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-rw-r--r--sound/core/pcm_lib.c85
-rw-r--r--sound/core/seq/seq_dummy.c31
-rw-r--r--sound/firewire/amdtp.c71
-rw-r--r--sound/firewire/amdtp.h5
-rw-r--r--sound/firewire/bebob/bebob_stream.c7
-rw-r--r--sound/firewire/fireworks/fireworks_stream.c5
-rw-r--r--sound/firewire/fireworks/fireworks_transaction.c2
-rw-r--r--sound/pci/hda/hda_controller.c24
-rw-r--r--sound/pci/hda/hda_intel.c5
-rw-r--r--sound/pci/hda/hda_priv.h1
-rw-r--r--sound/pci/hda/patch_hdmi.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c4
-rw-r--r--sound/soc/adi/axi-i2s.c2
-rw-r--r--sound/soc/atmel/Kconfig2
-rw-r--r--sound/soc/atmel/atmel-pcm-dma.c12
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c132
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c31
-rw-r--r--sound/soc/au1x/db1200.c19
-rw-r--r--sound/soc/au1x/dbdma2.c6
-rw-r--r--sound/soc/au1x/dma.c6
-rw-r--r--sound/soc/codecs/88pm860x-codec.c4
-rw-r--r--sound/soc/codecs/Kconfig4
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/ad193x.c4
-rw-r--r--sound/soc/codecs/ak4671.c2
-rw-r--r--sound/soc/codecs/alc5623.c8
-rw-r--r--sound/soc/codecs/alc5632.c12
-rw-r--r--sound/soc/codecs/arizona.c78
-rw-r--r--sound/soc/codecs/arizona.h5
-rw-r--r--sound/soc/codecs/bt-sco.c2
-rw-r--r--sound/soc/codecs/cs35l32.c4
-rw-r--r--sound/soc/codecs/cs42l52.c4
-rw-r--r--sound/soc/codecs/cs42l56.c4
-rw-r--r--sound/soc/codecs/cs42l73.c4
-rw-r--r--sound/soc/codecs/da732x.c4
-rw-r--r--sound/soc/codecs/max98357a.c138
-rw-r--r--sound/soc/codecs/pcm3008.c4
-rw-r--r--sound/soc/codecs/pcm512x-i2c.c4
-rw-r--r--sound/soc/codecs/pcm512x-spi.c4
-rw-r--r--sound/soc/codecs/pcm512x.c934
-rw-r--r--sound/soc/codecs/pcm512x.h109
-rw-r--r--sound/soc/codecs/rt286.c88
-rw-r--r--sound/soc/codecs/rt286.h7
-rw-r--r--sound/soc/codecs/rt5631.c28
-rw-r--r--sound/soc/codecs/rt5640.c12
-rw-r--r--sound/soc/codecs/rt5645.c255
-rw-r--r--sound/soc/codecs/rt5645.h87
-rw-r--r--sound/soc/codecs/rt5651.c18
-rw-r--r--sound/soc/codecs/rt5670.c137
-rw-r--r--sound/soc/codecs/rt5670.h80
-rw-r--r--sound/soc/codecs/rt5677.c267
-rw-r--r--sound/soc/codecs/sgtl5000.c27
-rw-r--r--sound/soc/codecs/sn95031.c33
-rw-r--r--sound/soc/codecs/sta32x.h2
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c9
-rw-r--r--sound/soc/codecs/tlv320aic3x.c4
-rw-r--r--sound/soc/codecs/tlv320dac33.c9
-rw-r--r--sound/soc/codecs/ts3a227e.c6
-rw-r--r--sound/soc/codecs/twl4030.c55
-rw-r--r--sound/soc/codecs/twl6040.c4
-rw-r--r--sound/soc/codecs/wm2000.c2
-rw-r--r--sound/soc/codecs/wm5100.c5
-rw-r--r--sound/soc/codecs/wm5102.c23
-rw-r--r--sound/soc/codecs/wm5110.c20
-rw-r--r--sound/soc/codecs/wm8350.c2
-rw-r--r--sound/soc/codecs/wm8400.c9
-rw-r--r--sound/soc/codecs/wm8731.c5
-rw-r--r--sound/soc/codecs/wm8750.c2
-rw-r--r--sound/soc/codecs/wm8770.c8
-rw-r--r--sound/soc/codecs/wm8900.c2
-rw-r--r--sound/soc/codecs/wm8903.c2
-rw-r--r--sound/soc/codecs/wm8904.c27
-rw-r--r--sound/soc/codecs/wm8955.c2
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c2
-rw-r--r--sound/soc/codecs/wm8960.c2
-rw-r--r--sound/soc/codecs/wm8961.c4
-rw-r--r--sound/soc/codecs/wm8962.c6
-rw-r--r--sound/soc/codecs/wm8988.c2
-rw-r--r--sound/soc/codecs/wm8990.c9
-rw-r--r--sound/soc/codecs/wm8991.c9
-rw-r--r--sound/soc/codecs/wm8993.c2
-rw-r--r--sound/soc/codecs/wm8994.c23
-rw-r--r--sound/soc/codecs/wm8995.c14
-rw-r--r--sound/soc/codecs/wm8996.c8
-rw-r--r--sound/soc/codecs/wm8997.c11
-rw-r--r--sound/soc/codecs/wm9081.c2
-rw-r--r--sound/soc/codecs/wm9090.c2
-rw-r--r--sound/soc/codecs/wm9705.c16
-rw-r--r--sound/soc/codecs/wm9712.c12
-rw-r--r--sound/soc/codecs/wm9713.c14
-rw-r--r--sound/soc/codecs/wm_adsp.c6
-rw-r--r--sound/soc/codecs/wm_hubs.c10
-rw-r--r--sound/soc/davinci/Kconfig3
-rw-r--r--sound/soc/davinci/davinci-evm.c6
-rw-r--r--sound/soc/davinci/davinci-mcasp.c103
-rw-r--r--sound/soc/dwc/Kconfig1
-rw-r--r--sound/soc/dwc/designware_i2s.c360
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c21
-rw-r--r--sound/soc/fsl/fsl_asrc.c5
-rw-r--r--sound/soc/fsl/fsl_asrc.h3
-rw-r--r--sound/soc/fsl/fsl_esai.c2
-rw-r--r--sound/soc/fsl/fsl_esai.h2
-rw-r--r--sound/soc/fsl/fsl_sai.c2
-rw-r--r--sound/soc/fsl/fsl_spdif.c17
-rw-r--r--sound/soc/fsl/fsl_ssi.c6
-rw-r--r--sound/soc/fsl/imx-spdif.c1
-rw-r--r--sound/soc/fsl/imx-wm8962.c1
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c12
-rw-r--r--sound/soc/fsl/wm1133-ev1.c12
-rw-r--r--sound/soc/generic/simple-card.c7
-rw-r--r--sound/soc/intel/Kconfig19
-rw-r--r--sound/soc/intel/Makefile2
-rw-r--r--sound/soc/intel/broadwell.c10
-rw-r--r--sound/soc/intel/byt-rt5640.c12
-rw-r--r--sound/soc/intel/bytcr_dpcm_rt5640.c3
-rw-r--r--sound/soc/intel/cht_bsw_rt5645.c326
-rw-r--r--sound/soc/intel/cht_bsw_rt5672.c15
-rw-r--r--sound/soc/intel/sst-baytrail-pcm.c6
-rw-r--r--sound/soc/intel/sst-dsp.c3
-rw-r--r--sound/soc/intel/sst-firmware.c18
-rw-r--r--sound/soc/intel/sst-haswell-dsp.c17
-rw-r--r--sound/soc/intel/sst-haswell-ipc.c207
-rw-r--r--sound/soc/intel/sst-haswell-ipc.h31
-rw-r--r--sound/soc/intel/sst-haswell-pcm.c167
-rw-r--r--sound/soc/intel/sst-mfld-platform-pcm.c7
-rw-r--r--sound/soc/intel/sst/sst.h3
-rw-r--r--sound/soc/intel/sst/sst_acpi.c13
-rw-r--r--sound/soc/intel/sst/sst_loader.c3
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c21
-rw-r--r--sound/soc/mxs/mxs-saif.c10
-rw-r--r--sound/soc/mxs/mxs-saif.h1
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c27
-rw-r--r--sound/soc/nuc900/nuc900-pcm.c6
-rw-r--r--sound/soc/omap/ams-delta.c18
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c1
-rw-r--r--sound/soc/omap/omap-mcbsp.c2
-rw-r--r--sound/soc/omap/omap-twl4030.c20
-rw-r--r--sound/soc/pxa/Kconfig2
-rw-r--r--sound/soc/pxa/corgi.c16
-rw-r--r--sound/soc/pxa/e740_wm9705.c20
-rw-r--r--sound/soc/pxa/e750_wm9705.c20
-rw-r--r--sound/soc/pxa/hx4700.c8
-rw-r--r--sound/soc/pxa/magician.c21
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c2
-rw-r--r--sound/soc/pxa/palm27x.c15
-rw-r--r--sound/soc/pxa/raumfeld.c35
-rw-r--r--sound/soc/pxa/spitz.c23
-rw-r--r--sound/soc/pxa/ttc-dkb.c4
-rw-r--r--sound/soc/pxa/zylonite.c12
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c9
-rw-r--r--sound/soc/rockchip/rockchip_i2s.h2
-rw-r--r--sound/soc/samsung/arndale_rt5631.c1
-rw-r--r--sound/soc/samsung/goni_wm8994.c23
-rw-r--r--sound/soc/samsung/h1940_uda1380.c15
-rw-r--r--sound/soc/samsung/jive_wm8750.c16
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c25
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c15
-rw-r--r--sound/soc/samsung/s3c24xx_simtec.c20
-rw-r--r--sound/soc/samsung/s3c24xx_uda134x.c12
-rw-r--r--sound/soc/samsung/smartq_wm8987.c16
-rw-r--r--sound/soc/samsung/smdk_wm8580.c21
-rw-r--r--sound/soc/samsung/smdk_wm8580pcm.c19
-rw-r--r--sound/soc/samsung/smdk_wm8994pcm.c16
-rw-r--r--sound/soc/sh/fsi.c9
-rw-r--r--sound/soc/sh/migor.c12
-rw-r--r--sound/soc/soc-ac97.c36
-rw-r--r--sound/soc/soc-compress.c9
-rw-r--r--sound/soc/soc-core.c203
-rw-r--r--sound/soc/soc-dapm.c125
-rw-r--r--sound/soc/soc-devres.c2
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c15
-rw-r--r--sound/soc/soc-pcm.c7
-rw-r--r--sound/soc/ux500/mop500_ab8500.c16
-rw-r--r--sound/usb/caiaq/audio.c2
-rw-r--r--sound/usb/mixer.c1
175 files changed, 3727 insertions, 1807 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index ec9e7866177f..446c00bd908b 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1015,6 +1015,60 @@ int snd_interval_list(struct snd_interval *i, unsigned int count,
EXPORT_SYMBOL(snd_interval_list);
+/**
+ * snd_interval_ranges - refine the interval value from the list of ranges
+ * @i: the interval value to refine
+ * @count: the number of elements in the list of ranges
+ * @ranges: the ranges list
+ * @mask: the bit-mask to evaluate
+ *
+ * Refines the interval value from the list of ranges.
+ * When mask is non-zero, only the elements corresponding to bit 1 are
+ * evaluated.
+ *
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
+ */
+int snd_interval_ranges(struct snd_interval *i, unsigned int count,
+ const struct snd_interval *ranges, unsigned int mask)
+{
+ unsigned int k;
+ struct snd_interval range_union;
+ struct snd_interval range;
+
+ if (!count) {
+ snd_interval_none(i);
+ return -EINVAL;
+ }
+ snd_interval_any(&range_union);
+ range_union.min = UINT_MAX;
+ range_union.max = 0;
+ for (k = 0; k < count; k++) {
+ if (mask && !(mask & (1 << k)))
+ continue;
+ snd_interval_copy(&range, &ranges[k]);
+ if (snd_interval_refine(&range, i) < 0)
+ continue;
+ if (snd_interval_empty(&range))
+ continue;
+
+ if (range.min < range_union.min) {
+ range_union.min = range.min;
+ range_union.openmin = 1;
+ }
+ if (range.min == range_union.min && !range.openmin)
+ range_union.openmin = 0;
+ if (range.max > range_union.max) {
+ range_union.max = range.max;
+ range_union.openmax = 1;
+ }
+ if (range.max == range_union.max && !range.openmax)
+ range_union.openmax = 0;
+ }
+ return snd_interval_refine(i, &range_union);
+}
+EXPORT_SYMBOL(snd_interval_ranges);
+
static int snd_interval_step(struct snd_interval *i, unsigned int step)
{
unsigned int n;
@@ -1221,6 +1275,37 @@ int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime,
EXPORT_SYMBOL(snd_pcm_hw_constraint_list);
+static int snd_pcm_hw_rule_ranges(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pcm_hw_constraint_ranges *r = rule->private;
+ return snd_interval_ranges(hw_param_interval(params, rule->var),
+ r->count, r->ranges, r->mask);
+}
+
+
+/**
+ * snd_pcm_hw_constraint_ranges - apply list of range constraints to a parameter
+ * @runtime: PCM runtime instance
+ * @cond: condition bits
+ * @var: hw_params variable to apply the list of range constraints
+ * @r: ranges
+ *
+ * Apply the list of range constraints to an interval parameter.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ const struct snd_pcm_hw_constraint_ranges *r)
+{
+ return snd_pcm_hw_rule_add(runtime, cond, var,
+ snd_pcm_hw_rule_ranges, (void *)r,
+ var, -1);
+}
+EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges);
+
static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index ec667f158f19..5d905d90d504 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -82,36 +82,6 @@ struct snd_seq_dummy_port {
static int my_client = -1;
/*
- * unuse callback - send ALL_SOUNDS_OFF and RESET_CONTROLLERS events
- * to subscribers.
- * Note: this callback is called only after all subscribers are removed.
- */
-static int
-dummy_unuse(void *private_data, struct snd_seq_port_subscribe *info)
-{
- struct snd_seq_dummy_port *p;
- int i;
- struct snd_seq_event ev;
-
- p = private_data;
- memset(&ev, 0, sizeof(ev));
- if (p->duplex)
- ev.source.port = p->connect;
- else
- ev.source.port = p->port;
- ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
- ev.type = SNDRV_SEQ_EVENT_CONTROLLER;
- for (i = 0; i < 16; i++) {
- ev.data.control.channel = i;
- ev.data.control.param = MIDI_CTL_ALL_SOUNDS_OFF;
- snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
- ev.data.control.param = MIDI_CTL_RESET_CONTROLLERS;
- snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
- }
- return 0;
-}
-
-/*
* event input callback - just redirect events to subscribers
*/
static int
@@ -175,7 +145,6 @@ create_port(int idx, int type)
| SNDRV_SEQ_PORT_TYPE_PORT;
memset(&pcb, 0, sizeof(pcb));
pcb.owner = THIS_MODULE;
- pcb.unuse = dummy_unuse;
pcb.event_input = dummy_input;
pcb.private_free = dummy_free;
pcb.private_data = rec;
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 3badc70124ab..0d580186ef1a 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -21,7 +21,19 @@
#define CYCLES_PER_SECOND 8000
#define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND)
-#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND 3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS 8
+
+#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */
/* isochronous header parameters */
#define ISO_DATA_LENGTH_SHIFT 16
@@ -78,8 +90,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
s->callbacked = false;
s->sync_slave = NULL;
- s->rx_blocks_for_midi = UINT_MAX;
-
return 0;
}
EXPORT_SYMBOL(amdtp_stream_init);
@@ -222,6 +232,14 @@ sfc_found:
for (i = 0; i < pcm_channels; i++)
s->pcm_positions[i] = i;
s->midi_position = s->pcm_channels;
+
+ /*
+ * We do not know the actual MIDI FIFO size of most devices. Just
+ * assume two bytes, i.e., one byte can be received over the bus while
+ * the previous one is transmitted over MIDI.
+ * (The value here is adjusted for midi_ratelimit_per_packet().)
+ */
+ s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
}
EXPORT_SYMBOL(amdtp_stream_set_parameters);
@@ -463,6 +481,36 @@ static void amdtp_fill_pcm_silence(struct amdtp_stream *s,
}
}
+/*
+ * To avoid sending MIDI bytes at too high a rate, assume that the receiving
+ * device has a FIFO, and track how much it is filled. This values increases
+ * by one whenever we send one byte in a packet, but the FIFO empties at
+ * a constant rate independent of our packet rate. One packet has syt_interval
+ * samples, so the number of bytes that empty out of the FIFO, per packet(!),
+ * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing
+ * fractional values, the values in midi_fifo_used[] are measured in bytes
+ * multiplied by the sample rate.
