_DMICx_SCLK_DIV_SHIFT)
+#define CS53L30_DMICx_SCLK_DIV (1 << CS53L30_DMICx_SCLK_DIV_SHIFT)
+#define CS53L30_CH_TYPE_SHIFT 0
+#define CS53L30_CH_TYPE_MASK (1 << CS53L30_CH_TYPE_SHIFT)
+#define CS53L30_CH_TYPE (1 << CS53L30_CH_TYPE_SHIFT)
+
+#define CS53L30_ADCDMICx_PDN_MASK 0xFF
+#define CS53L30_ADCDMICx_CTL1_DEFAULT (CS53L30_DMICx_PDN)
+
+/* R38/R46 (0x26/0x2E) CS53L30_ADCDMICx_CTL2 - ADC1/DMIC1 & ADC2/DMIC2 Control 2 */
+#define CS53L30_ADCx_NOTCH_DIS_SHIFT 7
+#define CS53L30_ADCx_NOTCH_DIS_MASK (1 << CS53L30_ADCx_NOTCH_DIS_SHIFT)
+#define CS53L30_ADCx_NOTCH_DIS (1 << CS53L30_ADCx_NOTCH_DIS_SHIFT)
+#define CS53L30_ADCxB_INV_SHIFT 5
+#define CS53L30_ADCxB_INV_MASK (1 << CS53L30_ADCxB_INV_SHIFT)
+#define CS53L30_ADCxB_INV (1 << CS53L30_ADCxB_INV_SHIFT)
+#define CS53L30_ADCxA_INV_SHIFT 4
+#define CS53L30_ADCxA_INV_MASK (1 << CS53L30_ADCxA_INV_SHIFT)
+#define CS53L30_ADCxA_INV (1 << CS53L30_ADCxA_INV_SHIFT)
+#define CS53L30_ADCxB_DIG_BOOST_SHIFT 1
+#define CS53L30_ADCxB_DIG_BOOST_MASK (1 << CS53L30_ADCxB_DIG_BOOST_SHIFT)
+#define CS53L30_ADCxB_DIG_BOOST (1 << CS53L30_ADCxB_DIG_BOOST_SHIFT)
+#define CS53L30_ADCxA_DIG_BOOST_SHIFT 0
+#define CS53L30_ADCxA_DIG_BOOST_MASK (1 << CS53L30_ADCxA_DIG_BOOST_SHIFT)
+#define CS53L30_ADCxA_DIG_BOOST (1 << CS53L30_ADCxA_DIG_BOOST_SHIFT)
+
+#define CS53L30_ADCDMIC1_CTL2_DEFAULT (0)
+
+/* R39/R47 (0x27/0x2F) CS53L30_ADCx_CTL3 - ADC1/ADC2 Control 3 */
+#define CS53L30_ADCx_HPF_EN_SHIFT 3
+#define CS53L30_ADCx_HPF_EN_MASK (1 << CS53L30_ADCx_HPF_EN_SHIFT)
+#define CS53L30_ADCx_HPF_EN (1 << CS53L30_ADCx_HPF_EN_SHIFT)
+#define CS53L30_ADCx_HPF_CF_SHIFT 1
+#define CS53L30_ADCx_HPF_CF_WIDTH 2
+#define CS53L30_ADCx_HPF_CF_MASK (((1 << CS53L30_ADCx_HPF_CF_WIDTH) - 1) << CS53L30_ADCx_HPF_CF_SHIFT)
+#define CS53L30_ADCx_HPF_CF_1HZ86 (0 << CS53L30_ADCx_HPF_CF_SHIFT)
+#define CS53L30_ADCx_HPF_CF_120HZ (1 << CS53L30_ADCx_HPF_CF_SHIFT)
+#define CS53L30_ADCx_HPF_CF_235HZ (2 << CS53L30_ADCx_HPF_CF_SHIFT)
+#define CS53L30_ADCx_HPF_CF_466HZ (3 << CS53L30_ADCx_HPF_CF_SHIFT)
+#define CS53L30_ADCx_NG_ALL_SHIFT 0
+#define CS53L30_ADCx_NG_ALL_MASK (1 << CS53L30_ADCx_NG_ALL_SHIFT)
+#define CS53L30_ADCx_NG_ALL (1 << CS53L30_ADCx_NG_ALL_SHIFT)
+
+#define CS53L30_ADCx_CTL3_DEFAULT (CS53L30_ADCx_HPF_EN)
+
+/* R40/R48 (0x28/0x30) CS53L30_ADCx_NG_CTL - ADC1/ADC2 Noise Gate Control */
+#define CS53L30_ADCxB_NG_SHIFT 7
+#define CS53L30_ADCxB_NG_MASK (1 << CS53L30_ADCxB_NG_SHIFT)
+#define CS53L30_ADCxB_NG (1 << CS53L30_ADCxB_NG_SHIFT)
+#define CS53L30_ADCxA_NG_SHIFT 6
+#define CS53L30_ADCxA_NG_MASK (1 << CS53L30_ADCxA_NG_SHIFT)
+#define CS53L30_ADCxA_NG (1 << CS53L30_ADCxA_NG_SHIFT)
+#define CS53L30_ADCx_NG_BOOST_SHIFT 5
+#define CS53L30_ADCx_NG_BOOST_MASK (1 << CS53L30_ADCx_NG_BOOST_SHIFT)
+#define CS53L30_ADCx_NG_BOOST (1 << CS53L30_ADCx_NG_BOOST_SHIFT)
+#define CS53L30_ADCx_NG_THRESH_SHIFT 2
+#define CS53L30_ADCx_NG_THRESH_WIDTH 3
+#define CS53L30_ADCx_NG_THRESH_MASK (((1 << CS53L30_ADCx_NG_THRESH_WIDTH) - 1) << CS53L30_ADCx_NG_THRESH_SHIFT)
+#define CS53L30_ADCx_NG_DELAY_SHIFT 0
+#define CS53L30_ADCx_NG_DELAY_WIDTH 2
+#define CS53L30_ADCx_NG_DELAY_MASK (((1 << CS53L30_ADCx_NG_DELAY_WIDTH) - 1) << CS53L30_ADCx_NG_DELAY_SHIFT)
+
+#define CS53L30_ADCx_NG_CTL_DEFAULT (0)
+
+/* R41/R42/R49/R50 (0x29/0x2A/0x31/0x32) CS53L30_ADCxy_AFE_CTL - ADC1A/1B/2A/2B AFE Control */
+#define CS53L30_ADCxy_PREAMP_SHIFT 6
+#define CS53L30_ADCxy_PREAMP_WIDTH 2
+#define CS53L30_ADCxy_PREAMP_MASK (((1 << CS53L30_ADCxy_PREAMP_WIDTH) - 1) << CS53L30_ADCxy_PREAMP_SHIFT)
+#define CS53L30_ADCxy_PGA_VOL_SHIFT 0
+#define CS53L30_ADCxy_PGA_VOL_WIDTH 6
+#define CS53L30_ADCxy_PGA_VOL_MASK (((1 << CS53L30_ADCxy_PGA_VOL_WIDTH) - 1) << CS53L30_ADCxy_PGA_VOL_SHIFT)
+
+#define CS53L30_ADCxy_AFE_CTL_DEFAULT (0)
+
+/* R43/R44/R51/R52 (0x2B/0x2C/0x33/0x34) CS53L30_ADCxy_DIG_VOL - ADC1A/1B/2A/2B Digital Volume */
+#define CS53L30_ADCxy_VOL_MUTE (0x80)
+
+#define CS53L30_ADCxy_DIG_VOL_DEFAULT (0x0)
+
+/* CS53L30_INT */
+#define CS53L30_PDN_DONE (1 << 7)
+#define CS53L30_THMS_TRIP (1 << 6)
+#define CS53L30_SYNC_DONE (1 << 5)
+#define CS53L30_ADC2B_OVFL (1 << 4)
+#define CS53L30_ADC2A_OVFL (1 << 3)
+#define CS53L30_ADC1B_OVFL (1 << 2)
+#define CS53L30_ADC1A_OVFL (1 << 1)
+#define CS53L30_MUTE_PIN (1 << 0)
+#define CS53L30_DEVICE_INT_MASK 0xFF
+
+#endif /* __CS53L30_H__ */
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index d6f4abbbf8a7..fb3885fe0afb 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -226,6 +226,7 @@ static int v253_open(struct tty_struct *tty)
if (!tty->disc_data)
return -ENODEV;
+ tty->receive_room = 16;
if (tty->ops->write(tty, v253_init, len) != len) {
ret = -EIO;
goto err;
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index 9459593eef13..f0057cd223a4 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -13,8 +13,8 @@
#include <linux/module.h>
#include <linux/platform_device.h>
-#include <linux/of_device.h>
-#include <linux/of_irq.h>
+#include <linux/i2c.h>
+#include <linux/property.h>
#include <linux/pm_wakeirq.h>
#include <linux/slab.h>
#include <linux/delay.h>
@@ -382,11 +382,11 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
}
/*
- * DT to pdata conversion
+ * DT/ACPI to pdata conversion
*/
static enum da7219_aad_micbias_pulse_lvl
- da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val)
+ da7219_aad_fw_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val)
{
switch (val) {
case 2800:
@@ -400,7 +400,7 @@ static enum da7219_aad_micbias_pulse_lvl
}
static enum da7219_aad_btn_cfg
- da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val)
+ da7219_aad_fw_btn_cfg(struct snd_soc_codec *codec, u32 val)
{
switch (val) {
case 2:
@@ -424,7 +424,7 @@ static enum da7219_aad_btn_cfg
}
static enum da7219_aad_mic_det_thr
- da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val)
+ da7219_aad_fw_mic_det_thr(struct snd_soc_codec *codec, u32 val)
{
switch (val) {
case 200:
@@ -442,7 +442,7 @@ static enum da7219_aad_mic_det_thr
}
static enum da7219_aad_jack_ins_deb
- da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val)
+ da7219_aad_fw_jack_ins_deb(struct snd_soc_codec *codec, u32 val)
{
switch (val) {
case 5:
@@ -468,7 +468,7 @@ static enum da7219_aad_jack_ins_deb
}
static enum da7219_aad_jack_det_rate
- da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str)
+ da7219_aad_fw_jack_det_rate(struct snd_soc_codec *codec, const char *str)
{
if (!strcmp(str, "32ms_64ms")) {
return DA7219_AAD_JACK_DET_RATE_32_64MS;
@@ -485,7 +485,7 @@ static enum da7219_aad_jack_det_rate
}
static enum da7219_aad_jack_rem_deb
- da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val)
+ da7219_aad_fw_jack_rem_deb(struct snd_soc_codec *codec, u32 val)
{
switch (val) {
case 1:
@@ -503,7 +503,7 @@ static enum da7219_aad_jack_rem_deb
}
static enum da7219_aad_btn_avg
- da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val)
+ da7219_aad_fw_btn_avg(struct snd_soc_codec *codec, u32 val)
{
switch (val) {
case 1:
@@ -521,7 +521,7 @@ static enum da7219_aad_btn_avg
}
static enum da7219_aad_adc_1bit_rpt
- da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val)
+ da7219_aad_fw_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val)
{
switch (val) {
case 1:
@@ -538,97 +538,96 @@ static enum da7219_aad_adc_1bit_rpt
}
}
-static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec)
+static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct snd_soc_codec *codec)
{
- struct device_node *np = codec->dev->of_node;
- struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad");
+ struct device *dev = codec->dev;
+ struct i2c_client *i2c = to_i2c_client(dev);
+ struct fwnode_handle *aad_np;
struct da7219_aad_pdata *aad_pdata;
- const char *of_str;
- u32 of_val32;
+ const char *fw_str;
+ u32 fw_val32;
+ aad_np = device_get_named_child_node(dev, "da7219_aad");
if (!aad_np)
return NULL;
aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL);
if (!aad_pdata)
- goto out;
+ return NULL;
- aad_pdata->irq = irq_of_parse_and_map(np, 0);
+ aad_pdata->irq = i2c->irq;
- if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl",
- &of_val32) >= 0)
+ if (fwnode_property_read_u32(aad_np, "dlg,micbias-pulse-lvl",
+ &fw_val32) >= 0)
aad_pdata->micbias_pulse_lvl =
- da7219_aad_of_micbias_pulse_lvl(codec, of_val32);
+ da7219_aad_fw_micbias_pulse_lvl(codec, fw_val32);
else
aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF;
- if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time",
- &of_val32) >= 0)
- aad_pdata->micbias_pulse_time = of_val32;
+ if (fwnode_property_read_u32(aad_np, "dlg,micbias-pulse-time",
+ &fw_val32) >= 0)
+ aad_pdata->micbias_pulse_time = fw_val32;
- if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0)
- aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32);
+ if (fwnode_property_read_u32(aad_np, "dlg,btn-cfg", &fw_val32) >= 0)
+ aad_pdata->btn_cfg = da7219_aad_fw_btn_cfg(codec, fw_val32);
else
aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS;
- if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0)
+ if (fwnode_property_read_u32(aad_np, "dlg,mic-det-thr", &fw_val32) >= 0)
aad_pdata->mic_det_thr =
- da7219_aad_of_mic_det_thr(codec, of_val32);
+ da7219_aad_fw_mic_det_thr(codec, fw_val32);
else
aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS;
- if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0)
+ if (fwnode_property_read_u32(aad_np, "dlg,jack-ins-deb", &fw_val32) >= 0)
aad_pdata->jack_ins_deb =
- da7219_aad_of_jack_ins_deb(codec, of_val32);
+ da7219_aad_fw_jack_ins_deb(codec, fw_val32);
else
aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS;
- if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str))
+ if (!fwnode_property_read_string(aad_np, "dlg,jack-det-rate", &fw_str))
aad_pdata->jack_det_rate =
- da7219_aad_of_jack_det_rate(codec, of_str);
+ da7219_aad_fw_jack_det_rate(codec, fw_str);
else
aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS;
- if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0)
+ if (fwnode_property_read_u32(aad_np, "dlg,jack-rem-deb", &fw_val32) >= 0)
aad_pdata->jack_rem_deb =
- da7219_aad_of_jack_rem_deb(codec, of_val32);
+ da7219_aad_fw_jack_rem_deb(codec, fw_val32);
else
aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS;
- if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0)
- aad_pdata->a_d_btn_thr = (u8) of_val32;
+ if (fwnode_property_read_u32(aad_np, "dlg,a-d-btn-thr", &fw_val32) >= 0)
+ aad_pdata->a_d_btn_thr = (u8) fw_val32;
else
aad_pdata->a_d_btn_thr = 0xA;
- if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0)
- aad_pdata->d_b_btn_thr = (u8) of_val32;
+ if (fwnode_property_read_u32(aad_np, "dlg,d-b-btn-thr", &fw_val32) >= 0)
+ aad_pdata->d_b_btn_thr = (u8) fw_val32;
else
aad_pdata->d_b_btn_thr = 0x16;
- if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0)
- aad_pdata->b_c_btn_thr = (u8) of_val32;
+ if (fwnode_property_read_u32(aad_np, "dlg,b-c-btn-thr", &fw_val32) >= 0)
+ aad_pdata->b_c_btn_thr = (u8) fw_val32;
else
aad_pdata->b_c_btn_thr = 0x21;
- if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0)
- aad_pdata->c_mic_btn_thr = (u8) of_val32;
+ if (fwnode_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &fw_val32) >= 0)
+ aad_pdata->c_mic_btn_thr = (u8) fw_val32;
else
aad_pdata->c_mic_btn_thr = 0x3E;
- if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0)
- aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32);
+ if (fwnode_property_read_u32(aad_np, "dlg,btn-avg", &fw_val32) >= 0)
+ aad_pdata->btn_avg = da7219_aad_fw_btn_avg(codec, fw_val32);
else
aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2;
- if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0)
+ if (fwnode_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &fw_val32) >= 0)
aad_pdata->adc_1bit_rpt =
- da7219_aad_of_adc_1bit_rpt(codec, of_val32);
+ da7219_aad_fw_adc_1bit_rpt(codec, fw_val32);
else
aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1;
-out:
- of_node_put(aad_np);
-
return aad_pdata;
}
@@ -769,9 +768,9 @@ int da7219_aad_init(struct snd_soc_codec *codec)
da7219->aad = da7219_aad;
da7219_aad->codec = codec;
- /* Handle any DT/platform data */
- if ((codec->dev->of_node) && (da7219->pdata))
- da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec);
+ /* Handle any DT/ACPI/platform data */
+ if (da7219->pdata && !da7219->pdata->aad_pdata)
+ da7219->pdata->aad_pdata = da7219_aad_fw_to_pdata(codec);
da7219_aad_handle_pdata(codec);
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index 5c93899f1f0e..50ea94317cb3 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -15,6 +15,7 @@
#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/of_device.h>
+#include <linux/property.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/pm.h>
@@ -1418,7 +1419,7 @@ static struct snd_soc_dai_driver da7219_dai = {
/*
- * DT
+ * DT/ACPI
*/
static const struct of_device_id da7219_of_match[] = {
@@ -1434,7 +1435,7 @@ static const struct acpi_device_id da7219_acpi_match[] = {
MODULE_DEVICE_TABLE(acpi, da7219_acpi_match);
static enum da7219_micbias_voltage
- da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
+ da7219_fw_micbias_lvl(struct device *dev, u32 val)
{
switch (val) {
case 1600:
@@ -1450,13 +1451,13 @@ static enum da7219_micbias_voltage
case 2600:
return DA7219_MICBIAS_2_6V;
default:
- dev_warn(codec->dev, "Invalid micbias level");
+ dev_warn(dev, "Invalid micbias level");
return DA7219_MICBIAS_2_2V;
}
}
static enum da7219_mic_amp_in_sel
- da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str)
+ da7219_fw_mic_amp_in_sel(struct device *dev, const char *str)
{
if (!strcmp(str, "diff")) {
return DA7219_MIC_AMP_IN_SEL_DIFF;
@@ -1465,29 +1466,29 @@ static enum da7219_mic_amp_in_sel
} else if (!strcmp(str, "se_n")) {
return DA7219_MIC_AMP_IN_SEL_SE_N;
} else {
- dev_warn(codec->dev, "Invalid mic input type selection");
+ dev_warn(dev, "Invalid mic input type selection");
return DA7219_MIC_AMP_IN_SEL_DIFF;
}
}
-static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec)
+static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_codec *codec)
{
- struct device_node *np = codec->dev->of_node;
+ struct device *dev = codec->dev;
struct da7219_pdata *pdata;
const char *of_str;
u32 of_val32;
- pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL);
+ pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
if (!pdata)
return NULL;
- if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0)
- pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32);
+ if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0)
+ pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32);
else
pdata->micbias_lvl = DA7219_MICBIAS_2_2V;
- if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str))
- pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str);
+ if (!device_property_read_string(dev, "dlg,mic-amp-in-sel", &of_str))
+ pdata->mic_amp_in_sel = da7219_fw_mic_amp_in_sel(dev, of_str);
else
pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF;
@@ -1662,11 +1663,10 @@ static int da7219_probe(struct snd_soc_codec *codec)
break;
}
- /* Handle DT/Platform data */
- if (codec->dev->of_node)
- da7219->pdata = da7219_of_to_pdata(codec);
- else
- da7219->pdata = dev_get_platdata(codec->dev);
+ /* Handle DT/ACPI/Platform data */
+ da7219->pdata = dev_get_platdata(codec->dev);
+ if (!da7219->pdata)
+ da7219->pdata = da7219_fw_to_pdata(codec);
da7219_handle_pdata(codec);
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 181cd3bf0b92..4e181b270d95 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1124,8 +1124,10 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll)
}
hdac_hdmi_parse_eld(edev, pin);
- print_hex_dump_bytes("ELD: ", DUMP_PREFIX_OFFSET,
- pin->eld.eld_buffer, pin->eld.eld_size);
+ print_hex_dump_debug("ELD: ",
+ DUMP_PREFIX_OFFSET, 16, 1,
+ pin->eld.eld_buffer, pin->eld.eld_size,
+ true);
} else {
pin->eld.monitor_present = false;
pin->eld.eld_valid = false;
@@ -1474,6 +1476,11 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec)
* exit, we call pm_runtime_suspend() so that will do for us
*/
hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+ if (!hlink) {
+ dev_err(&edev->hdac.dev, "hdac link not found\n");
+ return -EIO;
+ }
+
snd_hdac_ext_bus_link_get(edev->ebus, hlink);
ret = create_fill_widget_route_map(dapm);
@@ -1634,6 +1641,11 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
/* hold the ref while we probe */
hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+ if (!hlink) {
+ dev_err(&edev->hdac.dev, "hdac link not found\n");
+ return -EIO;
+ }
+
snd_hdac_ext_bus_link_get(edev->ebus, hlink);
hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL);
@@ -1744,6 +1756,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
}
hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+ if (!hlink) {
+ dev_err(dev, "hdac link not found\n");
+ return -EIO;
+ }
+
snd_hdac_ext_bus_link_put(ebus, hlink);
return 0;
@@ -1765,6 +1782,11 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
return 0;
hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+ if (!hlink) {
+ dev_err(dev, "hdac link not found\n");
+ return -EIO;
+ }
+
snd_hdac_ext_bus_link_get(ebus, hlink);
err = snd_hdac_display_power(bus, true);
@@ -1796,6 +1818,7 @@ static const struct dev_pm_ops hdac_hdmi_pm = {
static const struct hda_device_id hdmi_list[] = {
HDA_CODEC_EXT_ENTRY(0x80862809, 0x100000, "Skylake HDMI", 0),
HDA_CODEC_EXT_ENTRY(0x8086280a, 0x100000, "Broxton HDMI", 0),
+ HDA_CODEC_EXT_ENTRY(0x8086280b, 0x100000, "Kabylake HDMI", 0),
{}
};
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 8e36e883e453..f27d115626db 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -112,7 +112,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
return ret;
if (hcp->hcd.ops->audio_startup) {
- ret = hcp->hcd.ops->audio_startup(dai->dev->parent);
+ ret = hcp->hcd.ops->audio_startup(dai->dev->parent, hcp->hcd.data);
if (ret) {
mutex_lock(&hcp->current_stream_lock);
hcp->current_stream = NULL;
@@ -122,8 +122,8 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
}
if (hcp->hcd.ops->get_eld) {
- ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld,
- sizeof(hcp->eld));
+ ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data,
+ hcp->eld, sizeof(hcp->eld));
if (!ret) {
ret = snd_pcm_hw_constraint_eld(substream->runtime,
@@ -144,7 +144,7 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
WARN_ON(hcp->current_stream != substream);
- hcp->hcd.ops->audio_shutdown(dai->dev->parent);
+ hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data);
mutex_lock(&hcp->current_stream_lock);
hcp->current_stream = NULL;
@@ -195,8 +195,8 @@ static int hdmi_codec_hw_params(struct snd_pcm_substream *substream,
hp.sample_rate = params_rate(params);
hp.channels = params_channels(params);
- return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id],
- &hp);
+ return hcp->hcd.ops->hw_params(dai->dev->parent, hcp->hcd.data,
+ &hcp->daifmt[dai->id], &hp);
}
static int hdmi_codec_set_fmt(struct snd_soc_dai *dai,
@@ -280,7 +280,8 @@ static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
dev_dbg(dai->dev, "%s()\n", __func__);
if (hcp->hcd.ops->digital_mute)
- return hcp->hcd.ops->digital_mute(dai->dev->parent, mute);
+ return hcp->hcd.ops->digital_mute(dai->dev->parent,
+ hcp->hcd.data, mute);
return 0;
}
diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c
new file mode 100644
index 000000000000..68074c92a7c0
--- /dev/null
+++ b/sound/soc/codecs/max9860.c
@@ -0,0 +1,753 @@
+/*
+ * Driver for the MAX9860 Mono Audio Voice Codec
+ *
+ * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf
+ *
+ * The driver does not support sidetone since the DVST register field is
+ * backwards with the mute near the maximum level instead of the minimum.