+ */
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+ int used;
+
+ used = s->midi_fifo_used[port];
+ if (used == 0) /* common shortcut */
+ return true;
+
+ used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+ used = max(used, 0);
+ s->midi_fifo_used[port] = used;
+
+ return used < s->midi_fifo_limit;
+}
+
+static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
+{
+ s->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
+}
+
static void amdtp_fill_midi(struct amdtp_stream *s,
__be32 *buffer, unsigned int frames)
{
@@ -470,16 +518,21 @@ static void amdtp_fill_midi(struct amdtp_stream *s,
u8 *b;
for (f = 0; f < frames; f++) {
- buffer[s->midi_position] = 0;
b = (u8 *)&buffer[s->midi_position];
port = (s->data_block_counter + f) % 8;
- if ((f >= s->rx_blocks_for_midi) ||
- (s->midi[port] == NULL) ||
- (snd_rawmidi_transmit(s->midi[port], b + 1, 1) <= 0))
- b[0] = 0x80;
- else
+ if (f < MAX_MIDI_RX_BLOCKS &&
+ midi_ratelimit_per_packet(s, port) &&
+ s->midi[port] != NULL &&
+ snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) {
+ midi_rate_use_one_byte(s, port);
b[0] = 0x81;
+ } else {
+ b[0] = 0x80;
+ b[1] = 0;
+ }
+ b[2] = 0;
+ b[3] = 0;
buffer += s->data_block_quadlets;
}
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index e6e8926275b0..8a03a91e728b 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -148,13 +148,12 @@ struct amdtp_stream {
bool double_pcm_frames;
struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
+ int midi_fifo_limit;
+ int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
/* quirk: fixed interval of dbc between previos/current packets. */
unsigned int tx_dbc_interval;
- /* quirk: the first count of data blocks in an rx packet for MIDI */
- unsigned int rx_blocks_for_midi;
-
bool callbacked;
wait_queue_head_t callback_wait;
struct amdtp_stream *sync_slave;
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 1aab0a32870c..0ebcabfdc7ce 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -484,13 +484,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
amdtp_stream_destroy(&bebob->rx_stream);
destroy_both_connections(bebob);
}
- /*
- * The firmware for these devices ignore MIDI messages in more than
- * first 8 data blocks of an received AMDTP packet.
- */
- if (bebob->spec == &maudio_fw410_spec ||
- bebob->spec == &maudio_special_spec)
- bebob->rx_stream.rx_blocks_for_midi = 8;
end:
return err;
}
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index b985fc5ebdc6..4f440e163667 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -179,11 +179,6 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw)
destroy_stream(efw, &efw->tx_stream);
goto end;
}
- /*
- * Fireworks ignores MIDI messages in more than first 8 data
- * blocks of an received AMDTP packet.
- */
- efw->rx_stream.rx_blocks_for_midi = 8;
/* set IEC61883 compliant mode (actually not fully compliant...) */
err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883);
diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c
index 255dabc6fc33..2a85e4209f0b 100644
--- a/sound/firewire/fireworks/fireworks_transaction.c
+++ b/sound/firewire/fireworks/fireworks_transaction.c
@@ -124,7 +124,7 @@ copy_resp_to_buf(struct snd_efw *efw, void *data, size_t length, int *rcode)
spin_lock_irq(&efw->lock);
t = (struct snd_efw_transaction *)data;
- length = min_t(size_t, t->length * sizeof(t->length), length);
+ length = min_t(size_t, be32_to_cpu(t->length) * sizeof(u32), length);
if (efw->push_ptr < efw->pull_ptr)
capacity = (unsigned int)(efw->pull_ptr - efw->push_ptr);
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 8276a743e22e..0cfc9c8c4b4e 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip)
EXPORT_SYMBOL_GPL(azx_mixer_create);
+static bool is_input_stream(struct azx *chip, unsigned char index)
+{
+ return (index >= chip->capture_index_offset &&
+ index < chip->capture_index_offset + chip->capture_streams);
+}
+
/* initialize SD streams */
int azx_init_stream(struct azx *chip)
{
int i;
+ int in_stream_tag = 0;
+ int out_stream_tag = 0;
/* initialize each stream (aka device)
* assign the starting bdl address to each stream (device)
@@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip)
azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80);
/* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */
azx_dev->sd_int_sta_mask = 1 << i;
- /* stream tag: must be non-zero and unique */
azx_dev->index = i;
- azx_dev->stream_tag = i + 1;
+
+ /* stream tag must be unique throughout
+ * the stream direction group,
+ * valid values 1...15
+ * use separate stream tag if the flag
+ * AZX_DCAPS_SEPARATE_STREAM_TAG is used
+ */
+ if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG)
+ azx_dev->stream_tag =
+ is_input_stream(chip, i) ?
+ ++in_stream_tag :
+ ++out_stream_tag;
+ else
+ azx_dev->stream_tag = i + 1;
}
return 0;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2bf0b568e3de..d426a0bd6a5f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -299,6 +299,9 @@ enum {
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
AZX_DCAPS_SNOOP_TYPE(SCH))
+#define AZX_DCAPS_INTEL_SKYLAKE \
+ (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG)
+
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\
@@ -2027,7 +2030,7 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
/* Sunrise Point-LP */
{ PCI_DEVICE(0x8086, 0x9d70),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
/* Haswell */
{ PCI_DEVICE(0x8086, 0x0a0c),
.driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h
index aa484fdf4338..166e3e84b963 100644
--- a/sound/pci/hda/hda_priv.h
+++ b/sound/pci/hda/hda_priv.h
@@ -171,6 +171,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */
#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */
#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */
+#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */
enum {
AZX_SNOOP_TYPE_NONE ,
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 5f13d2d18079..b422e406a9cb 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3353,6 +3353,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
{ .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi },
{ .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi },
+{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi },
{ .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch },
{ .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
{ .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
@@ -3413,6 +3414,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0060");
MODULE_ALIAS("snd-hda-codec-id:10de0067");
MODULE_ALIAS("snd-hda-codec-id:10de0070");
MODULE_ALIAS("snd-hda-codec-id:10de0071");
+MODULE_ALIAS("snd-hda-codec-id:10de0072");
MODULE_ALIAS("snd-hda-codec-id:10de8001");
MODULE_ALIAS("snd-hda-codec-id:11069f80");
MODULE_ALIAS("snd-hda-codec-id:11069f81");
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4f6413e01c13..605d14003d25 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -568,9 +568,9 @@ static void stac_store_hints(struct hda_codec *codec)
spec->gpio_mask;
}
if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir))
- spec->gpio_mask &= spec->gpio_mask;
- if (get_int_hint(codec, "gpio_data", &spec->gpio_data))
spec->gpio_dir &= spec->gpio_mask;
+ if (get_int_hint(codec, "gpio_data", &spec->gpio_data))
+ spec->gpio_data &= spec->gpio_mask;
if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask))
spec->eapd_mask &= spec->gpio_mask;
if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute))
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
index 7752860f7230..4c23381727a1 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev)
if (ret)
goto err_clk_disable;
+ return 0;
+
err_clk_disable:
clk_disable_unprepare(i2s->clk);
return ret;
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index fb3878312bf8..1579e994acf8 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904
config SND_AT91_SOC_SAM9X5_WM8731
tristate "SoC Audio support for WM8731-based at91sam9x5 board"
- depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5
+ depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC
select SND_ATMEL_SOC_SSC
select SND_ATMEL_SOC_DMA
select SND_SOC_WM8731
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index 33fb3bb133df..b8e7bad05eb1 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -105,13 +105,11 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
return ret;
}
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- slave_config->dst_addr = ssc->phybase + SSC_THR;
- slave_config->dst_maxburst = 1;
- } else {
- slave_config->src_addr = ssc->phybase + SSC_RHR;
- slave_config->src_maxburst = 1;
- }
+ slave_config->dst_addr = ssc->phybase + SSC_THR;
+ slave_config->dst_maxburst = 1;
+
+ slave_config->src_addr = ssc->phybase + SSC_RHR;
+ slave_config->src_maxburst = 1;
prtd->dma_intr_handler = atmel_pcm_dma_irq;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 99ff35e2a25d..fb0b7e8b08ff 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -204,6 +204,13 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
ssc_readl(ssc_p->ssc->regs, SR));
+ /* Enable PMC peripheral clock for this SSC */
+ pr_debug("atmel_ssc_dai: Starting clock\n");
+ clk_enable(ssc_p->ssc->clk);
+
+ /* Reset the SSC to keep it at a clean status */
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dir = 0;
dir_mask = SSC_DIR_MASK_PLAYBACK;
@@ -250,11 +257,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
dma_params = ssc_p->dma_params[dir];
if (dma_params != NULL) {
- ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
- pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n",
- (dir ? "receive" : "transmit"),
- ssc_readl(ssc_p->ssc->regs, SR));
-
dma_params->ssc = NULL;
dma_params->substream = NULL;
ssc_p->dma_params[dir] = NULL;
@@ -266,10 +268,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
ssc_p->dir_mask &= ~dir_mask;
if (!ssc_p->dir_mask) {
if (ssc_p->initialized) {
- /* Shutdown the SSC clock. */
- pr_debug("atmel_ssc_dai: Stopping clock\n");
- clk_disable(ssc_p->ssc->clk);
-
free_irq(ssc_p->ssc->irq, ssc_p);
ssc_p->initialized = 0;
}
@@ -280,6 +278,10 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
}
spin_unlock_irq(&ssc_p->lock);
+
+ /* Shutdown the SSC clock. */
+ pr_debug("atmel_ssc_dai: Stopping clock\n");
+ clk_disable(ssc_p->ssc->clk);
}
@@ -348,7 +350,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
struct atmel_pcm_dma_params *dma_params;
int dir, channels, bits;
u32 tfmr, rfmr, tcmr, rcmr;
- int start_event;
int ret;
int fslen, fslen_ext;
@@ -451,25 +452,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
break;
case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
- /*
- * I2S format, CODEC supplies BCLK and LRC clocks.
- *
- * The SSC transmit clock is obtained from the BCLK signal on
- * on the TK line, and the SSC receive clock is
- * generated from the transmit clock.
- *
- * For single channel data, one sample is transferred
- * on the falling edge of the LRC clock.
- * For two channel data, one sample is
- * transferred on both edges of the LRC clock.
- */
- start_event = ((channels == 1)
- ? SSC_START_FALLING_RF
- : SSC_START_EDGE_RF);
-
+ /* I2S format, CODEC supplies BCLK and LRC clocks. */
rcmr = SSC_BF(RCMR_PERIOD, 0)
| SSC_BF(RCMR_STTDLY, START_DELAY)
- | SSC_BF(RCMR_START, start_event)
+ | SSC_BF(RCMR_START, SSC_START_FALLING_RF)
| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
@@ -478,14 +464,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
| SSC_BF(RFMR_FSLEN, 0)
- | SSC_BF(RFMR_DATNB, 0)
+ | SSC_BF(RFMR_DATNB, (channels - 1))
| SSC_BIT(RFMR_MSBF)
| SSC_BF(RFMR_LOOP, 0)
| SSC_BF(RFMR_DATLEN, (bits - 1));
tcmr = SSC_BF(TCMR_PERIOD, 0)
| SSC_BF(TCMR_STTDLY, START_DELAY)
- | SSC_BF(TCMR_START, start_event)
+ | SSC_BF(TCMR_START, SSC_START_FALLING_RF)
| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
| SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ?
@@ -495,7 +481,55 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(TFMR_FSDEN, 0)
| SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
| SSC_BF(TFMR_FSLEN, 0)
- | SSC_BF(TFMR_DATNB, 0)
+ | SSC_BF(TFMR_DATNB, (channels - 1))
+ | SSC_BIT(TFMR_MSBF)
+ | SSC_BF(TFMR_DATDEF, 0)
+ | SSC_BF(TFMR_DATLEN, (bits - 1));
+ break;
+
+ case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFS:
+ /* I2S format, CODEC supplies BCLK, SSC supplies LRCLK. */
+ if (bits > 16 && !ssc->pdata->has_fslen_ext) {
+ dev_err(dai->dev,
+ "sample size %d is too large for SSC device\n",
+ bits);
+ return -EINVAL;
+ }
+
+ fslen_ext = (bits - 1) / 16;
+ fslen = (bits - 1) % 16;
+
+ rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
+ | SSC_BF(RCMR_STTDLY, START_DELAY)
+ | SSC_BF(RCMR_START, SSC_START_FALLING_RF)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_PIN : SSC_CKS_CLOCK);
+
+ rfmr = SSC_BF(RFMR_FSLEN_EXT, fslen_ext)
+ | SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
+ | SSC_BF(RFMR_FSLEN, fslen)
+ | SSC_BF(RFMR_DATNB, (channels - 1))
+ | SSC_BIT(RFMR_MSBF)
+ | SSC_BF(RFMR_LOOP, 0)
+ | SSC_BF(RFMR_DATLEN, (bits - 1));
+
+ tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
+ | SSC_BF(TCMR_STTDLY, START_DELAY)
+ | SSC_BF(TCMR_START, SSC_START_FALLING_RF)
+ | SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(TCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_CLOCK : SSC_CKS_PIN);
+
+ tfmr = SSC_BF(TFMR_FSLEN_EXT, fslen_ext)
+ | SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_NEGATIVE)
+ | SSC_BF(TFMR_FSDEN, 0)
+ | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
+ | SSC_BF(TFMR_FSLEN, fslen)
+ | SSC_BF(TFMR_DATNB, (channels - 1))
| SSC_BIT(TFMR_MSBF)
| SSC_BF(TFMR_DATDEF, 0)
| SSC_BF(TFMR_DATLEN, (bits - 1));
@@ -512,7 +546,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
| SSC_BF(RCMR_STTDLY, 1)
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
- | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
@@ -527,7 +561,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
| SSC_BF(TCMR_STTDLY, 1)
| SSC_BF(TCMR_START, SSC_START_RISING_RF)
- | SSC_BF(TCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
@@ -545,10 +579,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
/*
* DSP/PCM Mode A format, CODEC supplies BCLK and LRC clocks.
*
- * The SSC transmit clock is obtained from the BCLK signal on
- * on the TK line, and the SSC receive clock is
- * generated from the transmit clock.
- *
* Data is transferred on first BCLK after LRC pulse rising
* edge.If stereo, the right channel data is contiguous with
* the left channel data.