+ *
+ * Author: Peter Rosin <peda@axentia.s>
+ * Copyright 2016 Axentia Technologies
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/i2c.h>
+#include <linux/regulator/consumer.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#include "max9860.h"
+
+struct max9860_priv {
+ struct regmap *regmap;
+ struct regulator *dvddio;
+ struct notifier_block dvddio_nb;
+ u8 psclk;
+ unsigned long pclk_rate;
+ int fmt;
+};
+
+static int max9860_dvddio_event(struct notifier_block *nb,
+ unsigned long event, void *data)
+{
+ struct max9860_priv *max9860 = container_of(nb, struct max9860_priv,
+ dvddio_nb);
+ if (event & REGULATOR_EVENT_DISABLE) {
+ regcache_mark_dirty(max9860->regmap);
+ regcache_cache_only(max9860->regmap, true);
+ }
+
+ return 0;
+}
+
+static const struct reg_default max9860_reg_defaults[] = {
+ { MAX9860_PWRMAN, 0x00 },
+ { MAX9860_INTEN, 0x00 },
+ { MAX9860_SYSCLK, 0x00 },
+ { MAX9860_AUDIOCLKHIGH, 0x00 },
+ { MAX9860_AUDIOCLKLOW, 0x00 },
+ { MAX9860_IFC1A, 0x00 },
+ { MAX9860_IFC1B, 0x00 },
+ { MAX9860_VOICEFLTR, 0x00 },
+ { MAX9860_DACATTN, 0x00 },
+ { MAX9860_ADCLEVEL, 0x00 },
+ { MAX9860_DACGAIN, 0x00 },
+ { MAX9860_MICGAIN, 0x00 },
+ { MAX9860_MICADC, 0x00 },
+ { MAX9860_NOISEGATE, 0x00 },
+};
+
+static bool max9860_readable(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX9860_INTRSTATUS ... MAX9860_MICGAIN:
+ case MAX9860_MICADC ... MAX9860_PWRMAN:
+ case MAX9860_REVISION:
+ return true;
+ }
+
+ return false;
+}
+
+static bool max9860_writeable(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX9860_INTEN ... MAX9860_MICGAIN:
+ case MAX9860_MICADC ... MAX9860_PWRMAN:
+ return true;
+ }
+
+ return false;
+}
+
+static bool max9860_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX9860_INTRSTATUS:
+ case MAX9860_MICREADBACK:
+ return true;
+ }
+
+ return false;
+}
+
+static bool max9860_precious(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX9860_INTRSTATUS:
+ return true;
+ }
+
+ return false;
+}
+
+static const struct regmap_config max9860_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .readable_reg = max9860_readable,
+ .writeable_reg = max9860_writeable,
+ .volatile_reg = max9860_volatile,
+ .precious_reg = max9860_precious,
+
+ .max_register = MAX9860_MAX_REGISTER,
+ .reg_defaults = max9860_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(max9860_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_RANGE(pam_tlv,
+ 0, MAX9860_PAM_MAX - 1, TLV_DB_SCALE_ITEM(-2000, 2000, 1),
+ MAX9860_PAM_MAX, MAX9860_PAM_MAX, TLV_DB_SCALE_ITEM(3000, 0, 0));
+static const DECLARE_TLV_DB_SCALE(pgam_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(anth_tlv, -7600, 400, 1);
+static const DECLARE_TLV_DB_SCALE(agcth_tlv, -1800, 100, 0);
+
+static const char * const agchld_text[] = {
+ "AGC Disabled", "50ms", "100ms", "400ms"
+};
+
+static SOC_ENUM_SINGLE_DECL(agchld_enum, MAX9860_MICADC,
+ MAX9860_AGCHLD_SHIFT, agchld_text);
+
+static const char * const agcsrc_text[] = {
+ "Left ADC", "Left/Right ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(agcsrc_enum, MAX9860_MICADC,
+ MAX9860_AGCSRC_SHIFT, agcsrc_text);
+
+static const char * const agcatk_text[] = {
+ "3ms", "12ms", "50ms", "200ms"
+};
+
+static SOC_ENUM_SINGLE_DECL(agcatk_enum, MAX9860_MICADC,
+ MAX9860_AGCATK_SHIFT, agcatk_text);
+
+static const char * const agcrls_text[] = {
+ "78ms", "156ms", "312ms", "625ms",
+ "1.25s", "2.5s", "5s", "10s"
+};
+
+static SOC_ENUM_SINGLE_DECL(agcrls_enum, MAX9860_MICADC,
+ MAX9860_AGCRLS_SHIFT, agcrls_text);
+
+static const char * const filter_text[] = {
+ "Disabled",
+ "Elliptical HP 217Hz notch (16kHz)",
+ "Butterworth HP 500Hz (16kHz)",
+ "Elliptical HP 217Hz notch (8kHz)",
+ "Butterworth HP 500Hz (8kHz)",
+ "Butterworth HP 200Hz (48kHz)"
+};
+
+static SOC_ENUM_SINGLE_DECL(avflt_enum, MAX9860_VOICEFLTR,
+ MAX9860_AVFLT_SHIFT, filter_text);
+
+static SOC_ENUM_SINGLE_DECL(dvflt_enum, MAX9860_VOICEFLTR,
+ MAX9860_DVFLT_SHIFT, filter_text);
+
+static const struct snd_kcontrol_new max9860_controls[] = {
+SOC_SINGLE_TLV("Master Playback Volume", MAX9860_DACATTN,
+ MAX9860_DVA_SHIFT, MAX9860_DVA_MUTE, 1, dva_tlv),
+SOC_SINGLE_TLV("DAC Gain Volume", MAX9860_DACGAIN,
+ MAX9860_DVG_SHIFT, MAX9860_DVG_MAX, 0, dvg_tlv),
+SOC_DOUBLE_TLV("Line Capture Volume", MAX9860_ADCLEVEL,
+ MAX9860_ADCLL_SHIFT, MAX9860_ADCRL_SHIFT, MAX9860_ADCxL_MIN, 1,
+ adc_tlv),
+
+SOC_ENUM("AGC Hold Time", agchld_enum),
+SOC_ENUM("AGC/Noise Gate Source", agcsrc_enum),
+SOC_ENUM("AGC Attack Time", agcatk_enum),
+SOC_ENUM("AGC Release Time", agcrls_enum),
+
+SOC_SINGLE_TLV("Noise Gate Threshold Volume", MAX9860_NOISEGATE,
+ MAX9860_ANTH_SHIFT, MAX9860_ANTH_MAX, 0, anth_tlv),
+SOC_SINGLE_TLV("AGC Signal Threshold Volume", MAX9860_NOISEGATE,
+ MAX9860_AGCTH_SHIFT, MAX9860_AGCTH_MIN, 1, agcth_tlv),
+
+SOC_SINGLE_TLV("Mic PGA Volume", MAX9860_MICGAIN,
+ MAX9860_PGAM_SHIFT, MAX9860_PGAM_MIN, 1, pgam_tlv),
+SOC_SINGLE_TLV("Mic Preamp Volume", MAX9860_MICGAIN,
+ MAX9860_PAM_SHIFT, MAX9860_PAM_MAX, 0, pam_tlv),
+
+SOC_ENUM("ADC Filter", avflt_enum),
+SOC_ENUM("DAC Filter", dvflt_enum),
+};
+
+static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("MICL"),
+SND_SOC_DAPM_INPUT("MICR"),
+
+SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0),
+SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0),
+
+SND_SOC_DAPM_OUTPUT("OUT"),
+
+SND_SOC_DAPM_SUPPLY("Supply", SND_SOC_NOPM, 0, 0,
+ NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DVDD", 0, 0),
+SND_SOC_DAPM_CLOCK_SUPPLY("mclk"),
+};
+
+static const struct snd_soc_dapm_route max9860_dapm_routes[] = {
+ { "ADCL", NULL, "MICL" },
+ { "ADCR", NULL, "MICR" },
+ { "AIFOUTL", NULL, "ADCL" },
+ { "AIFOUTR", NULL, "ADCR" },
+
+ { "DAC", NULL, "AIFINL" },
+ { "DAC", NULL, "AIFINR" },
+ { "OUT", NULL, "DAC" },
+
+ { "Supply", NULL, "AVDD" },
+ { "Supply", NULL, "DVDD" },
+ { "Supply", NULL, "mclk" },
+
+ { "DAC", NULL, "Supply" },
+ { "ADCL", NULL, "Supply" },
+ { "ADCR", NULL, "Supply" },
+};
+
+static int max9860_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec);
+ u8 master;
+ u8 ifc1a = 0;
+ u8 ifc1b = 0;
+ u8 sysclk = 0;
+ unsigned long n;
+ int ret;
+
+ dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n",
+ params_rate(params),
+ params_channels(params));
+
+ if (params_channels(params) == 2)
+ ifc1b |= MAX9860_ST;
+
+ switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ master = 0;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ master = MAX9860_MASTER;
+ break;
+ default:
+ return -EINVAL;
+ }
+ ifc1a |= master;
+
+ if (master) {
+ if (params_width(params) * params_channels(params) > 48)
+ ifc1b |= MAX9860_BSEL_64X;
+ else
+ ifc1b |= MAX9860_BSEL_48X;
+ }
+
+ switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ifc1a |= MAX9860_DDLY;
+ ifc1b |= MAX9860_ADLY;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ifc1a |= MAX9860_WCI;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ if (params_width(params) != 16) {
+ dev_err(codec->dev,
+ "DSP_A works for 16 bits per sample only.\n");
+ return -EINVAL;
+ }
+ ifc1a |= MAX9860_DDLY | MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM;
+ ifc1b |= MAX9860_ADLY;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ if (params_width(params) != 16) {
+ dev_err(codec->dev,
+ "DSP_B works for 16 bits per sample only.\n");
+ return -EINVAL;
+ }
+ ifc1a |= MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (max9860->fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ return -EINVAL;
+ }
+ ifc1a ^= MAX9860_WCI;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ return -EINVAL;
+ }
+ ifc1a ^= MAX9860_WCI;
+ /* fall through */
+ case SND_SOC_DAIFMT_IB_NF:
+ ifc1a ^= MAX9860_DBCI;
+ ifc1b ^= MAX9860_ABCI;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ dev_dbg(codec->dev, "IFC1A %02x\n", ifc1a);
+ ret = regmap_write(max9860->regmap, MAX9860_IFC1A, ifc1a);
+ if (ret) {
+ dev_err(codec->dev, "Failed to set IFC1A: %d\n", ret);
+ return ret;
+ }
+ dev_dbg(codec->dev, "IFC1B %02x\n", ifc1b);
+ ret = regmap_write(max9860->regmap, MAX9860_IFC1B, ifc1b);
+ if (ret) {
+ dev_err(codec->dev, "Failed to set IFC1B: %d\n", ret);
+ return ret;
+ }
+
+ /*
+ * Check if Integer Clock Mode is possible, but avoid it in slave mode
+ * since we then do not know if lrclk is derived from pclk and the
+ * datasheet mentions that the frequencies have to match exactly in
+ * order for this to work.
+ */
+ if (params_rate(params) == 8000 || params_rate(params) == 16000) {
+ if (master) {
+ switch (max9860->pclk_rate) {
+ case 12000000:
+ sysclk = MAX9860_FREQ_12MHZ;
+ break;
+ case 13000000:
+ sysclk = MAX9860_FREQ_13MHZ;
+ break;
+ case 19200000:
+ sysclk = MAX9860_FREQ_19_2MHZ;
+ break;
+ default:
+ /*
+ * Integer Clock Mode not possible. Leave
+ * sysclk at zero and fall through to the
+ * code below for PLL mode.
+ */
+ break;
+ }
+
+ if (sysclk && params_rate(params) == 16000)
+ sysclk |= MAX9860_16KHZ;
+ }
+ }
+
+ /*
+ * Largest possible n:
+ * 65536 * 96 * 48kHz / 10MHz -> 30199
+ * Smallest possible n:
+ * 65536 * 96 * 8kHz / 20MHz -> 2517
+ * Both fit nicely in the available 15 bits, no need to apply any mask.
+ */
+ n = DIV_ROUND_CLOSEST_ULL(65536ULL * 96 * params_rate(params),
+ max9860->pclk_rate);
+
+ if (!sysclk) {
+ /* PLL mode */
+ if (params_rate(params) > 24000)
+ sysclk |= MAX9860_16KHZ;
+
+ if (!master)
+ n |= 1; /* trigger rapid pll lock mode */
+ }
+
+ sysclk |= max9860->psclk;
+ dev_dbg(codec->dev, "SYSCLK %02x\n", sysclk);
+ ret = regmap_write(max9860->regmap,
+ MAX9860_SYSCLK, sysclk);
+ if (ret) {
+ dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+ dev_dbg(codec->dev, "N %lu\n", n);
+ ret = regmap_write(max9860->regmap,
+ MAX9860_AUDIOCLKHIGH, n >> 8);
+ if (ret) {
+ dev_err(codec->dev, "Failed to set NHI: %d\n", ret);
+ return ret;
+ }
+ ret = regmap_write(max9860->regmap,
+ MAX9860_AUDIOCLKLOW, n & 0xff);
+ if (ret) {
+ dev_err(codec->dev, "Failed to set NLO: %d\n", ret);
+ return ret;
+ }
+
+ if (!master) {
+ dev_dbg(codec->dev, "Enable PLL\n");
+ ret = regmap_update_bits(max9860->regmap, MAX9860_AUDIOCLKHIGH,
+ MAX9860_PLL, MAX9860_PLL);
+ if (ret) {
+ dev_err(codec->dev, "Failed to enable PLL: %d\n", ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ case SND_SOC_DAIFMT_CBS_CFS:
+ max9860->fmt = fmt;
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static const struct snd_soc_dai_ops max9860_dai_ops = {
+ .hw_params = max9860_hw_params,
+ .set_fmt = max9860_set_fmt,
+};
+
+static struct snd_soc_dai_driver max9860_dai = {
+ .name = "max9860-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &max9860_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int max9860_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct max9860_priv *max9860 = dev_get_drvdata(codec->dev);
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN,
+ MAX9860_SHDN, MAX9860_SHDN);
+ if (ret) {
+ dev_err(codec->dev, "Failed to remove SHDN: %d\n", ret);
+ return ret;
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN,
+ MAX9860_SHDN, 0);
+ if (ret) {
+ dev_err(codec->dev, "Failed to request SHDN: %d\n",
+ ret);
+ return ret;
+ }
+ break;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver max9860_codec_driver = {
+ .set_bias_level = max9860_set_bias_level,
+ .idle_bias_off = true,
+
+ .controls = max9860_controls,
+ .num_controls = ARRAY_SIZE(max9860_controls),
+ .dapm_widgets = max9860_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(max9860_dapm_widgets),
+ .dapm_routes = max9860_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes),
+};
+
+#ifdef CONFIG_PM
+static int max9860_suspend(struct device *dev)
+{
+ struct max9860_priv *max9860 = dev_get_drvdata(dev);
+ int ret;
+
+ ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK,
+ MAX9860_PSCLK, MAX9860_PSCLK_OFF);
+ if (ret) {
+ dev_err(dev, "Failed to disable clock: %d\n", ret);
+ return ret;
+ }
+
+ regulator_disable(max9860->dvddio);
+
+ return 0;
+}
+
+static int max9860_resume(struct device *dev)
+{
+ struct max9860_priv *max9860 = dev_get_drvdata(dev);
+ int ret;
+
+ ret = regulator_enable(max9860->dvddio);
+ if (ret) {
+ dev_err(dev, "Failed to enable DVDDIO: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(max9860->regmap, false);
+ ret = regcache_sync(max9860->regmap);
+ if (ret) {
+ dev_err(dev, "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK,
+ MAX9860_PSCLK, max9860->psclk);
+ if (ret) {
+ dev_err(dev, "Failed to enable clock: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops max9860_pm_ops = {
+ SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL)
+};
+
+static int max9860_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct max9860_priv *max9860;
+ int ret;
+ struct clk *mclk;
+ unsigned long mclk_rate;
+ int i;
+ int intr;
+
+ max9860 = devm_kzalloc(dev, sizeof(struct max9860_priv), GFP_KERNEL);
+ if (!max9860)
+ return -ENOMEM;
+
+ max9860->dvddio = devm_regulator_get(dev, "DVDDIO");
+ if (IS_ERR(max9860->dvddio)) {
+ ret = PTR_ERR(max9860->dvddio);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "Failed to get DVDDIO supply: %d\n", ret);
+ return ret;
+ }
+
+ max9860->dvddio_nb.notifier_call = max9860_dvddio_event;
+
+ ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb);
+ if (ret)
+ dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret);
+
+ ret = regulator_enable(max9860->dvddio);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable DVDDIO: %d\n", ret);
+ return ret;
+ }
+
+ max9860->regmap = devm_regmap_init_i2c(i2c, &max9860_regmap);
+ if (IS_ERR(max9860->regmap)) {
+ ret = PTR_ERR(max9860->regmap);
+ goto err_regulator;
+ }
+
+ dev_set_drvdata(dev, max9860);
+
+ /*
+ * mclk has to be in the 10MHz to 60MHz range.
+ * psclk is used to scale mclk into pclk so that
+ * pclk is in the 10MHz to 20MHz range.
+ */
+ mclk = clk_get(dev, "mclk");
+
+ if (IS_ERR(mclk)) {
+ ret = PTR_ERR(mclk);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "Failed to get MCLK: %d\n", ret);
+ goto err_regulator;
+ }
+
+ mclk_rate = clk_get_rate(mclk);
+ clk_put(mclk);
+
+ if (mclk_rate > 60000000 || mclk_rate < 10000000) {
+ dev_err(dev, "Bad mclk %luHz (needs 10MHz - 60MHz)\n",
+ mclk_rate);
+ ret = -EINVAL;
+ goto err_regulator;
+ }
+ if (mclk_rate >= 40000000)
+ max9860->psclk = 3;
+ else if (mclk_rate >= 20000000)
+ max9860->psclk = 2;
+ else
+ max9860->psclk = 1;
+ max9860->pclk_rate = mclk_rate >> (max9860->psclk - 1);
+ max9860->psclk <<= MAX9860_PSCLK_SHIFT;
+ dev_dbg(dev, "mclk %lu pclk %lu\n", mclk_rate, max9860->pclk_rate);
+
+ regcache_cache_bypass(max9860->regmap, true);
+ for (i = 0; i < max9860_regmap.num_reg_defaults; ++i) {
+ ret = regmap_write(max9860->regmap,
+ max9860_regmap.reg_defaults[i].reg,
+ max9860_regmap.reg_defaults[i].def);
+ if (ret) {
+ dev_err(dev, "Failed to initialize register %u: %d\n",
+ max9860_regmap.reg_defaults[i].reg, ret);
+ goto err_regulator;
+ }
+ }
+ regcache_cache_bypass(max9860->regmap, false);
+
+ ret = regmap_read(max9860->regmap, MAX9860_INTRSTATUS, &intr);
+ if (ret) {
+ dev_err(dev, "Failed to clear INTRSTATUS: %d\n", ret);
+ goto err_regulator;
+ }
+
+ pm_runtime_set_active(dev);
+ pm_runtime_enable(dev);
+ pm_runtime_idle(dev);
+
+ ret = snd_soc_register_codec(dev, &max9860_codec_driver,
+ &max9860_dai, 1);
+ if (ret) {
+ dev_err(dev, "Failed to register CODEC: %d\n", ret);
+ goto err_pm;
+ }
+
+ return 0;
+
+err_pm:
+ pm_runtime_disable(dev);
+err_regulator:
+ regulator_disable(max9860->dvddio);
+ return ret;
+}
+
+static int max9860_remove(struct i2c_client *i2c)
+{
+ struct device *dev = &i2c->dev;
+ struct max9860_priv *max9860 = dev_get_drvdata(dev);
+
+ snd_soc_unregister_codec(dev);
+ pm_runtime_disable(dev);
+ regulator_disable(max9860->dvddio);
+ return 0;
+}
+
+static const struct i2c_device_id max9860_i2c_id[] = {
+ { "max9860", },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, max9860_i2c_id);
+
+static const struct of_device_id max9860_of_match[] = {
+ { .compatible = "maxim,max9860", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, max9860_of_match);
+
+static struct i2c_driver max9860_i2c_driver = {
+ .probe = max9860_probe,
+ .remove = max9860_remove,
+ .id_table = max9860_i2c_id,
+ .driver = {
+ .name = "max9860",
+ .of_match_table = max9860_of_match,
+ .pm = &max9860_pm_ops,
+ },
+};
+
+module_i2c_driver(max9860_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC MAX9860 Mono Audio Voice Codec driver");
+MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/max9860.h b/sound/soc/codecs/max9860.h
new file mode 100644
index 000000000000..22041bd67a7d
--- /dev/null
+++ b/sound/soc/codecs/max9860.h
@@ -0,0 +1,162 @@
+/*
+ * Driver for the MAX9860 Mono Audio Voice Codec
+ *
+ * Author: Peter Rosin <peda@axentia.s>
+ * Copyright 2016 Axentia Technologies
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef _SND_SOC_MAX9860
+#define _SND_SOC_MAX9860
+
+#define MAX9860_INTRSTATUS 0x00
+#define MAX9860_MICREADBACK 0x01
+#define MAX9860_INTEN 0x02
+#define MAX9860_SYSCLK 0x03
+#define MAX9860_AUDIOCLKHIGH 0x04
+#define MAX9860_AUDIOCLKLOW 0x05
+#define MAX9860_IFC1A 0x06
+#define MAX9860_IFC1B 0x07
+#define MAX9860_VOICEFLTR 0x08
+#define MAX9860_DACATTN 0x09
+#define MAX9860_ADCLEVEL 0x0a
+#define MAX9860_DACGAIN 0x0b
+#define MAX9860_MICGAIN 0x0c
+#define MAX9860_RESERVED 0x0d
+#define MAX9860_MICADC 0x0e
+#define MAX9860_NOISEGATE 0x0f
+#define MAX9860_PWRMAN 0x10
+#define MAX9860_REVISION 0xff
+
+#define MAX9860_MAX_REGISTER 0xff
+
+/* INTRSTATUS */
+#define MAX9860_CLD 0x80
+#define MAX9860_SLD 0x40
+#define MAX9860_ULK 0x20
+
+/* MICREADBACK */
+#define MAX9860_NG 0xe0
+#define MAX9860_AGC 0x1f
+
+/* INTEN */
+#define MAX9860_ICLD 0x80
+#define MAX9860_ISLD 0x40
+#define MAX9860_IULK 0x20
+
+/* SYSCLK */
+#define MAX9860_PSCLK 0x30
+#define MAX9860_PSCLK_OFF 0x00
+#define MAX9860_PSCLK_SHIFT 4
+#define MAX9860_FREQ 0x06
+#define MAX9860_FREQ_NORMAL 0x00
+#define MAX9860_FREQ_12MHZ 0x02
+#define MAX9860_FREQ_13MHZ 0x04
+#define MAX9860_FREQ_19_2MHZ 0x06
+#define MAX9860_16KHZ 0x01
+
+/* AUDIOCLKHIGH */
+#define MAX9860_PLL 0x80
+#define MAX9860_NHI 0x7f
+
+/* AUDIOCLKLOW */
+#define MAX9860_NLO 0xff
+
+/* IFC1A */
+#define MAX9860_MASTER 0x80
+#define MAX9860_WCI 0x40
+#define MAX9860_DBCI 0x20
+#define MAX9860_DDLY 0x10
+#define MAX9860_HIZ 0x08
+#define MAX9860_TDM 0x04
+
+/* IFC1B */
+#define MAX9860_ABCI 0x20
+#define MAX9860_ADLY 0x10
+#define MAX9860_ST 0x08
+#define MAX9860_BSEL 0x07
+#define MAX9860_BSEL_OFF 0x00
+#define MAX9860_BSEL_64X 0x01
+#define MAX9860_BSEL_48X 0x02
+#define MAX9860_BSEL_PCLK_2 0x04
+#define MAX9860_BSEL_PCLK_4 0x05
+#define MAX9860_BSEL_PCLK_8 0x06
+#define MAX9860_BSEL_PCLK_16 0x07
+
+/* VOICEFLTR */
+#define MAX9860_AVFLT 0xf0
+#define MAX9860_AVFLT_SHIFT 4
+#define MAX9860_AVFLT_COUNT 6
+#define MAX9860_DVFLT 0x0f
+#define MAX9860_DVFLT_SHIFT 0
+#define MAX9860_DVFLT_COUNT 6
+
+/* DACATTN */
+#define MAX9860_DVA 0xfe
+#define MAX9860_DVA_SHIFT 1
+#define MAX9860_DVA_MUTE 0x5e
+
+/* ADCLEVEL */
+#define MAX9860_ADCRL 0xf0
+#define MAX9860_ADCRL_SHIFT 4
+#define MAX9860_ADCLL 0x0f
+#define MAX9860_ADCLL_SHIFT 0
+#define MAX9860_ADCxL_MIN 15
+
+/* DACGAIN */
+#define MAX9860_DVG 0x60
+#define MAX9860_DVG_SHIFT 5
+#define MAX9860_DVG_MAX 3
+#define MAX9860_DVST 0x1f
+#define MAX9860_DVST_SHIFT 0
+#define MAX9860_DVST_MIN 31
+
+/* MICGAIN */
+#define MAX9860_PAM 0x60
+#define MAX9860_PAM_SHIFT 5
+#define MAX9860_PAM_MAX 3
+#define MAX9860_PGAM 0x1f
+#define MAX9860_PGAM_SHIFT 0
+#define MAX9860_PGAM_MIN 20
+
+/* MICADC */
+#define MAX9860_AGCSRC 0x80
+#define MAX9860_AGCSRC_SHIFT 7
+#define MAX9860_AGCSRC_COUNT 2
+#define MAX9860_AGCRLS 0x70
+#define MAX9860_AGCRLS_SHIFT 4
+#define MAX9860_AGCRLS_COUNT 8
+#define MAX9860_AGCATK 0x0c
+#define MAX9860_AGCATK_SHIFT 2
+#define MAX9860_AGCATK_COUNT 4
+#define MAX9860_AGCHLD 0x03
+#define MAX9860_AGCHLD_OFF 0x00
+#define MAX9860_AGCHLD_SHIFT 0
+#define MAX9860_AGCHLD_COUNT 4
+
+/* NOISEGATE */
+#define MAX9860_ANTH 0xf0
+#define MAX9860_ANTH_SHIFT 4
+#define MAX9860_ANTH_MAX 15
+#define MAX9860_AGCTH 0x0f
+#define MAX9860_AGCTH_SHIFT 0
+#define MAX9860_AGCTH_MIN 15
+
+/* PWRMAN */
+#define MAX9860_SHDN 0x80
+#define MAX9860_DACEN 0x08
+#define MAX9860_DACEN_SHIFT 3
+#define MAX9860_ADCLEN 0x02
+#define MAX9860_ADCLEN_SHIFT 1
+#define MAX9860_ADCREN 0x01
+#define MAX9860_ADCREN_SHIFT 0
+
+#endif /* _SND_SOC_MAX9860 */
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 58325234285c..33e1fc2d1598 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -73,7 +73,7 @@ static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg)
return !((reg == 0x00) || (reg == 0x0f));
}
-static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg)
+static bool pcm1681_writeable_reg(struct device *dev, unsigned int reg)
{
return pcm1681_accessible_reg(dev, reg) &&
(reg != PCM1681_ZERO_DETECT_STATUS);
diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c
index 06a66579ca6d..88fbdd184aa0 100644
--- a/sound/soc/codecs/pcm179x.c
+++ b/sound/soc/codecs/pcm179x.c
@@ -59,7 +59,7 @@ static bool pcm179x_accessible_reg(struct device *dev, unsigned int reg)
return reg >= 0x10 && reg <= 0x17;
}
-static bool pcm179x_writeable_reg(struct device *dev, unsigned register reg)
+static bool pcm179x_writeable_reg(struct device *dev, unsigned int reg)
{
bool accessible;
diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c
index ed515677409b..8ba322a00363 100644
--- a/sound/soc/codecs/pcm5102a.c
+++ b/sound/soc/codecs/pcm5102a.c
@@ -57,7 +57,6 @@ static struct platform_driver pcm5102a_codec_driver = {
.remove = pcm5102a_remove,
.driver = {
.name = "pcm5102a-codec",
- .owner = THIS_MODULE,
.of_match_table = pcm5102a_of_match,
},
};
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 1bd31644a782..74c0e4eb3788 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -1100,6 +1100,13 @@ static const struct dmi_system_id force_combo_jack_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Skylake Client platform")
}
},
+ {
+ .ident = "Intel Kabylake RVP",
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform")
+ }
+ },
+
{ }
};
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 3c6594da6c9c..490bfe661346 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -63,6 +63,7 @@ static const struct reg_sequence init_list[] = {
{RT5645_PR_BASE + 0x20, 0x611f},
{RT5645_PR_BASE + 0x21, 0x4040},
{RT5645_PR_BASE + 0x23, 0x0004},
+ {RT5645_ASRC_4, 0x0120},
};
static const struct reg_sequence rt5650_init_list[] = {
@@ -157,7 +158,7 @@ static const struct reg_default rt5645_reg[] = {
{ 0x83, 0x0000 },
{ 0x84, 0x0000 },
{ 0x85, 0x0000 },
- { 0x8a, 0x0000 },
+ { 0x8a, 0x0120 },
{ 0x8e, 0x0004 },
{ 0x8f, 0x1100 },
{ 0x90, 0x0646 },
@@ -253,7 +254,7 @@ static const struct reg_default rt5650_reg[] = {
{ 0x2b, 0x5454 },
{ 0x2c, 0xaaa0 },
{ 0x2d, 0x0000 },
- { 0x2f, 0x1002 },
+ { 0x2f, 0x5002 },
{ 0x31, 0x5000 },
{ 0x32, 0x0000 },
{ 0x33, 0x0000 },
@@ -314,7 +315,7 @@ static const struct reg_default rt5650_reg[] = {
{ 0x83, 0x0000 },
{ 0x84, 0x0000 },
{ 0x85, 0x0000 },
- { 0x8a, 0x0000 },
+ { 0x8a, 0x0120 },
{ 0x8e, 0x0004 },
{ 0x8f, 0x1100 },
{ 0x90, 0x0646 },
@@ -440,6 +441,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg)
switch (reg) {
case RT5645_RESET:
+ case RT5645_PRIV_INDEX:
case RT5645_PRIV_DATA:
case RT5645_IN1_CTRL1:
case RT5645_IN1_CTRL2:
@@ -740,6 +742,14 @@ static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol,
return ret;
}
+static const char * const rt5645_dac1_vol_ctrl_mode_text[] = {
+ "immediately", "zero crossing", "soft ramp"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5645_dac1_vol_ctrl_mode, RT5645_PR_BASE,
+ RT5645_DA1_ZDET_SFT, rt5645_dac1_vol_ctrl_mode_text);
+
static const struct snd_kcontrol_new rt5645_snd_controls[] = {
/* Speaker Output Volume */
SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL,
@@ -806,6 +816,9 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT,
1, 1),
RT5645_HWEQ("Speaker HWEQ"),
+
+ /* Digital Soft Volume Control */
+ SOC_ENUM("DAC1 Digital Volume Control Func", rt5645_dac1_vol_ctrl_mode),
};
/**
@@ -3531,6 +3544,7 @@ MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id);
static const struct acpi_device_id rt5645_acpi_match[] = {
{ "10EC5645", 0 },
{ "10EC5650", 0 },
+ { "10EC5640", 0 },
{},
};
MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match);
@@ -3561,6 +3575,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"),
},
},
+ {
+ .ident = "Microsoft Surface 3",
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Surface 3"),
+ },
+ },
{ }
};
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 205e0715c99a..cfc5f97549eb 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -2018,6 +2018,9 @@
/* Codec Private Register definition */
+/* DAC ADC Digital Volume (0x00) */
+#define RT5645_DA1_ZDET_SFT 6
+
/* 3D Speaker Control (0x63) */
#define RT5645_3D_SPK_MASK (0x1 << 15)
#define RT5645_3D_SPK_SFT 15
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 49a9e7049e2b..0af5ddbef1da 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -619,7 +619,7 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = {
RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL,
RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
- 39, 0, out_vol_tlv),
+ 39, 1, out_vol_tlv),
/* OUTPUT Control */
SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1,
RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1),
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index da60e3fe5ee7..846deed6af41 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1713,6 +1713,7 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
{ "MICSUPP", NULL, "SYSCLK" },
+ { "DRC1 Signal Activity", NULL, "SYSCLK" },
{ "DRC1 Signal Activity", NULL, "DRC1L" },
{ "DRC1 Signal Activity", NULL, "DRC1R" },
};
@@ -1872,7 +1873,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = {
.capture = {
.stream_name = "Audio Trace CPU",
.channels_min = 1,
- .channels_max = 6,
+ .channels_max = 4,
.rates = WM5102_RATES,
.formats = WM5102_FORMATS,
},
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index b5820e4d5471..156547026a40 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1104,6 +1104,11 @@ SND_SOC_DAPM_INPUT("IN4R"),
SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
+SND_SOC_DAPM_OUTPUT("DSP Voice Trigger"),
+
+SND_SOC_DAPM_SWITCH("DSP3 Voice Trigger", SND_SOC_NOPM, 2, 0,
+ &arizona_voice_trigger_switch[2]),
+
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
@@ -1723,6 +1728,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
{ "OUT2L", NULL, "SYSCLK" },
{ "OUT2R", NULL, "SYSCLK" },
{ "OUT3L", NULL, "SYSCLK" },
+ { "OUT3R", NULL, "SYSCLK" },
{ "OUT4L", NULL, "SYSCLK" },
{ "OUT4R", NULL, "SYSCLK" },
{ "OUT5L", NULL, "SYSCLK" },
@@ -1997,10 +2003,16 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
{ "MICSUPP", NULL, "SYSCLK" },
+ { "DRC1 Signal Activity", NULL, "SYSCLK" },
+ { "DRC2 Signal Activity", NULL, "SYSCLK" },
{ "DRC1 Signal Activity", NULL, "DRC1L" },
{ "DRC1 Signal Activity", NULL, "DRC1R" },
{ "DRC2 Signal Activity", NULL, "DRC2L" },
{ "DRC2 Signal Activity", NULL, "DRC2R" },
+
+ { "DSP Voice Trigger", NULL, "SYSCLK" },
+ { "DSP Voice Trigger", NULL, "DSP3 Voice Trigger" },
+ { "DSP3 Voice Trigger", "Switch", "DSP3" },
};
static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
@@ -2222,6 +2234,7 @@ static irqreturn_t wm5110_adsp2_irq(int irq, void *data)
{
struct wm5110_priv *priv = data;
struct arizona *arizona = priv->core.arizona;
+ struct arizona_voice_trigger_info info;
int serviced = 0;
int i, ret;
@@ -2229,6 +2242,12 @@ static irqreturn_t wm5110_adsp2_irq(int irq, void *data)
ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
if (ret != -ENODEV)
serviced++;
+ if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) {
+ info.core = i;
+ arizona_call_notifiers(arizona,
+ ARIZONA_NOTIFY_VOICE_TRIGGER,
+ &info);
+ }
}
if (!serviced) {
@@ -2251,6 +2270,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
arizona_init_spk(codec);
arizona_init_gpio(codec);
arizona_init_mono(codec);
+ arizona_init_notifiers(codec);
ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1,
"ADSP2 Compressed IRQ", wm5110_adsp2_irq,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index f6f9395ea38e..1c600819f768 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -743,6 +743,7 @@ static const struct regmap_config wm8940_regmap = {
.max_register = WM8940_MONOMIX,
.reg_defaults = wm8940_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm8940_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
.readable_reg = wm8940_readable_register,
.volatile_reg = wm8940_volatile_register,
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 449f66636205..3a5c896a2d13 100644
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -1166,6 +1166,7 @@ static const struct snd_soc_dapm_route wm8998_dapm_routes[] = {
{ "MICSUPP", NULL, "SYSCLK" },
+ { "DRC1 Signal Activity", NULL, "SYSCLK" },
{ "DRC1 Signal Activity", NULL, "DRC1L" },
{ "DRC1 Signal Activity", NULL, "DRC1R" },
};
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index a07bd7c2c587..21fbe7d07063 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -394,6 +394,7 @@ static const struct {
int compr_direction;
int num_caps;
const struct wm_adsp_fw_caps *caps;
+ bool voice_trigger;
} wm_adsp_fw[WM_ADSP_NUM_FW] = {
[WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" },
[WM_ADSP_FW_HIFI] = { .file = "hifi" },
@@ -406,6 +407,7 @@ static const struct {
.compr_direction = SND_COMPRESS_CAPTURE,
.num_caps = ARRAY_SIZE(ctrl_caps),
.caps = ctrl_caps,
+ .voice_trigger = true,
},
[WM_ADSP_FW_ASR] = { .file = "asr" },
[WM_ADSP_FW_TRACE] = {
@@ -2366,13 +2368,15 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
dsp->running = false;
regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
- ADSP2_SYS_ENA | ADSP2_CORE_ENA |
- ADSP2_START, 0);
+ ADSP2_CORE_ENA | ADSP2_START, 0);
/* Make sure DMAs are quiesced */
+ regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0);
regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0);
regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_2, 0);
- regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0);
+
+ regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
+ ADSP2_SYS_ENA, 0);
list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
@@ -2998,6 +3002,9 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
goto out;
}
+ if (wm_adsp_fw[dsp->fw].voice_trigger && buf->irq_count == 2)
+ ret = WM_ADSP_COMPR_VOICE_TRIGGER;
+
out_notify:
if (compr && compr->stream)
snd_compr_fragment_elapsed(compr->stream);
@@ -3037,12 +3044,8 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
buf = compr->buf;
- if (!compr->buf) {
- ret = -ENXIO;
- goto out;
- }
-
- if (compr->buf->error) {
+ if (!compr->buf || compr->buf->error) {
+ snd_compr_stop_error(stream, SNDRV_PCM_STATE_XRUN);
ret = -EIO;
goto out;
}
@@ -3060,8 +3063,12 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
*/
if (buf->avail < wm_adsp_compr_frag_words(compr)) {
ret = wm_adsp_buffer_get_error(buf);
- if (ret < 0)
+ if (ret < 0) {
+ if (compr->buf->error)
+ snd_compr_stop_error(stream,
+ SNDRV_PCM_STATE_XRUN);
goto out;
+ }
ret = wm_adsp_buffer_reenable_irq(buf);
if (ret < 0) {
@@ -3156,11 +3163,10 @@ static int wm_adsp_compr_read(struct wm_adsp_compr *compr,
adsp_dbg(dsp, "Requested read of %zu bytes\n", count);
- if (!compr->buf)
- return -ENXIO;
-
- if (compr->buf->error)
+ if (!compr->buf || compr->buf->error) {
+ snd_compr_stop_error(compr->stream, SNDRV_PCM_STATE_XRUN);
return -EIO;
+ }
count /= WM_ADSP_DATA_WORD_SIZE;
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index feb61e2c4bb4..be3b5bcb7f17 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -19,6 +19,10 @@
#include "wmfw.h"
+/* Return values for wm_adsp_compr_handle_irq */
+#define WM_ADSP_COMPR_OK 0
+#define WM_ADSP_COMPR_VOICE_TRIGGER 1
+
struct wm_adsp_region {
int type;
unsigned int base;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 0f66fda2c772..05c2d33aa74d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1513,8 +1513,9 @@ static struct davinci_mcasp_pdata am33xx_mcasp_pdata = {
};
static struct davinci_mcasp_pdata dra7_mcasp_pdata = {
- .tx_dma_offset = 0x200,
- .rx_dma_offset = 0x284,
+ /* The CFG port offset will be calculated if it is needed */
+ .tx_dma_offset = 0,
+ .rx_dma_offset = 0,
.version = MCASP_VERSION_4,
};
@@ -1598,7 +1599,14 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of(
pdata = pdev->dev.platform_data;
return pdata;
} else if (match) {
- pdata = (struct davinci_mcasp_pdata*) match->data;
+ pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata),
+ GFP_KERNEL);
+ if (!pdata) {
+ dev_err(&pdev->dev,
+ "Failed to allocate memory for pdata\n");
+ ret = -ENOMEM;
+ return pdata;
+ }
} else {
/* control shouldn't reach here. something is wrong */
ret = -EINVAL;
@@ -1734,6 +1742,52 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
return PCM_EDMA;
}
+static u32 davinci_mcasp_txdma_offset(struct davinci_mcasp_pdata *pdata)
+{
+ int i;
+ u32 offset = 0;
+
+ if (pdata->version != MCASP_VERSION_4)
+ return pdata->tx_dma_offset;
+
+ for (i = 0; i < pdata->num_serializer; i++) {
+ if (pdata->serial_dir[i] == TX_MODE) {
+ if (!offset) {
+ offset = DAVINCI_MCASP_TXBUF_REG(i);
+ } else {
+ pr_err("%s: Only one serializer allowed!\n",
+ __func__);
+ break;
+ }
+ }
+ }
+
+ return offset;
+}
+
+static u32 davinci_mcasp_rxdma_offset(struct davinci_mcasp_pdata *pdata)
+{
+ int i;
+ u32 offset = 0;
+
+ if (pdata->version != MCASP_VERSION_4)
+ return pdata->rx_dma_offset;
+
+ for (i = 0; i < pdata->num_serializer; i++) {
+ if (pdata->serial_dir[i] == RX_MODE) {
+ if (!offset) {
+ offset = DAVINCI_MCASP_RXBUF_REG(i);
+ } else {
+ pr_err("%s: Only one serializer allowed!\n",
+ __func__);
+ break;
+ }
+ }
+ }
+
+ return offset;
+}
+
static int davinci_mcasp_probe(struct platform_device *pdev)
{
struct snd_dmaengine_dai_dma_data *dma_data;
@@ -1862,7 +1916,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
if (dat)
dma_data->addr = dat->start;
else
- dma_data->addr = mem->start + pdata->tx_dma_offset;
+ dma_data->addr = mem->start + davinci_mcasp_txdma_offset(pdata);
dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -1883,7 +1937,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
if (dat)
dma_data->addr = dat->start;
else
- dma_data->addr = mem->start + pdata->rx_dma_offset;
+ dma_data->addr =
+ mem->start + davinci_mcasp_rxdma_offset(pdata);
dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE];
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 1e8787fb3fb7..afddc8010c54 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -85,9 +85,9 @@
(n << 2))
/* Transmit Buffer for Serializer n */
-#define DAVINCI_MCASP_TXBUF_REG 0x200
+#define DAVINCI_MCASP_TXBUF_REG(n) (0x200 + (n << 2))
/* Receive Buffer for Serializer n */
-#define DAVINCI_MCASP_RXBUF_REG 0x280
+#define DAVINCI_MCASP_RXBUF_REG(n) (0x280 + (n << 2))
/* McASP FIFO Registers */
#define DAVINCI_MCASP_V2_AFIFO_BASE (0x1010)
diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig
index d50e08517dce..c297efe43861 100644
--- a/sound/soc/dwc/Kconfig
+++ b/sound/soc/dwc/Kconfig
@@ -7,4 +7,13 @@ config SND_DESIGNWARE_I2S
Synopsys desigwnware I2S device. The device supports upto
maximum of 8 channels each for play and record.
+config SND_DESIGNWARE_PCM
+ tristate "PCM PIO extension for I2S driver"
+ depends on SND_DESIGNWARE_I2S
+ help
+ Say Y, M or N if you want to add a custom ALSA extension that registers
+ a PCM and uses PIO to transfer data.
+
+ This functionality is specially suited for I2S devices that don't have
+ DMA support.
diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile
index 319371f690f4..38f1ca31c5fa 100644
--- a/sound/soc/dwc/Makefile
+++ b/sound/soc/dwc/Makefile
@@ -1,3 +1,5 @@
# SYNOPSYS Platform Support
obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o
-
+ifdef CONFIG_SND_DESIGNWARE_PCM
+obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_pcm.o
+endif
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 0db69b7e9617..dc97f4349e66 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -24,90 +24,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/dmaengine_pcm.h>
-
-/* common register for all channel */
-#define IER 0x000
-#define IRER 0x004
-#define ITER 0x008
-#define CER 0x00C
-#define CCR 0x010
-#define RXFFR 0x014
-#define TXFFR 0x018
-
-/* I2STxRxRegisters for all channels */
-#define LRBR_LTHR(x) (0x40 * x + 0x020)
-#define RRBR_RTHR(x) (0x40 * x + 0x024)
-#define RER(x) (0x40 * x + 0x028)
-#define TER(x) (0x40 * x + 0x02C)
-#define RCR(x) (0x40 * x + 0x030)
-#define TCR(x) (0x40 * x + 0x034)
-#define ISR(x) (0x40 * x + 0x038)
-#define IMR(x) (0x40 * x + 0x03C)
-#define ROR(x) (0x40 * x + 0x040)
-#define TOR(x) (0x40 * x + 0x044)
-#define RFCR(x) (0x40 * x + 0x048)
-#define TFCR(x) (0x40 * x + 0x04C)
-#define RFF(x) (0x40 * x + 0x050)
-#define TFF(x) (0x40 * x + 0x054)
-
-/* I2SCOMPRegisters */
-#define I2S_COMP_PARAM_2 0x01F0
-#define I2S_COMP_PARAM_1 0x01F4
-#define I2S_COMP_VERSION 0x01F8
-#define I2S_COMP_TYPE 0x01FC
-
-/*
- * Component parameter register fields - define the I2S block's
- * configuration.
- */
-#define COMP1_TX_WORDSIZE_3(r) (((r) & GENMASK(27, 25)) >> 25)
-#define COMP1_TX_WORDSIZE_2(r) (((r) & GENMASK(24, 22)) >> 22)
-#define COMP1_TX_WORDSIZE_1(r) (((r) & GENMASK(21, 19)) >> 19)
-#define COMP1_TX_WORDSIZE_0(r) (((r) & GENMASK(18, 16)) >> 16)
-#define COMP1_TX_CHANNELS(r) (((r) & GENMASK(10, 9)) >> 9)
-#define COMP1_RX_CHANNELS(r) (((r) & GENMASK(8, 7)) >> 7)
-#define COMP1_RX_ENABLED(r) (((r) & BIT(6)) >> 6)
-#define COMP1_TX_ENABLED(r) (((r) & BIT(5)) >> 5)
-#define COMP1_MODE_EN(r) (((r) & BIT(4)) >> 4)
-#define COMP1_FIFO_DEPTH_GLOBAL(r) (((r) & GENMASK(3, 2)) >> 2)
-#define COMP1_APB_DATA_WIDTH(r) (((r) & GENMASK(1, 0)) >> 0)
-
-#define COMP2_RX_WORDSIZE_3(r) (((r) & GENMASK(12, 10)) >> 10)
-#define COMP2_RX_WORDSIZE_2(r) (((r) & GENMASK(9, 7)) >> 7)
-#define COMP2_RX_WORDSIZE_1(r) (((r) & GENMASK(5, 3)) >> 3)
-#define COMP2_RX_WORDSIZE_0(r) (((r) & GENMASK(2, 0)) >> 0)
-
-/* Number of entries in WORDSIZE and DATA_WIDTH parameter registers */
-#define COMP_MAX_WORDSIZE (1 << 3)
-#define COMP_MAX_DATA_WIDTH (1 << 2)
-
-#define MAX_CHANNEL_NUM 8
-#define MIN_CHANNEL_NUM 2
-
-union dw_i2s_snd_dma_data {
- struct i2s_dma_data pd;
- struct snd_dmaengine_dai_dma_data dt;
-};
-
-struct dw_i2s_dev {
- void __iomem *i2s_base;
- struct clk *clk;
- int active;
- unsigned int capability;
- unsigned int quirks;
- unsigned int i2s_reg_comp1;
- unsigned int i2s_reg_comp2;
- struct device *dev;
- u32 ccr;
- u32 xfer_resolution;
- u32 fifo_th;
-
- /* data related to DMA transfers b/w i2s and DMAC */
- union dw_i2s_snd_dma_data play_dma_data;
- union dw_i2s_snd_dma_data capture_dma_data;
- struct i2s_clk_config_data config;
- int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
-};
+#include "local.h"
static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val)
{
@@ -145,51 +62,115 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
}
}
-static void i2s_start(struct dw_i2s_dev *dev,
- struct snd_pcm_substream *substream)
+static inline void i2s_disable_irqs(struct dw_i2s_dev *dev, u32 stream,
+ int chan_nr)
+{
+ u32 i, irq;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < (chan_nr / 2); i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30);
+ }
+ } else {
+ for (i = 0; i < (chan_nr / 2); i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03);
+ }
+ }
+}
+
+static inline void i2s_enable_irqs(struct dw_i2s_dev *dev, u32 stream,
+ int chan_nr)
{
- struct i2s_clk_config_data *config = &dev->config;
u32 i, irq;
- i2s_write_reg(dev->i2s_base, IER, 1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- for (i = 0; i < (config->chan_nr / 2); i++) {
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < (chan_nr / 2); i++) {
irq = i2s_read_reg(dev->i2s_base, IMR(i));
i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
}
- i2s_write_reg(dev->i2s_base, ITER, 1);
} else {
- for (i = 0; i < (config->chan_nr / 2); i++) {
+ for (i = 0; i < (chan_nr / 2); i++) {
irq = i2s_read_reg(dev->i2s_base, IMR(i));
i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
}
- i2s_write_reg(dev->i2s_base, IRER, 1);
+ }
+}
+
+static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
+{
+ struct dw_i2s_dev *dev = dev_id;
+ bool irq_valid = false;
+ u32 isr[4];
+ int i;
+
+ for (i = 0; i < 4; i++)
+ isr[i] = i2s_read_reg(dev->i2s_base, ISR(i));
+
+ i2s_clear_irqs(dev, SNDRV_PCM_STREAM_PLAYBACK);
+ i2s_clear_irqs(dev, SNDRV_PCM_STREAM_CAPTURE);
+
+ for (i = 0; i < 4; i++) {
+ /*
+ * Check if TX fifo is empty. If empty fill FIFO with samples
+ * NOTE: Only two channels supported
+ */
+ if ((isr[i] & ISR_TXFE) && (i == 0) && dev->use_pio) {
+ dw_pcm_push_tx(dev);
+ irq_valid = true;
+ }
+
+ /* Data available. Record mode not supported in PIO mode */
+ if (isr[i] & ISR_RXDA)
+ irq_valid = true;
+
+ /* Error Handling: TX */
+ if (isr[i] & ISR_TXFO) {
+ dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i);
+ irq_valid = true;
+ }
+
+ /* Error Handling: TX */
+ if (isr[i] & ISR_RXFO) {
+ dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i);
+ irq_valid = true;
+ }
}
+ if (irq_valid)
+ return IRQ_HANDLED;
+ else
+ return IRQ_NONE;
+}
+
+static void i2s_start(struct dw_i2s_dev *dev,
+ struct snd_pcm_substream *substream)
+{
+ struct i2s_clk_config_data *config = &dev->config;
+
+ i2s_write_reg(dev->i2s_base, IER, 1);
+ i2s_enable_irqs(dev, substream->stream, config->chan_nr);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ i2s_write_reg(dev->i2s_base, ITER, 1);
+ else
+ i2s_write_reg(dev->i2s_base, IRER, 1);
+
i2s_write_reg(dev->i2s_base, CER, 1);
}
static void i2s_stop(struct dw_i2s_dev *dev,
struct snd_pcm_substream *substream)
{
- u32 i = 0, irq;
i2s_clear_irqs(dev, substream->stream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
i2s_write_reg(dev->i2s_base, ITER, 0);
-
- for (i = 0; i < 4; i++) {
- irq = i2s_read_reg(dev->i2s_base, IMR(i));
- i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30);
- }
- } else {
+ else
i2s_write_reg(dev->i2s_base, IRER, 0);
- for (i = 0; i < 4; i++) {
- irq = i2s_read_reg(dev->i2s_base, IMR(i));
- i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03);
- }
- }
+ i2s_disable_irqs(dev, substream->stream, 8);
if (!dev->active) {
i2s_write_reg(dev->i2s_base, CER, 0);
@@ -223,7 +204,7 @@ static int dw_i2s_startup(struct snd_pcm_substream *substream,
static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
{
- u32 ch_reg, irq;
+ u32 ch_reg;
struct i2s_clk_config_data *config = &dev->config;
@@ -235,16 +216,12 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
dev->xfer_resolution);
i2s_write_reg(dev->i2s_base, TFCR(ch_reg),
dev->fifo_th - 1);
- irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
- i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30);
i2s_write_reg(dev->i2s_base, TER(ch_reg), 1);
} else {
i2s_write_reg(dev->i2s_base, RCR(ch_reg),
dev->xfer_resolution);
i2s_write_reg(dev->i2s_base, RFCR(ch_reg),
dev->fifo_th - 1);
- irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
- i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03);
i2s_write_reg(dev->i2s_base, RER(ch_reg), 1);
}
@@ -278,7 +255,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
break;
default:
- dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt");
+ dev_err(dev->dev, "designware-i2s: unsupported PCM fmt");
return -EINVAL;
}
@@ -626,7 +603,7 @@ static int dw_i2s_probe(struct platform_device *pdev)
const struct i2s_platform_data *pdata = pdev->dev.platform_data;
struct dw_i2s_dev *dev;
struct resource *res;
- int ret;
+ int ret, irq;
struct snd_soc_dai_driver *dw_i2s_dai;
const char *clk_id;
@@ -651,6 +628,16 @@ static int dw_i2s_probe(struct platform_device *pdev)
dev->dev = &pdev->dev;
+ irq = platform_get_irq(pdev, 0);
+ if (irq >= 0) {
+ ret = devm_request_irq(&pdev->dev, irq, i2s_irq_handler, 0,
+ pdev->name, dev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to request irq\n");
+ return ret;
+ }
+ }
+
dev->i2s_reg_comp1 = I2S_COMP_PARAM_1;
dev->i2s_reg_comp2 = I2S_COMP_PARAM_2;
if (pdata) {
@@ -697,12 +684,24 @@ static int dw_i2s_probe(struct platform_device *pdev)
if (!pdata) {
ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
- if (ret) {
+ if (ret == -EPROBE_DEFER) {
+ dev_err(&pdev->dev,
+ "failed to register PCM, deferring probe\n");
+ return ret;
+ } else if (ret) {
dev_err(&pdev->dev,
- "Could not register PCM: %d\n", ret);
- goto err_clk_disable;
+ "Could not register DMA PCM: %d\n"
+ "falling back to PIO mode\n", ret);
+ ret = dw_pcm_register(pdev);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "Could not register PIO PCM: %d\n",
+ ret);
+ goto err_clk_disable;
+ }
}
}
+
pm_runtime_enable(&pdev->dev);
return 0;
diff --git a/sound/soc/dwc/designware_pcm.c b/sound/soc/dwc/designware_pcm.c
new file mode 100644
index 000000000000..4a83a22fa3cb
--- /dev/null
+++ b/sound/soc/dwc/designware_pcm.c
@@ -0,0 +1,225 @@
+/*
+ * ALSA SoC Synopsys PIO PCM for I2S driver
+ *
+ * sound/soc/dwc/designware_pcm.c
+ *
+ * Copyright (C) 2016 Synopsys
+ * Jose Abreu <joabreu@synopsys.com>
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#include <linux/io.h>
+#include <linux/rcupdate.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "local.h"
+
+#define BUFFER_BYTES_MAX (3 * 2 * 8 * PERIOD_BYTES_MIN)
+#define PERIOD_BYTES_MIN 4096
+#define PERIODS_MIN 2
+
+#define dw_pcm_tx_fn(sample_bits) \
+static unsigned int dw_pcm_tx_##sample_bits(struct dw_i2s_dev *dev, \
+ struct snd_pcm_runtime *runtime, unsigned int tx_ptr, \
+ bool *period_elapsed) \
+{ \
+ const u##sample_bits (*p)[2] = (void *)runtime->dma_area; \
+ unsigned int period_pos = tx_ptr % runtime->period_size; \
+ int i; \
+\
+ for (i = 0; i < dev->fifo_th; i++) { \
+ iowrite32(p[tx_ptr][0], dev->i2s_base + LRBR_LTHR(0)); \
+ iowrite32(p[tx_ptr][1], dev->i2s_base + RRBR_RTHR(0)); \
+ period_pos++; \
+ if (++tx_ptr >= runtime->buffer_size) \
+ tx_ptr = 0; \
+ } \
+ *period_elapsed = period_pos >= runtime->period_size; \
+ return tx_ptr; \
+}
+
+dw_pcm_tx_fn(16);
+dw_pcm_tx_fn(32);
+
+#undef dw_pcm_tx_fn
+
+static const struct snd_pcm_hardware dw_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .rate_min = 32000,
+ .rate_max = 48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = BUFFER_BYTES_MAX,
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN,
+ .periods_min = PERIODS_MIN,
+ .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN,
+ .fifo_size = 16,
+};
+
+void dw_pcm_push_tx(struct dw_i2s_dev *dev)
+{
+ struct snd_pcm_substream *tx_substream;
+ bool tx_active, period_elapsed;
+
+ rcu_read_lock();
+ tx_substream = rcu_dereference(dev->tx_substream);
+ tx_active = tx_substream && snd_pcm_running(tx_substream);
+ if (tx_active) {
+ unsigned int tx_ptr = READ_ONCE(dev->tx_ptr);
+ unsigned int new_tx_ptr = dev->tx_fn(dev, tx_substream->runtime,
+ tx_ptr, &period_elapsed);
+ cmpxchg(&dev->tx_ptr, tx_ptr, new_tx_ptr);
+
+ if (period_elapsed)
+ snd_pcm_period_elapsed(tx_substream);
+ }
+ rcu_read_unlock();
+}
+EXPORT_SYMBOL_GPL(dw_pcm_push_tx);
+
+static int dw_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+
+ snd_soc_set_runtime_hwparams(substream, &dw_pcm_hardware);
+ snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ runtime->private_data = dev;
+
+ return 0;
+}
+
+static int dw_pcm_close(struct snd_pcm_substream *substream)
+{
+ synchronize_rcu();
+ return 0;
+}
+
+static int dw_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct dw_i2s_dev *dev = runtime->private_data;
+ int ret;
+
+ switch (params_channels(hw_params)) {
+ case 2:
+ break;
+ default:
+ dev_err(dev->dev, "invalid channels number\n");
+ return -EINVAL;
+ }
+
+ switch (params_format(hw_params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ dev->tx_fn = dw_pcm_tx_16;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ dev->tx_fn = dw_pcm_tx_32;
+ break;
+ default:
+ dev_err(dev->dev, "invalid format\n");
+ return -EINVAL;
+ }
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) {
+ dev_err(dev->dev, "only playback is available\n");
+ return -EINVAL;
+ }
+
+ ret = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (ret < 0)
+ return ret;
+ else
+ return 0;
+}
+
+static int dw_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int dw_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct dw_i2s_dev *dev = runtime->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ WRITE_ONCE(dev->tx_ptr, 0);
+ rcu_assign_pointer(dev->tx_substream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ rcu_assign_pointer(dev->tx_substream, NULL);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t dw_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct dw_i2s_dev *dev = runtime->private_data;
+ snd_pcm_uframes_t pos = READ_ONCE(dev->tx_ptr);
+
+ return pos < runtime->buffer_size ? pos : 0;
+}
+
+static int dw_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ size_t size = dw_pcm_hardware.buffer_bytes_max;
+
+ return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL), size, size);
+}
+
+static void dw_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static const struct snd_pcm_ops dw_pcm_ops = {
+ .open = dw_pcm_open,
+ .close = dw_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = dw_pcm_hw_params,
+ .hw_free = dw_pcm_hw_free,
+ .trigger = dw_pcm_trigger,
+ .pointer = dw_pcm_pointer,
+};
+
+static const struct snd_soc_platform_driver dw_pcm_platform = {
+ .pcm_new = dw_pcm_new,
+ .pcm_free = dw_pcm_free,
+ .ops = &dw_pcm_ops,
+};
+
+int dw_pcm_register(struct platform_device *pdev)
+{
+ return devm_snd_soc_register_platform(&pdev->dev, &dw_pcm_platform);
+}
+EXPORT_SYMBOL_GPL(dw_pcm_register);
diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h
new file mode 100644
index 000000000000..68afd7577343
--- /dev/null
+++ b/sound/soc/dwc/local.h
@@ -0,0 +1,128 @@
+/*
+ * Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux@gmail.com)
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#ifndef __DESIGNWARE_LOCAL_H
+#define __DESIGNWARE_LOCAL_H
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/types.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm.h>
+#include <sound/designware_i2s.h>
+
+/* common register for all channel */
+#define IER 0x000
+#define IRER 0x004
+#define ITER 0x008
+#define CER 0x00C
+#define CCR 0x010
+#define RXFFR 0x014
+#define TXFFR 0x018
+
+/* Interrupt status register fields */
+#define ISR_TXFO BIT(5)
+#define ISR_TXFE BIT(4)
+#define ISR_RXFO BIT(1)
+#define ISR_RXDA BIT(0)
+
+/* I2STxRxRegisters for all channels */
+#define LRBR_LTHR(x) (0x40 * x + 0x020)
+#define RRBR_RTHR(x) (0x40 * x + 0x024)
+#define RER(x) (0x40 * x + 0x028)
+#define TER(x) (0x40 * x + 0x02C)
+#define RCR(x) (0x40 * x + 0x030)
+#define TCR(x) (0x40 * x + 0x034)
+#define ISR(x) (0x40 * x + 0x038)
+#define IMR(x) (0x40 * x + 0x03C)
+#define ROR(x) (0x40 * x + 0x040)
+#define TOR(x) (0x40 * x + 0x044)
+#define RFCR(x) (0x40 * x + 0x048)
+#define TFCR(x) (0x40 * x + 0x04C)
+#define RFF(x) (0x40 * x + 0x050)
+#define TFF(x) (0x40 * x + 0x054)
+
+/* I2SCOMPRegisters */
+#define I2S_COMP_PARAM_2 0x01F0
+#define I2S_COMP_PARAM_1 0x01F4
+#define I2S_COMP_VERSION 0x01F8
+#define I2S_COMP_TYPE 0x01FC
+
+/*
+ * Component parameter register fields - define the I2S block's
+ * configuration.