@@ -556,7 +586,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rcmr = SSC_BF(RCMR_PERIOD, 0)
| SSC_BF(RCMR_STTDLY, START_DELAY)
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
- | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
SSC_CKS_PIN : SSC_CKS_CLOCK);
@@ -597,23 +627,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rcmr, rfmr, tcmr, tfmr);
if (!ssc_p->initialized) {
-
- /* Enable PMC peripheral clock for this SSC */
- pr_debug("atmel_ssc_dai: Starting clock\n");
- clk_enable(ssc_p->ssc->clk);
-
- /* Reset the SSC and its PDC registers */
- ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
-
- ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
-
- ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
+ if (!ssc_p->ssc->pdata->use_dma) {
+ ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
+
+ ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
+ }
ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0,
ssc_p->name, ssc_p);
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 66b66d0e7514..f5ad214663f9 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -47,7 +47,6 @@
#include <sound/soc.h>
#include <asm/mach-types.h>
-#include <mach/hardware.h>
#include "../codecs/wm8731.h"
#include "atmel-pcm.h"
@@ -64,33 +63,6 @@
static struct clk *mclk;
-static int at91sam9g20ek_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;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret;
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static struct snd_soc_ops at91sam9g20ek_ops = {
- .hw_params = at91sam9g20ek_hw_params,
-};
-
static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
@@ -173,7 +145,8 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
.init = at91sam9g20ek_wm8731_init,
.platform_name = "at91rm9200_ssc.0",
.codec_name = "wm8731.0-001b",
- .ops = &at91sam9g20ek_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
};
static struct snd_soc_card snd_soc_at91sam9g20ek = {
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index a747ac0b399f..c75995f2779c 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -91,27 +91,12 @@ static int db1200_i2s_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret;
/* WM8731 has its own 12MHz crystal */
snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
12000000, SND_SOC_CLOCK_IN);
- /* codec is bitclock and lrclk master */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- goto out;
-
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- goto out;
-
- ret = 0;
-out:
- return ret;
+ return 0;
}
static struct snd_soc_ops db1200_i2s_wm8731_ops = {
@@ -125,6 +110,8 @@ static struct snd_soc_dai_link db1200_i2s_dai = {
.cpu_dai_name = "au1xpsc_i2s.1",
.platform_name = "au1xpsc-pcm.1",
.codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &db1200_i2s_wm8731_ops,
};
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index b06b8d8128c6..dd94fea72d5d 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -315,11 +315,6 @@ static struct snd_pcm_ops au1xpsc_pcm_ops = {
.pointer = au1xpsc_pcm_pointer,
};
-static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm)
-{
- snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
@@ -335,7 +330,6 @@ static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_platform_driver au1xpsc_soc_platform = {
.ops = &au1xpsc_pcm_ops,
.pcm_new = au1xpsc_pcm_new,
- .pcm_free = au1xpsc_pcm_free_dma_buffers,
};
static int au1xpsc_pcm_drvprobe(struct platform_device *pdev)
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
index 6ffaaff469c7..24cc7f40d87a 100644
--- a/sound/soc/au1x/dma.c
+++ b/sound/soc/au1x/dma.c
@@ -287,11 +287,6 @@ static struct snd_pcm_ops alchemy_pcm_ops = {
.pointer = alchemy_pcm_pointer,
};
-static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm)
-{
- snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm *pcm = rtd->pcm;
@@ -305,7 +300,6 @@ static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_platform_driver alchemy_pcm_soc_platform = {
.ops = &alchemy_pcm_ops,
.pcm_new = alchemy_pcm_new,
- .pcm_free = alchemy_pcm_free_dma_buffers,
};
static int alchemy_pcm_drvprobe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index a2bf27f4baab..a0f265327fdf 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -386,7 +386,7 @@ static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol,
static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
/*
* In order to avoid current on the load, mute power-on and power-off
@@ -403,7 +403,7 @@ static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int dac = 0;
int data;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8349f982a586..6ecac1e4428e 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,6 +69,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98090 if I2C
select SND_SOC_MAX98095 if I2C
+ select SND_SOC_MAX98357A
select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
@@ -456,6 +457,9 @@ config SND_SOC_MAX98090
config SND_SOC_MAX98095
tristate
+config SND_SOC_MAX98357A
+ tristate
+
config SND_SOC_MAX9850
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index bbdfd1e1c182..69b8666d187a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -64,6 +64,7 @@ snd-soc-max9768-objs := max9768.o
snd-soc-max98088-objs := max98088.o
snd-soc-max98090-objs := max98090.o
snd-soc-max98095-objs := max98095.o
+snd-soc-max98357a-objs := max98357a.o
snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
@@ -245,6 +246,7 @@ obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o
obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o
obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o
+obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 387530b0b0fd..17c953595660 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -333,8 +333,8 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec)
regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0);
/* de-emphasis: 48kHz, powedown dac */
regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A);
- /* powerdown dac, dac in tdm mode */
- regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x41);
+ /* dac in tdm mode */
+ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40);
/* high-pass filter enable */
regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
/* sata delay=1, adc aux mode */
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 686cacb0e835..632e89f793a7 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -163,7 +163,7 @@ static const struct snd_kcontrol_new ak4671_snd_controls[] = {
static int ak4671_out2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index bdf8c5ac8ca4..0e357996864b 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -55,18 +55,20 @@ static inline int alc5623_reset(struct snd_soc_codec *codec)
static int amp_mixer_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);
+
/* to power-on/off class-d amp generators/speaker */
/* need to write to 'index-46h' register : */
/* so write index num (here 0x46) to reg 0x6a */
/* and then 0xffff/0 to reg 0x6c */
- snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
+ snd_soc_write(codec, ALC5623_HID_CTRL_INDEX, 0x46);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
+ snd_soc_write(codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
break;
case SND_SOC_DAPM_POST_PMD:
- snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
+ snd_soc_write(codec, ALC5623_HID_CTRL_DATA, 0);
break;
}
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index d1fdbc266631..db3283abbe18 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -116,18 +116,20 @@ static inline int alc5632_reset(struct regmap *map)
static int amp_mixer_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);
+
/* to power-on/off class-d amp generators/speaker */
/* need to write to 'index-46h' register : */
/* so write index num (here 0x46) to reg 0x6a */
/* and then 0xffff/0 to reg 0x6c */
- snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46);
+ snd_soc_write(codec, ALC5632_HID_CTRL_INDEX, 0x46);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF);
+ snd_soc_write(codec, ALC5632_HID_CTRL_DATA, 0xFFFF);
break;
case SND_SOC_DAPM_POST_PMD:
- snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0);
+ snd_soc_write(codec, ALC5632_HID_CTRL_DATA, 0);
break;
}
@@ -1066,7 +1068,7 @@ static int alc5632_probe(struct snd_soc_codec *codec)
return 0;
}
-static struct snd_soc_codec_driver soc_codec_device_alc5632 = {
+static const struct snd_soc_codec_driver soc_codec_device_alc5632 = {
.probe = alc5632_probe,
.resume = alc5632_resume,
.set_bias_level = alc5632_set_bias_level,
@@ -1080,7 +1082,7 @@ static struct snd_soc_codec_driver soc_codec_device_alc5632 = {
.num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes),
};
-static struct regmap_config alc5632_regmap = {
+static const struct regmap_config alc5632_regmap = {
.reg_bits = 8,
.val_bits = 16,
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 9550d7433ad0..29202610dd0d 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -84,7 +84,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
- struct snd_soc_codec *codec = w->codec;
+ 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;
@@ -692,7 +692,8 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
int event)
{
- struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
unsigned int reg;
if (w->shift % 2)
@@ -705,25 +706,25 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
priv->in_pending++;
break;
case SND_SOC_DAPM_POST_PMU:
- snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0);
+ snd_soc_update_bits(codec, reg, ARIZONA_IN1L_MUTE, 0);
/* If this is the last input pending then allow VU */
priv->in_pending--;
if (priv->in_pending == 0) {
msleep(1);
- arizona_in_set_vu(w->codec, 1);
+ arizona_in_set_vu(codec, 1);
}
break;
case SND_SOC_DAPM_PRE_PMD:
- snd_soc_update_bits(w->codec, reg,
+ snd_soc_update_bits(codec, reg,
ARIZONA_IN1L_MUTE | ARIZONA_IN_VU,
ARIZONA_IN1L_MUTE | ARIZONA_IN_VU);
break;
case SND_SOC_DAPM_POST_PMD:
/* Disable volume updates if no inputs are enabled */
- reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES);
+ reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES);
if (reg == 0)
- arizona_in_set_vu(w->codec, 0);
+ arizona_in_set_vu(codec, 0);
}
return 0;
@@ -734,7 +735,25 @@ int arizona_out_ev(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 arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+
switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ switch (w->shift) {
+ case ARIZONA_OUT1L_ENA_SHIFT:
+ case ARIZONA_OUT1R_ENA_SHIFT:
+ case ARIZONA_OUT2L_ENA_SHIFT:
+ case ARIZONA_OUT2R_ENA_SHIFT:
+ case ARIZONA_OUT3L_ENA_SHIFT:
+ case ARIZONA_OUT3R_ENA_SHIFT:
+ priv->out_up_pending++;
+ priv->out_up_delay += 17;
+ break;
+ default:
+ break;
+ }
+ break;
case SND_SOC_DAPM_POST_PMU:
switch (w->shift) {
case ARIZONA_OUT1L_ENA_SHIFT:
@@ -743,13 +762,50 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
case ARIZONA_OUT2R_ENA_SHIFT:
case ARIZONA_OUT3L_ENA_SHIFT:
case ARIZONA_OUT3R_ENA_SHIFT:
- msleep(17);
+ priv->out_up_pending--;
+ if (!priv->out_up_pending) {
+ msleep(priv->out_up_delay);
+ priv->out_up_delay = 0;
+ }
break;
default:
break;
}
break;
+ case SND_SOC_DAPM_PRE_PMD:
+ switch (w->shift) {
+ case ARIZONA_OUT1L_ENA_SHIFT:
+ case ARIZONA_OUT1R_ENA_SHIFT:
+ case ARIZONA_OUT2L_ENA_SHIFT:
+ case ARIZONA_OUT2R_ENA_SHIFT:
+ case ARIZONA_OUT3L_ENA_SHIFT:
+ case ARIZONA_OUT3R_ENA_SHIFT:
+ priv->out_down_pending++;
+ priv->out_down_delay++;
+ break;
+ default:
+ break;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ switch (w->shift) {
+ case ARIZONA_OUT1L_ENA_SHIFT:
+ case ARIZONA_OUT1R_ENA_SHIFT:
+ case ARIZONA_OUT2L_ENA_SHIFT:
+ case ARIZONA_OUT2R_ENA_SHIFT:
+ case ARIZONA_OUT3L_ENA_SHIFT:
+ case ARIZONA_OUT3R_ENA_SHIFT:
+ priv->out_down_pending--;
+ if (!priv->out_down_pending) {
+ msleep(priv->out_down_delay);
+ priv->out_down_delay = 0;
+ }
+ break;
+ default:
+ break;
+ }
+ break;
}
return 0;
@@ -760,7 +816,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
- struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
struct arizona *arizona = priv->arizona;
unsigned int mask = 1 << w->shift;
unsigned int val;
@@ -772,6 +829,9 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w,
case SND_SOC_DAPM_PRE_PMD:
val = 0;
break;
+ case SND_SOC_DAPM_PRE_PMU:
+ case SND_SOC_DAPM_POST_PMD:
+ return arizona_out_ev(w, kcontrol, event);
default:
return -EINVAL;
}
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 942cfb197b6d..11ff899b0272 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -77,6 +77,11 @@ struct arizona_priv {
int num_inputs;
unsigned int in_pending;
+ unsigned int out_up_pending;
+ unsigned int out_up_delay;
+ unsigned int out_down_pending;
+ unsigned int out_down_delay;
+
unsigned int spk_ena:2;
unsigned int spk_ena_pending:1;
};
diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c
index 5075bf0a7276..e7238b8904bc 100644
--- a/sound/soc/codecs/bt-sco.c
+++ b/sound/soc/codecs/bt-sco.c
@@ -86,5 +86,5 @@ static struct platform_driver bt_sco_driver = {
module_platform_driver(bt_sco_driver);
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
-MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver");
+MODULE_DESCRIPTION("ASoC generic bluetooth sco link driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c
index ec55c590afd0..f2b8aad21274 100644
--- a/sound/soc/codecs/cs35l32.c
+++ b/sound/soc/codecs/cs35l32.c
@@ -264,7 +264,7 @@ static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec,
CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO_MASK, val);
}
-static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs35l32 = {
.set_sysclk = cs35l32_codec_set_sysclk,
.dapm_widgets = cs35l32_dapm_widgets,
@@ -288,7 +288,7 @@ static const struct reg_default cs35l32_monitor_patch[] = {
{ 0x00, 0x00 },
};
-static struct regmap_config cs35l32_regmap = {
+static const struct regmap_config cs35l32_regmap = {
.reg_bits = 8,
.val_bits = 8,
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 35fbef743fbe..1589e7a881d8 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -1103,7 +1103,7 @@ static int cs42l52_remove(struct snd_soc_codec *codec)
return 0;
}
-static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs42l52 = {
.probe = cs42l52_probe,
.remove = cs42l52_remove,
.set_bias_level = cs42l52_set_bias_level,
@@ -1130,7 +1130,7 @@ static const struct reg_default cs42l52_threshold_patch[] = {
};
-static struct regmap_config cs42l52_regmap = {
+static const struct regmap_config cs42l52_regmap = {
.reg_bits = 8,
.val_bits = 8,
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 2ddc7ac10ad7..cbc654fe48c7 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -1164,7 +1164,7 @@ static int cs42l56_remove(struct snd_soc_codec *codec)
return 0;
}
-static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs42l56 = {
.probe = cs42l56_probe,
.remove = cs42l56_remove,
.set_bias_level = cs42l56_set_bias_level,
@@ -1179,7 +1179,7 @@ static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = {
.num_controls = ARRAY_SIZE(cs42l56_snd_controls),
};
-static struct regmap_config cs42l56_regmap = {
+static const struct regmap_config cs42l56_regmap = {
.reg_bits = 8,
.val_bits = 8,
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 7c55537c69cf..8ecedba79606 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1347,7 +1347,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec)
return 0;
}
-static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs42l73 = {
.probe = cs42l73_probe,
.set_bias_level = cs42l73_set_bias_level,
.suspend_bias_off = true,
@@ -1361,7 +1361,7 @@ static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = {
.num_controls = ARRAY_SIZE(cs42l73_snd_controls),
};
-static struct regmap_config cs42l73_regmap = {
+static const struct regmap_config cs42l73_regmap = {
.reg_bits = 8,
.val_bits = 8,
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index 61b2f9a2eef1..ffe96175a8a5 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -609,7 +609,7 @@ static const struct snd_kcontrol_new da732x_snd_controls[] = {
static int da732x_adc_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -663,7 +663,7 @@ static int da732x_adc_event(struct snd_soc_dapm_widget *w,
static int da732x_out_pga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
new file mode 100644
index 000000000000..1806333ea29e
--- /dev/null
+++ b/sound/soc/codecs/max98357a.c
@@ -0,0 +1,138 @@
+/* Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ *
+ * max98357a.c -- MAX98357A ALSA SoC Codec driver
+ */
+
+#include <linux/module.h>
+#include <linux/gpio.h>
+#include <sound/soc.h>
+
+#define DRV_NAME "max98357a"
+
+static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ gpiod_set_value(sdmode, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ gpiod_set_value(sdmode, 0);
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("SDMode", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_OUTPUT("Speaker"),
+};
+
+static const struct snd_soc_dapm_route max98357a_dapm_routes[] = {
+ {"Speaker", NULL, "SDMode"},
+};
+
+static int max98357a_codec_probe(struct snd_soc_codec *codec)
+{
+ struct gpio_desc *sdmode;
+
+ sdmode = devm_gpiod_get(codec->dev, "sdmode");
+ if (IS_ERR(sdmode)) {
+ dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n",
+ __func__, PTR_ERR(sdmode));
+ return PTR_ERR(sdmode);
+ }
+ gpiod_direction_output(sdmode, 0);
+ snd_soc_codec_set_drvdata(codec, sdmode);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver max98357a_codec_driver = {
+ .probe = max98357a_codec_probe,
+ .dapm_widgets = max98357a_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets),
+ .dapm_routes = max98357a_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(max98357a_dapm_routes),
+};
+
+static struct snd_soc_dai_ops max98357a_dai_ops = {
+ .trigger = max98357a_daiops_trigger,
+};
+
+static struct snd_soc_dai_driver max98357a_dai_driver = {
+ .name = DRV_NAME,
+ .playback = {
+ .stream_name = DRV_NAME "-playback",
+ .formats = SNDRV_PCM_FMTBIT_S16 |
+ SNDRV_PCM_FMTBIT_S24 |
+ SNDRV_PCM_FMTBIT_S32,
+ .rates = SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &max98357a_dai_ops,
+};
+
+static int max98357a_platform_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ ret = snd_soc_register_codec(&pdev->dev, &max98357a_codec_driver,
+ &max98357a_dai_driver, 1);
+ if (ret)
+ dev_err(&pdev->dev, "%s() error registering codec driver: %d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static int max98357a_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+
+ return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id max98357a_device_id[] = {
+ { .compatible = "maxim," DRV_NAME, },
+ {}
+};
+MODULE_DEVICE_TABLE(of, max98357a_device_id);
+#endif
+
+static struct platform_driver max98357a_platform_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .of_match_table = of_match_ptr(max98357a_device_id),
+ },
+ .probe = max98357a_platform_probe,
+ .remove = max98357a_platform_remove,
+};
+module_platform_driver(max98357a_platform_driver);
+
+MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index 7e73fa4b3183..8fb445f33f6f 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -32,7 +32,7 @@ static int pcm3008_dac_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct pcm3008_setup_data *setup = codec->dev->platform_data;
gpio_set_value_cansleep(setup->pdda_pin,
@@ -45,7 +45,7 @@ static int pcm3008_adc_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct pcm3008_setup_data *setup = codec->dev->platform_data;
gpio_set_value_cansleep(setup->pdad_pin,
diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c
index d0547fa275fc..dcdfac0ffeb1 100644
--- a/sound/soc/codecs/pcm512x-i2c.c
+++ b/sound/soc/codecs/pcm512x-i2c.c
@@ -46,6 +46,8 @@ static int pcm512x_i2c_remove(struct i2c_client *i2c)
static const struct i2c_device_id pcm512x_i2c_id[] = {
{ "pcm5121", },
{ "pcm5122", },
+ { "pcm5141", },
+ { "pcm5142", },
{ }
};
MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id);
@@ -53,6 +55,8 @@ MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id);
static const struct of_device_id pcm512x_of_match[] = {
{ .compatible = "ti,pcm5121", },
{ .compatible = "ti,pcm5122", },
+ { .compatible = "ti,pcm5141", },
+ { .compatible = "ti,pcm5142", },
{ }
};
MODULE_DEVICE_TABLE(of, pcm512x_of_match);
diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c
index f297058c0038..7b64a9cef704 100644
--- a/sound/soc/codecs/pcm512x-spi.c
+++ b/sound/soc/codecs/pcm512x-spi.c
@@ -43,6 +43,8 @@ static int pcm512x_spi_remove(struct spi_device *spi)
static const struct spi_device_id pcm512x_spi_id[] = {
{ "pcm5121", },
{ "pcm5122", },
+ { "pcm5141", },
+ { "pcm5142", },
{ },
};
MODULE_DEVICE_TABLE(spi, pcm512x_spi_id);
@@ -50,6 +52,8 @@ MODULE_DEVICE_TABLE(spi, pcm512x_spi_id);
static const struct of_device_id pcm512x_of_match[] = {
{ .compatible = "ti,pcm5121", },
{ .compatible = "ti,pcm5122", },
+ { .compatible = "ti,pcm5141", },
+ { .compatible = "ti,pcm5142", },
{ }
};
MODULE_DEVICE_TABLE(of, pcm512x_of_match);
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index e5f2fb884bf3..9974f201a08f 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -21,12 +21,19 @@
#include <linux/pm_runtime.h>
#include <linux/regmap.h>
#include <linux/regulator/consumer.h>
+#include <linux/gcd.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
#include <sound/tlv.h>
#include "pcm512x.h"
+#define DIV_ROUND_DOWN_ULL(ll, d) \
+ ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; })
+#define DIV_ROUND_CLOSEST_ULL(ll, d) \
+ ({ unsigned long long _tmp = (ll)+(d)/2; do_div(_tmp, d); _tmp; })
+
#define PCM512x_NUM_SUPPLIES 3
static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = {
"AVDD",
@@ -39,6 +46,14 @@ struct pcm512x_priv {
struct clk *sclk;
struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES];
struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES];
+ int fmt;
+ int pll_in;
+ int pll_out;
+ int pll_r;
+ int pll_j;
+ int pll_d;
+ int pll_p;
+ unsigned long real_pll;
};
/*
@@ -69,6 +84,7 @@ static const struct reg_default pcm512x_reg_defaults[] = {
{ PCM512x_MUTE, 0x00 },
{ PCM512x_DSP, 0x00 },
{ PCM512x_PLL_REF, 0x00 },
+ { PCM512x_DAC_REF, 0x00 },
{ PCM512x_DAC_ROUTING, 0x11 },
{ PCM512x_DSP_PROGRAM, 0x01 },
{ PCM512x_CLKDET, 0x00 },
@@ -87,6 +103,25 @@ static const struct reg_default pcm512x_reg_defaults[] = {
{ PCM512x_ANALOG_GAIN_BOOST, 0x00 },
{ PCM512x_VCOM_CTRL_1, 0x00 },
{ PCM512x_VCOM_CTRL_2, 0x01 },
+ { PCM512x_BCLK_LRCLK_CFG, 0x00 },
+ { PCM512x_MASTER_MODE, 0x7c },
+ { PCM512x_GPIO_DACIN, 0x00 },
+ { PCM512x_GPIO_PLLIN, 0x00 },
+ { PCM512x_SYNCHRONIZE, 0x10 },
+ { PCM512x_PLL_COEFF_0, 0x00 },
+ { PCM512x_PLL_COEFF_1, 0x00 },
+ { PCM512x_PLL_COEFF_2, 0x00 },
+ { PCM512x_PLL_COEFF_3, 0x00 },
+ { PCM512x_PLL_COEFF_4, 0x00 },
+ { PCM512x_DSP_CLKDIV, 0x00 },
+ { PCM512x_DAC_CLKDIV, 0x00 },
+ { PCM512x_NCP_CLKDIV, 0x00 },
+ { PCM512x_OSR_CLKDIV, 0x00 },
+ { PCM512x_MASTER_CLKDIV_1, 0x00 },
+ { PCM512x_MASTER_CLKDIV_2, 0x00 },
+ { PCM512x_FS_SPEED_MODE, 0x00 },
+ { PCM512x_IDAC_1, 0x01 },
+ { PCM512x_IDAC_2, 0x00 },
};
static bool pcm512x_readable(struct device *dev, unsigned int reg)
@@ -103,6 +138,10 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg)
case PCM512x_DSP_GPIO_INPUT:
case PCM512x_MASTER_MODE:
case PCM512x_PLL_REF:
+ case PCM512x_DAC_REF:
+ case PCM512x_GPIO_DACIN:
+ case PCM512x_GPIO_PLLIN:
+ case PCM512x_SYNCHRONIZE:
case PCM512x_PLL_COEFF_0:
case PCM512x_PLL_COEFF_1:
case PCM512x_PLL_COEFF_2:
@@ -143,6 +182,7 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg)
case PCM512x_RATE_DET_2:
case PCM512x_RATE_DET_3:
case PCM512x_RATE_DET_4:
+ case PCM512x_CLOCK_STATUS:
case PCM512x_ANALOG_MUTE_DET:
case PCM512x_GPIN:
case PCM512x_DIGITAL_MUTE_DET:
@@ -154,6 +194,8 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg)
case PCM512x_VCOM_CTRL_1:
case PCM512x_VCOM_CTRL_2:
case PCM512x_CRAM_CTRL:
+ case PCM512x_FLEX_A:
+ case PCM512x_FLEX_B:
return true;
default:
/* There are 256 raw register addresses */
@@ -170,6 +212,7 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg)
case PCM512x_RATE_DET_2:
case PCM512x_RATE_DET_3:
case PCM512x_RATE_DET_4:
+ case PCM512x_CLOCK_STATUS:
case PCM512x_ANALOG_MUTE_DET:
case PCM512x_GPIN:
case PCM512x_DIGITAL_MUTE_DET:
@@ -188,8 +231,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0);
static const char * const pcm512x_dsp_program_texts[] = {
"FIR interpolation with de-emphasis",
"Low latency IIR with de-emphasis",
- "Fixed process flow",
"High attenuation with de-emphasis",
+ "Fixed process flow",
"Ringing-less low latency FIR",
};
@@ -277,7 +320,7 @@ SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r),
SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3,
PCM512x_ACTL_SHIFT, 1, 0),
SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT,
- PCM512x_AMLR_SHIFT, 1, 0),
+ PCM512x_AMRE_SHIFT, 1, 0),
SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf),
SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds),
@@ -303,6 +346,136 @@ static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = {
{ "OUTR", NULL, "DACR" },
};
+static const u32 pcm512x_dai_rates[] = {
+ 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
+ 88200, 96000, 176400, 192000, 384000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_slave = {
+ .count = ARRAY_SIZE(pcm512x_dai_rates),
+ .list = pcm512x_dai_rates,
+};
+
+static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval ranges[2];
+ int frame_size;
+
+ frame_size = snd_soc_params_to_frame_size(params);
+ if (frame_size < 0)
+ return frame_size;
+
+ switch (frame_size) {
+ case 32:
+ /* No hole when the frame size is 32. */
+ return 0;
+ case 48:
+ case 64:
+ /* There is only one hole in the range of supported
+ * rates, but it moves with the frame size.
+ */
+ memset(ranges, 0, sizeof(ranges));
+ ranges[0].min = 8000;
+ ranges[0].max = 25000000 / frame_size / 2;
+ ranges[1].min = DIV_ROUND_UP(16000000, frame_size);
+ ranges[1].max = 384000;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return snd_interval_ranges(hw_param_interval(params, rule->var),
+ ARRAY_SIZE(ranges), ranges, 0);
+}
+
+static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = dai->dev;
+ struct snd_pcm_hw_constraint_ratnums *constraints_no_pll;
+ struct snd_ratnum *rats_no_pll;
+
+ if (IS_ERR(pcm512x->sclk)) {
+ dev_err(dev, "Need SCLK for master mode: %ld\n",
+ PTR_ERR(pcm512x->sclk));
+ return PTR_ERR(pcm512x->sclk);
+ }
+
+ if (pcm512x->pll_out)
+ return snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ pcm512x_hw_rule_rate,
+ NULL,
+ SNDRV_PCM_HW_PARAM_FRAME_BITS,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+
+ constraints_no_pll = devm_kzalloc(dev, sizeof(*constraints_no_pll),
+ GFP_KERNEL);
+ if (!constraints_no_pll)
+ return -ENOMEM;
+ constraints_no_pll->nrats = 1;
+ rats_no_pll = devm_kzalloc(dev, sizeof(*rats_no_pll), GFP_KERNEL);
+ if (!rats_no_pll)
+ return -ENOMEM;
+ constraints_no_pll->rats = rats_no_pll;
+ rats_no_pll->num = clk_get_rate(pcm512x->sclk) / 64;
+ rats_no_pll->den_min = 1;
+ rats_no_pll->den_max = 128;
+ rats_no_pll->den_step = 1;
+
+ return snd_pcm_hw_constraint_ratnums(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ constraints_no_pll);
+}
+
+static int pcm512x_dai_startup_slave(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = dai->dev;
+ struct regmap *regmap = pcm512x->regmap;
+
+ if (IS_ERR(pcm512x->sclk)) {
+ dev_info(dev, "No SCLK, using BCLK: %ld\n",
+ PTR_ERR(pcm512x->sclk));
+
+ /* Disable reporting of missing SCLK as an error */
+ regmap_update_bits(regmap, PCM512x_ERROR_DETECT,
+ PCM512x_IDCH, PCM512x_IDCH);
+
+ /* Switch PLL input to BCLK */
+ regmap_update_bits(regmap, PCM512x_PLL_REF,
+ PCM512x_SREF, PCM512x_SREF_BCK);
+ }
+
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_slave);
+}
+
+static int pcm512x_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+ switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ return pcm512x_dai_startup_master(substream, dai);
+
+ case SND_SOC_DAIFMT_CBS_CFS:
+ return pcm512x_dai_startup_slave(substream, dai);
+
+ default:
+ return -EINVAL;
+ }
+}
+
static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -340,17 +513,717 @@ static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
+static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai,
+ unsigned long bclk_rate)
+{
+ struct device *dev = dai->dev;
+ unsigned long sck_rate;
+ int pow2;
+
+ /* 64 MHz <= pll_rate <= 100 MHz, VREF mode */
+ /* 16 MHz <= sck_rate <= 25 MHz, VREF mode */
+
+ /* select sck_rate as a multiple of bclk_rate but still with
+ * as many factors of 2 as possible, as that makes it easier
+ * to find a fast DAC rate
+ */
+ pow2 = 1 << fls((25000000 - 16000000) / bclk_rate);
+ for (; pow2; pow2 >>= 1) {
+ sck_rate = rounddown(25000000, bclk_rate * pow2);
+ if (sck_rate >= 16000000)
+ break;
+ }
+ if (!pow2) {
+ dev_err(dev, "Impossible to generate a suitable SCK\n");
+ return 0;
+ }
+
+ dev_dbg(dev, "sck_rate %lu\n", sck_rate);
+ return sck_rate;
+}
+
+/* pll_rate = pllin_rate * R * J.D / P
+ * 1 <= R <= 16
+ * 1 <= J <= 63
+ * 0 <= D <= 9999
+ * 1 <= P <= 15
+ * 64 MHz <= pll_rate <= 100 MHz
+ * if D == 0
+ * 1 MHz <= pllin_rate / P <= 20 MHz
+ * else if D > 0
+ * 6.667 MHz <= pllin_rate / P <= 20 MHz
+ * 4 <= J <= 11
+ * R = 1
+ */
+static int pcm512x_find_pll_coeff(struct snd_soc_dai *dai,
+ unsigned long pllin_rate,
+ unsigned long pll_rate)
+{
+ struct device *dev = dai->dev;
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+ unsigned long common;
+ int R, J, D, P;
+ unsigned long K; /* 10000 * J.D */
+ unsigned long num;
+ unsigned long den;
+
+ common = gcd(pll_rate, pllin_rate);
+ dev_dbg(dev, "pll %lu pllin %lu common %lu\n",
+ pll_rate, pllin_rate, common);
+ num = pll_rate / common;
+ den = pllin_rate / common;
+
+ /* pllin_rate / P (or here, den) cannot be greater than 20 MHz */
+ if (pllin_rate / den > 20000000 && num < 8) {
+ num *= 20000000 / (pllin_rate / den);
+ den *= 20000000 / (pllin_rate / den);
+ }
+ dev_dbg(dev, "num / den = %lu / %lu\n", num, den);
+
+ P = den;
+ if (den <= 15 && num <= 16 * 63
+ && 1000000 <= pllin_rate / P && pllin_rate / P <= 20000000) {
+ /* Try the case with D = 0 */
+ D = 0;
+ /* factor 'num' into J and R, such that R <= 16 and J <= 63 */
+ for (R = 16; R; R--) {
+ if (num % R)
+ continue;
+ J = num / R;
+ if (J == 0 || J > 63)
+ continue;
+
+ dev_dbg(dev, "R * J / P = %d * %d / %d\n", R, J, P);
+ pcm512x->real_pll = pll_rate;
+ goto done;
+ }
+ /* no luck */
+ }
+
+ R = 1;
+
+ if (num > 0xffffffffUL / 10000)
+ goto fallback;
+
+ /* Try to find an exact pll_rate using the D > 0 case */
+ common = gcd(10000 * num, den);
+ num = 10000 * num / common;
+ den /= common;
+ dev_dbg(dev, "num %lu den %lu common %lu\n", num, den, common);
+
+ for (P = den; P <= 15; P++) {
+ if (pllin_rate / P < 6667000 || 200000000 < pllin_rate / P)
+ continue;
+ if (num * P % den)
+ continue;
+ K = num * P / den;
+ /* J == 12 is ok if D == 0 */
+ if (K < 40000 || K > 120000)
+ continue;
+
+ J = K / 10000;
+ D = K % 10000;
+ dev_dbg(dev, "J.D / P = %d.%04d / %d\n", J, D, P);
+ pcm512x->real_pll = pll_rate;
+ goto done;
+ }
+
+ /* Fall back to an approximate pll_rate */
+
+fallback:
+ /* find smallest possible P */
+ P = DIV_ROUND_UP(pllin_rate, 20000000);
+ if (!P)
+ P = 1;
+ else if (P > 15) {
+ dev_err(dev, "Need a slower clock as pll-input\n");
+ return -EINVAL;
+ }
+ if (pllin_rate / P < 6667000) {
+ dev_err(dev, "Need a faster clock as pll-input\n");
+ return -EINVAL;
+ }
+ K = DIV_ROUND_CLOSEST_ULL(10000ULL * pll_rate * P, pllin_rate);
+ if (K < 40000)
+ K = 40000;
+ /* J == 12 is ok if D == 0 */
+ if (K > 120000)
+ K = 120000;
+ J = K / 10000;
+ D = K % 10000;
+ dev_dbg(dev, "J.D / P ~ %d.%04d / %d\n", J, D, P);
+ pcm512x->real_pll = DIV_ROUND_DOWN_ULL((u64)K * pllin_rate, 10000 * P);
+
+done:
+ pcm512x->pll_r = R;
+ pcm512x->pll_j = J;
+ pcm512x->pll_d = D;
+ pcm512x->pll_p = P;
+ return 0;
+}
+
+static unsigned long pcm512x_pllin_dac_rate(struct snd_soc_dai *dai,
+ unsigned long osr_rate,
+ unsigned long pllin_rate)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+ unsigned long dac_rate;
+
+ if (!pcm512x->pll_out)
+ return 0; /* no PLL to bypass, force SCK as DAC input */
+
+ if (pllin_rate % osr_rate)
+ return 0; /* futile, quit early */
+
+ /* run DAC no faster than 6144000 Hz */
+ for (dac_rate = rounddown(6144000, osr_rate);
+ dac_rate;
+ dac_rate -= osr_rate) {
+
+ if (pllin_rate / dac_rate > 128)
+ return 0; /* DAC divider would be too big */
+
+ if (!(pllin_rate % dac_rate))
+ return dac_rate;
+
+ dac_rate -= osr_rate;
+ }
+
+ return 0;
+}
+
+static int pcm512x_set_dividers(struct snd_soc_dai *dai,
+ struct snd_pcm_hw_params *params)
+{
+ struct device *dev = dai->dev;
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+ unsigned long pllin_rate = 0;
+ unsigned long pll_rate;
+ unsigned long sck_rate;
+ unsigned long mck_rate;
+ unsigned long bclk_rate;
+ unsigned long sample_rate;
+ unsigned long osr_rate;
+ unsigned long dacsrc_rate;
+ int bclk_div;
+ int lrclk_div;
+ int dsp_div;
+ int dac_div;
+ unsigned long dac_rate;
+ int ncp_div;
+ int osr_div;
+ int ret;
+ int idac;
+ int fssp;
+ int gpio;
+
+ lrclk_div = snd_soc_params_to_frame_size(params);
+ if (lrclk_div == 0) {
+ dev_err(dev, "No LRCLK?\n");
+ return -EINVAL;
+ }
+
+ if (!pcm512x->pll_out) {
+ sck_rate = clk_get_rate(pcm512x->sclk);
+ bclk_div = params->rate_den * 64 / lrclk_div;
+ bclk_rate = DIV_ROUND_CLOSEST(sck_rate, bclk_div);
+
+ mck_rate = sck_rate;
+ } else {
+ ret = snd_soc_params_to_bclk(params);
+ if (ret < 0) {
+ dev_err(dev, "Failed to find suitable BCLK: %d\n", ret);
+ return ret;
+ }
+ if (ret == 0) {
+ dev_err(dev, "No BCLK?\n");
+ return -EINVAL;
+ }
+ bclk_rate = ret;
+
+ pllin_rate = clk_get_rate(pcm512x->sclk);
+
+ sck_rate = pcm512x_find_sck(dai, bclk_rate);
+ if (!sck_rate)
+ return -EINVAL;
+ pll_rate = 4 * sck_rate;
+
+ ret = pcm512x_find_pll_coeff(dai, pllin_rate, pll_rate);
+ if (ret != 0)
+ return ret;
+
+ ret = regmap_write(pcm512x->regmap,
+ PCM512x_PLL_COEFF_0, pcm512x->pll_p - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write PLL P: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap,
+ PCM512x_PLL_COEFF_1, pcm512x->pll_j);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write PLL J: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap,
+ PCM512x_PLL_COEFF_2, pcm512x->pll_d >> 8);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write PLL D msb: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap,
+ PCM512x_PLL_COEFF_3, pcm512x->pll_d & 0xff);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write PLL D lsb: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap,
+ PCM512x_PLL_COEFF_4, pcm512x->pll_r - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write PLL R: %d\n", ret);
+ return ret;
+ }
+
+ mck_rate = pcm512x->real_pll;
+
+ bclk_div = DIV_ROUND_CLOSEST(sck_rate, bclk_rate);
+ }
+
+ if (bclk_div > 128) {
+ dev_err(dev, "Failed to find BCLK divider\n");
+ return -EINVAL;
+ }
+
+ /* the actual rate */
+ sample_rate = sck_rate / bclk_div / lrclk_div;
+ osr_rate = 16 * sample_rate;
+
+ /* run DSP no faster than 50 MHz */
+ dsp_div = mck_rate > 50000000 ? 2 : 1;
+
+ dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate);
+ if (dac_rate) {
+ /* the desired clock rate is "compatible" with the pll input
+ * clock, so use that clock as dac input instead of the pll
+ * output clock since the pll will introduce jitter and thus
+ * noise.