+ */
+#define COMP1_TX_WORDSIZE_3(r) (((r) & GENMASK(27, 25)) >> 25)
+#define COMP1_TX_WORDSIZE_2(r) (((r) & GENMASK(24, 22)) >> 22)
+#define COMP1_TX_WORDSIZE_1(r) (((r) & GENMASK(21, 19)) >> 19)
+#define COMP1_TX_WORDSIZE_0(r) (((r) & GENMASK(18, 16)) >> 16)
+#define COMP1_TX_CHANNELS(r) (((r) & GENMASK(10, 9)) >> 9)
+#define COMP1_RX_CHANNELS(r) (((r) & GENMASK(8, 7)) >> 7)
+#define COMP1_RX_ENABLED(r) (((r) & BIT(6)) >> 6)
+#define COMP1_TX_ENABLED(r) (((r) & BIT(5)) >> 5)
+#define COMP1_MODE_EN(r) (((r) & BIT(4)) >> 4)
+#define COMP1_FIFO_DEPTH_GLOBAL(r) (((r) & GENMASK(3, 2)) >> 2)
+#define COMP1_APB_DATA_WIDTH(r) (((r) & GENMASK(1, 0)) >> 0)
+
+#define COMP2_RX_WORDSIZE_3(r) (((r) & GENMASK(12, 10)) >> 10)
+#define COMP2_RX_WORDSIZE_2(r) (((r) & GENMASK(9, 7)) >> 7)
+#define COMP2_RX_WORDSIZE_1(r) (((r) & GENMASK(5, 3)) >> 3)
+#define COMP2_RX_WORDSIZE_0(r) (((r) & GENMASK(2, 0)) >> 0)
+
+/* Number of entries in WORDSIZE and DATA_WIDTH parameter registers */
+#define COMP_MAX_WORDSIZE (1 << 3)
+#define COMP_MAX_DATA_WIDTH (1 << 2)
+
+#define MAX_CHANNEL_NUM 8
+#define MIN_CHANNEL_NUM 2
+
+union dw_i2s_snd_dma_data {
+ struct i2s_dma_data pd;
+ struct snd_dmaengine_dai_dma_data dt;
+};
+
+struct dw_i2s_dev {
+ void __iomem *i2s_base;
+ struct clk *clk;
+ int active;
+ unsigned int capability;
+ unsigned int quirks;
+ unsigned int i2s_reg_comp1;
+ unsigned int i2s_reg_comp2;
+ struct device *dev;
+ u32 ccr;
+ u32 xfer_resolution;
+ u32 fifo_th;
+
+ /* data related to DMA transfers b/w i2s and DMAC */
+ union dw_i2s_snd_dma_data play_dma_data;
+ union dw_i2s_snd_dma_data capture_dma_data;
+ struct i2s_clk_config_data config;
+ int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
+
+ /* data related to PIO transfers (TX) */
+ bool use_pio;
+ struct snd_pcm_substream __rcu *tx_substream;
+ unsigned int (*tx_fn)(struct dw_i2s_dev *dev,
+ struct snd_pcm_runtime *runtime, unsigned int tx_ptr,
+ bool *period_elapsed);
+ unsigned int tx_ptr;
+};
+
+#if IS_ENABLED(CONFIG_SND_DESIGNWARE_PCM)
+void dw_pcm_push_tx(struct dw_i2s_dev *dev);
+int dw_pcm_register(struct platform_device *pdev);
+#else
+void dw_pcm_push_tx(struct dw_i2s_dev *dev) { }
+int dw_pcm_register(struct platform_device *pdev)
+{
+ return -EINVAL;
+}
+#endif
+
+#endif
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 35aabf9dc503..19bdcac71775 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -4,6 +4,7 @@ comment "Common SoC Audio options for Freescale CPUs:"
config SND_SOC_FSL_ASRC
tristate "Asynchronous Sample Rate Converter (ASRC) module support"
+ depends on HAS_DMA
select REGMAP_MMIO
select SND_SOC_GENERIC_DMAENGINE_PCM
help
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 151849f79863..beec7934a265 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -172,7 +172,7 @@ static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name)
if (*pos >= size * 2) {
*pos = 0;
} else if (unlikely((*pos % size) + 3 > size)) {
- dev_err(&pdev->dev, "User bit receivce buffer overflow\n");
+ dev_err(&pdev->dev, "User bit receive buffer overflow\n");
return;
}
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 632ecc0e3956..bedec4a32581 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -952,16 +952,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
ssi_private->i2s_mode = CCSR_SSI_SCR_NET;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
+ regmap_update_bits(regs, CCSR_SSI_STCCR,
+ CCSR_SSI_SxCCR_DC_MASK,
+ CCSR_SSI_SxCCR_DC(2));
+ regmap_update_bits(regs, CCSR_SSI_SRCCR,
+ CCSR_SSI_SxCCR_DC_MASK,
+ CCSR_SSI_SxCCR_DC(2));
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFS:
case SND_SOC_DAIFMT_CBS_CFS:
ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER;
- regmap_update_bits(regs, CCSR_SSI_STCCR,
- CCSR_SSI_SxCCR_DC_MASK,
- CCSR_SSI_SxCCR_DC(2));
- regmap_update_bits(regs, CCSR_SSI_SRCCR,
- CCSR_SSI_SxCCR_DC_MASK,
- CCSR_SSI_SxCCR_DC(2));
break;
case SND_SOC_DAIFMT_CBM_CFM:
ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE;
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 91c15abb625e..a20c3dfbcb5d 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -7,7 +7,7 @@ config SND_MFLD_MACHINE
help
This adds support for ASoC machine driver for Intel(R) MID Medfield platform
used as alsa device in audio substem in Intel(R) MID devices
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
config SND_SST_MFLD_PLATFORM
@@ -25,7 +25,6 @@ config SND_SST_IPC_ACPI
tristate
select SND_SST_IPC
select SND_SOC_INTEL_SST
- depends on ACPI
config SND_SOC_INTEL_SST
tristate
@@ -33,6 +32,12 @@ config SND_SOC_INTEL_SST
select SND_SOC_INTEL_SST_MATCH if ACPI
depends on (X86 || COMPILE_TEST)
+# firmware stuff depends DW_DMAC_CORE; since there is no depends-on from
+# the reverse selection, each machine driver needs to select
+# SND_SOC_INTEL_SST_FIRMWARE carefully depending on DW_DMAC_CORE
+config SND_SOC_INTEL_SST_FIRMWARE
+ tristate
+
config SND_SOC_INTEL_SST_ACPI
tristate
@@ -48,16 +53,33 @@ config SND_SOC_INTEL_BAYTRAIL
config SND_SOC_INTEL_HASWELL_MACH
tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM
- depends on DW_DMAC_CORE=y
+ depends on DW_DMAC_CORE
select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SST_FIRMWARE
select SND_SOC_INTEL_HASWELL
select SND_SOC_RT5640
help
This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell
Ultrabook platforms.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
+config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
+ tristate "ASoC Audio driver for Broxton with DA7219 and MAX98357A in I2S Mode"
+ depends on X86 && ACPI && I2C
+ select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SKYLAKE
+ select SND_SOC_DA7219
+ select SND_SOC_MAX98357A
+ select SND_SOC_DMIC
+ select SND_SOC_HDAC_HDMI
+ select SND_HDA_DSP_LOADER
+ help
+ This adds support for ASoC machine driver for Broxton-P platforms
+ with DA7219 + MAX98357A I2S audio codec.
+ Say Y if you have such a device.
+ If unsure select "N".
+
config SND_SOC_INTEL_BXT_RT298_MACH
tristate "ASoC Audio driver for Broxton with RT298 I2S mode"
depends on X86 && ACPI && I2C
@@ -70,26 +92,28 @@ config SND_SOC_INTEL_BXT_RT298_MACH
help
This adds support for ASoC machine driver for Broxton platforms
with RT286 I2S audio codec.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
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_SST_IPC_ACPI = n)
+ depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n)
select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SST_FIRMWARE
select SND_SOC_INTEL_BAYTRAIL
select SND_SOC_RT5640
help
This adds audio driver for Intel Baytrail platform based boards
with the RT5640 audio codec. This driver is deprecated, use
- SND_SOC_INTEL_BYTCR_RT5640_MACH instead for better functionality
+ SND_SOC_INTEL_BYTCR_RT5640_MACH instead for better functionality.
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 && (SND_SST_IPC_ACPI = n)
+ depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n)
select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SST_FIRMWARE
select SND_SOC_INTEL_BAYTRAIL
select SND_SOC_MAX98090
help
@@ -100,19 +124,20 @@ config SND_SOC_INTEL_BROADWELL_MACH
tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
depends on X86_INTEL_LPSS && I2C && DW_DMAC && \
I2C_DESIGNWARE_PLATFORM
- depends on DW_DMAC_CORE=y
+ depends on DW_DMAC_CORE
select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SST_FIRMWARE
select SND_SOC_INTEL_HASWELL
select SND_SOC_RT286
help
This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell
Ultrabook platforms.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_BYTCR_RT5640_MACH
tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec"
- depends on X86 && I2C
+ depends on X86 && I2C && ACPI
select SND_SOC_RT5640
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
@@ -120,12 +145,12 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5640 audio codec.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_BYTCR_RT5651_MACH
tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec"
- depends on X86 && I2C
+ depends on X86 && I2C && ACPI
select SND_SOC_RT5651
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
@@ -133,12 +158,12 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5651 audio codec.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
- depends on X86_INTEL_LPSS && I2C
+ depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_RT5670
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
@@ -146,12 +171,12 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5672 audio codec.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec"
- depends on X86_INTEL_LPSS && I2C
+ depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_RT5645
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
@@ -163,16 +188,16 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec"
- depends on X86_INTEL_LPSS && I2C
+ depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_MAX98090
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.
- If unsure select "N".
+ 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.
+ If unsure select "N".
config SND_SOC_INTEL_SKYLAKE
tristate
@@ -192,7 +217,7 @@ config SND_SOC_INTEL_SKL_RT286_MACH
help
This adds support for ASoC machine driver for Skylake platforms
with RT286 I2S audio codec.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH
@@ -207,7 +232,7 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH
help
This adds support for ASoC Onboard Codec I2S machine driver. This will
create an alsa sound card for NAU88L25 + SSM4567.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH
@@ -222,5 +247,5 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH
help
This adds support for ASoC Onboard Codec I2S machine driver. This will
create an alsa sound card for NAU88L25 + MAX98357A.
- Say Y if you have such a device
+ Say Y if you have such a device.
If unsure select "N".
diff --git a/sound/soc/intel/atom/sst-mfld-platform-compress.c b/sound/soc/intel/atom/sst-mfld-platform-compress.c
index 395168986462..1bead81bb510 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-compress.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-compress.c
@@ -182,24 +182,29 @@ static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
case SNDRV_PCM_TRIGGER_START:
if (stream->compr_ops->stream_start)
return stream->compr_ops->stream_start(sst->dev, stream->id);
+ break;
case SNDRV_PCM_TRIGGER_STOP:
if (stream->compr_ops->stream_drop)
return stream->compr_ops->stream_drop(sst->dev, stream->id);
+ break;
case SND_COMPR_TRIGGER_DRAIN:
if (stream->compr_ops->stream_drain)
return stream->compr_ops->stream_drain(sst->dev, stream->id);
+ break;
case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
if (stream->compr_ops->stream_partial_drain)
return stream->compr_ops->stream_partial_drain(sst->dev, stream->id);
+ break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (stream->compr_ops->stream_pause)
return stream->compr_ops->stream_pause(sst->dev, stream->id);
+ break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (stream->compr_ops->stream_pause_release)
return stream->compr_ops->stream_pause_release(sst->dev, stream->id);
- default:
- return -EINVAL;
+ break;
}
+ return -EINVAL;
}
static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c
index 3bc4b63b2f9d..4d3184971227 100644
--- a/sound/soc/intel/atom/sst/sst_acpi.c
+++ b/sound/soc/intel/atom/sst/sst_acpi.c
@@ -28,6 +28,7 @@
#include <linux/firmware.h>
#include <linux/pm_runtime.h>
#include <linux/pm_qos.h>
+#include <linux/dmi.h>
#include <linux/acpi.h>
#include <asm/platform_sst_audio.h>
#include <sound/core.h>
@@ -237,6 +238,9 @@ static int sst_acpi_probe(struct platform_device *pdev)
dev_err(dev, "No matching machine driver found\n");
return -ENODEV;
}
+ if (mach->machine_quirk)
+ mach = mach->machine_quirk(mach);
+
pdata = mach->pdata;
ret = kstrtouint(id->id, 16, &dev_id);
@@ -320,6 +324,44 @@ static int sst_acpi_remove(struct platform_device *pdev)
return 0;
}
+static unsigned long cht_machine_id;
+
+#define CHT_SURFACE_MACH 1
+
+static int cht_surface_quirk_cb(const struct dmi_system_id *id)
+{
+ cht_machine_id = CHT_SURFACE_MACH;
+ return 1;
+}
+
+
+static const struct dmi_system_id cht_table[] = {
+ {
+ .callback = cht_surface_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Microsoft Corporation"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Surface 3"),
+ },
+ },
+};
+
+
+static struct sst_acpi_mach cht_surface_mach = {
+ "10EC5640", "cht-bsw-rt5645", "intel/fw_sst_22a8.bin", "cht-bsw", NULL,
+ &chv_platform_data };
+
+static struct sst_acpi_mach *cht_quirk(void *arg)
+{
+ struct sst_acpi_mach *mach = arg;
+
+ dmi_check_system(cht_table);
+
+ if (cht_machine_id == CHT_SURFACE_MACH)
+ return &cht_surface_mach;
+ else
+ return mach;
+}
+
static struct sst_acpi_mach sst_acpi_bytcr[] = {
{"10EC5640", "bytcr_rt5640", "intel/fw_sst_0f28.bin", "bytcr_rt5640", NULL,
&byt_rvp_platform_data },
@@ -343,7 +385,7 @@ static struct sst_acpi_mach sst_acpi_chv[] = {
{"193C9890", "cht-bsw-max98090", "intel/fw_sst_22a8.bin", "cht-bsw", NULL,
&chv_platform_data },
/* some CHT-T platforms rely on RT5640, use Baytrail machine driver */
- {"10EC5640", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", NULL,
+ {"10EC5640", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", cht_quirk,
&chv_platform_data },
{},
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index a8506774f510..dac03a06bfd8 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
snd-soc-sst-broadwell-objs := broadwell.o
+snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o
snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
@@ -15,6 +16,7 @@ snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
new file mode 100644
index 000000000000..3774b117d365
--- /dev/null
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -0,0 +1,460 @@
+/*
+ * Intel Broxton-P I2S Machine Driver
+ *
+ * Copyright (C) 2016, Intel Corporation. All rights reserved.
+ *
+ * Modified from:
+ * Intel Skylake I2S Machine driver
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../../codecs/da7219.h"
+#include "../../codecs/da7219-aad.h"
+
+#define BXT_DIALOG_CODEC_DAI "da7219-hifi"
+#define BXT_MAXIM_CODEC_DAI "HiFi"
+#define DUAL_CHANNEL 2
+
+static struct snd_soc_jack broxton_headset;
+
+enum {
+ BXT_DPCM_AUDIO_PB = 0,
+ BXT_DPCM_AUDIO_CP,
+ BXT_DPCM_AUDIO_REF_CP,
+ BXT_DPCM_AUDIO_HDMI1_PB,
+ BXT_DPCM_AUDIO_HDMI2_PB,
+ BXT_DPCM_AUDIO_HDMI3_PB,
+};
+
+static const struct snd_kcontrol_new broxton_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Spk"),
+};
+
+static const struct snd_soc_dapm_widget broxton_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_SPK("Spk", NULL),
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+ SND_SOC_DAPM_SPK("HDMI1", NULL),
+ SND_SOC_DAPM_SPK("HDMI2", NULL),
+ SND_SOC_DAPM_SPK("HDMI3", NULL),
+};
+
+static const struct snd_soc_dapm_route broxton_map[] = {
+ /* HP jack connectors - unknown if we have jack detection */
+ {"Headphone Jack", NULL, "HPL"},
+ {"Headphone Jack", NULL, "HPR"},
+
+ /* speaker */
+ {"Spk", NULL, "Speaker"},
+
+ /* other jacks */
+ {"MIC", NULL, "Headset Mic"},
+
+ /* digital mics */
+ {"DMic", NULL, "SoC DMIC"},
+
+ /* CODEC BE connections */
+ {"HiFi Playback", NULL, "ssp5 Tx"},
+ {"ssp5 Tx", NULL, "codec0_out"},
+
+ {"Playback", NULL, "ssp1 Tx"},
+ {"ssp1 Tx", NULL, "codec1_out"},
+
+ {"codec0_in", NULL, "ssp1 Rx"},
+ {"ssp1 Rx", NULL, "Capture"},
+
+ {"HDMI1", NULL, "hif5 Output"},
+ {"HDMI2", NULL, "hif6 Output"},
+ {"HDMI3", NULL, "hif7 Output"},
+
+ {"hifi3", NULL, "iDisp3 Tx"},
+ {"iDisp3 Tx", NULL, "iDisp3_out"},
+ {"hifi2", NULL, "iDisp2 Tx"},
+ {"iDisp2 Tx", NULL, "iDisp2_out"},
+ {"hifi1", NULL, "iDisp1 Tx"},
+ {"iDisp1 Tx", NULL, "iDisp1_out"},
+
+ /* DMIC */
+ {"dmic01_hifi", NULL, "DMIC01 Rx"},
+ {"DMIC01 Rx", NULL, "DMIC AIF"},
+};
+
+static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+ /* The ADSP will convert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = DUAL_CHANNEL;
+
+ /* set SSP to 24 bit */
+ snd_mask_none(fmt);
+ snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+ return 0;
+}
+
+static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ /*
+ * Headset buttons map to the google Reference headset.
+ * These can be configured by userspace.