+ */
+ dev_dbg(dev, "using pll input as dac input\n");
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_DAC_REF,
+ PCM512x_SDAC, PCM512x_SDAC_GPIO);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to set gpio as dacref: %d\n", ret);
+ return ret;
+ }
+
+ gpio = PCM512x_GREF_GPIO1 + pcm512x->pll_in - 1;
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_DACIN,
+ PCM512x_GREF, gpio);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to set gpio %d as dacin: %d\n",
+ pcm512x->pll_in, ret);
+ return ret;
+ }
+
+ dacsrc_rate = pllin_rate;
+ } else {
+ /* run DAC no faster than 6144000 Hz */
+ unsigned long dac_mul = 6144000 / osr_rate;
+ unsigned long sck_mul = sck_rate / osr_rate;
+
+ for (; dac_mul; dac_mul--) {
+ if (!(sck_mul % dac_mul))
+ break;
+ }
+ if (!dac_mul) {
+ dev_err(dev, "Failed to find DAC rate\n");
+ return -EINVAL;
+ }
+
+ dac_rate = dac_mul * osr_rate;
+ dev_dbg(dev, "dac_rate %lu sample_rate %lu\n",
+ dac_rate, sample_rate);
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_DAC_REF,
+ PCM512x_SDAC, PCM512x_SDAC_SCK);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to set sck as dacref: %d\n", ret);
+ return ret;
+ }
+
+ dacsrc_rate = sck_rate;
+ }
+
+ dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate);
+ if (dac_div > 128) {
+ dev_err(dev, "Failed to find DAC divider\n");
+ return -EINVAL;
+ }
+
+ ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000);
+ if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) {
+ /* run NCP no faster than 2048000 Hz, but why? */
+ ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000);
+ if (ncp_div > 128) {
+ dev_err(dev, "Failed to find NCP divider\n");
+ return -EINVAL;
+ }
+ }
+
+ osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate);
+ if (osr_div > 128) {
+ dev_err(dev, "Failed to find OSR divider\n");
+ return -EINVAL;
+ }
+
+ idac = mck_rate / (dsp_div * sample_rate);
+
+ ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write DSP divider: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap, PCM512x_DAC_CLKDIV, dac_div - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write DAC divider: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap, PCM512x_NCP_CLKDIV, ncp_div - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write NCP divider: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap, PCM512x_OSR_CLKDIV, osr_div - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write OSR divider: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap,
+ PCM512x_MASTER_CLKDIV_1, bclk_div - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write BCLK divider: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap,
+ PCM512x_MASTER_CLKDIV_2, lrclk_div - 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write LRCLK divider: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap, PCM512x_IDAC_1, idac >> 8);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write IDAC msb divider: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap, PCM512x_IDAC_2, idac & 0xff);
+ if (ret != 0) {
+ dev_err(dev, "Failed to write IDAC lsb divider: %d\n", ret);
+ return ret;
+ }
+
+ if (sample_rate <= 48000)
+ fssp = PCM512x_FSSP_48KHZ;
+ else if (sample_rate <= 96000)
+ fssp = PCM512x_FSSP_96KHZ;
+ else if (sample_rate <= 192000)
+ fssp = PCM512x_FSSP_192KHZ;
+ else
+ fssp = PCM512x_FSSP_384KHZ;
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_FS_SPEED_MODE,
+ PCM512x_FSSP, fssp);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set fs speed: %d\n", ret);
+ return ret;
+ }
+
+ dev_dbg(codec->dev, "DSP divider %d\n", dsp_div);
+ dev_dbg(codec->dev, "DAC divider %d\n", dac_div);
+ dev_dbg(codec->dev, "NCP divider %d\n", ncp_div);
+ dev_dbg(codec->dev, "OSR divider %d\n", osr_div);
+ dev_dbg(codec->dev, "BCK divider %d\n", bclk_div);
+ dev_dbg(codec->dev, "LRCK divider %d\n", lrclk_div);
+ dev_dbg(codec->dev, "IDAC %d\n", idac);
+ dev_dbg(codec->dev, "1<<FSSP %d\n", 1 << fssp);
+
+ return 0;
+}
+
+static int pcm512x_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 pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+ int alen;
+ int gpio;
+ int clock_output;
+ int master_mode;
+ int ret;
+
+ dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n",
+ params_rate(params),
+ params_channels(params));
+
+ switch (snd_pcm_format_width(params_format(params))) {
+ case 16:
+ alen = PCM512x_ALEN_16;
+ break;
+ case 20:
+ alen = PCM512x_ALEN_20;
+ break;
+ case 24:
+ alen = PCM512x_ALEN_24;
+ break;
+ case 32:
+ alen = PCM512x_ALEN_32;
+ break;
+ default:
+ dev_err(codec->dev, "Bad frame size: %d\n",
+ snd_pcm_format_width(params_format(params)));
+ return -EINVAL;
+ }
+
+ switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ret = regmap_update_bits(pcm512x->regmap,
+ PCM512x_BCLK_LRCLK_CFG,
+ PCM512x_BCKP
+ | PCM512x_BCKO | PCM512x_LRKO,
+ 0);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to enable slave mode: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT,
+ PCM512x_DCAS, 0);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to enable clock divider autoset: %d\n",
+ ret);
+ return ret;
+ }
+ return 0;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ clock_output = PCM512x_BCKO | PCM512x_LRKO;
+ master_mode = PCM512x_RLRK | PCM512x_RBCK;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ clock_output = PCM512x_BCKO;
+ master_mode = PCM512x_RBCK;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_I2S_1,
+ PCM512x_ALEN, alen);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set frame size: %d\n", ret);
+ return ret;
+ }
+
+ if (pcm512x->pll_out) {
+ ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_A, 0x11);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set FLEX_A: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_B, 0xff);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set FLEX_B: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT,
+ PCM512x_IDFS | PCM512x_IDBK
+ | PCM512x_IDSK | PCM512x_IDCH
+ | PCM512x_IDCM | PCM512x_DCAS
+ | PCM512x_IPLK,
+ PCM512x_IDFS | PCM512x_IDBK
+ | PCM512x_IDSK | PCM512x_IDCH
+ | PCM512x_DCAS);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to ignore auto-clock failures: %d\n",
+ ret);
+ return ret;
+ }
+ } else {
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT,
+ PCM512x_IDFS | PCM512x_IDBK
+ | PCM512x_IDSK | PCM512x_IDCH
+ | PCM512x_IDCM | PCM512x_DCAS
+ | PCM512x_IPLK,
+ PCM512x_IDFS | PCM512x_IDBK
+ | PCM512x_IDSK | PCM512x_IDCH
+ | PCM512x_DCAS | PCM512x_IPLK);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to ignore auto-clock failures: %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_EN,
+ PCM512x_PLLE, 0);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to disable pll: %d\n", ret);
+ return ret;
+ }
+ }
+
+ ret = pcm512x_set_dividers(dai, params);
+ if (ret != 0)
+ return ret;
+
+ if (pcm512x->pll_out) {
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_REF,
+ PCM512x_SREF, PCM512x_SREF_GPIO);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to set gpio as pllref: %d\n", ret);
+ return ret;
+ }
+
+ gpio = PCM512x_GREF_GPIO1 + pcm512x->pll_in - 1;
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_PLLIN,
+ PCM512x_GREF, gpio);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to set gpio %d as pllin: %d\n",
+ pcm512x->pll_in, ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_EN,
+ PCM512x_PLLE, PCM512x_PLLE);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable pll: %d\n", ret);
+ return ret;
+ }
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_BCLK_LRCLK_CFG,
+ PCM512x_BCKP | PCM512x_BCKO | PCM512x_LRKO,
+ clock_output);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable clock output: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_MASTER_MODE,
+ PCM512x_RLRK | PCM512x_RBCK,
+ master_mode);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable master mode: %d\n", ret);
+ return ret;
+ }
+
+ if (pcm512x->pll_out) {
+ gpio = PCM512x_G1OE << (pcm512x->pll_out - 1);
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN,
+ gpio, gpio);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable gpio %d: %d\n",
+ pcm512x->pll_out, ret);
+ return ret;
+ }
+
+ gpio = PCM512x_GPIO_OUTPUT_1 + pcm512x->pll_out - 1;
+ ret = regmap_update_bits(pcm512x->regmap, gpio,
+ PCM512x_GxSL, PCM512x_GxSL_PLLCK);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to output pll on %d: %d\n",
+ ret, pcm512x->pll_out);
+ return ret;
+ }
+
+ gpio = PCM512x_G1OE << (4 - 1);
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN,
+ gpio, gpio);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable gpio %d: %d\n",
+ 4, ret);
+ return ret;
+ }
+
+ gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1;
+ ret = regmap_update_bits(pcm512x->regmap, gpio,
+ PCM512x_GxSL, PCM512x_GxSL_PLLLK);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to output pll lock on %d: %d\n",
+ ret, 4);
+ return ret;
+ }
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE,
+ PCM512x_RQSY, PCM512x_RQSY_HALT);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to halt clocks: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE,
+ PCM512x_RQSY, PCM512x_RQSY_RESUME);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to resume clocks: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+ pcm512x->fmt = fmt;
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops pcm512x_dai_ops = {
+ .startup = pcm512x_dai_startup,
+ .hw_params = pcm512x_hw_params,
+ .set_fmt = pcm512x_set_fmt,
+};
+
static struct snd_soc_dai_driver pcm512x_dai = {
.name = "pcm512x-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 8000,
+ .rate_max = 384000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S32_LE
},
+ .ops = &pcm512x_dai_ops,
};
static struct snd_soc_codec_driver pcm512x_codec_driver = {
@@ -448,21 +1321,9 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap)
}
pcm512x->sclk = devm_clk_get(dev, NULL);
- if (IS_ERR(pcm512x->sclk)) {
- if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER)
- return -EPROBE_DEFER;
-
- dev_info(dev, "No SCLK, using BCLK: %ld\n",
- PTR_ERR(pcm512x->sclk));
-
- /* Disable reporting of missing SCLK as an error */
- regmap_update_bits(regmap, PCM512x_ERROR_DETECT,
- PCM512x_IDCH, PCM512x_IDCH);
-
- /* Switch PLL input to BCLK */
- regmap_update_bits(regmap, PCM512x_PLL_REF,
- PCM512x_SREF, PCM512x_SREF);
- } else {
+ if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+ if (!IS_ERR(pcm512x->sclk)) {
ret = clk_prepare_enable(pcm512x->sclk);
if (ret != 0) {
dev_err(dev, "Failed to enable SCLK: %d\n", ret);
@@ -483,6 +1344,43 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap)
pm_runtime_enable(dev);
pm_runtime_idle(dev);
+#ifdef CONFIG_OF
+ if (dev->of_node) {
+ const struct device_node *np = dev->of_node;
+ u32 val;
+
+ if (of_property_read_u32(np, "pll-in", &val) >= 0) {
+ if (val > 6) {
+ dev_err(dev, "Invalid pll-in\n");
+ ret = -EINVAL;
+ goto err_clk;
+ }
+ pcm512x->pll_in = val;
+ }
+
+ if (of_property_read_u32(np, "pll-out", &val) >= 0) {
+ if (val > 6) {
+ dev_err(dev, "Invalid pll-out\n");
+ ret = -EINVAL;
+ goto err_clk;
+ }
+ pcm512x->pll_out = val;
+ }
+
+ if (!pcm512x->pll_in != !pcm512x->pll_out) {
+ dev_err(dev,
+ "Error: both pll-in and pll-out, or none\n");
+ ret = -EINVAL;
+ goto err_clk;
+ }
+ if (pcm512x->pll_in && pcm512x->pll_in == pcm512x->pll_out) {
+ dev_err(dev, "Error: pll-in == pll-out\n");
+ ret = -EINVAL;
+ goto err_clk;
+ }
+ }
+#endif
+
ret = snd_soc_register_codec(dev, &pcm512x_codec_driver,
&pcm512x_dai, 1);
if (ret != 0) {
diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h
index 6ee76aaca09a..b7c310207223 100644
--- a/sound/soc/codecs/pcm512x.h
+++ b/sound/soc/codecs/pcm512x.h
@@ -37,6 +37,10 @@
#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10)
#define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12)
#define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13)
+#define PCM512x_DAC_REF (PCM512x_PAGE_BASE(0) + 14)
+#define PCM512x_GPIO_DACIN (PCM512x_PAGE_BASE(0) + 16)
+#define PCM512x_GPIO_PLLIN (PCM512x_PAGE_BASE(0) + 18)
+#define PCM512x_SYNCHRONIZE (PCM512x_PAGE_BASE(0) + 19)
#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20)
#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21)
#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22)
@@ -77,6 +81,7 @@
#define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92)
#define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93)
#define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94)
+#define PCM512x_CLOCK_STATUS (PCM512x_PAGE_BASE(0) + 95)
#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108)
#define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119)
#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120)
@@ -91,7 +96,10 @@
#define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1)
-#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1)
+#define PCM512x_FLEX_A (PCM512x_PAGE_BASE(253) + 63)
+#define PCM512x_FLEX_B (PCM512x_PAGE_BASE(253) + 64)
+
+#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(253) + 64)
/* Page 0, Register 1 - reset */
#define PCM512x_RSTR (1 << 0)
@@ -108,8 +116,8 @@
#define PCM512x_RQML_SHIFT 4
/* Page 0, Register 4 - PLL */
-#define PCM512x_PLCE (1 << 0)
-#define PCM512x_RLCE_SHIFT 0
+#define PCM512x_PLLE (1 << 0)
+#define PCM512x_PLLE_SHIFT 0
#define PCM512x_PLCK (1 << 4)
#define PCM512x_PLCK_SHIFT 4
@@ -119,8 +127,66 @@
#define PCM512x_DEMP (1 << 4)
#define PCM512x_DEMP_SHIFT 4
+/* Page 0, Register 8 - GPIO output enable */
+#define PCM512x_G1OE (1 << 0)
+#define PCM512x_G2OE (1 << 1)
+#define PCM512x_G3OE (1 << 2)
+#define PCM512x_G4OE (1 << 3)
+#define PCM512x_G5OE (1 << 4)
+#define PCM512x_G6OE (1 << 5)
+
+/* Page 0, Register 9 - BCK, LRCLK configuration */
+#define PCM512x_LRKO (1 << 0)
+#define PCM512x_LRKO_SHIFT 0
+#define PCM512x_BCKO (1 << 4)
+#define PCM512x_BCKO_SHIFT 4
+#define PCM512x_BCKP (1 << 5)
+#define PCM512x_BCKP_SHIFT 5
+
+/* Page 0, Register 12 - Master mode BCK, LRCLK reset */
+#define PCM512x_RLRK (1 << 0)
+#define PCM512x_RLRK_SHIFT 0
+#define PCM512x_RBCK (1 << 1)
+#define PCM512x_RBCK_SHIFT 1
+
/* Page 0, Register 13 - PLL reference */
-#define PCM512x_SREF (1 << 4)
+#define PCM512x_SREF (7 << 4)
+#define PCM512x_SREF_SHIFT 4
+#define PCM512x_SREF_SCK (0 << 4)
+#define PCM512x_SREF_BCK (1 << 4)
+#define PCM512x_SREF_GPIO (3 << 4)
+
+/* Page 0, Register 14 - DAC reference */
+#define PCM512x_SDAC (7 << 4)
+#define PCM512x_SDAC_SHIFT 4
+#define PCM512x_SDAC_MCK (0 << 4)
+#define PCM512x_SDAC_PLL (1 << 4)
+#define PCM512x_SDAC_SCK (3 << 4)
+#define PCM512x_SDAC_BCK (4 << 4)
+#define PCM512x_SDAC_GPIO (5 << 4)
+
+/* Page 0, Register 16, 18 - GPIO source for DAC, PLL */
+#define PCM512x_GREF (7 << 0)
+#define PCM512x_GREF_SHIFT 0
+#define PCM512x_GREF_GPIO1 (0 << 0)
+#define PCM512x_GREF_GPIO2 (1 << 0)
+#define PCM512x_GREF_GPIO3 (2 << 0)
+#define PCM512x_GREF_GPIO4 (3 << 0)
+#define PCM512x_GREF_GPIO5 (4 << 0)
+#define PCM512x_GREF_GPIO6 (5 << 0)
+
+/* Page 0, Register 19 - synchronize */
+#define PCM512x_RQSY (1 << 0)
+#define PCM512x_RQSY_RESUME (0 << 0)
+#define PCM512x_RQSY_HALT (1 << 0)
+
+/* Page 0, Register 34 - fs speed mode */
+#define PCM512x_FSSP (3 << 0)
+#define PCM512x_FSSP_SHIFT 0
+#define PCM512x_FSSP_48KHZ (0 << 0)
+#define PCM512x_FSSP_96KHZ (1 << 0)
+#define PCM512x_FSSP_192KHZ (2 << 0)
+#define PCM512x_FSSP_384KHZ (3 << 0)
/* Page 0, Register 37 - Error detection */
#define PCM512x_IPLK (1 << 0)
@@ -131,6 +197,20 @@
#define PCM512x_IDBK (1 << 5)
#define PCM512x_IDFS (1 << 6)
+/* Page 0, Register 40 - I2S configuration */
+#define PCM512x_ALEN (3 << 0)
+#define PCM512x_ALEN_SHIFT 0
+#define PCM512x_ALEN_16 (0 << 0)
+#define PCM512x_ALEN_20 (1 << 0)
+#define PCM512x_ALEN_24 (2 << 0)
+#define PCM512x_ALEN_32 (3 << 0)
+#define PCM512x_AFMT (3 << 4)
+#define PCM512x_AFMT_SHIFT 4
+#define PCM512x_AFMT_I2S (0 << 4)
+#define PCM512x_AFMT_DSP (1 << 4)
+#define PCM512x_AFMT_RTJ (2 << 4)
+#define PCM512x_AFMT_LTJ (3 << 4)
+
/* Page 0, Register 42 - DAC routing */
#define PCM512x_AUPR_SHIFT 0
#define PCM512x_AUPL_SHIFT 4
@@ -152,7 +232,26 @@
/* Page 0, Register 65 - Digital mute enables */
#define PCM512x_ACTL_SHIFT 2
#define PCM512x_AMLE_SHIFT 1
-#define PCM512x_AMLR_SHIFT 0
+#define PCM512x_AMRE_SHIFT 0
+
+/* Page 0, Register 80-85, GPIO output selection */
+#define PCM512x_GxSL (31 << 0)
+#define PCM512x_GxSL_SHIFT 0
+#define PCM512x_GxSL_OFF (0 << 0)
+#define PCM512x_GxSL_DSP (1 << 0)
+#define PCM512x_GxSL_REG (2 << 0)
+#define PCM512x_GxSL_AMUTB (3 << 0)
+#define PCM512x_GxSL_AMUTL (4 << 0)
+#define PCM512x_GxSL_AMUTR (5 << 0)
+#define PCM512x_GxSL_CLKI (6 << 0)
+#define PCM512x_GxSL_SDOUT (7 << 0)
+#define PCM512x_GxSL_ANMUL (8 << 0)
+#define PCM512x_GxSL_ANMUR (9 << 0)
+#define PCM512x_GxSL_PLLLK (10 << 0)
+#define PCM512x_GxSL_CPCLK (11 << 0)
+#define PCM512x_GxSL_UV0_7 (14 << 0)
+#define PCM512x_GxSL_UV0_3 (15 << 0)
+#define PCM512x_GxSL_PLLCK (16 << 0)
/* Page 1, Register 2 - analog volume control */
#define PCM512x_RAGN_SHIFT 0
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 2cd4fe463102..f374840a5a7c 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -34,6 +34,7 @@
#include "rt286.h"
#define RT286_VENDOR_ID 0x10ec0286
+#define RT288_VENDOR_ID 0x10ec0288
struct rt286_priv {
struct regmap *regmap;
@@ -305,6 +306,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
*hp = false;
*mic = false;
+ if (!rt286->codec)
+ return -EINVAL;
if (rt286->pdata.cbj_en) {
regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
*hp = buf & 0x80000000;
@@ -403,7 +406,8 @@ EXPORT_SYMBOL_GPL(rt286_mic_detect);
static int is_mclk_mode(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(source->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
if (rt286->clk_id == RT286_SCLK_S_MCLK)
return 1;
@@ -417,6 +421,8 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0);
static const struct snd_kcontrol_new rt286_snd_controls[] = {
SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN,
RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+ SOC_DOUBLE_R("ADC0 Capture Switch", RT286_ADCL_GAIN,
+ RT286_ADCR_GAIN, 7, 1, 1),
SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN,
RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv),
SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN,
@@ -500,7 +506,7 @@ SOC_DAPM_ENUM("SPO source", rt286_spo_enum);
static int rt286_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -522,7 +528,7 @@ static int rt286_spk_event(struct snd_soc_dapm_widget *w,
static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -538,36 +544,10 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static int rt286_adc_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = w->codec;
- unsigned int nid;
-
- nid = (w->reg >> 20) & 0xff;
-
- switch (event) {
- case SND_SOC_DAPM_POST_PMU:
- snd_soc_update_bits(codec,
- VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
- 0x7080, 0x7000);
- break;
- case SND_SOC_DAPM_PRE_PMD:
- snd_soc_update_bits(codec,
- VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
- 0x7080, 0x7080);
- break;
- default:
- return 0;
- }
-
- return 0;
-}
-
static int rt286_vref_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -585,7 +565,7 @@ static int rt286_vref_event(struct snd_soc_dapm_widget *w,
static int rt286_ldo2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -604,7 +584,7 @@ static int rt286_ldo2_event(struct snd_soc_dapm_widget *w,
static int rt286_mic1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -667,12 +647,10 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0),
/* ADC Mux */
- SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1,
- &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
- SND_SOC_DAPM_POST_PMU),
- SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1,
- &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
- SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MUX("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1,
+ &rt286_adc0_mux),
+ SND_SOC_DAPM_MUX("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1,
+ &rt286_adc1_mux),
/* Audio Interface */
SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
@@ -861,10 +839,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream,
RT286_I2S_CTRL1, 0x0018, d_len_code << 3);
dev_dbg(codec->dev, "format val = 0x%x\n", val);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
- else
- snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
+ snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
+ snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
return 0;
}
@@ -1196,6 +1172,7 @@ static const struct regmap_config rt286_regmap = {
static const struct i2c_device_id rt286_i2c_id[] = {
{"rt286", 0},
+ {"rt288", 0},
{}
};
MODULE_DEVICE_TABLE(i2c, rt286_i2c_id);
@@ -1216,6 +1193,17 @@ static struct dmi_system_id force_combo_jack_table[] = {
{ }
};
+static struct dmi_system_id dmi_dell_dino[] = {
+ {
+ .ident = "Dell Dino",
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
+ DMI_MATCH(DMI_BOARD_NAME, "0144P8")
+ }
+ },
+ { }
+};
+
static int rt286_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1238,7 +1226,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
regmap_read(rt286->regmap,
RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret);
- if (ret != RT286_VENDOR_ID) {
+ if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) {
dev_err(&i2c->dev,
"Device with ID register %x is not rt286\n", ret);
return -ENODEV;
@@ -1251,7 +1239,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
if (pdata)
rt286->pdata = *pdata;
- if (dmi_check_system(force_combo_jack_table))
+ if (dmi_check_system(force_combo_jack_table) ||
+ dmi_check_system(dmi_dell_dino))
rt286->pdata.cbj_en = true;
regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
@@ -1290,6 +1279,17 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
+ if (dmi_check_system(dmi_dell_dino)) {
+ regmap_update_bits(rt286->regmap,
+ RT286_SET_GPIO_MASK, 0x40, 0x40);
+ regmap_update_bits(rt286->regmap,
+ RT286_SET_GPIO_DIRECTION, 0x40, 0x40);
+ regmap_update_bits(rt286->regmap,
+ RT286_SET_GPIO_DATA, 0x40, 0x40);
+ regmap_update_bits(rt286->regmap,
+ RT286_GPIO_CTRL, 0xc, 0x8);
+ }
+
if (rt286->i2c->irq) {
ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq,
IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286);
diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h
index b539b7320a79..7130edb152ef 100644
--- a/sound/soc/codecs/rt286.h
+++ b/sound/soc/codecs/rt286.h
@@ -117,6 +117,12 @@
VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0)
#define RT286_PROC_COEF\
VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0)
+#define RT286_SET_GPIO_MASK\
+ VERB_CMD(AC_VERB_SET_GPIO_MASK, RT286_AUDIO_FUNCTION_GROUP, 0)
+#define RT286_SET_GPIO_DIRECTION\
+ VERB_CMD(AC_VERB_SET_GPIO_DIRECTION, RT286_AUDIO_FUNCTION_GROUP, 0)
+#define RT286_SET_GPIO_DATA\
+ VERB_CMD(AC_VERB_SET_GPIO_DATA, RT286_AUDIO_FUNCTION_GROUP, 0)
/* Index registers */
#define RT286_A_BIAS_CTRL1 0x01
@@ -131,6 +137,7 @@
#define RT286_POWER_CTRL3 0x0f
#define RT286_MIC1_DET_CTRL 0x19
#define RT286_MISC_CTRL1 0x20
+#define RT286_GPIO_CTRL 0x29
#define RT286_IRQ_CTRL 0x33
#define RT286_PLL_CTRL1 0x49
#define RT286_CBJ_CTRL1 0x4f
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 6d7b7ca7d530..c61852742ee3 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -287,70 +287,78 @@ static const struct snd_kcontrol_new rt5631_snd_controls[] = {
static int check_sysclk1_source(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
- reg = snd_soc_read(source->codec, RT5631_GLOBAL_CLK_CTRL);
+ reg = snd_soc_read(codec, RT5631_GLOBAL_CLK_CTRL);
return reg & RT5631_SYSCLK_SOUR_SEL_PLL;
}
static int check_dmic_used(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(source->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
return rt5631->dmic_used_flag;
}
static int check_dacl_to_outmixl(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
- reg = snd_soc_read(source->codec, RT5631_OUTMIXER_L_CTRL);
+ reg = snd_soc_read(codec, RT5631_OUTMIXER_L_CTRL);
return !(reg & RT5631_M_DAC_L_TO_OUTMIXER_L);
}
static int check_dacr_to_outmixr(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
- reg = snd_soc_read(source->codec, RT5631_OUTMIXER_R_CTRL);
+ reg = snd_soc_read(codec, RT5631_OUTMIXER_R_CTRL);
return !(reg & RT5631_M_DAC_R_TO_OUTMIXER_R);
}
static int check_dacl_to_spkmixl(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
- reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL);
+ reg = snd_soc_read(codec, RT5631_SPK_MIXER_CTRL);
return !(reg & RT5631_M_DAC_L_TO_SPKMIXER_L);
}
static int check_dacr_to_spkmixr(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
- reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL);
+ reg = snd_soc_read(codec, RT5631_SPK_MIXER_CTRL);
return !(reg & RT5631_M_DAC_R_TO_SPKMIXER_R);
}
static int check_adcl_select(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
- reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER);
+ reg = snd_soc_read(codec, RT5631_ADC_REC_MIXER);
return !(reg & RT5631_M_MIC1_TO_RECMIXER_L);
}
static int check_adcr_select(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
- reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER);
+ reg = snd_soc_read(codec, RT5631_ADC_REC_MIXER);
return !(reg & RT5631_M_MIC2_TO_RECMIXER_R);
}
@@ -556,7 +564,7 @@ static void depop_seq_mute_stage(struct snd_soc_codec *codec, int enable)
static int hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -590,7 +598,7 @@ static int hp_event(struct snd_soc_dapm_widget *w,
static int set_dmic_params(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
switch (rt5631->rx_rate) {
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index c3f2decd643c..178e55d4d481 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -458,7 +458,7 @@ static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = {
static int set_dmic_clk(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
int idx = -EINVAL;
@@ -475,9 +475,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int val;
- val = snd_soc_read(source->codec, RT5640_GLB_CLK);
+ val = snd_soc_read(codec, RT5640_GLB_CLK);
val &= RT5640_SCLK_SRC_MASK;
if (val == RT5640_SCLK_SRC_PLL1)
return 1;
@@ -963,7 +964,7 @@ static void rt5640_pmu_depop(struct snd_soc_codec *codec)
static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -987,7 +988,7 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -1003,7 +1004,7 @@ static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -2124,6 +2125,7 @@ MODULE_DEVICE_TABLE(of, rt5640_of_match);
static struct acpi_device_id rt5640_acpi_match[] = {
{ "INT33CA", 0 },
{ "10EC5640", 0 },
+ { "10EC5642", 0 },
{ },
};
MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 27141e2df878..c9a4c5be083b 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -31,6 +31,7 @@
#include "rt5645.h"
#define RT5645_DEVICE_ID 0x6308
+#define RT5650_DEVICE_ID 0x6419
#define RT5645_PR_RANGE_BASE (0xff + 1)
#define RT5645_PR_SPACING 0x100
@@ -59,6 +60,10 @@ static const struct reg_default init_list[] = {
};
#define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list)
+static const struct reg_default rt5650_init_list[] = {
+ {0xf6, 0x0100},
+};
+
static const struct reg_default rt5645_reg[] = {
{ 0x00, 0x0000 },
{ 0x01, 0xc8c8 },
@@ -86,6 +91,7 @@ static const struct reg_default rt5645_reg[] = {
{ 0x2a, 0x5656 },
{ 0x2b, 0x5454 },
{ 0x2c, 0xaaa0 },
+ { 0x2d, 0x0000 },
{ 0x2f, 0x1002 },
{ 0x31, 0x5000 },
{ 0x32, 0x0000 },
@@ -193,6 +199,8 @@ static const struct reg_default rt5645_reg[] = {
{ 0xdb, 0x0003 },
{ 0xdc, 0x0049 },
{ 0xdd, 0x001b },
+ { 0xdf, 0x0008 },
+ { 0xe0, 0x4000 },
{ 0xe6, 0x8000 },
{ 0xe7, 0x0200 },
{ 0xec, 0xb300 },
@@ -242,6 +250,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg)
case RT5645_IRQ_CTRL3:
case RT5645_INT_IRQ_ST:
case RT5645_IL_CMD:
+ case RT5650_4BTN_IL_CMD1:
case RT5645_VENDOR_ID:
case RT5645_VENDOR_ID1:
case RT5645_VENDOR_ID2:
@@ -287,6 +296,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg)
case RT5645_STO_DAC_MIXER:
case RT5645_MONO_DAC_MIXER:
case RT5645_DIG_MIXER:
+ case RT5650_A_DAC_SOUR:
case RT5645_DIG_INF1_DATA:
case RT5645_PDM_OUT_CTRL:
case RT5645_REC_L1_MIXER:
@@ -378,6 +388,8 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg)
case RT5645_IL_CMD:
case RT5645_IL_CMD2:
case RT5645_IL_CMD3:
+ case RT5650_4BTN_IL_CMD1:
+ case RT5650_4BTN_IL_CMD2:
case RT5645_DRC1_HL_CTRL1:
case RT5645_DRC2_HL_CTRL1:
case RT5645_ADC_MONO_HP_CTRL1:
@@ -527,7 +539,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
static int set_dmic_clk(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
int idx = -EINVAL;
@@ -544,9 +556,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int val;
- val = snd_soc_read(source->codec, RT5645_GLB_CLK);
+ val = snd_soc_read(codec, RT5645_GLB_CLK);
val &= RT5645_SCLK_SRC_MASK;
if (val == RT5645_SCLK_SRC_PLL1)
return 1;
@@ -557,6 +570,7 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
static int is_using_asrc(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg, shift, val;
switch (source->shift) {
@@ -588,7 +602,7 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source,
return 0;
}
- val = (snd_soc_read(source->codec, reg) >> shift) & 0xf;
+ val = (snd_soc_read(codec, reg) >> shift) & 0xf;
switch (val) {
case 1:
case 2:
@@ -601,6 +615,87 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source,
}
+/**
+ * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters
+ * @codec: SoC audio codec device.
+ * @filter_mask: mask of filters.
+ * @clk_src: clock source
+ *
+ * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5645 can
+ * only support standard 32fs or 64fs i2s format, ASRC should be enabled to
+ * support special i2s clock format such as Intel's 100fs(100 * sampling rate).
+ * ASRC function will track i2s clock and generate a corresponding system clock
+ * for codec. This function provides an API to select the clock source for a
+ * set of filters specified by the mask. And the codec driver will turn on ASRC
+ * for these filters if ASRC is selected as their clock source.