+ */
+ ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3, &broxton_headset,
+ NULL, 0);
+ if (ret) {
+ dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+ return ret;
+ }
+
+ da7219_aad_jack_det(codec, &broxton_headset);
+
+ snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC");
+
+ return ret;
+}
+
+static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *dai = rtd->codec_dai;
+
+ return hdac_hdmi_jack_init(dai, BXT_DPCM_AUDIO_HDMI1_PB + dai->id);
+}
+
+static int broxton_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dapm_context *dapm;
+ struct snd_soc_component *component = rtd->cpu_dai->component;
+
+ dapm = snd_soc_component_get_dapm(component);
+ snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+
+ return 0;
+}
+
+static unsigned int rates[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static unsigned int channels[] = {
+ DUAL_CHANNEL,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+};
+
+static int bxt_fe_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ /*
+ * On this platform for PCM device we support,
+ * 48Khz
+ * stereo
+ * 16 bit audio
+ */
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+
+ return 0;
+}
+
+static const struct snd_soc_ops broxton_da7219_fe_ops = {
+ .startup = bxt_fe_startup,
+};
+
+static int broxton_da7219_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->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
+
+ ret = snd_soc_dai_set_pll(codec_dai, 0,
+ DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
+ return -EIO;
+ }
+
+ return ret;
+}
+
+static int broxton_da7219_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, 0,
+ DA7219_SYSCLK_MCLK, 0, 0);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
+ return -EIO;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_ops broxton_da7219_ops = {
+ .hw_params = broxton_da7219_hw_params,
+ .hw_free = broxton_da7219_hw_free,
+};
+
+/* broxton digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broxton_dais[] = {
+ /* Front End DAI links */
+ [BXT_DPCM_AUDIO_PB]
+ {
+ .name = "Bxt Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .nonatomic = 1,
+ .init = broxton_da7219_fe_init,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .ops = &broxton_da7219_fe_ops,
+ },
+ [BXT_DPCM_AUDIO_CP]
+ {
+ .name = "Bxt Audio Capture Port",
+ .stream_name = "Audio Record",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .nonatomic = 1,
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ .ops = &broxton_da7219_fe_ops,
+ },
+ [BXT_DPCM_AUDIO_REF_CP]
+ {
+ .name = "Bxt Audio Reference cap",
+ .stream_name = "Refcap",
+ .cpu_dai_name = "Reference Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI1_PB]
+ {
+ .name = "Bxt HDMI Port1",
+ .stream_name = "Hdmi1",
+ .cpu_dai_name = "HDMI1 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI2_PB]
+ {
+ .name = "Bxt HDMI Port2",
+ .stream_name = "Hdmi2",
+ .cpu_dai_name = "HDMI2 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI3_PB]
+ {
+ .name = "Bxt HDMI Port3",
+ .stream_name = "Hdmi3",
+ .cpu_dai_name = "HDMI3 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ /* Back End DAI links */
+ {
+ /* SSP5 - Codec */
+ .name = "SSP5-Codec",
+ .id = 0,
+ .cpu_dai_name = "SSP5 Pin",
+ .platform_name = "0000:00:0e.0",
+ .no_pcm = 1,
+ .codec_name = "MX98357A:00",
+ .codec_dai_name = BXT_MAXIM_CODEC_DAI,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = broxton_ssp_fixup,
+ .dpcm_playback = 1,
+ },
+ {
+ /* SSP1 - Codec */
+ .name = "SSP1-Codec",
+ .id = 1,
+ .cpu_dai_name = "SSP1 Pin",
+ .platform_name = "0000:00:0e.0",
+ .no_pcm = 1,
+ .codec_name = "i2c-DLGS7219:00",
+ .codec_dai_name = BXT_DIALOG_CODEC_DAI,
+ .init = broxton_da7219_codec_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = broxton_ssp_fixup,
+ .ops = &broxton_da7219_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "dmic01",
+ .id = 2,
+ .cpu_dai_name = "DMIC01 Pin",
+ .codec_name = "dmic-codec",
+ .codec_dai_name = "dmic-hifi",
+ .platform_name = "0000:00:0e.0",
+ .ignore_suspend = 1,
+ .dpcm_capture = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp1",
+ .id = 3,
+ .cpu_dai_name = "iDisp1 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi1",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp2",
+ .id = 4,
+ .cpu_dai_name = "iDisp2 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi2",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp3",
+ .id = 5,
+ .cpu_dai_name = "iDisp3 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi3",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+};
+
+/* broxton audio machine driver for SPT + da7219 */
+static struct snd_soc_card broxton_audio_card = {
+ .name = "bxtda7219max",
+ .owner = THIS_MODULE,
+ .dai_link = broxton_dais,
+ .num_links = ARRAY_SIZE(broxton_dais),
+ .controls = broxton_controls,
+ .num_controls = ARRAY_SIZE(broxton_controls),
+ .dapm_widgets = broxton_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(broxton_widgets),
+ .dapm_routes = broxton_map,
+ .num_dapm_routes = ARRAY_SIZE(broxton_map),
+ .fully_routed = true,
+};
+
+static int broxton_audio_probe(struct platform_device *pdev)
+{
+ broxton_audio_card.dev = &pdev->dev;
+ return devm_snd_soc_register_card(&pdev->dev, &broxton_audio_card);
+}
+
+static struct platform_driver broxton_audio = {
+ .probe = broxton_audio_probe,
+ .driver = {
+ .name = "bxt_da7219_max98357a_i2s",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+module_platform_driver(broxton_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode");
+MODULE_AUTHOR("Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>");
+MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com>");
+MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
+MODULE_AUTHOR("Conrad Cooke <conrad.cooke@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s");
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index f4787515c0ed..253d7bfbf511 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -33,6 +33,7 @@ enum {
BXT_DPCM_AUDIO_PB = 0,
BXT_DPCM_AUDIO_CP,
BXT_DPCM_AUDIO_REF_CP,
+ BXT_DPCM_AUDIO_DMIC_CP,
BXT_DPCM_AUDIO_HDMI1_PB,
BXT_DPCM_AUDIO_HDMI2_PB,
BXT_DPCM_AUDIO_HDMI3_PB,
@@ -88,6 +89,7 @@ static const struct snd_soc_dapm_route broxton_rt298_map[] = {
/* CODEC BE connections */
{ "AIF1 Playback", NULL, "ssp5 Tx"},
{ "ssp5 Tx", NULL, "codec0_out"},
+ { "ssp5 Tx", NULL, "codec1_out"},
{ "codec0_in", NULL, "ssp5 Rx" },
{ "ssp5 Rx", NULL, "AIF1 Capture" },
@@ -104,6 +106,17 @@ static const struct snd_soc_dapm_route broxton_rt298_map[] = {
};
+static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dapm_context *dapm;
+ struct snd_soc_component *component = rtd->cpu_dai->component;
+
+ dapm = snd_soc_component_get_dapm(component);
+ snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+
+ return 0;
+}
+
static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
@@ -118,6 +131,9 @@ static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
return ret;
rt298_mic_detect(codec, &broxton_headset);
+
+ snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC");
+
return 0;
}
@@ -169,6 +185,89 @@ static struct snd_soc_ops broxton_rt298_ops = {
.hw_params = broxton_rt298_hw_params,
};
+static unsigned int rates[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ if (params_channels(params) == 2)
+ channels->min = channels->max = 2;
+ else
+ channels->min = channels->max = 4;
+
+ return 0;
+}
+
+static unsigned int channels_dmic[] = {
+ 2, 4,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_dmic_channels = {
+ .count = ARRAY_SIZE(channels_dmic),
+ .list = channels_dmic,
+ .mask = 0,
+};
+
+static int broxton_dmic_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.channels_max = 4;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_dmic_channels);
+
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+}
+
+static struct snd_soc_ops broxton_dmic_ops = {
+ .startup = broxton_dmic_startup,
+};
+
+static unsigned int channels[] = {
+ 2,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+};
+
+static int bxt_fe_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ /*
+ * on this platform for PCM device we support:
+ * 48Khz
+ * stereo
+ */
+
+ runtime->hw.channels_max = 2;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+
+ return 0;
+}
+
+static const struct snd_soc_ops broxton_rt286_fe_ops = {
+ .startup = bxt_fe_startup,
+};
+
/* broxton digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link broxton_rt298_dais[] = {
/* Front End DAI links */
@@ -182,8 +281,10 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = {
.dynamic = 1,
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
+ .init = broxton_rt298_fe_init,
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_playback = 1,
+ .ops = &broxton_rt286_fe_ops,
},
[BXT_DPCM_AUDIO_CP]
{
@@ -197,6 +298,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = {
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
+ .ops = &broxton_rt286_fe_ops,
},
[BXT_DPCM_AUDIO_REF_CP]
{
@@ -211,6 +313,20 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = {
.nonatomic = 1,
.dynamic = 1,
},
+ [BXT_DPCM_AUDIO_DMIC_CP]
+ {
+ .name = "Bxt Audio DMIC cap",
+ .stream_name = "dmiccap",
+ .cpu_dai_name = "DMIC Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ .ops = &broxton_dmic_ops,
+ },
[BXT_DPCM_AUDIO_HDMI1_PB]
{
.name = "Bxt HDMI Port1",
@@ -276,6 +392,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = {
.codec_name = "dmic-codec",
.codec_dai_name = "dmic-hifi",
.platform_name = "0000:00:0e.0",
+ .be_hw_params_fixup = broxton_dmic_fixup,
.ignore_suspend = 1,
.dpcm_capture = 1,
.no_pcm = 1,
@@ -341,6 +458,7 @@ static struct platform_driver broxton_audio = {
.probe = broxton_audio_probe,
.driver = {
.name = "bxt_alc298s_i2s",
+ .pm = &snd_soc_pm_ops,
},
};
module_platform_driver(broxton_audio)
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index d7ef292c402d..56056ed7fcfd 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -30,6 +30,7 @@
#include <sound/jack.h>
#include "../../codecs/rt5645.h"
#include "../atom/sst-atom-controls.h"
+#include "../common/sst-acpi.h"
#define CHT_PLAT_CLK_3_HZ 19200000
#define CHT_CODEC_DAI "rt5645-aif1"
@@ -340,10 +341,13 @@ static struct snd_soc_card snd_soc_card_chtrt5650 = {
};
static struct cht_acpi_card snd_soc_cards[] = {
+ {"10EC5640", CODEC_TYPE_RT5645, &snd_soc_card_chtrt5645},
{"10EC5645", CODEC_TYPE_RT5645, &snd_soc_card_chtrt5645},
{"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650},
};
+static char cht_rt5640_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */
+
static int snd_cht_mc_probe(struct platform_device *pdev)
{
int ret_val = 0;
@@ -351,6 +355,9 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
struct cht_mc_private *drv;
struct snd_soc_card *card = snd_soc_cards[0].soc_card;
char codec_name[16];
+ struct sst_acpi_mach *mach;
+ const char *i2c_name = NULL;
+ int dai_index = 0;
drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
if (!drv)
@@ -366,12 +373,23 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
}
}
card->dev = &pdev->dev;
+ mach = card->dev->platform_data;
sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id);
/* set correct codec name */
for (i = 0; i < ARRAY_SIZE(cht_dailink); i++)
- if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00"))
+ if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00")) {
card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL);
+ dai_index = i;
+ }
+
+ /* fixup codec name based on HID */
+ i2c_name = sst_acpi_find_name_from_hid(mach->id);
+ if (i2c_name != NULL) {
+ snprintf(cht_rt5640_codec_name, sizeof(cht_rt5640_codec_name),
+ "%s%s", "i2c-", i2c_name);
+ cht_dailink[dai_index].codec_name = cht_rt5640_codec_name;
+ }
snd_soc_card_set_drvdata(card, drv);
ret_val = devm_snd_soc_register_card(&pdev->dev, card);
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index d2808652b974..25db5be7fdfa 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -23,12 +23,15 @@
#include <sound/soc.h>
#include "../../codecs/nau8825.h"
#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi"
#define SKL_MAXIM_CODEC_DAI "HiFi"
+#define DMIC_CH(p) p->list[p->count-1]
static struct snd_soc_jack skylake_headset;
static struct snd_soc_card skylake_audio_card;
+static const struct snd_pcm_hw_constraint_list *dmic_constraints;
struct skl_hdmi_pcm {
struct list_head head;
@@ -339,7 +342,7 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
- if (params_channels(params) == 2)
+ if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
channels->min = channels->max = 2;
else
channels->min = channels->max = 4;
@@ -357,13 +360,23 @@ static struct snd_pcm_hw_constraint_list constraints_dmic_channels = {
.mask = 0,
};
+static const unsigned int dmic_2ch[] = {
+ 2,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = {
+ .count = ARRAY_SIZE(dmic_2ch),
+ .list = dmic_2ch,
+ .mask = 0,
+};
+
static int skylake_dmic_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.channels_max = 4;
+ runtime->hw.channels_max = DMIC_CH(dmic_constraints);
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_dmic_channels);
+ dmic_constraints);
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
@@ -382,8 +395,22 @@ static struct snd_pcm_hw_constraint_list constraints_16000 = {
.list = rates_16000,
};
+static const unsigned int ch_mono[] = {
+ 1,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_refcap = {
+ .count = ARRAY_SIZE(ch_mono),
+ .list = ch_mono,
+};
+
static int skylake_refcap_startup(struct snd_pcm_substream *substream)
{
+ substream->runtime->hw.channels_max = 1;
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_refcap);
+
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_16000);
@@ -610,6 +637,7 @@ static struct snd_soc_card skylake_audio_card = {
static int skylake_audio_probe(struct platform_device *pdev)
{
struct skl_nau8825_private *ctx;
+ struct skl_machine_pdata *pdata;
ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
if (!ctx)
@@ -620,15 +648,27 @@ static int skylake_audio_probe(struct platform_device *pdev)
skylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
+ pdata = dev_get_drvdata(&pdev->dev);
+ if (pdata)
+ dmic_constraints = pdata->dmic_num == 2 ?
+ &constraints_dmic_2ch : &constraints_dmic_channels;
+
return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
}
+static const struct platform_device_id skl_board_ids[] = {
+ { .name = "skl_n88l25_m98357a" },
+ { .name = "kbl_n88l25_m98357a" },
+ { }
+};
+
static struct platform_driver skylake_audio = {
.probe = skylake_audio_probe,
.driver = {
- .name = "skl_nau88l25_max98357a_i2s",
+ .name = "skl_n88l25_m98357a",
.pm = &snd_soc_pm_ops,
},
+ .id_table = skl_board_ids,
};
module_platform_driver(skylake_audio)
@@ -637,4 +677,5 @@ module_platform_driver(skylake_audio)
MODULE_DESCRIPTION("Audio Machine driver-NAU88L25 & MAX98357A in I2S mode");
MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com");
MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:skl_nau88l25_max98357a_i2s");
+MODULE_ALIAS("platform:skl_n88l25_m98357a");
+MODULE_ALIAS("platform:kbl_n88l25_m98357a");
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index e19aa99c4f72..69c5d5da4e86 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -27,12 +27,15 @@
#include <sound/pcm_params.h>
#include "../../codecs/nau8825.h"
#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi"
#define SKL_SSM_CODEC_DAI "ssm4567-hifi"
+#define DMIC_CH(p) p->list[p->count-1]
static struct snd_soc_jack skylake_headset;
static struct snd_soc_card skylake_audio_card;
+static const struct snd_pcm_hw_constraint_list *dmic_constraints;
struct skl_hdmi_pcm {
struct list_head head;
@@ -367,7 +370,7 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
- if (params_channels(params) == 2)
+ if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
channels->min = channels->max = 2;
else
channels->min = channels->max = 4;
@@ -405,13 +408,23 @@ static struct snd_pcm_hw_constraint_list constraints_dmic_channels = {
.mask = 0,
};
+static const unsigned int dmic_2ch[] = {
+ 2,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = {
+ .count = ARRAY_SIZE(dmic_2ch),
+ .list = dmic_2ch,
+ .mask = 0,
+};
+
static int skylake_dmic_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.channels_max = 4;
+ runtime->hw.channels_max = DMIC_CH(dmic_constraints);
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- &constraints_dmic_channels);
+ dmic_constraints);
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
@@ -430,8 +443,22 @@ static struct snd_pcm_hw_constraint_list constraints_16000 = {
.list = rates_16000,
};
+static const unsigned int ch_mono[] = {
+ 1,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_refcap = {
+ .count = ARRAY_SIZE(ch_mono),
+ .list = ch_mono,
+};
+
static int skylake_refcap_startup(struct snd_pcm_substream *substream)
{
+ substream->runtime->hw.channels_max = 1;
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_refcap);
+
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_16000);
@@ -662,6 +689,7 @@ static struct snd_soc_card skylake_audio_card = {
static int skylake_audio_probe(struct platform_device *pdev)
{
struct skl_nau88125_private *ctx;
+ struct skl_machine_pdata *pdata;
ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
if (!ctx)
@@ -672,15 +700,27 @@ static int skylake_audio_probe(struct platform_device *pdev)
skylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
+ pdata = dev_get_drvdata(&pdev->dev);
+ if (pdata)
+ dmic_constraints = pdata->dmic_num == 2 ?
+ &constraints_dmic_2ch : &constraints_dmic_channels;
+
return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
}
+static const struct platform_device_id skl_board_ids[] = {
+ { .name = "skl_n88l25_s4567" },
+ { .name = "kbl_n88l25_s4567" },
+ { }
+};
+
static struct platform_driver skylake_audio = {
.probe = skylake_audio_probe,
.driver = {
- .name = "skl_nau88l25_ssm4567_i2s",
+ .name = "skl_n88l25_s4567",
.pm = &snd_soc_pm_ops,
},
+ .id_table = skl_board_ids,
};
module_platform_driver(skylake_audio)
@@ -693,4 +733,5 @@ MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>");
MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>");
MODULE_DESCRIPTION("Intel Audio Machine driver for SKL with NAU88L25 and SSM4567 in I2S Mode");
MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:skl_nau88l25_ssm4567_i2s");
+MODULE_ALIAS("platform:skl_n88l25_s4567");
+MODULE_ALIAS("platform:kbl_n88l25_s4567");
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 426b48233fdb..88c61e8cb87f 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -505,12 +505,20 @@ static int skylake_audio_probe(struct platform_device *pdev)
return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286);
}
+static const struct platform_device_id skl_board_ids[] = {
+ { .name = "skl_alc286s_i2s" },
+ { .name = "kbl_alc286s_i2s" },
+ { }
+};
+
static struct platform_driver skylake_audio = {
.probe = skylake_audio_probe,
.driver = {
.name = "skl_alc286s_i2s",
.pm = &snd_soc_pm_ops,
},
+ .id_table = skl_board_ids,
+
};
module_platform_driver(skylake_audio)
@@ -520,3 +528,4 @@ MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah@intel.com>");
MODULE_DESCRIPTION("Intel SST Audio for Skylake");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:skl_alc286s_i2s");
+MODULE_ALIAS("platform:kbl_alc286s_i2s");
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index fbbb25c2ceed..1a35149bcad7 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -2,9 +2,9 @@ snd-soc-sst-dsp-objs := sst-dsp.o
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
+snd-soc-sst-firmware-objs := 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-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o
+obj-$(CONFIG_SND_SOC_INTEL_SST_FIRMWARE) += snd-soc-sst-firmware.o
diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h
index 8398cb227ba9..5d2949324d0e 100644
--- a/sound/soc/intel/common/sst-acpi.h
+++ b/sound/soc/intel/common/sst-acpi.h
@@ -20,7 +20,7 @@
#if IS_ENABLED(CONFIG_ACPI)
const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]);
#else
-inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN])
+static inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN])
{
return NULL;
}
@@ -40,6 +40,6 @@ struct sst_acpi_mach {
/* board name */
const char *board;
- void (*machine_quirk)(void);
+ struct sst_acpi_mach * (*machine_quirk)(void *arg);
void *pdata;
};
diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h
index 97dc1ae05e69..d13c84364c3c 100644
--- a/sound/soc/intel/common/sst-dsp-priv.h
+++ b/sound/soc/intel/common/sst-dsp-priv.h
@@ -383,10 +383,6 @@ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset,
u32 index, void *private);
void sst_mem_block_unregister_all(struct sst_dsp *dsp);
-/* Create/Free DMA resources */
-int sst_dma_new(struct sst_dsp *sst);
-void sst_dma_free(struct sst_dma *dma);
-
u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset,
enum sst_mem_type type);
#endif
diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c
index b5bbdf4fe93a..c00ede4ea4d7 100644
--- a/sound/soc/intel/common/sst-dsp.c
+++ b/sound/soc/intel/common/sst-dsp.c
@@ -285,7 +285,7 @@ int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask,
}
reg = sst_dsp_shim_read_unlocked(ctx, offset);
- dev_info(ctx->dev, "FW Poll Status: reg=%#x %s %s\n", reg, operation,
+ dev_dbg(ctx->dev, "FW Poll Status: reg=%#x %s %s\n", reg, operation,
(time < timeout) ? "successful" : "timedout");
ret = time < timeout ? 0 : -ETIME;
@@ -420,73 +420,6 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes)
}
EXPORT_SYMBOL_GPL(sst_dsp_inbox_read);
-#ifdef CONFIG_DW_DMAC_CORE
-struct sst_dsp *sst_dsp_new(struct device *dev,
- struct sst_dsp_device *sst_dev, struct sst_pdata *pdata)
-{
- struct sst_dsp *sst;
- int err;
-
- dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id);
-
- sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL);
- if (sst == NULL)
- return NULL;
-
- spin_lock_init(&sst->spinlock);
- mutex_init(&sst->mutex);
- sst->dev = dev;
- sst->dma_dev = pdata->dma_dev;
- sst->thread_context = sst_dev->thread_context;
- sst->sst_dev = sst_dev;
- sst->id = pdata->id;
- sst->irq = pdata->irq;
- sst->ops = sst_dev->ops;
- sst->pdata = pdata;
- INIT_LIST_HEAD(&sst->used_block_list);
- INIT_LIST_HEAD(&sst->free_block_list);
- INIT_LIST_HEAD(&sst->module_list);
- INIT_LIST_HEAD(&sst->fw_list);
- INIT_LIST_HEAD(&sst->scratch_block_list);
-
- /* Initialise SST Audio DSP */
- if (sst->ops->init) {
- err = sst->ops->init(sst, pdata);
- if (err < 0)
- return NULL;
- }
-
- /* Register the ISR */
- err = request_threaded_irq(sst->irq, sst->ops->irq_handler,
- sst_dev->thread, IRQF_SHARED, "AudioDSP", sst);
- if (err)
- goto irq_err;
-
- err = sst_dma_new(sst);
- if (err)
- dev_warn(dev, "sst_dma_new failed %d\n", err);
-
- return sst;
-
-irq_err:
- if (sst->ops->free)
- sst->ops->free(sst);
-
- return NULL;
-}
-EXPORT_SYMBOL_GPL(sst_dsp_new);
-
-void sst_dsp_free(struct sst_dsp *sst)
-{
- free_irq(sst->irq, sst);
- if (sst->ops->free)
- sst->ops->free(sst);
-
- sst_dma_free(sst->dma);
-}
-EXPORT_SYMBOL_GPL(sst_dsp_free);
-#endif
-
/* Module information */
MODULE_AUTHOR("Liam Girdwood");
MODULE_DESCRIPTION("Intel SST Core");
diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h
index 0b84c719ec48..859f0de00339 100644
--- a/sound/soc/intel/common/sst-dsp.h
+++ b/sound/soc/intel/common/sst-dsp.h
@@ -216,7 +216,7 @@ struct sst_pdata {
void *dsp;
};
-#ifdef CONFIG_DW_DMAC_CORE
+#if IS_ENABLED(CONFIG_DW_DMAC_CORE)
/* Initialization */
struct sst_dsp *sst_dsp_new(struct device *dev,
struct sst_dsp_device *sst_dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 25993527370b..a086c35f91bb 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -1211,3 +1211,71 @@ u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset,
}
}
EXPORT_SYMBOL_GPL(sst_dsp_get_offset);
+
+struct sst_dsp *sst_dsp_new(struct device *dev,
+ struct sst_dsp_device *sst_dev, struct sst_pdata *pdata)
+{
+ struct sst_dsp *sst;
+ int err;
+
+ dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id);
+
+ sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL);
+ if (sst == NULL)
+ return NULL;
+
+ spin_lock_init(&sst->spinlock);
+ mutex_init(&sst->mutex);
+ sst->dev = dev;
+ sst->dma_dev = pdata->dma_dev;
+ sst->thread_context = sst_dev->thread_context;
+ sst->sst_dev = sst_dev;
+ sst->id = pdata->id;
+ sst->irq = pdata->irq;
+ sst->ops = sst_dev->ops;
+ sst->pdata = pdata;
+ INIT_LIST_HEAD(&sst->used_block_list);
+ INIT_LIST_HEAD(&sst->free_block_list);
+ INIT_LIST_HEAD(&sst->module_list);
+ INIT_LIST_HEAD(&sst->fw_list);
+ INIT_LIST_HEAD(&sst->scratch_block_list);
+
+ /* Initialise SST Audio DSP */
+ if (sst->ops->init) {
+ err = sst->ops->init(sst, pdata);
+ if (err < 0)
+ return NULL;
+ }
+
+ /* Register the ISR */
+ err = request_threaded_irq(sst->irq, sst->ops->irq_handler,
+ sst_dev->thread, IRQF_SHARED, "AudioDSP", sst);
+ if (err)
+ goto irq_err;
+
+ err = sst_dma_new(sst);
+ if (err)
+ dev_warn(dev, "sst_dma_new failed %d\n", err);
+
+ return sst;
+
+irq_err:
+ if (sst->ops->free)
+ sst->ops->free(sst);
+
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_new);
+
+void sst_dsp_free(struct sst_dsp *sst)
+{
+ free_irq(sst->irq, sst);
+ if (sst->ops->free)
+ sst->ops->free(sst);
+
+ sst_dma_free(sst->dma);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_free);
+
+MODULE_DESCRIPTION("Intel SST Firmware Loader");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c
index 994256b39b9c..3154525c2b83 100644
--- a/sound/soc/intel/haswell/sst-haswell-pcm.c
+++ b/sound/soc/intel/haswell/sst-haswell-pcm.c
@@ -819,7 +819,6 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream)
mutex_lock(&pcm_data->mutex);
pm_runtime_get_sync(pdata->dev);
- snd_soc_pcm_set_drvdata(rtd, pcm_data);
pcm_data->substream = substream;
snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware);
diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index c28f5d0e1d99..60fbc9bbe473 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
# Skylake IPC Support
snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \
- skl-sst.o bxt-sst.o
+ skl-sst.o bxt-sst.o skl-sst-utils.o
obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o
diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index 965ce40ce752..2663781278aa 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -37,11 +37,19 @@
#define BXT_ADSP_SRAM1_BASE 0xA0000
+#define BXT_INSTANCE_ID 0
+#define BXT_BASE_FW_MODULE_ID 0
+
static unsigned int bxt_get_errorcode(struct sst_dsp *ctx)
{
return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE);
}
+/*
+ * First boot sequence has some extra steps. Core 0 waits for power
+ * status on core 1, so power up core 1 also momentarily, keep it in
+ * reset/stall and then turn it off
+ */
static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
const void *fwdata, u32 fwsize)
{
@@ -49,7 +57,7 @@ static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
u32 reg;
stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab);
- if (stream_tag < 0) {
+ if (stream_tag <= 0) {
dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n",
stream_tag);
return stream_tag;
@@ -58,17 +66,27 @@ static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
ctx->dsp_ops.stream_tag = stream_tag;
memcpy(ctx->dmab.area, fwdata, fwsize);
- /* Purge FW request */
+ /* Step 1: Power up core 0 and core1 */
+ ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK |
+ SKL_DSP_CORE_MASK(1));
+ if (ret < 0) {
+ dev_err(ctx->dev, "dsp core0/1 power up failed\n");
+ goto base_fw_load_failed;
+ }
+
+ /* Step 2: Purge FW request */
sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY |
- BXT_IPC_PURGE_FW | (stream_tag - 1));
+ (BXT_IPC_PURGE_FW | ((stream_tag - 1) << 9)));
- ret = skl_dsp_enable_core(ctx);
+ /* Step 3: Unset core0 reset state & unstall/run core0 */
+ ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK);
if (ret < 0) {
- dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret);
+ dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret);
ret = -EIO;
goto base_fw_load_failed;
}
+ /* Step 4: Wait for DONE Bit */
for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
reg = sst_dsp_shim_read(ctx, SKL_ADSP_REG_HIPCIE);
@@ -88,10 +106,18 @@ static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
SKL_ADSP_REG_HIPCIE_DONE);
}
- /* enable Interrupt */
+ /* Step 5: power down core1 */
+ ret = skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1));
+ if (ret < 0) {
+ dev_err(ctx->dev, "dsp core1 power down failed\n");
+ goto base_fw_load_failed;
+ }
+
+ /* Step 6: Enable Interrupt */
skl_ipc_int_enable(ctx);
skl_ipc_op_int_enable(ctx);
+ /* Step 7: Wait for ROM init */
for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
if (SKL_FW_INIT ==
(sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS) &
@@ -112,7 +138,8 @@ static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
base_fw_load_failed:
ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag);
- skl_dsp_disable_core(ctx);
+ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1));
+ skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
return ret;
}
@@ -130,23 +157,41 @@ static int sst_transfer_fw_host_dma(struct sst_dsp *ctx)
return ret;
}
+#define BXT_ADSP_FW_BIN_HDR_OFFSET 0x2000
+
static int bxt_load_base_firmware(struct sst_dsp *ctx)
{
- const struct firmware *fw = NULL;
+ struct firmware stripped_fw;
struct skl_sst *skl = ctx->thread_context;
int ret;
- ret = request_firmware(&fw, ctx->fw_name, ctx->dev);
+ ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev);
if (ret < 0) {
dev_err(ctx->dev, "Request firmware failed %d\n", ret);
goto sst_load_base_firmware_failed;
}
- ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+ /* check for extended manifest */
+ if (ctx->fw == NULL)
+ goto sst_load_base_firmware_failed;
+
+ ret = snd_skl_parse_uuids(ctx, BXT_ADSP_FW_BIN_HDR_OFFSET);
+ if (ret < 0)
+ goto sst_load_base_firmware_failed;
+
+ stripped_fw.data = ctx->fw->data;
+ stripped_fw.size = ctx->fw->size;
+ skl_dsp_strip_extended_manifest(&stripped_fw);
+
+ ret = sst_bxt_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
/* Retry Enabling core and ROM load. Retry seemed to help */
if (ret < 0) {
- ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+ ret = sst_bxt_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
if (ret < 0) {
+ dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+ sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+ sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+
dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret);
goto sst_load_base_firmware_failed;
}
@@ -159,83 +204,135 @@ static int bxt_load_base_firmware(struct sst_dsp *ctx)
sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
- skl_dsp_disable_core(ctx);
+ skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
} else {
dev_dbg(ctx->dev, "Firmware download successful\n");
ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
if (ret == 0) {
dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n");
- skl_dsp_disable_core(ctx);
+ skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
ret = -EIO;
} else {
- skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
ret = 0;
+ skl->fw_loaded = true;
}
}
sst_load_base_firmware_failed:
- release_firmware(fw);
+ release_firmware(ctx->fw);
return ret;
}
-static int bxt_set_dsp_D0(struct sst_dsp *ctx)
+static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id)
{
struct skl_sst *skl = ctx->thread_context;
int ret;
+ struct skl_ipc_dxstate_info dx;
+ unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
- skl->boot_complete = false;
-
- ret = skl_dsp_enable_core(ctx);
- if (ret < 0) {
- dev_err(ctx->dev, "enable dsp core failed ret: %d\n", ret);
+ if (skl->fw_loaded == false) {
+ skl->boot_complete = false;
+ ret = bxt_load_base_firmware(ctx);
+ if (ret < 0)
+ dev_err(ctx->dev, "reload fw failed: %d\n", ret);
return ret;
}
- /* enable interrupt */
- skl_ipc_int_enable(ctx);
- skl_ipc_op_int_enable(ctx);
+ /* If core 0 is being turned on, turn on core 1 as well */
+ if (core_id == SKL_DSP_CORE0_ID)
+ ret = skl_dsp_core_power_up(ctx, core_mask |
+ SKL_DSP_CORE_MASK(1));
+ else
+ ret = skl_dsp_core_power_up(ctx, core_mask);
+
+ if (ret < 0)
+ goto err;
+
+ if (core_id == SKL_DSP_CORE0_ID) {
+
+ /*
+ * Enable interrupt after SPA is set and before
+ * DSP is unstalled
+ */
+ skl_ipc_int_enable(ctx);
+ skl_ipc_op_int_enable(ctx);
+ skl->boot_complete = false;
+ }
- ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
- msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
- if (ret == 0) {
- dev_err(ctx->dev, "ipc: error DSP boot timeout\n");
- dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
- sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
- sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
- return -EIO;
+ ret = skl_dsp_start_core(ctx, core_mask);
+ if (ret < 0)
+ goto err;
+
+ if (core_id == SKL_DSP_CORE0_ID) {
+ ret = wait_event_timeout(skl->boot_wait,
+ skl->boot_complete,
+ msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+
+ /* If core 1 was turned on for booting core 0, turn it off */
+ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1));
+ if (ret == 0) {
+ dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__);
+ dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+ sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+ sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+ dev_err(ctx->dev, "Failed to set core0 to D0 state\n");
+ ret = -EIO;
+ goto err;
+ }
+ }
+
+ /* Tell FW if additional core in now On */
+
+ if (core_id != SKL_DSP_CORE0_ID) {
+ dx.core_mask = core_mask;
+ dx.dx_mask = core_mask;
+
+ ret = skl_ipc_set_dx(&skl->ipc, BXT_INSTANCE_ID,
+ BXT_BASE_FW_MODULE_ID, &dx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "IPC set_dx for core %d fail: %d\n",
+ core_id, ret);
+ goto err;
+ }
}
- skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+ skl->cores.state[core_id] = SKL_DSP_RUNNING;
return 0;
+err:
+ if (core_id == SKL_DSP_CORE0_ID)
+ core_mask |= SKL_DSP_CORE_MASK(1);
+ skl_dsp_disable_core(ctx, core_mask);
+
+ return ret;
}
-static int bxt_set_dsp_D3(struct sst_dsp *ctx)
+static int bxt_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id)
{
+ int ret;
struct skl_ipc_dxstate_info dx;
struct skl_sst *skl = ctx->thread_context;
- int ret = 0;
+ unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
- if (!is_skl_dsp_running(ctx))
- return ret;
-
- dx.core_mask = SKL_DSP_CORE0_MASK;
+ dx.core_mask = core_mask;
dx.dx_mask = SKL_IPC_D3_MASK;
- ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID,
- SKL_BASE_FW_MODULE_ID, &dx);
- if (ret < 0) {
- dev_err(ctx->dev, "Failed to set DSP to D3 state: %d\n", ret);
- return ret;
- }
+ dev_dbg(ctx->dev, "core mask=%x dx_mask=%x\n",
+ dx.core_mask, dx.dx_mask);
+
+ ret = skl_ipc_set_dx(&skl->ipc, BXT_INSTANCE_ID,
+ BXT_BASE_FW_MODULE_ID, &dx);
+ if (ret < 0)
+ dev_err(ctx->dev,
+ "Failed to set DSP to D3:core id = %d;Continue reset\n",
+ core_id);
- ret = skl_dsp_disable_core(ctx);
+ ret = skl_dsp_disable_core(ctx, core_mask);
if (ret < 0) {
- dev_err(ctx->dev, "disbale dsp core failed: %d\n", ret);
- ret = -EIO;
+ dev_err(ctx->dev, "Failed to disable core %d", ret);
+ return ret;
}
-
- skl_dsp_set_state_locked(ctx, SKL_DSP_RESET);
+ skl->cores.state[core_id] = SKL_DSP_RESET;
return 0;
}
@@ -274,6 +371,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
skl->dev = dev;
skl_dev.thread_context = skl;
+ INIT_LIST_HEAD(&skl->uuid_list);
skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq);
if (!skl->dsp) {
@@ -291,10 +389,12 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ),
SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
+ INIT_LIST_HEAD(&sst->module_list);
ret = skl_ipc_init(dev, skl);
if (ret)
return ret;
+ skl->cores.count = 2;
skl->boot_complete = false;
init_waitqueue_head(&skl->boot_wait);
@@ -304,6 +404,8 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
return ret;
}
+ skl_dsp_init_core_state(sst);
+
if (dsp)
*dsp = skl;
@@ -314,6 +416,7 @@ EXPORT_SYMBOL_GPL(bxt_sst_dsp_init);
void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
{
+ skl_freeup_uuid_list(ctx);
skl_ipc_free(&ctx->ipc);
ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 226db84ba20f..44ab595ce21a 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -206,6 +206,12 @@ static const struct skl_dsp_ops dsp_ops[] = {
.cleanup = skl_sst_dsp_cleanup
},
{
+ .id = 0x9d71,
+ .loader_ops = skl_get_loader_ops,
+ .init = skl_sst_dsp_init,
+ .cleanup = skl_sst_dsp_cleanup
+ },
+ {
.id = 0x5a98,
.loader_ops = bxt_get_loader_ops,
.init = bxt_sst_dsp_init,
@@ -730,7 +736,7 @@ static int skl_set_module_format(struct skl_sst *ctx,
dev_dbg(ctx->dev, "Module type=%d config size: %d bytes\n",
module_config->id.module_id, param_size);
- print_hex_dump(KERN_DEBUG, "Module params:", DUMP_PREFIX_OFFSET, 8, 4,
+ print_hex_dump_debug("Module params:", DUMP_PREFIX_OFFSET, 8, 4,
*param_data, param_size, false);
return 0;
}
@@ -1046,7 +1052,7 @@ int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
dev_dbg(ctx->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id);
- /* If pipe is not started, do not try to stop the pipe in FW. */
+ /* If pipe is started, do stop the pipe in FW. */
if (pipe->state > SKL_PIPE_STARTED) {
ret = skl_set_pipe_state(ctx, pipe, PPL_PAUSED);
if (ret < 0) {
@@ -1055,18 +1061,20 @@ int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
}
pipe->state = SKL_PIPE_PAUSED;
- } else {
- /* If pipe was not created in FW, do not try to delete it */
- if (pipe->state < SKL_PIPE_CREATED)
- return 0;
+ }
- ret = skl_ipc_delete_pipeline(&ctx->ipc, pipe->ppl_id);
- if (ret < 0)
- dev_err(ctx->dev, "Failed to delete pipeline\n");
+ /* If pipe was not created in FW, do not try to delete it */
+ if (pipe->state < SKL_PIPE_CREATED)
+ return 0;
- pipe->state = SKL_PIPE_INVALID;
+ ret = skl_ipc_delete_pipeline(&ctx->ipc, pipe->ppl_id);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Failed to delete pipeline\n");
+ return ret;
}
+ pipe->state = SKL_PIPE_INVALID;
+
return ret;
}
@@ -1125,7 +1133,30 @@ int skl_stop_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
return ret;
}
- pipe->state = SKL_PIPE_CREATED;
+ pipe->state = SKL_PIPE_PAUSED;
+
+ return 0;
+}
+
+/*
+ * Reset the pipeline by sending set pipe state IPC this will reset the DMA
+ * from the DSP side
+ */
+int skl_reset_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
+{
+ int ret;
+
+ /* If pipe was not created in FW, do not try to pause or delete */
+ if (pipe->state < SKL_PIPE_PAUSED)
+ return 0;
+
+ ret = skl_set_pipe_state(ctx, pipe, PPL_RESET);
+ if (ret < 0) {
+ dev_dbg(ctx->dev, "Failed to reset pipe ret=%d\n", ret);
+ return ret;
+ }
+
+ pipe->state = SKL_PIPE_RESET;
return 0;
}
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 7d73648e5f9a..3f8e6f0b7eb5 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -17,6 +17,7 @@
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
*/
+#include <linux/pci.h>
#include "skl.h"
/* Unique identification for getting NHLT blobs */
@@ -149,6 +150,45 @@ struct nhlt_specific_cfg
return NULL;
}
+int skl_get_dmic_geo(struct skl *skl)
+{
+ struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
+ struct nhlt_endpoint *epnt;
+ struct nhlt_dmic_array_config *cfg;
+ struct device *dev = &skl->pci->dev;
+ unsigned int dmic_geo = 0;
+ u8 j;
+
+ epnt = (struct nhlt_endpoint *)nhlt->desc;
+
+ for (j = 0; j < nhlt->endpoint_count; j++) {
+ if (epnt->linktype == NHLT_LINK_DMIC) {
+ cfg = (struct nhlt_dmic_array_config *)
+ (epnt->config.caps);
+ switch (cfg->array_type) {
+ case NHLT_MIC_ARRAY_2CH_SMALL:
+ case NHLT_MIC_ARRAY_2CH_BIG:
+ dmic_geo |= MIC_ARRAY_2CH;
+ break;
+
+ case NHLT_MIC_ARRAY_4CH_1ST_GEOM:
+ case NHLT_MIC_ARRAY_4CH_L_SHAPED:
+ case NHLT_MIC_ARRAY_4CH_2ND_GEOM:
+ dmic_geo |= MIC_ARRAY_4CH;
+ break;
+
+ default:
+ dev_warn(dev, "undefined DMIC array_type 0x%0x\n",
+ cfg->array_type);
+
+ }
+ }
+ epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length);
+ }
+
+ return dmic_geo;
+}
+
static void skl_nhlt_trim_space(struct skl *skl)
{
char *s = skl->tplg_name;
diff --git a/sound/soc/intel/skylake/skl-nhlt.h b/sound/soc/intel/skylake/skl-nhlt.h
index 3769f9fefe2b..116534e7b3c5 100644
--- a/sound/soc/intel/skylake/skl-nhlt.h
+++ b/sound/soc/intel/skylake/skl-nhlt.h
@@ -103,4 +103,26 @@ struct nhlt_resource_desc {
u64 length;
} __packed;
+#define MIC_ARRAY_2CH 2
+#define MIC_ARRAY_4CH 4
+
+struct nhlt_tdm_config {
+ u8 virtual_slot;
+ u8 config_type;
+} __packed;
+
+struct nhlt_dmic_array_config {
+ struct nhlt_tdm_config tdm_config;
+ u8 array_type;
+} __packed;
+
+enum {
+ NHLT_MIC_ARRAY_2CH_SMALL = 0xa,
+ NHLT_MIC_ARRAY_2CH_BIG = 0xb,
+ NHLT_MIC_ARRAY_4CH_1ST_GEOM = 0xc,
+ NHLT_MIC_ARRAY_4CH_L_SHAPED = 0xd,
+ NHLT_MIC_ARRAY_4CH_2ND_GEOM = 0xe,
+ NHLT_MIC_ARRAY_VENDOR_DEFINED = 0xf,
+};
+
#endif
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 7c81b31748ff..6e05bf8622f7 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -227,16 +227,25 @@ static int skl_pcm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
+ struct skl *skl = get_skl_ctx(dai->dev);
unsigned int format_val;
int err;
+ struct skl_module_cfg *mconfig;
dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+ mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
+
format_val = skl_get_format(substream, dai);
dev_dbg(dai->dev, "stream_tag=%d formatvalue=%d\n",
hdac_stream(stream)->stream_tag, format_val);
snd_hdac_stream_reset(hdac_stream(stream));
+ /* In case of XRUN recovery, reset the FW pipe to clean state */
+ if (mconfig && (substream->runtime->status->state ==
+ SNDRV_PCM_STATE_XRUN))
+ skl_reset_pipe(skl->skl_sst, mconfig->pipe);
+
err = snd_hdac_stream_set_params(hdac_stream(stream), format_val);
if (err < 0)
return err;
@@ -521,6 +530,8 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream,
struct skl_dma_params *dma_params;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct hdac_ext_link *link;
+ struct skl *skl = get_skl_ctx(dai->dev);
+ struct skl_module_cfg *mconfig = NULL;
dma_params = (struct skl_dma_params *)
snd_soc_dai_get_dma_data(codec_dai, substream);
@@ -535,6 +546,12 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream,
snd_hdac_ext_link_stream_reset(link_dev);
+ /* In case of XRUN recovery, reset the FW pipe to clean state */
+ mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream);
+ if (mconfig && (substream->runtime->status->state ==
+ SNDRV_PCM_STATE_XRUN))
+ skl_reset_pipe(skl->skl_sst, mconfig->pipe);
+
snd_hdac_ext_link_stream_setup(link_dev, format_val);
snd_hdac_ext_link_set_stream_id(link, hdac_stream(link_dev)->stream_tag);
@@ -1009,51 +1026,11 @@ static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream,
return 0;
}
-/* calculate runtime delay from LPIB */
-static int skl_get_delay_from_lpib(struct hdac_ext_bus *ebus,
- struct hdac_ext_stream *sstream,
- unsigned int pos)
-{
- struct hdac_bus *bus = ebus_to_hbus(ebus);
- struct hdac_stream *hstream = hdac_stream(sstream);
- struct snd_pcm_substream *substream = hstream->substream;
- int stream = substream->stream;
- unsigned int lpib_pos = snd_hdac_stream_get_pos_lpib(hstream);
- int delay;
-
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- delay = pos - lpib_pos;
- else
- delay = lpib_pos - pos;
-
- if (delay < 0) {
- if (delay >= hstream->delay_negative_threshold)
- delay = 0;
- else
- delay += hstream->bufsize;
- }
-
- if (hstream->bufsize == delay)
- delay = 0;
-
- if (delay >= hstream->period_bytes) {
- dev_info(bus->dev,
- "Unstable LPIB (%d >= %d); disabling LPIB delay counting\n",
- delay, hstream->period_bytes);
- delay = 0;
- }
-
- return bytes_to_frames(substream->runtime, delay);
-}
-
-static unsigned int skl_get_position(struct hdac_ext_stream *hstream,
- int codec_delay)
+static snd_pcm_uframes_t skl_platform_pcm_pointer
+ (struct snd_pcm_substream *substream)
{
- struct hdac_stream *hstr = hdac_stream(hstream);
- struct snd_pcm_substream *substream = hstr->substream;
- struct hdac_ext_bus *ebus;
+ struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream);
unsigned int pos;
- int delay;
/* use the position buffer as default */
pos = snd_hdac_stream_get_pos_posbuf(hdac_stream(hstream));
@@ -1061,23 +1038,7 @@ static unsigned int skl_get_position(struct hdac_ext_stream *hstream,
if (pos >= hdac_stream(hstream)->bufsize)
pos = 0;
- if (substream->runtime) {
- ebus = get_bus_ctx(substream);
- delay = skl_get_delay_from_lpib(ebus, hstream, pos)
- + codec_delay;
- substream->runtime->delay += delay;
- }
-
- return pos;
-}
-
-static snd_pcm_uframes_t skl_platform_pcm_pointer
- (struct snd_pcm_substream *substream)
-{
- struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream);
-
- return bytes_to_frames(substream->runtime,
- skl_get_position(hstream, 0));
+ return bytes_to_frames(substream->runtime, pos);
}
static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream,
@@ -1180,9 +1141,17 @@ static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd)
static int skl_platform_soc_probe(struct snd_soc_platform *platform)
{
struct hdac_ext_bus *ebus = dev_get_drvdata(platform->dev);
+ struct skl *skl = ebus_to_skl(ebus);
+ int ret;
- if (ebus->ppcap)
- return skl_tplg_init(platform, ebus);
+ if (ebus->ppcap) {
+ ret = skl_tplg_init(platform, ebus);
+ if (ret < 0) {
+ dev_err(platform->dev, "Failed to init topology!\n");
+ return ret;
+ }
+ skl->platform = platform;
+ }
return 0;
}
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index 13c19855ee1a..c3deefab65d6 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -34,33 +34,84 @@ void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state)
mutex_unlock(&ctx->mutex);
}
-static int skl_dsp_core_set_reset_state(struct sst_dsp *ctx)
+/*
+ * Initialize core power state and usage count. To be called after
+ * successful first boot. Hence core 0 will be running and other cores
+ * will be reset
+ */
+void skl_dsp_init_core_state(struct sst_dsp *ctx)
+{
+ struct skl_sst *skl = ctx->thread_context;
+ int i;
+
+ skl->cores.state[SKL_DSP_CORE0_ID] = SKL_DSP_RUNNING;
+ skl->cores.usage_count[SKL_DSP_CORE0_ID] = 1;
+
+ for (i = SKL_DSP_CORE0_ID + 1; i < SKL_DSP_CORES_MAX; i++) {
+ skl->cores.state[i] = SKL_DSP_RESET;
+ skl->cores.usage_count[i] = 0;
+ }
+}
+
+/* Get the mask for all enabled cores */
+unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx)
+{
+ struct skl_sst *skl = ctx->thread_context;
+ unsigned int core_mask, en_cores_mask;
+ u32 val;
+
+ core_mask = SKL_DSP_CORES_MASK(skl->cores.count);
+
+ val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS);
+
+ /* Cores having CPA bit set */
+ en_cores_mask = (val & SKL_ADSPCS_CPA_MASK(core_mask)) >>
+ SKL_ADSPCS_CPA_SHIFT;
+
+ /* And cores having CRST bit cleared */
+ en_cores_mask &= (~val & SKL_ADSPCS_CRST_MASK(core_mask)) >>
+ SKL_ADSPCS_CRST_SHIFT;
+
+ /* And cores having CSTALL bit cleared */
+ en_cores_mask &= (~val & SKL_ADSPCS_CSTALL_MASK(core_mask)) >>
+ SKL_ADSPCS_CSTALL_SHIFT;
+ en_cores_mask &= core_mask;
+
+ dev_dbg(ctx->dev, "DSP enabled cores mask = %x\n", en_cores_mask);
+
+ return en_cores_mask;
+}
+
+static int
+skl_dsp_core_set_reset_state(struct sst_dsp *ctx, unsigned int core_mask)
{
int ret;
/* update bits */
sst_dsp_shim_update_bits_unlocked(ctx,
- SKL_ADSP_REG_ADSPCS, SKL_ADSPCS_CRST_MASK,
- SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK));
+ SKL_ADSP_REG_ADSPCS, SKL_ADSPCS_CRST_MASK(core_mask),
+ SKL_ADSPCS_CRST_MASK(core_mask));
/* poll with timeout to check if operation successful */
ret = sst_dsp_register_poll(ctx,
SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_CRST_MASK,
- SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK),
+ SKL_ADSPCS_CRST_MASK(core_mask),
+ SKL_ADSPCS_CRST_MASK(core_mask),
SKL_DSP_RESET_TO,
"Set reset");
if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) &
- SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) !=
- SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) {
- dev_err(ctx->dev, "Set reset state failed\n");
+ SKL_ADSPCS_CRST_MASK(core_mask)) !=
+ SKL_ADSPCS_CRST_MASK(core_mask)) {
+ dev_err(ctx->dev, "Set reset state failed: core_mask %x\n",
+ core_mask);
ret = -EIO;
}
return ret;
}
-static int skl_dsp_core_unset_reset_state(struct sst_dsp *ctx)
+int skl_dsp_core_unset_reset_state(
+ struct sst_dsp *ctx, unsigned int core_mask)
{
int ret;
@@ -68,152 +119,160 @@ static int skl_dsp_core_unset_reset_state(struct sst_dsp *ctx)
/* update bits */
sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_CRST_MASK, 0);
+ SKL_ADSPCS_CRST_MASK(core_mask), 0);
/* poll with timeout to check if operation successful */
ret = sst_dsp_register_poll(ctx,
SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_CRST_MASK,
+ SKL_ADSPCS_CRST_MASK(core_mask),
0,
SKL_DSP_RESET_TO,
"Unset reset");
if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) &
- SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) != 0) {
- dev_err(ctx->dev, "Unset reset state failed\n");
+ SKL_ADSPCS_CRST_MASK(core_mask)) != 0) {
+ dev_err(ctx->dev, "Unset reset state failed: core_mask %x\n",
+ core_mask);
ret = -EIO;
}
return ret;
}
-static bool is_skl_dsp_core_enable(struct sst_dsp *ctx)
+static bool
+is_skl_dsp_core_enable(struct sst_dsp *ctx, unsigned int core_mask)
{
int val;
bool is_enable;
val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS);
- is_enable = ((val & SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK)) &&
- (val & SKL_ADSPCS_SPA(SKL_DSP_CORES_MASK)) &&
- !(val & SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) &&
- !(val & SKL_ADSPCS_CSTALL(SKL_DSP_CORES_MASK)));
+ is_enable = ((val & SKL_ADSPCS_CPA_MASK(core_mask)) &&
+ (val & SKL_ADSPCS_SPA_MASK(core_mask)) &&
+ !(val & SKL_ADSPCS_CRST_MASK(core_mask)) &&
+ !(val & SKL_ADSPCS_CSTALL_MASK(core_mask)));
+
+ dev_dbg(ctx->dev, "DSP core(s) enabled? %d : core_mask %x\n",
+ is_enable, core_mask);
- dev_dbg(ctx->dev, "DSP core is enabled=%d\n", is_enable);
return is_enable;
}
-static int skl_dsp_reset_core(struct sst_dsp *ctx)
+static int skl_dsp_reset_core(struct sst_dsp *ctx, unsigned int core_mask)
{
/* stall core */
- sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_ADSPCS,
- sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) &
- SKL_ADSPCS_CSTALL(SKL_DSP_CORES_MASK));
+ sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS,
+ SKL_ADSPCS_CSTALL_MASK(core_mask),
+ SKL_ADSPCS_CSTALL_MASK(core_mask));
/* set reset state */
- return skl_dsp_core_set_reset_state(ctx);
+ return skl_dsp_core_set_reset_state(ctx, core_mask);
}
-static int skl_dsp_start_core(struct sst_dsp *ctx)
+int skl_dsp_start_core(struct sst_dsp *ctx, unsigned int core_mask)
{
int ret;
/* unset reset state */
- ret = skl_dsp_core_unset_reset_state(ctx);
- if (ret < 0) {
- dev_dbg(ctx->dev, "dsp unset reset fails\n");
+ ret = skl_dsp_core_unset_reset_state(ctx, core_mask);
+ if (ret < 0)
return ret;
- }
/* run core */
- dev_dbg(ctx->dev, "run core...\n");
- sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_ADSPCS,
- sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) &
- ~SKL_ADSPCS_CSTALL(SKL_DSP_CORES_MASK));
-
- if (!is_skl_dsp_core_enable(ctx)) {
- skl_dsp_reset_core(ctx);
- dev_err(ctx->dev, "DSP core enable failed\n");
+ dev_dbg(ctx->dev, "unstall/run core: core_mask = %x\n", core_mask);
+ sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS,
+ SKL_ADSPCS_CSTALL_MASK(core_mask), 0);
+
+ if (!is_skl_dsp_core_enable(ctx, core_mask)) {
+ skl_dsp_reset_core(ctx, core_mask);
+ dev_err(ctx->dev, "DSP start core failed: core_mask %x\n",
+ core_mask);
ret = -EIO;
}
return ret;
}
-static int skl_dsp_core_power_up(struct sst_dsp *ctx)
+int skl_dsp_core_power_up(struct sst_dsp *ctx, unsigned int core_mask)
{
int ret;
/* update bits */
sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_SPA_MASK, SKL_ADSPCS_SPA(SKL_DSP_CORES_MASK));
+ SKL_ADSPCS_SPA_MASK(core_mask),
+ SKL_ADSPCS_SPA_MASK(core_mask));
/* poll with timeout to check if operation successful */
ret = sst_dsp_register_poll(ctx,
SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_CPA_MASK,
- SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK),
+ SKL_ADSPCS_CPA_MASK(core_mask),
+ SKL_ADSPCS_CPA_MASK(core_mask),
SKL_DSP_PU_TO,
"Power up");
if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) &
- SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK)) !=
- SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK)) {
- dev_err(ctx->dev, "DSP core power up failed\n");
+ SKL_ADSPCS_CPA_MASK(core_mask)) !=
+ SKL_ADSPCS_CPA_MASK(core_mask)) {
+ dev_err(ctx->dev, "DSP core power up failed: core_mask %x\n",
+ core_mask);
ret = -EIO;
}
return ret;
}
-static int skl_dsp_core_power_down(struct sst_dsp *ctx)
+int skl_dsp_core_power_down(struct sst_dsp *ctx, unsigned int core_mask)
{
/* update bits */
sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_SPA_MASK, 0);
+ SKL_ADSPCS_SPA_MASK(core_mask), 0);
/* poll with timeout to check if operation successful */
return sst_dsp_register_poll(ctx,
SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_CPA_MASK,
+ SKL_ADSPCS_CPA_MASK(core_mask),
0,
SKL_DSP_PD_TO,
"Power down");
}
-int skl_dsp_enable_core(struct sst_dsp *ctx)
+int skl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core_mask)
{
int ret;
/* power up */
- ret = skl_dsp_core_power_up(ctx);
+ ret = skl_dsp_core_power_up(ctx, core_mask);
if (ret < 0) {
- dev_dbg(ctx->dev, "dsp core power up failed\n");
+ dev_err(ctx->dev, "dsp core power up failed: core_mask %x\n",
+ core_mask);
return ret;
}
- return skl_dsp_start_core(ctx);
+ return skl_dsp_start_core(ctx, core_mask);
}
-int skl_dsp_disable_core(struct sst_dsp *ctx)
+int skl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core_mask)
{
int ret;
- ret = skl_dsp_reset_core(ctx);
+ ret = skl_dsp_reset_core(ctx, core_mask);
if (ret < 0) {
- dev_err(ctx->dev, "dsp core reset failed\n");
+ dev_err(ctx->dev, "dsp core reset failed: core_mask %x\n",
+ core_mask);
return ret;
}
/* power down core*/
- ret = skl_dsp_core_power_down(ctx);
+ ret = skl_dsp_core_power_down(ctx, core_mask);
if (ret < 0) {
- dev_err(ctx->dev, "dsp core power down failed\n");
+ dev_err(ctx->dev, "dsp core power down fail mask %x: %d\n",
+ core_mask, ret);
return ret;
}
- if (is_skl_dsp_core_enable(ctx)) {
- dev_err(ctx->dev, "DSP core disable failed\n");
+ if (is_skl_dsp_core_enable(ctx, core_mask)) {
+ dev_err(ctx->dev, "dsp core disable fail mask %x: %d\n",
+ core_mask, ret);
ret = -EIO;
}
@@ -224,28 +283,25 @@ int skl_dsp_boot(struct sst_dsp *ctx)
{
int ret;
- if (is_skl_dsp_core_enable(ctx)) {
- dev_dbg(ctx->dev, "dsp core is already enabled, so reset the dap core\n");
- ret = skl_dsp_reset_core(ctx);
+ if (is_skl_dsp_core_enable(ctx, SKL_DSP_CORE0_MASK)) {
+ ret = skl_dsp_reset_core(ctx, SKL_DSP_CORE0_MASK);
if (ret < 0) {
- dev_err(ctx->dev, "dsp reset failed\n");
+ dev_err(ctx->dev, "dsp core0 reset fail: %d\n", ret);
return ret;
}
- ret = skl_dsp_start_core(ctx);
+ ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK);
if (ret < 0) {
- dev_err(ctx->dev, "dsp start failed\n");
+ dev_err(ctx->dev, "dsp core0 start fail: %d\n", ret);
return ret;
}
} else {
- dev_dbg(ctx->dev, "disable and enable to make sure DSP is invalid state\n");
- ret = skl_dsp_disable_core(ctx);
-
+ ret = skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
if (ret < 0) {
- dev_err(ctx->dev, "dsp disable core failes\n");
+ dev_err(ctx->dev, "dsp core0 disable fail: %d\n", ret);
return ret;
}
- ret = skl_dsp_enable_core(ctx);
+ ret = skl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK);
}
return ret;
@@ -281,16 +337,74 @@ irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id)
return result;
}
+/*
+ * skl_dsp_get_core/skl_dsp_put_core will be called inside DAPM context
+ * within the dapm mutex. Hence no separate lock is used.
+ */
+int skl_dsp_get_core(struct sst_dsp *ctx, unsigned int core_id)
+{
+ struct skl_sst *skl = ctx->thread_context;
+ int ret = 0;
+
+ if (core_id >= skl->cores.count) {
+ dev_err(ctx->dev, "invalid core id: %d\n", core_id);
+ return -EINVAL;
+ }
+
+ if (skl->cores.state[core_id] == SKL_DSP_RESET) {
+ ret = ctx->fw_ops.set_state_D0(ctx, core_id);
+ if (ret < 0) {
+ dev_err(ctx->dev, "unable to get core%d\n", core_id);
+ return ret;
+ }
+ }
+
+ skl->cores.usage_count[core_id]++;
+
+ dev_dbg(ctx->dev, "core id %d state %d usage_count %d\n",
+ core_id, skl->cores.state[core_id],
+ skl->cores.usage_count[core_id]);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(skl_dsp_get_core);
+
+int skl_dsp_put_core(struct sst_dsp *ctx, unsigned int core_id)
+{
+ struct skl_sst *skl = ctx->thread_context;
+ int ret = 0;
+
+ if (core_id >= skl->cores.count) {
+ dev_err(ctx->dev, "invalid core id: %d\n", core_id);
+ return -EINVAL;
+ }
+
+ if (--skl->cores.usage_count[core_id] == 0) {
+ ret = ctx->fw_ops.set_state_D3(ctx, core_id);
+ if (ret < 0) {
+ dev_err(ctx->dev, "unable to put core %d: %d\n",
+ core_id, ret);
+ skl->cores.usage_count[core_id]++;
+ }
+ }
+
+ dev_dbg(ctx->dev, "core id %d state %d usage_count %d\n",
+ core_id, skl->cores.state[core_id],
+ skl->cores.usage_count[core_id]);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(skl_dsp_put_core);
int skl_dsp_wake(struct sst_dsp *ctx)
{
- return ctx->fw_ops.set_state_D0(ctx);
+ return skl_dsp_get_core(ctx, SKL_DSP_CORE0_ID);
}
EXPORT_SYMBOL_GPL(skl_dsp_wake);
int skl_dsp_sleep(struct sst_dsp *ctx)
{
- return ctx->fw_ops.set_state_D3(ctx);
+ return skl_dsp_put_core(ctx, SKL_DSP_CORE0_ID);
}
EXPORT_SYMBOL_GPL(skl_dsp_sleep);
@@ -337,9 +451,7 @@ void skl_dsp_free(struct sst_dsp *dsp)
free_irq(dsp->irq, dsp);
skl_ipc_op_int_disable(dsp);
- skl_ipc_int_disable(dsp);
-
- skl_dsp_disable_core(dsp);
+ skl_dsp_disable_core(dsp, SKL_DSP_CORE0_MASK);
}
EXPORT_SYMBOL_GPL(skl_dsp_free);
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index deabe7308d3b..0f8629ef79ac 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -19,6 +19,7 @@
#include <linux/interrupt.h>
#include <sound/memalloc.h>
#include "skl-sst-cldma.h"
+#include "skl-tplg-interface.h"
struct sst_dsp;
struct skl_sst;
@@ -76,35 +77,53 @@ struct sst_dsp_device;
#define SKL_ADSPIC_IPC 1
#define SKL_ADSPIS_IPC 1
+/* Core ID of core0 */
+#define SKL_DSP_CORE0_ID 0
+
+/* Mask for a given core index, c = 0.. number of supported cores - 1 */
+#define SKL_DSP_CORE_MASK(c) BIT(c)
+
+/*
+ * Core 0 mask = SKL_DSP_CORE_MASK(0); Defined separately
+ * since Core0 is primary core and it is used often
+ */
+#define SKL_DSP_CORE0_MASK BIT(0)
+
+/*
+ * Mask for a given number of cores
+ * nc = number of supported cores
+ */
+#define SKL_DSP_CORES_MASK(nc) GENMASK((nc - 1), 0)
+
/* ADSPCS - Audio DSP Control & Status */
-#define SKL_DSP_CORES 1
-#define SKL_DSP_CORE0_MASK 1
-#define SKL_DSP_CORES_MASK ((1 << SKL_DSP_CORES) - 1)
-
-/* Core Reset - asserted high */
-#define SKL_ADSPCS_CRST_SHIFT 0
-#define SKL_ADSPCS_CRST_MASK (SKL_DSP_CORES_MASK << SKL_ADSPCS_CRST_SHIFT)
-#define SKL_ADSPCS_CRST(x) ((x << SKL_ADSPCS_CRST_SHIFT) & SKL_ADSPCS_CRST_MASK)
-
-/* Core run/stall - when set to '1' core is stalled */
-#define SKL_ADSPCS_CSTALL_SHIFT 8
-#define SKL_ADSPCS_CSTALL_MASK (SKL_DSP_CORES_MASK << \
- SKL_ADSPCS_CSTALL_SHIFT)
-#define SKL_ADSPCS_CSTALL(x) ((x << SKL_ADSPCS_CSTALL_SHIFT) & \
- SKL_ADSPCS_CSTALL_MASK)
-
-/* Set Power Active - when set to '1' turn cores on */
-#define SKL_ADSPCS_SPA_SHIFT 16
-#define SKL_ADSPCS_SPA_MASK (SKL_DSP_CORES_MASK << SKL_ADSPCS_SPA_SHIFT)
-#define SKL_ADSPCS_SPA(x) ((x << SKL_ADSPCS_SPA_SHIFT) & SKL_ADSPCS_SPA_MASK)
-
-/* Current Power Active - power status of cores, set by hardware */
-#define SKL_ADSPCS_CPA_SHIFT 24
-#define SKL_ADSPCS_CPA_MASK (SKL_DSP_CORES_MASK << SKL_ADSPCS_CPA_SHIFT)
-#define SKL_ADSPCS_CPA(x) ((x << SKL_ADSPCS_CPA_SHIFT) & SKL_ADSPCS_CPA_MASK)
-
-#define SST_DSP_POWER_D0 0x0 /* full On */
-#define SST_DSP_POWER_D3 0x3 /* Off */
+
+/*
+ * Core Reset - asserted high
+ * CRST Mask for a given core mask pattern, cm
+ */
+#define SKL_ADSPCS_CRST_SHIFT 0
+#define SKL_ADSPCS_CRST_MASK(cm) ((cm) << SKL_ADSPCS_CRST_SHIFT)
+
+/*
+ * Core run/stall - when set to '1' core is stalled
+ * CSTALL Mask for a given core mask pattern, cm
+ */
+#define SKL_ADSPCS_CSTALL_SHIFT 8
+#define SKL_ADSPCS_CSTALL_MASK(cm) ((cm) << SKL_ADSPCS_CSTALL_SHIFT)
+
+/*
+ * Set Power Active - when set to '1' turn cores on
+ * SPA Mask for a given core mask pattern, cm
+ */
+#define SKL_ADSPCS_SPA_SHIFT 16
+#define SKL_ADSPCS_SPA_MASK(cm) ((cm) << SKL_ADSPCS_SPA_SHIFT)
+
+/*
+ * Current Power Active - power status of cores, set by hardware
+ * CPA Mask for a given core mask pattern, cm
+ */
+#define SKL_ADSPCS_CPA_SHIFT 24
+#define SKL_ADSPCS_CPA_MASK(cm) ((cm) << SKL_ADSPCS_CPA_SHIFT)
enum skl_dsp_states {
SKL_DSP_RUNNING = 1,
@@ -115,8 +134,8 @@ struct skl_dsp_fw_ops {
int (*load_fw)(struct sst_dsp *ctx);
/* FW module parser/loader */
int (*parse_fw)(struct sst_dsp *ctx);
- int (*set_state_D0)(struct sst_dsp *ctx);
- int (*set_state_D3)(struct sst_dsp *ctx);
+ int (*set_state_D0)(struct sst_dsp *ctx, unsigned int core_id);
+ int (*set_state_D3)(struct sst_dsp *ctx, unsigned int core_id);
unsigned int (*get_fw_errcode)(struct sst_dsp *ctx);
int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name);
int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id);
@@ -157,14 +176,26 @@ int skl_cldma_prepare(struct sst_dsp *ctx);
void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state);
struct sst_dsp *skl_dsp_ctx_init(struct device *dev,
struct sst_dsp_device *sst_dev, int irq);
-int skl_dsp_enable_core(struct sst_dsp *ctx);
-int skl_dsp_disable_core(struct sst_dsp *ctx);
bool is_skl_dsp_running(struct sst_dsp *ctx);
+
+unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx);
+void skl_dsp_init_core_state(struct sst_dsp *ctx);
+int skl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core_mask);
+int skl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core_mask);
+int skl_dsp_core_power_up(struct sst_dsp *ctx, unsigned int core_mask);
+int skl_dsp_core_power_down(struct sst_dsp *ctx, unsigned int core_mask);
+int skl_dsp_core_unset_reset_state(struct sst_dsp *ctx,
+ unsigned int core_mask);
+int skl_dsp_start_core(struct sst_dsp *ctx, unsigned int core_mask);
+
irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id);
int skl_dsp_wake(struct sst_dsp *ctx);
int skl_dsp_sleep(struct sst_dsp *ctx);
void skl_dsp_free(struct sst_dsp *dsp);
+int skl_dsp_get_core(struct sst_dsp *ctx, unsigned int core_id);
+int skl_dsp_put_core(struct sst_dsp *ctx, unsigned int core_id);
+
int skl_dsp_boot(struct sst_dsp *ctx);
int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
@@ -175,4 +206,11 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
+int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid,
+ struct skl_dfw_module *dfw_config);
+int snd_skl_parse_uuids(struct sst_dsp *ctx, unsigned int offset);
+void skl_freeup_uuid_list(struct skl_sst *ctx);
+
+int skl_dsp_strip_extended_manifest(struct firmware *fw);
+
#endif /*__SKL_SST_DSP_H__*/
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index 543460293b00..96f2f6889b18 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -363,7 +363,7 @@ static void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
/* first process the header */
switch (reply) {
case IPC_GLB_REPLY_SUCCESS:
- dev_info(ipc->dev, "ipc FW reply %x: success\n", header.primary);
+ dev_dbg(ipc->dev, "ipc FW reply %x: success\n", header.primary);
/* copy the rx data from the mailbox */
sst_dsp_inbox_read(ipc->dsp, msg->rx_data, msg->rx_size);
break;
@@ -692,7 +692,7 @@ int skl_ipc_init_instance(struct sst_generic_ipc *ipc,
/* param_block_size must be in dwords */
u16 param_block_size = msg->param_data_size / sizeof(u32);
- print_hex_dump(KERN_DEBUG, NULL, DUMP_PREFIX_NONE,
+ print_hex_dump_debug("Param data:", DUMP_PREFIX_NONE,
16, 4, buffer, param_block_size, false);
header.primary = IPC_MSG_TARGET(IPC_MOD_MSG);
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h
index d59d1ba62a43..2e3d4e80ef97 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.h
+++ b/sound/soc/intel/skylake/skl-sst-ipc.h
@@ -45,6 +45,14 @@ struct skl_ipc_header {
u32 extension;
};
+#define SKL_DSP_CORES_MAX 2
+
+struct skl_dsp_cores {
+ unsigned int count;
+ enum skl_dsp_states state[SKL_DSP_CORES_MAX];
+ int usage_count[SKL_DSP_CORES_MAX];
+};
+
struct skl_sst {
struct device *dev;
struct sst_dsp *dsp;
@@ -60,6 +68,15 @@ struct skl_sst {
void (*enable_miscbdcge)(struct device *dev, bool enable);
/*Is CGCTL.MISCBDCGE disabled*/
bool miscbdcg_disabled;
+
+ /* Populate module information */
+ struct list_head uuid_list;
+
+ /* Is firmware loaded */
+ bool fw_loaded;
+
+ /* multi-core */
+ struct skl_dsp_cores cores;
};
struct skl_ipc_init_instance_msg {
@@ -136,5 +153,6 @@ void skl_ipc_int_disable(struct sst_dsp *dsp);
bool skl_ipc_int_status(struct sst_dsp *dsp);
void skl_ipc_free(struct sst_generic_ipc *ipc);
int skl_ipc_init(struct device *dev, struct skl_sst *skl);
+void skl_clear_module_cnt(struct sst_dsp *ctx);
#endif /* __SKL_IPC_H */
diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c
new file mode 100644
index 000000000000..25fcb796bd86
--- /dev/null
+++ b/sound/soc/intel/skylake/skl-sst-utils.c
@@ -0,0 +1,256 @@
+/*
+ * skl-sst-utils.c - SKL sst utils functions
+ *
+ * Copyright (C) 2016 Intel Corp
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/uuid.h>
+#include "skl-sst-dsp.h"
+#include "../common/sst-dsp.h"
+#include "../common/sst-dsp-priv.h"
+#include "skl-sst-ipc.h"
+
+
+#define UUID_STR_SIZE 37
+#define DEFAULT_HASH_SHA256_LEN 32
+
+/* FW Extended Manifest Header id = $AE1 */
+#define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124
+
+struct skl_dfw_module_mod {
+ char name[100];
+ struct skl_dfw_module skl_dfw_mod;
+};
+
+struct UUID {
+ u8 id[16];
+};
+
+union seg_flags {
+ u32 ul;
+ struct {
+ u32 contents : 1;
+ u32 alloc : 1;
+ u32 load : 1;
+ u32 read_only : 1;
+ u32 code : 1;
+ u32 data : 1;
+ u32 _rsvd0 : 2;
+ u32 type : 4;
+ u32 _rsvd1 : 4;
+ u32 length : 16;
+ } r;
+} __packed;
+
+struct segment_desc {
+ union seg_flags flags;
+ u32 v_base_addr;
+ u32 file_offset;
+};
+
+struct module_type {
+ u32 load_type : 4;
+ u32 auto_start : 1;
+ u32 domain_ll : 1;
+ u32 domain_dp : 1;
+ u32 rsvd : 25;
+} __packed;
+
+struct adsp_module_entry {
+ u32 struct_id;
+ u8 name[8];
+ struct UUID uuid;
+ struct module_type type;
+ u8 hash1[DEFAULT_HASH_SHA256_LEN];
+ u32 entry_point;
+ u16 cfg_offset;
+ u16 cfg_count;
+ u32 affinity_mask;
+ u16 instance_max_count;
+ u16 instance_bss_size;
+ struct segment_desc segments[3];
+} __packed;
+
+struct adsp_fw_hdr {
+ u32 id;
+ u32 len;
+ u8 name[8];
+ u32 preload_page_count;
+ u32 fw_image_flags;
+ u32 feature_mask;
+ u16 major;
+ u16 minor;
+ u16 hotfix;
+ u16 build;
+ u32 num_modules;
+ u32 hw_buf_base;
+ u32 hw_buf_length;
+ u32 load_offset;
+} __packed;
+
+struct uuid_module {
+ uuid_le uuid;
+ int id;
+ int is_loadable;
+
+ struct list_head list;
+};
+
+struct skl_ext_manifest_hdr {
+ u32 id;
+ u32 len;
+ u16 version_major;
+ u16 version_minor;
+ u32 entries;
+};
+
+int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid,
+ struct skl_dfw_module *dfw_config)
+{
+ struct uuid_module *module;
+ uuid_le *uuid_mod;
+
+ uuid_mod = (uuid_le *)uuid;
+
+ list_for_each_entry(module, &ctx->uuid_list, list) {
+ if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) {
+ dfw_config->module_id = module->id;
+ dfw_config->is_loadable = module->is_loadable;
+
+ return 0;
+ }
+ }
+
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_skl_get_module_info);
+
+/*
+ * Parse the firmware binary to get the UUID, module id
+ * and loadable flags
+ */
+int snd_skl_parse_uuids(struct sst_dsp *ctx, unsigned int offset)
+{
+ struct adsp_fw_hdr *adsp_hdr;
+ struct adsp_module_entry *mod_entry;
+ int i, num_entry;
+ uuid_le *uuid_bin;
+ const char *buf;
+ struct skl_sst *skl = ctx->thread_context;
+ struct uuid_module *module;
+ struct firmware stripped_fw;
+ unsigned int safe_file;
+
+ /* Get the FW pointer to derive ADSP header */
+ stripped_fw.data = ctx->fw->data;
+ stripped_fw.size = ctx->fw->size;
+
+ skl_dsp_strip_extended_manifest(&stripped_fw);
+
+ buf = stripped_fw.data;
+
+ /* check if we have enough space in file to move to header */
+ safe_file = sizeof(*adsp_hdr) + offset;
+ if (stripped_fw.size <= safe_file) {
+ dev_err(ctx->dev, "Small fw file size, No space for hdr\n");
+ return -EINVAL;
+ }
+
+ adsp_hdr = (struct adsp_fw_hdr *)(buf + offset);
+
+ /* check 1st module entry is in file */
+ safe_file += adsp_hdr->len + sizeof(*mod_entry);
+ if (stripped_fw.size <= safe_file) {
+ dev_err(ctx->dev, "Small fw file size, No module entry\n");
+ return -EINVAL;
+ }
+
+ mod_entry = (struct adsp_module_entry *)
+ (buf + offset + adsp_hdr->len);
+
+ num_entry = adsp_hdr->num_modules;
+
+ /* check all entries are in file */
+ safe_file += num_entry * sizeof(*mod_entry);
+ if (stripped_fw.size <= safe_file) {
+ dev_err(ctx->dev, "Small fw file size, No modules\n");
+ return -EINVAL;
+ }
+
+
+ /*
+ * Read the UUID(GUID) from FW Manifest.
+ *
+ * The 16 byte UUID format is: XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXX
+ * Populate the UUID table to store module_id and loadable flags
+ * for the module.
+ */
+
+ for (i = 0; i < num_entry; i++, mod_entry++) {
+ module = kzalloc(sizeof(*module), GFP_KERNEL);
+ if (!module)
+ return -ENOMEM;
+
+ uuid_bin = (uuid_le *)mod_entry->uuid.id;
+ memcpy(&module->uuid, uuid_bin, sizeof(module->uuid));
+
+ module->id = i;
+ module->is_loadable = mod_entry->type.load_type;
+
+ list_add_tail(&module->list, &skl->uuid_list);
+
+ dev_dbg(ctx->dev,
+ "Adding uuid :%pUL mod id: %d Loadable: %d\n",
+ &module->uuid, module->id, module->is_loadable);
+ }
+
+ return 0;
+}
+
+void skl_freeup_uuid_list(struct skl_sst *ctx)
+{
+ struct uuid_module *uuid, *_uuid;
+
+ list_for_each_entry_safe(uuid, _uuid, &ctx->uuid_list, list) {
+ list_del(&uuid->list);
+ kfree(uuid);
+ }
+}
+
+/*
+ * some firmware binary contains some extended manifest. This needs
+ * to be stripped in that case before we load and use that image.
+ *
+ * Get the module id for the module by checking
+ * the table for the UUID for the module
+ */
+int skl_dsp_strip_extended_manifest(struct firmware *fw)
+{
+ struct skl_ext_manifest_hdr *hdr;
+
+ /* check if fw file is greater than header we are looking */
+ if (fw->size < sizeof(hdr)) {
+ pr_err("%s: Firmware file small, no hdr\n", __func__);
+ return -EINVAL;
+ }
+
+ hdr = (struct skl_ext_manifest_hdr *)fw->data;
+
+ if (hdr->id == SKL_EXT_MANIFEST_HEADER_MAGIC) {
+ fw->size -= hdr->len;
+ fw->data += hdr->len;
+ }
+
+ return 0;
+}
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 13ec8d53b526..588f899ceb65 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -68,10 +68,13 @@ static int skl_transfer_firmware(struct sst_dsp *ctx,
return ret;
}
+#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284
+
static int skl_load_base_firmware(struct sst_dsp *ctx)
{
int ret = 0, i;
struct skl_sst *skl = ctx->thread_context;
+ struct firmware stripped_fw;
u32 reg;
skl->boot_complete = false;
@@ -81,11 +84,25 @@ static int skl_load_base_firmware(struct sst_dsp *ctx)
ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev);
if (ret < 0) {
dev_err(ctx->dev, "Request firmware failed %d\n", ret);
- skl_dsp_disable_core(ctx);
return -EIO;
}
}
+ ret = snd_skl_parse_uuids(ctx, SKL_ADSP_FW_BIN_HDR_OFFSET);
+ if (ret < 0) {
+ dev_err(ctx->dev,
+ "UUID parsing err: %d\n", ret);
+ release_firmware(ctx->fw);
+ skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
+ return ret;
+ }
+
+ /* check for extended manifest */
+ stripped_fw.data = ctx->fw->data;
+ stripped_fw.size = ctx->fw->size;
+
+ skl_dsp_strip_extended_manifest(&stripped_fw);
+
ret = skl_dsp_boot(ctx);
if (ret < 0) {
dev_err(ctx->dev, "Boot dsp core failed ret: %d", ret);
@@ -119,7 +136,7 @@ static int skl_load_base_firmware(struct sst_dsp *ctx)
goto transfer_firmware_failed;
}
- ret = skl_transfer_firmware(ctx, ctx->fw->data, ctx->fw->size);
+ ret = skl_transfer_firmware(ctx, stripped_fw.data, stripped_fw.size);
if (ret < 0) {
dev_err(ctx->dev, "Transfer firmware failed%d\n", ret);
goto transfer_firmware_failed;
@@ -133,67 +150,87 @@ static int skl_load_base_firmware(struct sst_dsp *ctx)
}
dev_dbg(ctx->dev, "Download firmware successful%d\n", ret);
- skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+ skl->fw_loaded = true;
}
return 0;
transfer_firmware_failed:
ctx->cl_dev.ops.cl_cleanup_controller(ctx);
skl_load_base_firmware_failed:
- skl_dsp_disable_core(ctx);
+ skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
release_firmware(ctx->fw);
ctx->fw = NULL;
return ret;
}
-static int skl_set_dsp_D0(struct sst_dsp *ctx)
+static int skl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id)
{
int ret;
+ struct skl_ipc_dxstate_info dx;
+ struct skl_sst *skl = ctx->thread_context;
+ unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
- ret = skl_load_base_firmware(ctx);
- if (ret < 0) {
- dev_err(ctx->dev, "unable to load firmware\n");
- return ret;
+ /* If core0 is being turned on, we need to load the FW */
+ if (core_id == SKL_DSP_CORE0_ID) {
+ ret = skl_load_base_firmware(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "unable to load firmware\n");
+ return ret;
+ }
}
- skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+ /*
+ * If any core other than core 0 is being moved to D0, enable the
+ * core and send the set dx IPC for the core.
+ */
+ if (core_id != SKL_DSP_CORE0_ID) {
+ ret = skl_dsp_enable_core(ctx, core_mask);
+ if (ret < 0)
+ return ret;
+
+ dx.core_mask = core_mask;
+ dx.dx_mask = core_mask;
+
+ ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID,
+ SKL_BASE_FW_MODULE_ID, &dx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Failed to set dsp to D0:core id= %d\n",
+ core_id);
+ skl_dsp_disable_core(ctx, core_mask);
+ }
+ }
+
+ skl->cores.state[core_id] = SKL_DSP_RUNNING;
return ret;
}
-static int skl_set_dsp_D3(struct sst_dsp *ctx)
+static int skl_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id)
{
int ret;
struct skl_ipc_dxstate_info dx;
struct skl_sst *skl = ctx->thread_context;
+ unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
- dev_dbg(ctx->dev, "In %s:\n", __func__);
- mutex_lock(&ctx->mutex);
- if (!is_skl_dsp_running(ctx)) {
- mutex_unlock(&ctx->mutex);
- return 0;
- }
- mutex_unlock(&ctx->mutex);
-
- dx.core_mask = SKL_DSP_CORE0_MASK;
+ dx.core_mask = core_mask;
dx.dx_mask = SKL_IPC_D3_MASK;
+
ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID, SKL_BASE_FW_MODULE_ID, &dx);
if (ret < 0)
- dev_err(ctx->dev,
- "D3 request to FW failed, continuing reset: %d", ret);
-
- /* disable Interrupt */
- ctx->cl_dev.ops.cl_cleanup_controller(ctx);
- skl_cldma_int_disable(ctx);
- skl_ipc_op_int_disable(ctx);
- skl_ipc_int_disable(ctx);
-
- ret = skl_dsp_disable_core(ctx);
- if (ret < 0) {
- dev_err(ctx->dev, "disable dsp core failed ret: %d\n", ret);
- ret = -EIO;
+ dev_err(ctx->dev, "set Dx core %d fail: %d\n", core_id, ret);
+
+ if (core_id == SKL_DSP_CORE0_ID) {
+ /* disable Interrupt */
+ ctx->cl_dev.ops.cl_cleanup_controller(ctx);
+ skl_cldma_int_disable(ctx);
+ skl_ipc_op_int_disable(ctx);
+ skl_ipc_int_disable(ctx);
}
- skl_dsp_set_state_locked(ctx, SKL_DSP_RESET);
+ ret = skl_dsp_disable_core(ctx, core_mask);
+ if (ret < 0)
+ return ret;
+
+ skl->cores.state[core_id] = SKL_DSP_RESET;
return ret;
}
@@ -360,6 +397,19 @@ static int skl_unload_module(struct sst_dsp *ctx, u16 mod_id)
return ret;
}
+void skl_clear_module_cnt(struct sst_dsp *ctx)
+{
+ struct skl_module_table *module;
+
+ if (list_empty(&ctx->module_list))
+ return;
+
+ list_for_each_entry(module, &ctx->module_list, list) {
+ module->usage_cnt = 0;
+ }
+}
+EXPORT_SYMBOL_GPL(skl_clear_module_cnt);
+
static void skl_clear_module_table(struct sst_dsp *ctx)
{
struct skl_module_table *module, *tmp;
@@ -409,6 +459,7 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
skl->dev = dev;
skl_dev.thread_context = skl;
+ INIT_LIST_HEAD(&skl->uuid_list);
skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq);
if (!skl->dsp) {
@@ -432,12 +483,16 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
if (ret)
return ret;
+ skl->cores.count = 2;
+
ret = sst->fw_ops.load_fw(sst);
if (ret < 0) {
dev_err(dev, "Load base fw failed : %d", ret);
goto cleanup;
}
+ skl_dsp_init_core_state(sst);
+
if (dsp)
*dsp = skl;
@@ -452,6 +507,7 @@ EXPORT_SYMBOL_GPL(skl_sst_dsp_init);
void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
{
skl_clear_module_table(ctx->dsp);
+ skl_freeup_uuid_list(ctx);
skl_ipc_free(&ctx->ipc);
ctx->dsp->ops->free(ctx->dsp);
if (ctx->boot_complete) {
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 3e036b0349b9..cc0150fc2601 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -379,43 +379,6 @@ static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w,
}
/*
- * A pipe can have multiple modules, each of them will be a DAPM widget as
- * well. While managing a pipeline we need to get the list of all the
- * widgets in a pipelines, so this helper - skl_tplg_get_pipe_widget() helps
- * to get the SKL type widgets in that pipeline
- */
-static int skl_tplg_alloc_pipe_widget(struct device *dev,
- struct snd_soc_dapm_widget *w, struct skl_pipe *pipe)
-{
- struct skl_module_cfg *src_module = NULL;
- struct snd_soc_dapm_path *p = NULL;
- struct skl_pipe_module *p_module = NULL;
-
- p_module = devm_kzalloc(dev, sizeof(*p_module), GFP_KERNEL);
- if (!p_module)
- return -ENOMEM;
-
- p_module->w = w;
- list_add_tail(&p_module->node, &pipe->w_list);
-
- snd_soc_dapm_widget_for_each_sink_path(w, p) {
- if ((p->sink->priv == NULL)
- && (!is_skl_dsp_widget_type(w)))
- continue;
-
- if ((p->sink->priv != NULL) && p->connect
- && is_skl_dsp_widget_type(p->sink)) {
-
- src_module = p->sink->priv;
- if (pipe->ppl_id == src_module->pipe->ppl_id)
- skl_tplg_alloc_pipe_widget(dev,
- p->sink, pipe);
- }
- }
- return 0;
-}
-
-/*
* some modules can have multiple params set from user control and
* need to be set after module is initialized. If set_param flag is
* set module params will be done after module is initialised.