+ */
+int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec,
+ unsigned int filter_mask, unsigned int clk_src)
+{
+ unsigned int asrc2_mask = 0;
+ unsigned int asrc2_value = 0;
+ unsigned int asrc3_mask = 0;
+ unsigned int asrc3_value = 0;
+
+ switch (clk_src) {
+ case RT5645_CLK_SEL_SYS:
+ case RT5645_CLK_SEL_I2S1_ASRC:
+ case RT5645_CLK_SEL_I2S2_ASRC:
+ case RT5645_CLK_SEL_SYS2:
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ if (filter_mask & RT5645_DA_STEREO_FILTER) {
+ asrc2_mask |= RT5645_DA_STO_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5645_DA_STO_CLK_SEL_MASK)
+ | (clk_src << RT5645_DA_STO_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5645_DA_MONO_L_FILTER) {
+ asrc2_mask |= RT5645_DA_MONOL_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5645_DA_MONOL_CLK_SEL_MASK)
+ | (clk_src << RT5645_DA_MONOL_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5645_DA_MONO_R_FILTER) {
+ asrc2_mask |= RT5645_DA_MONOR_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5645_DA_MONOR_CLK_SEL_MASK)
+ | (clk_src << RT5645_DA_MONOR_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5645_AD_STEREO_FILTER) {
+ asrc2_mask |= RT5645_AD_STO1_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5645_AD_STO1_CLK_SEL_MASK)
+ | (clk_src << RT5645_AD_STO1_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5645_AD_MONO_L_FILTER) {
+ asrc3_mask |= RT5645_AD_MONOL_CLK_SEL_MASK;
+ asrc3_value = (asrc3_value & ~RT5645_AD_MONOL_CLK_SEL_MASK)
+ | (clk_src << RT5645_AD_MONOL_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5645_AD_MONO_R_FILTER) {
+ asrc3_mask |= RT5645_AD_MONOR_CLK_SEL_MASK;
+ asrc3_value = (asrc3_value & ~RT5645_AD_MONOR_CLK_SEL_MASK)
+ | (clk_src << RT5645_AD_MONOR_CLK_SEL_SFT);
+ }
+
+ if (asrc2_mask)
+ snd_soc_update_bits(codec, RT5645_ASRC_2,
+ asrc2_mask, asrc2_value);
+
+ if (asrc3_mask)
+ snd_soc_update_bits(codec, RT5645_ASRC_3,
+ asrc3_mask, asrc3_value);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(rt5645_sel_asrc_clk_src);
+
/* Digital Mixer */
static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = {
SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER,
@@ -1007,6 +1102,44 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5645_if1_adc_in_mux =
SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum);
+/* MX-2d [3] [2] */
+static const char * const rt5650_a_dac1_src[] = {
+ "DAC1", "Stereo DAC Mixer"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5650_a_dac1_l_enum, RT5650_A_DAC_SOUR,
+ RT5650_A_DAC1_L_IN_SFT, rt5650_a_dac1_src);
+
+static const struct snd_kcontrol_new rt5650_a_dac1_l_mux =
+ SOC_DAPM_ENUM("A DAC1 L source", rt5650_a_dac1_l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5650_a_dac1_r_enum, RT5650_A_DAC_SOUR,
+ RT5650_A_DAC1_R_IN_SFT, rt5650_a_dac1_src);
+
+static const struct snd_kcontrol_new rt5650_a_dac1_r_mux =
+ SOC_DAPM_ENUM("A DAC1 R source", rt5650_a_dac1_r_enum);
+
+/* MX-2d [1] [0] */
+static const char * const rt5650_a_dac2_src[] = {
+ "Stereo DAC Mixer", "Mono DAC Mixer"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5650_a_dac2_l_enum, RT5650_A_DAC_SOUR,
+ RT5650_A_DAC2_L_IN_SFT, rt5650_a_dac2_src);
+
+static const struct snd_kcontrol_new rt5650_a_dac2_l_mux =
+ SOC_DAPM_ENUM("A DAC2 L source", rt5650_a_dac2_l_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5650_a_dac2_r_enum, RT5650_A_DAC_SOUR,
+ RT5650_A_DAC2_R_IN_SFT, rt5650_a_dac2_src);
+
+static const struct snd_kcontrol_new rt5650_a_dac2_r_mux =
+ SOC_DAPM_ENUM("A DAC2 R source", rt5650_a_dac2_r_enum);
+
/* MX-2F [13:12] */
static const char * const rt5645_if2_adc_in_src[] = {
"IF_ADC1", "IF_ADC2", "VAD_ADC"
@@ -1144,18 +1277,23 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
static int rt5645_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
hp_amp_power(codec, 1);
/* headphone unmute sequence */
- snd_soc_update_bits(codec, RT5645_DEPOP_M3, RT5645_CP_FQ1_MASK |
- RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK,
- (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) |
- (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) |
- (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT));
+ if (rt5645->codec_type == CODEC_TYPE_RT5650) {
+ snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737);
+ } else {
+ snd_soc_update_bits(codec, RT5645_DEPOP_M3,
+ RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK |
+ RT5645_CP_FQ3_MASK,
+ (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) |
+ (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) |
+ (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT));
+ }
regmap_write(rt5645->regmap,
RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00);
snd_soc_update_bits(codec, RT5645_DEPOP_M1,
@@ -1175,12 +1313,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w,
case SND_SOC_DAPM_PRE_PMD:
/* headphone mute sequence */
- snd_soc_update_bits(codec, RT5645_DEPOP_M3,
- RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK |
- RT5645_CP_FQ3_MASK,
- (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) |
- (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) |
- (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT));
+ if (rt5645->codec_type == CODEC_TYPE_RT5650) {
+ snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737);
+ } else {
+ snd_soc_update_bits(codec, RT5645_DEPOP_M3,
+ RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK |
+ RT5645_CP_FQ3_MASK,
+ (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) |
+ (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) |
+ (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT));
+ }
regmap_write(rt5645->regmap,
RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00);
snd_soc_update_bits(codec, RT5645_DEPOP_M1,
@@ -1205,7 +1347,7 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w,
static int rt5645_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -1232,7 +1374,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w,
static int rt5645_lout_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -1262,7 +1404,7 @@ static int rt5645_lout_event(struct snd_soc_dapm_widget *w,
static int rt5645_bst2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -1574,6 +1716,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("SPOR"),
};
+static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = {
+ SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM,
+ 0, 0, &rt5650_a_dac1_l_mux),
+ SND_SOC_DAPM_MUX("A DAC1 R Mux", SND_SOC_NOPM,
+ 0, 0, &rt5650_a_dac1_r_mux),
+ SND_SOC_DAPM_MUX("A DAC2 L Mux", SND_SOC_NOPM,
+ 0, 0, &rt5650_a_dac2_l_mux),
+ SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM,
+ 0, 0, &rt5650_a_dac2_r_mux),
+};
+
static const struct snd_soc_dapm_route rt5645_dapm_routes[] = {
{ "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc },
{ "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc },
@@ -1779,13 +1932,9 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = {
{ "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" },
{ "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" },
- { "DAC L1", NULL, "Stereo DAC MIXL" },
{ "DAC L1", NULL, "PLL1", is_sys_clk_from_pll },
- { "DAC R1", NULL, "Stereo DAC MIXR" },
{ "DAC R1", NULL, "PLL1", is_sys_clk_from_pll },
- { "DAC L2", NULL, "Mono DAC MIXL" },
{ "DAC L2", NULL, "PLL1", is_sys_clk_from_pll },
- { "DAC R2", NULL, "Mono DAC MIXR" },
{ "DAC R2", NULL, "PLL1", is_sys_clk_from_pll },
{ "SPK MIXL", "BST1 Switch", "BST1" },
@@ -1874,6 +2023,30 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = {
{ "SPOR", NULL, "SPK amp" },
};
+static const struct snd_soc_dapm_route rt5650_specific_dapm_routes[] = {
+ { "A DAC1 L Mux", "DAC1", "DAC1 MIXL"},
+ { "A DAC1 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"},
+ { "A DAC1 R Mux", "DAC1", "DAC1 MIXR"},
+ { "A DAC1 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"},
+
+ { "A DAC2 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"},
+ { "A DAC2 L Mux", "Mono DAC Mixer", "Mono DAC MIXL"},
+ { "A DAC2 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"},
+ { "A DAC2 R Mux", "Mono DAC Mixer", "Mono DAC MIXR"},
+
+ { "DAC L1", NULL, "A DAC1 L Mux" },
+ { "DAC R1", NULL, "A DAC1 R Mux" },
+ { "DAC L2", NULL, "A DAC2 L Mux" },
+ { "DAC R2", NULL, "A DAC2 R Mux" },
+};
+
+static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = {
+ { "DAC L1", NULL, "Stereo DAC MIXL" },
+ { "DAC R1", NULL, "Stereo DAC MIXR" },
+ { "DAC L2", NULL, "Mono DAC MIXL" },
+ { "DAC R2", NULL, "Mono DAC MIXR" },
+};
+
static int rt5645_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
@@ -2293,6 +2466,22 @@ static int rt5645_probe(struct snd_soc_codec *codec)
rt5645->codec = codec;
+ switch (rt5645->codec_type) {
+ case CODEC_TYPE_RT5645:
+ snd_soc_dapm_add_routes(&codec->dapm,
+ rt5645_specific_dapm_routes,
+ ARRAY_SIZE(rt5645_specific_dapm_routes));
+ break;
+ case CODEC_TYPE_RT5650:
+ snd_soc_dapm_new_controls(&codec->dapm,
+ rt5650_specific_dapm_widgets,
+ ARRAY_SIZE(rt5650_specific_dapm_widgets));
+ snd_soc_dapm_add_routes(&codec->dapm,
+ rt5650_specific_dapm_routes,
+ ARRAY_SIZE(rt5650_specific_dapm_routes));
+ break;
+ }
+
rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
@@ -2424,6 +2613,7 @@ static const struct regmap_config rt5645_regmap = {
static const struct i2c_device_id rt5645_i2c_id[] = {
{ "rt5645", 0 },
+ { "rt5650", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id);
@@ -2456,9 +2646,18 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
}
regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val);
- if (val != RT5645_DEVICE_ID) {
+
+ switch (val) {
+ case RT5645_DEVICE_ID:
+ rt5645->codec_type = CODEC_TYPE_RT5645;
+ break;
+ case RT5650_DEVICE_ID:
+ rt5645->codec_type = CODEC_TYPE_RT5650;
+ break;
+ default:
dev_err(&i2c->dev,
- "Device with ID register %x is not rt5645\n", val);
+ "Device with ID register %x is not rt5645 or rt5650\n",
+ val);
return -ENODEV;
}
@@ -2469,6 +2668,14 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
if (ret != 0)
dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+ if (rt5645->codec_type == CODEC_TYPE_RT5650) {
+ ret = regmap_register_patch(rt5645->regmap, rt5650_init_list,
+ ARRAY_SIZE(rt5650_init_list));
+ if (ret != 0)
+ dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n",
+ ret);
+ }
+
if (rt5645->pdata.in2_diff)
regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL,
RT5645_IN_DF2, RT5645_IN_DF2);
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index a815e36a2bdb..dbfd98c22f4d 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -47,6 +47,7 @@
#define RT5645_STO_DAC_MIXER 0x2a
#define RT5645_MONO_DAC_MIXER 0x2b
#define RT5645_DIG_MIXER 0x2c
+#define RT5650_A_DAC_SOUR 0x2d
#define RT5645_DIG_INF1_DATA 0x2f
/* Mixer - PDM */
#define RT5645_PDM_OUT_CTRL 0x31
@@ -150,6 +151,8 @@
#define RT5645_IL_CMD 0xdb
#define RT5645_IL_CMD2 0xdc
#define RT5645_IL_CMD3 0xdd
+#define RT5650_4BTN_IL_CMD1 0xdf
+#define RT5650_4BTN_IL_CMD2 0xe0
#define RT5645_DRC1_HL_CTRL1 0xe7
#define RT5645_DRC2_HL_CTRL1 0xe9
#define RT5645_MUTI_DRC_CTRL1 0xea
@@ -472,6 +475,12 @@
#define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4)
#define RT5645_DAC_L2_DAC_R_VOL_SFT 4
+/* Analog DAC1/2 Input Source Control (0x2d) */
+#define RT5650_A_DAC1_L_IN_SFT 3
+#define RT5650_A_DAC1_R_IN_SFT 2
+#define RT5650_A_DAC2_L_IN_SFT 1
+#define RT5650_A_DAC2_R_IN_SFT 0
+
/* Digital Interface Data Control (0x2f) */
#define RT5645_IF1_ADC2_IN_SEL (0x1 << 15)
#define RT5645_IF1_ADC2_IN_SFT 15
@@ -1111,50 +1120,27 @@
#define RT5645_DMIC_2_M_NOR (0x0 << 8)
#define RT5645_DMIC_2_M_ASYN (0x1 << 8)
+/* ASRC clock source selection (0x84, 0x85) */
+#define RT5645_CLK_SEL_SYS (0x0)
+#define RT5645_CLK_SEL_I2S1_ASRC (0x1)
+#define RT5645_CLK_SEL_I2S2_ASRC (0x2)
+#define RT5645_CLK_SEL_SYS2 (0x5)
+
/* ASRC Control 2 (0x84) */
-#define RT5645_MDA_L_M_MASK (0x1 << 15)
-#define RT5645_MDA_L_M_SFT 15
-#define RT5645_MDA_L_M_NOR (0x0 << 15)
-#define RT5645_MDA_L_M_ASYN (0x1 << 15)
-#define RT5645_MDA_R_M_MASK (0x1 << 14)
-#define RT5645_MDA_R_M_SFT 14
-#define RT5645_MDA_R_M_NOR (0x0 << 14)
-#define RT5645_MDA_R_M_ASYN (0x1 << 14)
-#define RT5645_MAD_L_M_MASK (0x1 << 13)
-#define RT5645_MAD_L_M_SFT 13
-#define RT5645_MAD_L_M_NOR (0x0 << 13)
-#define RT5645_MAD_L_M_ASYN (0x1 << 13)
-#define RT5645_MAD_R_M_MASK (0x1 << 12)
-#define RT5645_MAD_R_M_SFT 12
-#define RT5645_MAD_R_M_NOR (0x0 << 12)
-#define RT5645_MAD_R_M_ASYN (0x1 << 12)
-#define RT5645_ADC_M_MASK (0x1 << 11)
-#define RT5645_ADC_M_SFT 11
-#define RT5645_ADC_M_NOR (0x0 << 11)
-#define RT5645_ADC_M_ASYN (0x1 << 11)
-#define RT5645_STO_DAC_M_MASK (0x1 << 5)
-#define RT5645_STO_DAC_M_SFT 5
-#define RT5645_STO_DAC_M_NOR (0x0 << 5)
-#define RT5645_STO_DAC_M_ASYN (0x1 << 5)
-#define RT5645_I2S1_R_D_MASK (0x1 << 4)
-#define RT5645_I2S1_R_D_SFT 4
-#define RT5645_I2S1_R_D_DIS (0x0 << 4)
-#define RT5645_I2S1_R_D_EN (0x1 << 4)
-#define RT5645_I2S2_R_D_MASK (0x1 << 3)
-#define RT5645_I2S2_R_D_SFT 3
-#define RT5645_I2S2_R_D_DIS (0x0 << 3)
-#define RT5645_I2S2_R_D_EN (0x1 << 3)
-#define RT5645_PRE_SCLK_MASK (0x3)
-#define RT5645_PRE_SCLK_SFT 0
-#define RT5645_PRE_SCLK_512 (0x0)
-#define RT5645_PRE_SCLK_1024 (0x1)
-#define RT5645_PRE_SCLK_2048 (0x2)
+#define RT5645_DA_STO_CLK_SEL_MASK (0xf << 12)
+#define RT5645_DA_STO_CLK_SEL_SFT 12
+#define RT5645_DA_MONOL_CLK_SEL_MASK (0xf << 8)
+#define RT5645_DA_MONOL_CLK_SEL_SFT 8
+#define RT5645_DA_MONOR_CLK_SEL_MASK (0xf << 4)
+#define RT5645_DA_MONOR_CLK_SEL_SFT 4
+#define RT5645_AD_STO1_CLK_SEL_MASK (0xf << 0)
+#define RT5645_AD_STO1_CLK_SEL_SFT 0
/* ASRC Control 3 (0x85) */
-#define RT5645_I2S1_RATE_MASK (0xf << 12)
-#define RT5645_I2S1_RATE_SFT 12
-#define RT5645_I2S2_RATE_MASK (0xf << 8)
-#define RT5645_I2S2_RATE_SFT 8
+#define RT5645_AD_MONOL_CLK_SEL_MASK (0xf << 4)
+#define RT5645_AD_MONOL_CLK_SEL_SFT 4
+#define RT5645_AD_MONOR_CLK_SEL_MASK (0xf << 0)
+#define RT5645_AD_MONOR_CLK_SEL_SFT 0
/* ASRC Control 4 (0x89) */
#define RT5645_I2S1_PD_MASK (0x7 << 12)
@@ -2175,6 +2161,24 @@ enum {
RT5645_DMIC_DATA_GPIO11,
};
+enum {
+ CODEC_TYPE_RT5645,
+ CODEC_TYPE_RT5650,
+};
+
+/* filter mask */
+enum {
+ RT5645_DA_STEREO_FILTER = 0x1,
+ RT5645_DA_MONO_L_FILTER = (0x1 << 1),
+ RT5645_DA_MONO_R_FILTER = (0x1 << 2),
+ RT5645_AD_STEREO_FILTER = (0x1 << 3),
+ RT5645_AD_MONO_L_FILTER = (0x1 << 4),
+ RT5645_AD_MONO_R_FILTER = (0x1 << 5),
+};
+
+int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec,
+ unsigned int filter_mask, unsigned int clk_src);
+
struct rt5645_priv {
struct snd_soc_codec *codec;
struct rt5645_platform_data pdata;
@@ -2184,6 +2188,7 @@ struct rt5645_priv {
struct snd_soc_jack *mic_jack;
struct delayed_work jack_detect_work;
+ int codec_type;
int sysclk;
int sysclk_src;
int lrck[RT5645_AIFS];
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index bb0a3ab5416c..9f4c7be6d798 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -376,7 +376,7 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = {
static int set_dmic_clk(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec);
int idx = -EINVAL;
@@ -394,9 +394,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
static int is_sysclk_from_pll(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int val;
- val = snd_soc_read(source->codec, RT5651_GLB_CLK);
+ val = snd_soc_read(codec, RT5651_GLB_CLK);
val &= RT5651_SCLK_SRC_MASK;
if (val == RT5651_SCLK_SRC_PLL1)
return 1;
@@ -731,7 +732,7 @@ static const struct snd_kcontrol_new rt5651_pdm_r_mux =
static int rt5651_amp_power_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -769,7 +770,7 @@ static int rt5651_amp_power_event(struct snd_soc_dapm_widget *w,
static int rt5651_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -813,7 +814,8 @@ static int rt5651_hp_event(struct snd_soc_dapm_widget *w,
static int rt5651_hp_post_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -833,7 +835,7 @@ static int rt5651_hp_post_event(struct snd_soc_dapm_widget *w,
static int rt5651_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -856,7 +858,7 @@ static int rt5651_bst1_event(struct snd_soc_dapm_widget *w,
static int rt5651_bst2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -879,7 +881,7 @@ static int rt5651_bst2_event(struct snd_soc_dapm_widget *w,
static int rt5651_bst3_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 8a0833de1665..7b3d6b5992f1 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -14,10 +14,12 @@
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
+#include <linux/pm_runtime.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/acpi.h>
#include <linux/spi/spi.h>
+#include <linux/dmi.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -498,7 +500,7 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = {
static int set_dmic_clk(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
int idx = -EINVAL;
@@ -515,9 +517,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int val;
- val = snd_soc_read(source->codec, RT5670_GLB_CLK);
+ val = snd_soc_read(codec, RT5670_GLB_CLK);
val &= RT5670_SCLK_SRC_MASK;
if (val == RT5670_SCLK_SRC_PLL1)
return 1;
@@ -528,6 +531,7 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
static int is_using_asrc(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg, shift, val;
switch (source->shift) {
@@ -563,7 +567,7 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source,
return 0;
}
- val = (snd_soc_read(source->codec, reg) >> shift) & 0xf;
+ val = (snd_soc_read(codec, reg) >> shift) & 0xf;
switch (val) {
case 1:
case 2:
@@ -588,6 +592,89 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source,
return 0;
}
+
+/**
+ * rt5670_sel_asrc_clk_src - select ASRC clock source for a set of filters
+ * @codec: SoC audio codec device.
+ * @filter_mask: mask of filters.
+ * @clk_src: clock source
+ *
+ * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5670 can
+ * only support standard 32fs or 64fs i2s format, ASRC should be enabled to
+ * support special i2s clock format such as Intel's 100fs(100 * sampling rate).
+ * ASRC function will track i2s clock and generate a corresponding system clock
+ * for codec. This function provides an API to select the clock source for a
+ * set of filters specified by the mask. And the codec driver will turn on ASRC
+ * for these filters if ASRC is selected as their clock source.