@@ -448,7 +411,7 @@ static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w,
if (bc->set_params == SKL_PARAM_SET) {
ret = skl_set_module_params(ctx,
- (u32 *)bc->params, bc->max,
+ (u32 *)bc->params, bc->size,
bc->param_id, mconfig);
if (ret < 0)
return ret;
@@ -483,7 +446,7 @@ static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w)
continue;
mconfig->formats_config.caps = (u32 *)&bc->params;
- mconfig->formats_config.caps_size = bc->max;
+ mconfig->formats_config.caps_size = bc->size;
break;
}
@@ -514,8 +477,6 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
if (!skl_is_pipe_mcps_avail(skl, mconfig))
return -ENOMEM;
- skl_tplg_alloc_pipe_mcps(skl, mconfig);
-
if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) {
ret = ctx->dsp->fw_ops.load_mod(ctx->dsp,
mconfig->id.module_id, mconfig->guid);
@@ -539,6 +500,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
if (ret < 0)
return ret;
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
ret = skl_tplg_set_module_params(w, ctx);
if (ret < 0)
return ret;
@@ -591,9 +553,6 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
if (!skl_is_pipe_mem_avail(skl, mconfig))
return -ENOMEM;
- skl_tplg_alloc_pipe_mem(skl, mconfig);
- skl_tplg_alloc_pipe_mcps(skl, mconfig);
-
/*
* Create a list of modules for pipe.
* This list contains modules from source to sink
@@ -602,19 +561,8 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
if (ret < 0)
return ret;
- /*
- * we create a w_list of all widgets in that pipe. This list is not
- * freed on PMD event as widgets within a pipe are static. This
- * saves us cycles to get widgets in pipe every time.
- *
- * So if we have already initialized all the widgets of a pipeline
- * we skip, so check for list_empty and create the list if empty
- */
- if (list_empty(&s_pipe->w_list)) {
- ret = skl_tplg_alloc_pipe_widget(ctx->dev, w, s_pipe);
- if (ret < 0)
- return ret;
- }
+ skl_tplg_alloc_pipe_mem(skl, mconfig);
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
/* Init all pipe modules from source to sink */
ret = skl_tplg_init_pipe_modules(skl, s_pipe);
@@ -949,13 +897,17 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
struct skl_pipe *s_pipe = mconfig->pipe;
int ret = 0;
+ if (s_pipe->state == SKL_PIPE_INVALID)
+ return -EINVAL;
+
skl_tplg_free_pipe_mcps(skl, mconfig);
skl_tplg_free_pipe_mem(skl, mconfig);
list_for_each_entry(w_module, &s_pipe->w_list, node) {
dst_module = w_module->w->priv;
- skl_tplg_free_pipe_mcps(skl, dst_module);
+ if (mconfig->m_state >= SKL_MODULE_INIT_DONE)
+ skl_tplg_free_pipe_mcps(skl, dst_module);
if (src_module == NULL) {
src_module = dst_module;
continue;
@@ -1102,7 +1054,7 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol,
if (w->power)
skl_get_module_params(skl->skl_sst, (u32 *)bc->params,
- bc->max, bc->param_id, mconfig);
+ bc->size, bc->param_id, mconfig);
/* decrement size for TLV header */
size -= 2 * sizeof(u32);
@@ -1136,6 +1088,10 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol,
struct skl *skl = get_skl_ctx(w->dapm->dev);
if (ac->params) {
+ if (size > ac->max)
+ return -EINVAL;
+
+ ac->size = size;
/*
* if the param_is is of type Vendor, firmware expects actual
* parameter id and size from the control.
@@ -1151,7 +1107,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol,
if (w->power)
return skl_set_module_params(skl->skl_sst,
- (u32 *)ac->params, ac->max,
+ (u32 *)ac->params, ac->size,
ac->param_id, mconfig);
}
@@ -1159,6 +1115,39 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol,
}
/*
+ * Fill the dma id for host and link. In case of passthrough
+ * pipeline, this will both host and link in the same
+ * pipeline, so need to copy the link and host based on dev_type
+ */
+static void skl_tplg_fill_dma_id(struct skl_module_cfg *mcfg,
+ struct skl_pipe_params *params)
+{
+ struct skl_pipe *pipe = mcfg->pipe;
+
+ if (pipe->passthru) {
+ switch (mcfg->dev_type) {
+ case SKL_DEVICE_HDALINK:
+ pipe->p_params->link_dma_id = params->link_dma_id;
+ break;
+
+ case SKL_DEVICE_HDAHOST:
+ pipe->p_params->host_dma_id = params->host_dma_id;
+ break;
+
+ default:
+ break;
+ }
+ pipe->p_params->s_fmt = params->s_fmt;
+ pipe->p_params->ch = params->ch;
+ pipe->p_params->s_freq = params->s_freq;
+ pipe->p_params->stream = params->stream;
+
+ } else {
+ memcpy(pipe->p_params, params, sizeof(*params));
+ }
+}
+
+/*
* The FE params are passed by hw_params of the DAI.
* On hw_params, the params are stored in Gateway module of the FE and we
* need to calculate the format in DSP module configuration, that
@@ -1168,10 +1157,9 @@ int skl_tplg_update_pipe_params(struct device *dev,
struct skl_module_cfg *mconfig,
struct skl_pipe_params *params)
{
- struct skl_pipe *pipe = mconfig->pipe;
struct skl_module_fmt *format = NULL;
- memcpy(pipe->p_params, params, sizeof(*params));
+ skl_tplg_fill_dma_id(mconfig, params);
if (params->stream == SNDRV_PCM_STREAM_PLAYBACK)
format = &mconfig->in_fmt[0];
@@ -1358,12 +1346,11 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai,
struct skl_module_cfg *mconfig,
struct skl_pipe_params *params)
{
- struct skl_pipe *pipe = mconfig->pipe;
struct nhlt_specific_cfg *cfg;
struct skl *skl = get_skl_ctx(dai->dev);
int link_type = skl_tplg_be_link_type(mconfig->dev_type);
- memcpy(pipe->p_params, params, sizeof(*params));
+ skl_tplg_fill_dma_id(mconfig, params);
if (link_type == NHLT_LINK_HDA)
return 0;
@@ -1554,6 +1541,55 @@ static void skl_tplg_fill_fmt(struct skl_module_fmt *dst_fmt,
}
}
+static void skl_clear_pin_config(struct snd_soc_platform *platform,
+ struct snd_soc_dapm_widget *w)
+{
+ int i;
+ struct skl_module_cfg *mconfig;
+ struct skl_pipe *pipe;
+
+ if (!strncmp(w->dapm->component->name, platform->component.name,
+ strlen(platform->component.name))) {
+ mconfig = w->priv;
+ pipe = mconfig->pipe;
+ for (i = 0; i < mconfig->max_in_queue; i++) {
+ mconfig->m_in_pin[i].in_use = false;
+ mconfig->m_in_pin[i].pin_state = SKL_PIN_UNBIND;
+ }
+ for (i = 0; i < mconfig->max_out_queue; i++) {
+ mconfig->m_out_pin[i].in_use = false;
+ mconfig->m_out_pin[i].pin_state = SKL_PIN_UNBIND;
+ }
+ pipe->state = SKL_PIPE_INVALID;
+ mconfig->m_state = SKL_MODULE_UNINIT;
+ }
+}
+
+void skl_cleanup_resources(struct skl *skl)
+{
+ struct skl_sst *ctx = skl->skl_sst;
+ struct snd_soc_platform *soc_platform = skl->platform;
+ struct snd_soc_dapm_widget *w;
+ struct snd_soc_card *card;
+
+ if (soc_platform == NULL)
+ return;
+
+ card = soc_platform->component.card;
+ if (!card || !card->instantiated)
+ return;
+
+ skl->resource.mem = 0;
+ skl->resource.mcps = 0;
+
+ list_for_each_entry(w, &card->widgets, list) {
+ if (is_skl_dsp_widget_type(w) && (w->priv != NULL))
+ skl_clear_pin_config(soc_platform, w);
+ }
+
+ skl_clear_module_cnt(ctx->dsp);
+}
+
/*
* Topology core widget load callback
*
@@ -1585,6 +1621,10 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
w->priv = mconfig;
memcpy(&mconfig->guid, &dfw_config->uuid, 16);
+ ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config);
+ if (ret < 0)
+ return ret;
+
mconfig->id.module_id = dfw_config->module_id;
mconfig->id.instance_id = dfw_config->instance_id;
mconfig->mcps = dfw_config->max_mcps;
@@ -1683,6 +1723,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be,
ac->max = dfw_ac->max;
ac->param_id = dfw_ac->param_id;
ac->set_params = dfw_ac->set_params;
+ ac->size = dfw_ac->max;
if (ac->max) {
ac->params = (char *) devm_kzalloc(dev, ac->max, GFP_KERNEL);
@@ -1733,6 +1774,60 @@ static struct snd_soc_tplg_ops skl_tplg_ops = {
.bytes_ext_ops_count = ARRAY_SIZE(skl_tlv_ops),
};
+/*
+ * A pipe can have multiple modules, each of them will be a DAPM widget as
+ * well. While managing a pipeline we need to get the list of all the
+ * widgets in a pipelines, so this helper - skl_tplg_create_pipe_widget_list()
+ * helps to get the SKL type widgets in that pipeline
+ */
+static int skl_tplg_create_pipe_widget_list(struct snd_soc_platform *platform)
+{
+ struct snd_soc_dapm_widget *w;
+ struct skl_module_cfg *mcfg = NULL;
+ struct skl_pipe_module *p_module = NULL;
+ struct skl_pipe *pipe;
+
+ list_for_each_entry(w, &platform->component.card->widgets, list) {
+ if (is_skl_dsp_widget_type(w) && w->priv != NULL) {
+ mcfg = w->priv;
+ pipe = mcfg->pipe;
+
+ p_module = devm_kzalloc(platform->dev,
+ sizeof(*p_module), GFP_KERNEL);
+ if (!p_module)
+ return -ENOMEM;
+
+ p_module->w = w;
+ list_add_tail(&p_module->node, &pipe->w_list);
+ }
+ }
+
+ return 0;
+}
+
+static void skl_tplg_set_pipe_type(struct skl *skl, struct skl_pipe *pipe)
+{
+ struct skl_pipe_module *w_module;
+ struct snd_soc_dapm_widget *w;
+ struct skl_module_cfg *mconfig;
+ bool host_found = false, link_found = false;
+
+ list_for_each_entry(w_module, &pipe->w_list, node) {
+ w = w_module->w;
+ mconfig = w->priv;
+
+ if (mconfig->dev_type == SKL_DEVICE_HDAHOST)
+ host_found = true;
+ else if (mconfig->dev_type != SKL_DEVICE_NONE)
+ link_found = true;
+ }
+
+ if (host_found && link_found)
+ pipe->passthru = true;
+ else
+ pipe->passthru = false;
+}
+
/* This will be read from topology manifest, currently defined here */
#define SKL_MAX_MCPS 30000000
#define SKL_FW_MAX_MEM 1000000
@@ -1746,6 +1841,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
const struct firmware *fw;
struct hdac_bus *bus = ebus_to_hbus(ebus);
struct skl *skl = ebus_to_skl(ebus);
+ struct skl_pipeline *ppl;
ret = request_firmware(&fw, skl->tplg_name, bus->dev);
if (ret < 0) {
@@ -1775,6 +1871,12 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
skl->resource.max_mem = SKL_FW_MAX_MEM;
skl->tplg = fw;
+ ret = skl_tplg_create_pipe_widget_list(platform);
+ if (ret < 0)
+ return ret;
+
+ list_for_each_entry(ppl, &skl->ppl_list, node)
+ skl_tplg_set_pipe_type(skl, ppl->pipe);
return 0;
}
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index e4b399cd7868..22d3ef83817d 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -244,7 +244,8 @@ enum skl_pipe_state {
SKL_PIPE_INVALID = 0,
SKL_PIPE_CREATED = 1,
SKL_PIPE_PAUSED = 2,
- SKL_PIPE_STARTED = 3
+ SKL_PIPE_STARTED = 3,
+ SKL_PIPE_RESET = 4
};
struct skl_pipe_module {
@@ -270,6 +271,7 @@ struct skl_pipe {
struct skl_pipe_params *p_params;
enum skl_pipe_state state;
struct list_head w_list;
+ bool passthru;
};
enum skl_module_state {
@@ -319,6 +321,7 @@ struct skl_algo_data {
u32 param_id;
u32 set_params;
u32 max;
+ u32 size;
char *params;
};
@@ -357,6 +360,8 @@ int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
int skl_stop_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
+int skl_reset_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
+
int skl_init_module(struct skl_sst *ctx, struct skl_module_cfg *module_config);
int skl_bind_modules(struct skl_sst *ctx, struct skl_module_cfg
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 06d8c263c68f..cd59536a761d 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -35,6 +35,8 @@
#include "skl-sst-dsp.h"
#include "skl-sst-ipc.h"
+static struct skl_machine_pdata skl_dmic_data;
+
/*
* initialize the PCI registers
*/
@@ -184,6 +186,7 @@ static int _skl_suspend(struct hdac_ext_bus *ebus)
{
struct skl *skl = ebus_to_skl(ebus);
struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct pci_dev *pci = to_pci_dev(bus->dev);
int ret;
snd_hdac_ext_bus_link_power_down_all(ebus);
@@ -193,9 +196,12 @@ static int _skl_suspend(struct hdac_ext_bus *ebus)
return ret;
snd_hdac_bus_stop_chip(bus);
+ update_pci_dword(pci, AZX_PCIREG_PGCTL,
+ AZX_PGCTL_LSRMD_MASK, AZX_PGCTL_LSRMD_MASK);
skl_enable_miscbdcge(bus->dev, false);
snd_hdac_bus_enter_link_reset(bus);
skl_enable_miscbdcge(bus->dev, true);
+ skl_cleanup_resources(skl);
return 0;
}
@@ -242,6 +248,7 @@ static int skl_suspend(struct device *dev)
ret = _skl_suspend(ebus);
if (ret < 0)
return ret;
+ skl->skl_sst->fw_loaded = false;
}
if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
@@ -397,6 +404,10 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data)
platform_device_put(pdev);
return -EIO;
}
+
+ if (mach->pdata)
+ dev_set_drvdata(&pdev->dev, mach->pdata);
+
skl->i2s_dev = pdev;
return 0;
@@ -657,6 +668,8 @@ static int skl_probe(struct pci_dev *pci,
skl->pci_id = pci->device;
+ device_disable_async_suspend(bus->dev);
+
skl->nhlt = skl_nhlt_init(bus->dev);
if (skl->nhlt == NULL)
@@ -666,6 +679,8 @@ static int skl_probe(struct pci_dev *pci,
pci_set_drvdata(skl->pci, ebus);
+ skl_dmic_data.dmic_num = skl_get_dmic_geo(skl);
+
/* check if dsp is there */
if (ebus->ppcap) {
err = skl_machine_device_register(skl,
@@ -713,7 +728,7 @@ static int skl_probe(struct pci_dev *pci,
list_for_each_entry(hlink, &ebus->hlink_list, list)
snd_hdac_ext_bus_link_put(ebus, hlink);
- /*configure PM */
+ /* configure PM */
pm_runtime_put_noidle(bus->dev);
pm_runtime_allow(bus->dev);
@@ -766,8 +781,7 @@ static void skl_remove(struct pci_dev *pci)
struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
struct skl *skl = ebus_to_skl(ebus);
- if (skl->tplg)
- release_firmware(skl->tplg);
+ release_firmware(skl->tplg);
if (pci_dev_run_wake(pci))
pm_runtime_get_noresume(&pci->dev);
@@ -786,15 +800,23 @@ static void skl_remove(struct pci_dev *pci)
static struct sst_acpi_mach sst_skl_devdata[] = {
{ "INT343A", "skl_alc286s_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL },
- { "INT343B", "skl_nau88l25_ssm4567_i2s", "intel/dsp_fw_release.bin",
- NULL, NULL, NULL },
- { "MX98357A", "skl_nau88l25_max98357a_i2s", "intel/dsp_fw_release.bin",
- NULL, NULL, NULL },
+ { "INT343B", "skl_n88l25_s4567", "intel/dsp_fw_release.bin",
+ NULL, NULL, &skl_dmic_data },
+ { "MX98357A", "skl_n88l25_m98357a", "intel/dsp_fw_release.bin",
+ NULL, NULL, &skl_dmic_data },
{}
};
static struct sst_acpi_mach sst_bxtp_devdata[] = {
{ "INT343A", "bxt_alc298s_i2s", "intel/dsp_fw_bxtn.bin", NULL, NULL, NULL },
+ { "DLGS7219", "bxt_da7219_max98357a_i2s", "intel/dsp_fw_bxtn.bin", NULL, NULL, NULL },
+};
+
+static struct sst_acpi_mach sst_kbl_devdata[] = {
+ { "INT343A", "kbl_alc286s_i2s", "intel/dsp_fw_kbl.bin", NULL, NULL, NULL },
+ { "INT343B", "kbl_n88l25_s4567", "intel/dsp_fw_kbl.bin", NULL, NULL, &skl_dmic_data },
+ { "MX98357A", "kbl_n88l25_m98357a", "intel/dsp_fw_kbl.bin", NULL, NULL, &skl_dmic_data },
+ {}
};
/* PCI IDs */
@@ -805,6 +827,9 @@ static const struct pci_device_id skl_ids[] = {
/* BXT-P */
{ PCI_DEVICE(0x8086, 0x5a98),
.driver_data = (unsigned long)&sst_bxtp_devdata},
+ /* KBL */
+ { PCI_DEVICE(0x8086, 0x9D71),
+ .driver_data = (unsigned long)&sst_kbl_devdata},
{ 0, }
};
MODULE_DEVICE_TABLE(pci, skl_ids);
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 4b4b3876aea9..9064e5b0d676 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -48,6 +48,8 @@
#define AZX_REG_VS_SDXEFIFOS_XBASE 0x1094
#define AZX_REG_VS_SDXEFIFOS_XINTERVAL 0x20
+#define AZX_PCIREG_PGCTL 0x44
+#define AZX_PGCTL_LSRMD_MASK (1 << 4)
#define AZX_PCIREG_CGCTL 0x48
#define AZX_CGCTL_MISCBDCGE_MASK (1 << 6)
@@ -65,6 +67,7 @@ struct skl {
unsigned int init_failed:1; /* delayed init failed */
struct platform_device *dmic_dev;
struct platform_device *i2s_dev;
+ struct snd_soc_platform *platform;
struct nhlt_acpi_table *nhlt; /* nhlt ptr */
struct skl_sst *skl_sst; /* sst skl ctx */
@@ -90,6 +93,11 @@ struct skl_dma_params {
u8 stream_tag;
};
+/* to pass dmic data */
+struct skl_machine_pdata {
+ u32 dmic_num;
+};
+
struct skl_dsp_ops {
int id;
struct skl_dsp_loader_ops (*loader_ops)(void);
@@ -108,9 +116,11 @@ void skl_nhlt_free(struct nhlt_acpi_table *addr);
struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance,
u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn);
+int skl_get_dmic_geo(struct skl *skl);
int skl_nhlt_update_topology_bin(struct skl *skl);
int skl_init_dsp(struct skl *skl);
int skl_free_dsp(struct skl *skl);
int skl_suspend_dsp(struct skl *skl);
int skl_resume_dsp(struct skl *skl);
+void skl_cleanup_resources(struct skl *skl);
#endif /* __SOUND_SOC_SKL_H */
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 49354d17ea55..c4c51a4d3c8f 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -518,7 +518,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
}
}
- rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr);
+ rsnd_mod_bset(adg_mod, SSICKR, 0x80FF0000, ckr);
rsnd_mod_write(adg_mod, BRRA, rbga);
rsnd_mod_write(adg_mod, BRRB, rbgb);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 875733c52953..d2df46c14c68 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -530,14 +530,15 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream,
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
if (platform->driver->compr_ops && platform->driver->compr_ops->pointer)
- platform->driver->compr_ops->pointer(cstream, tstamp);
+ ret = platform->driver->compr_ops->pointer(cstream, tstamp);
mutex_unlock(&rtd->pcm_mutex);
- return 0;
+ return ret;
}
static int soc_compr_copy(struct snd_compr_stream *cstream,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c4464858bf01..8698c26773b3 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list,
*/
static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
struct list_head *list, enum snd_soc_dapm_direction dir,
- int (*fn)(struct snd_soc_dapm_widget *, struct list_head *))
+ int (*fn)(struct snd_soc_dapm_widget *, struct list_head *,
+ bool (*custom_stop_condition)(struct snd_soc_dapm_widget *,
+ enum snd_soc_dapm_direction)),
+ bool (*custom_stop_condition)(struct snd_soc_dapm_widget *,
+ enum snd_soc_dapm_direction))
{
enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir);
struct snd_soc_dapm_path *path;
@@ -1088,6 +1092,11 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
if (list)
list_add_tail(&widget->work_list, list);
+ if (custom_stop_condition && custom_stop_condition(widget, dir)) {
+ widget->endpoints[dir] = 1;
+ return widget->endpoints[dir];
+ }
+
if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) {
widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget);
return widget->endpoints[dir];
@@ -1106,7 +1115,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
if (path->connect) {
path->walking = 1;
- con += fn(path->node[dir], list);
+ con += fn(path->node[dir], list, custom_stop_condition);
path->walking = 0;
}
}
@@ -1119,23 +1128,37 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
/*
* Recursively check for a completed path to an active or physically connected
* output widget. Returns number of complete paths.
+ *
+ * Optionally, can be supplied with a function acting as a stopping condition.
+ * This function takes the dapm widget currently being examined and the walk
+ * direction as an arguments, it should return true if the walk should be
+ * stopped and false otherwise.
*/
static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
- struct list_head *list)
+ struct list_head *list,
+ bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i,
+ enum snd_soc_dapm_direction))
{
return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT,
- is_connected_output_ep);
+ is_connected_output_ep, custom_stop_condition);
}
/*
* Recursively check for a completed path to an active or physically connected
* input widget. Returns number of complete paths.
+ *
+ * Optionally, can be supplied with a function acting as a stopping condition.
+ * This function takes the dapm widget currently being examined and the walk
+ * direction as an arguments, it should return true if the walk should be
+ * stopped and false otherwise.
*/
static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
- struct list_head *list)
+ struct list_head *list,
+ bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i,
+ enum snd_soc_dapm_direction))
{
return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN,
- is_connected_input_ep);
+ is_connected_input_ep, custom_stop_condition);
}
/**
@@ -1143,15 +1166,24 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
* @dai: the soc DAI.
* @stream: stream direction.
* @list: list of active widgets for this stream.
+ * @custom_stop_condition: (optional) a function meant to stop the widget graph
+ * walk based on custom logic.
*
* Queries DAPM graph as to whether an valid audio stream path exists for
* the initial stream specified by name. This takes into account
* current mixer and mux kcontrol settings. Creates list of valid widgets.
*
+ * Optionally, can be supplied with a function acting as a stopping condition.
+ * This function takes the dapm widget currently being examined and the walk
+ * direction as an arguments, it should return true if the walk should be
+ * stopped and false otherwise.
+ *
* Returns the number of valid paths or negative error.
*/
int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
- struct snd_soc_dapm_widget_list **list)
+ struct snd_soc_dapm_widget_list **list,
+ bool (*custom_stop_condition)(struct snd_soc_dapm_widget *,
+ enum snd_soc_dapm_direction))
{
struct snd_soc_card *card = dai->component->card;
struct snd_soc_dapm_widget *w;
@@ -1171,9 +1203,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
}
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- paths = is_connected_output_ep(dai->playback_widget, &widgets);
+ paths = is_connected_output_ep(dai->playback_widget, &widgets,
+ custom_stop_condition);
else
- paths = is_connected_input_ep(dai->capture_widget, &widgets);
+ paths = is_connected_input_ep(dai->capture_widget, &widgets,
+ custom_stop_condition);
/* Drop starting point */
list_del(widgets.next);
@@ -1268,8 +1302,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
DAPM_UPDATE_STAT(w, power_checks);
- in = is_connected_input_ep(w, NULL);
- out = is_connected_output_ep(w, NULL);
+ in = is_connected_input_ep(w, NULL, NULL);
+ out = is_connected_output_ep(w, NULL, NULL);
return out != 0 && in != 0;
}
@@ -1928,8 +1962,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
in = 0;
out = 0;
} else {
- in = is_connected_input_ep(w, NULL);
- out = is_connected_output_ep(w, NULL);
+ in = is_connected_input_ep(w, NULL, NULL);
+ out = is_connected_output_ep(w, NULL, NULL);
}
ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d",
@@ -3282,6 +3316,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
mutex_unlock(&dapm->card->dapm_mutex);
return w;
}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
struct snd_soc_dapm_widget *
snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index aa99dac31b3b..60d702f8b9f0 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1287,6 +1287,46 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list,
return 0;
}
+static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
+ enum snd_soc_dapm_direction dir)
+{
+ struct snd_soc_card *card = widget->dapm->card;
+ struct snd_soc_pcm_runtime *rtd;
+ int i;
+
+ if (dir == SND_SOC_DAPM_DIR_OUT) {
+ list_for_each_entry(rtd, &card->rtd_list, list) {
+ if (!rtd->dai_link->no_pcm)
+ continue;
+
+ if (rtd->cpu_dai->playback_widget == widget)
+ return true;
+
+ for (i = 0; i < rtd->num_codecs; ++i) {
+ struct snd_soc_dai *dai = rtd->codec_dais[i];
+ if (dai->playback_widget == widget)
+ return true;
+ }
+ }
+ } else { /* SND_SOC_DAPM_DIR_IN */
+ list_for_each_entry(rtd, &card->rtd_list, list) {
+ if (!rtd->dai_link->no_pcm)
+ continue;
+
+ if (rtd->cpu_dai->capture_widget == widget)
+ return true;
+
+ for (i = 0; i < rtd->num_codecs; ++i) {
+ struct snd_soc_dai *dai = rtd->codec_dais[i];
+ if (dai->capture_widget == widget)
+ return true;
+ }
+ }
+ }
+
+ return false;
+}
+
int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
int stream, struct snd_soc_dapm_widget_list **list)
{
@@ -1294,7 +1334,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
int paths;
/* get number of valid DAI paths and their widgets */
- paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list);
+ paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list,
+ dpcm_end_walk_at_be);
dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths,
stream ? "capture" : "playback");
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 69860da473ea..9e5276d6dda0 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -556,7 +556,6 @@ static int usb_audio_probe(struct usb_interface *intf,
goto __error;
}
chip = usb_chip[i];
- dev_set_drvdata(&dev->dev, chip);
atomic_inc(&chip->active); /* avoid autopm */
break;
}
@@ -582,6 +581,7 @@ static int usb_audio_probe(struct usb_interface *intf,
goto __error;
}
}
+ dev_set_drvdata(&dev->dev, chip);
/*
* For devices with more than one control interface, we assume the