+ */
+int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec,
+ unsigned int filter_mask, unsigned int clk_src)
+{
+ unsigned int asrc2_mask = 0, asrc2_value = 0;
+ unsigned int asrc3_mask = 0, asrc3_value = 0;
+
+ if (clk_src > RT5670_CLK_SEL_SYS3)
+ return -EINVAL;
+
+ if (filter_mask & RT5670_DA_STEREO_FILTER) {
+ asrc2_mask |= RT5670_DA_STO_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5670_DA_STO_CLK_SEL_MASK)
+ | (clk_src << RT5670_DA_STO_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5670_DA_MONO_L_FILTER) {
+ asrc2_mask |= RT5670_DA_MONOL_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5670_DA_MONOL_CLK_SEL_MASK)
+ | (clk_src << RT5670_DA_MONOL_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5670_DA_MONO_R_FILTER) {
+ asrc2_mask |= RT5670_DA_MONOR_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5670_DA_MONOR_CLK_SEL_MASK)
+ | (clk_src << RT5670_DA_MONOR_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5670_AD_STEREO_FILTER) {
+ asrc2_mask |= RT5670_AD_STO1_CLK_SEL_MASK;
+ asrc2_value = (asrc2_value & ~RT5670_AD_STO1_CLK_SEL_MASK)
+ | (clk_src << RT5670_AD_STO1_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5670_AD_MONO_L_FILTER) {
+ asrc3_mask |= RT5670_AD_MONOL_CLK_SEL_MASK;
+ asrc3_value = (asrc3_value & ~RT5670_AD_MONOL_CLK_SEL_MASK)
+ | (clk_src << RT5670_AD_MONOL_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5670_AD_MONO_R_FILTER) {
+ asrc3_mask |= RT5670_AD_MONOR_CLK_SEL_MASK;
+ asrc3_value = (asrc3_value & ~RT5670_AD_MONOR_CLK_SEL_MASK)
+ | (clk_src << RT5670_AD_MONOR_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5670_UP_RATE_FILTER) {
+ asrc3_mask |= RT5670_UP_CLK_SEL_MASK;
+ asrc3_value = (asrc3_value & ~RT5670_UP_CLK_SEL_MASK)
+ | (clk_src << RT5670_UP_CLK_SEL_SFT);
+ }
+
+ if (filter_mask & RT5670_DOWN_RATE_FILTER) {
+ asrc3_mask |= RT5670_DOWN_CLK_SEL_MASK;
+ asrc3_value = (asrc3_value & ~RT5670_DOWN_CLK_SEL_MASK)
+ | (clk_src << RT5670_DOWN_CLK_SEL_SFT);
+ }
+
+ if (asrc2_mask)
+ snd_soc_update_bits(codec, RT5670_ASRC_2,
+ asrc2_mask, asrc2_value);
+
+ if (asrc3_mask)
+ snd_soc_update_bits(codec, RT5670_ASRC_3,
+ asrc3_mask, asrc3_value);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(rt5670_sel_asrc_clk_src);
+
/* Digital Mixer */
static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = {
SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER,
@@ -1146,7 +1233,7 @@ static const struct snd_kcontrol_new rt5670_vad_adc_mux =
static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -1182,7 +1269,7 @@ static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w,
static int rt5670_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -1232,7 +1319,7 @@ static int rt5670_hp_event(struct snd_soc_dapm_widget *w,
static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -1255,7 +1342,7 @@ static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
static int rt5670_bst2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -2188,6 +2275,13 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai,
if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src)
return 0;
+ if (rt5670->pdata.jd_mode) {
+ if (clk_id == RT5670_SCLK_S_PLL1)
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+ else
+ snd_soc_dapm_disable_pin(&codec->dapm, "PLL1");
+ snd_soc_dapm_sync(&codec->dapm);
+ }
switch (clk_id) {
case RT5670_SCLK_S_MCLK:
reg_val |= RT5670_SCLK_SRC_MCLK;
@@ -2549,6 +2643,17 @@ static struct acpi_device_id rt5670_acpi_match[] = {
MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match);
#endif
+static const struct dmi_system_id dmi_platform_intel_braswell[] = {
+ {
+ .ident = "Intel Braswell",
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
+ DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"),
+ },
+ },
+ {}
+};
+
static int rt5670_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -2568,6 +2673,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
if (pdata)
rt5670->pdata = *pdata;
+ if (dmi_check_system(dmi_platform_intel_braswell)) {
+ rt5670->pdata.dmic_en = true;
+ rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P;
+ rt5670->pdata.jd_mode = 1;
+ }
+
rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap);
if (IS_ERR(rt5670->regmap)) {
ret = PTR_ERR(rt5670->regmap);
@@ -2609,6 +2720,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
}
if (rt5670->pdata.jd_mode) {
+ regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK,
+ RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK);
+ rt5670->sysclk = 0;
+ rt5670->sysclk_src = RT5670_SCLK_S_RCCLK;
regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1,
RT5670_PWR_MB, RT5670_PWR_MB);
regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2,
@@ -2716,18 +2831,26 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
}
+ pm_runtime_enable(&i2c->dev);
+ pm_request_idle(&i2c->dev);
+
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670,
rt5670_dai, ARRAY_SIZE(rt5670_dai));
if (ret < 0)
goto err;
+ pm_runtime_put(&i2c->dev);
+
return 0;
err:
+ pm_runtime_disable(&i2c->dev);
+
return ret;
}
static int rt5670_i2c_remove(struct i2c_client *i2c)
{
+ pm_runtime_disable(&i2c->dev);
snd_soc_unregister_codec(&i2c->dev);
return 0;
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index d11b9c207e26..21f8e18c13c4 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -1023,50 +1023,33 @@
#define RT5670_DMIC_2_M_NOR (0x0 << 8)
#define RT5670_DMIC_2_M_ASYN (0x1 << 8)
+/* ASRC clock source selection (0x84, 0x85) */
+#define RT5670_CLK_SEL_SYS (0x0)
+#define RT5670_CLK_SEL_I2S1_ASRC (0x1)
+#define RT5670_CLK_SEL_I2S2_ASRC (0x2)
+#define RT5670_CLK_SEL_I2S3_ASRC (0x3)
+#define RT5670_CLK_SEL_SYS2 (0x5)
+#define RT5670_CLK_SEL_SYS3 (0x6)
+
/* ASRC Control 2 (0x84) */
-#define RT5670_MDA_L_M_MASK (0x1 << 15)
-#define RT5670_MDA_L_M_SFT 15
-#define RT5670_MDA_L_M_NOR (0x0 << 15)
-#define RT5670_MDA_L_M_ASYN (0x1 << 15)
-#define RT5670_MDA_R_M_MASK (0x1 << 14)
-#define RT5670_MDA_R_M_SFT 14
-#define RT5670_MDA_R_M_NOR (0x0 << 14)
-#define RT5670_MDA_R_M_ASYN (0x1 << 14)
-#define RT5670_MAD_L_M_MASK (0x1 << 13)
-#define RT5670_MAD_L_M_SFT 13
-#define RT5670_MAD_L_M_NOR (0x0 << 13)
-#define RT5670_MAD_L_M_ASYN (0x1 << 13)
-#define RT5670_MAD_R_M_MASK (0x1 << 12)
-#define RT5670_MAD_R_M_SFT 12
-#define RT5670_MAD_R_M_NOR (0x0 << 12)
-#define RT5670_MAD_R_M_ASYN (0x1 << 12)
-#define RT5670_ADC_M_MASK (0x1 << 11)
-#define RT5670_ADC_M_SFT 11
-#define RT5670_ADC_M_NOR (0x0 << 11)
-#define RT5670_ADC_M_ASYN (0x1 << 11)
-#define RT5670_STO_DAC_M_MASK (0x1 << 5)
-#define RT5670_STO_DAC_M_SFT 5
-#define RT5670_STO_DAC_M_NOR (0x0 << 5)
-#define RT5670_STO_DAC_M_ASYN (0x1 << 5)
-#define RT5670_I2S1_R_D_MASK (0x1 << 4)
-#define RT5670_I2S1_R_D_SFT 4
-#define RT5670_I2S1_R_D_DIS (0x0 << 4)
-#define RT5670_I2S1_R_D_EN (0x1 << 4)
-#define RT5670_I2S2_R_D_MASK (0x1 << 3)
-#define RT5670_I2S2_R_D_SFT 3
-#define RT5670_I2S2_R_D_DIS (0x0 << 3)
-#define RT5670_I2S2_R_D_EN (0x1 << 3)
-#define RT5670_PRE_SCLK_MASK (0x3)
-#define RT5670_PRE_SCLK_SFT 0
-#define RT5670_PRE_SCLK_512 (0x0)
-#define RT5670_PRE_SCLK_1024 (0x1)
-#define RT5670_PRE_SCLK_2048 (0x2)
+#define RT5670_DA_STO_CLK_SEL_MASK (0xf << 12)
+#define RT5670_DA_STO_CLK_SEL_SFT 12
+#define RT5670_DA_MONOL_CLK_SEL_MASK (0xf << 8)
+#define RT5670_DA_MONOL_CLK_SEL_SFT 8
+#define RT5670_DA_MONOR_CLK_SEL_MASK (0xf << 4)
+#define RT5670_DA_MONOR_CLK_SEL_SFT 4
+#define RT5670_AD_STO1_CLK_SEL_MASK (0xf << 0)
+#define RT5670_AD_STO1_CLK_SEL_SFT 0
/* ASRC Control 3 (0x85) */
-#define RT5670_I2S1_RATE_MASK (0xf << 12)
-#define RT5670_I2S1_RATE_SFT 12
-#define RT5670_I2S2_RATE_MASK (0xf << 8)
-#define RT5670_I2S2_RATE_SFT 8
+#define RT5670_UP_CLK_SEL_MASK (0xf << 12)
+#define RT5670_UP_CLK_SEL_SFT 12
+#define RT5670_DOWN_CLK_SEL_MASK (0xf << 8)
+#define RT5670_DOWN_CLK_SEL_SFT 8
+#define RT5670_AD_MONOL_CLK_SEL_MASK (0xf << 4)
+#define RT5670_AD_MONOL_CLK_SEL_SFT 4
+#define RT5670_AD_MONOR_CLK_SEL_MASK (0xf << 0)
+#define RT5670_AD_MONOR_CLK_SEL_SFT 0
/* ASRC Control 4 (0x89) */
#define RT5670_I2S1_PD_MASK (0x7 << 12)
@@ -1983,6 +1966,21 @@ enum {
RT5670_DMIC_DATA_GPIO5,
};
+/* filter mask */
+enum {
+ RT5670_DA_STEREO_FILTER = 0x1,
+ RT5670_DA_MONO_L_FILTER = (0x1 << 1),
+ RT5670_DA_MONO_R_FILTER = (0x1 << 2),
+ RT5670_AD_STEREO_FILTER = (0x1 << 3),
+ RT5670_AD_MONO_L_FILTER = (0x1 << 4),
+ RT5670_AD_MONO_R_FILTER = (0x1 << 5),
+ RT5670_UP_RATE_FILTER = (0x1 << 6),
+ RT5670_DOWN_RATE_FILTER = (0x1 << 7),
+};
+
+int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec,
+ unsigned int filter_mask, unsigned int clk_src);
+
struct rt5670_priv {
struct snd_soc_codec *codec;
struct rt5670_platform_data pdata;
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 81fe1464d268..26fc538f03b1 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -784,8 +784,8 @@ static unsigned int bst_tlv[] = {
static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component);
ucontrol->value.integer.value[0] = rt5677->dsp_vad_en;
@@ -795,8 +795,9 @@ static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol,
static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component);
+ struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0];
@@ -895,7 +896,7 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = {
static int set_dmic_clk(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
int idx = rl6231_calc_dmic_clk(rt5677->sysclk);
@@ -910,7 +911,8 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(source->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
unsigned int val;
regmap_read(rt5677->regmap, RT5677_GLB_CLK1, &val);
@@ -921,6 +923,101 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
return 0;
}
+static int is_using_asrc(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ unsigned int reg, shift, val;
+
+ if (source->reg == RT5677_ASRC_1) {
+ switch (source->shift) {
+ case 12:
+ reg = RT5677_ASRC_4;
+ shift = 0;
+ break;
+ case 13:
+ reg = RT5677_ASRC_4;
+ shift = 4;
+ break;
+ case 14:
+ reg = RT5677_ASRC_4;
+ shift = 8;
+ break;
+ case 15:
+ reg = RT5677_ASRC_4;
+ shift = 12;
+ break;
+ default:
+ return 0;
+ }
+ } else {
+ switch (source->shift) {
+ case 0:
+ reg = RT5677_ASRC_6;
+ shift = 8;
+ break;
+ case 1:
+ reg = RT5677_ASRC_6;
+ shift = 12;
+ break;
+ case 2:
+ reg = RT5677_ASRC_5;
+ shift = 0;
+ break;
+ case 3:
+ reg = RT5677_ASRC_5;
+ shift = 4;
+ break;
+ case 4:
+ reg = RT5677_ASRC_5;
+ shift = 8;
+ break;
+ case 5:
+ reg = RT5677_ASRC_5;
+ shift = 12;
+ break;
+ case 12:
+ reg = RT5677_ASRC_3;
+ shift = 0;
+ break;
+ case 13:
+ reg = RT5677_ASRC_3;
+ shift = 4;
+ break;
+ case 14:
+ reg = RT5677_ASRC_3;
+ shift = 12;
+ break;
+ default:
+ return 0;
+ }
+ }
+
+ regmap_read(rt5677->regmap, reg, &val);
+ val = (val >> shift) & 0xf;
+
+ switch (val) {
+ case 1 ... 6:
+ return 1;
+ default:
+ return 0;
+ }
+
+}
+
+static int can_use_asrc(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+
+ if (rt5677->sysclk > rt5677->lrck[RT5677_AIF1] * 384)
+ return 1;
+
+ return 0;
+}
+
/* Digital Mixer */
static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = {
SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER,
@@ -2030,7 +2127,7 @@ static const struct snd_kcontrol_new rt5677_if2_dac7_tdm_sel_mux =
static int rt5677_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -2054,7 +2151,7 @@ static int rt5677_bst1_event(struct snd_soc_dapm_widget *w,
static int rt5677_bst2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -2078,14 +2175,18 @@ static int rt5677_bst2_event(struct snd_soc_dapm_widget *w,
static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
switch (event) {
- case SND_SOC_DAPM_POST_PMU:
+ case SND_SOC_DAPM_PRE_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2);
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0);
break;
+
default:
return 0;
}
@@ -2096,14 +2197,18 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w,
static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
switch (event) {
- case SND_SOC_DAPM_POST_PMU:
+ case SND_SOC_DAPM_PRE_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2);
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0);
break;
+
default:
return 0;
}
@@ -2114,7 +2219,7 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w,
static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -2141,7 +2246,7 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w,
static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
unsigned int value;
@@ -2164,7 +2269,7 @@ static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w,
static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
unsigned int value;
@@ -2187,7 +2292,7 @@ static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w,
static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -2211,9 +2316,50 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
- 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU),
+ 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT,
- 0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU),
+ 0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+
+ /* ASRC */
+ SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5677_ASRC_1, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5677_ASRC_1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("I2S3 ASRC", 1, RT5677_ASRC_1, 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("I2S4 ASRC", 1, RT5677_ASRC_1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5677_ASRC_2, 14, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO2 L ASRC", 1, RT5677_ASRC_2, 13, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO2 R ASRC", 1, RT5677_ASRC_2, 12, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO3 L ASRC", 1, RT5677_ASRC_1, 15, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO3 R ASRC", 1, RT5677_ASRC_1, 14, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO4 L ASRC", 1, RT5677_ASRC_1, 13, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO4 R ASRC", 1, RT5677_ASRC_1, 12, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5677_ASRC_2, 11, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC STO2 ASRC", 1, RT5677_ASRC_2, 10, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC STO3 ASRC", 1, RT5677_ASRC_2, 9, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC STO4 ASRC", 1, RT5677_ASRC_2, 8, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5677_ASRC_2, 7, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5677_ASRC_2, 6, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5677_ASRC_2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5677_ASRC_2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO3 ASRC", 1, RT5677_ASRC_2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO4 ASRC", 1, RT5677_ASRC_2, 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5677_ASRC_2, 1, 0, NULL,
+ 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5677_ASRC_2, 0, 0, NULL,
+ 0),
/* Input Side */
/* micbias */
@@ -2645,10 +2791,18 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
/* DAC Mixer */
SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2,
RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0),
- SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5677_PWR_DIG2,
+ SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2,
RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0),
- SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5677_PWR_DIG2,
+ SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2,
RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2,
+ RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2,
+ RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2,
+ RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2,
+ RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0,
rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)),
@@ -2721,6 +2875,31 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
+ { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", can_use_asrc },
+ { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", can_use_asrc },
+ { "Stereo3 DMIC Mux", NULL, "DMIC STO3 ASRC", can_use_asrc },
+ { "Stereo4 DMIC Mux", NULL, "DMIC STO4 ASRC", can_use_asrc },
+ { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", can_use_asrc },
+ { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", can_use_asrc },
+ { "I2S1", NULL, "I2S1 ASRC", can_use_asrc},
+ { "I2S2", NULL, "I2S2 ASRC", can_use_asrc},
+ { "I2S3", NULL, "I2S3 ASRC", can_use_asrc},
+ { "I2S4", NULL, "I2S4 ASRC", can_use_asrc},
+
+ { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc },
+ { "dac mono2 left filter", NULL, "DAC MONO2 L ASRC", is_using_asrc },
+ { "dac mono2 right filter", NULL, "DAC MONO2 R ASRC", is_using_asrc },
+ { "dac mono3 left filter", NULL, "DAC MONO3 L ASRC", is_using_asrc },
+ { "dac mono3 right filter", NULL, "DAC MONO3 R ASRC", is_using_asrc },
+ { "dac mono4 left filter", NULL, "DAC MONO4 L ASRC", is_using_asrc },
+ { "dac mono4 right filter", NULL, "DAC MONO4 R ASRC", is_using_asrc },
+ { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc },
+ { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc },
+ { "adc stereo3 filter", NULL, "ADC STO3 ASRC", is_using_asrc },
+ { "adc stereo4 filter", NULL, "ADC STO4 ASRC", is_using_asrc },
+ { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc },
+ { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc },
+
{ "DMIC1", NULL, "DMIC L1" },
{ "DMIC1", NULL, "DMIC R1" },
{ "DMIC2", NULL, "DMIC L2" },
@@ -2851,8 +3030,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" },
{ "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" },
- { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll },
-
{ "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" },
{ "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" },
{ "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll },
@@ -2873,8 +3050,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" },
{ "Stereo2 ADC MIXL", NULL, "adc stereo2 filter" },
- { "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll },
-
{ "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" },
{ "Stereo2 ADC MIXR", NULL, "adc stereo2 filter" },
{ "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll },
@@ -2889,8 +3064,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "Stereo3 ADC MIXL", NULL, "Sto3 ADC MIXL" },
{ "Stereo3 ADC MIXL", NULL, "adc stereo3 filter" },
- { "adc stereo3 filter", NULL, "PLL1", is_sys_clk_from_pll },
-
{ "Stereo3 ADC MIXR", NULL, "Sto3 ADC MIXR" },
{ "Stereo3 ADC MIXR", NULL, "adc stereo3 filter" },
{ "adc stereo3 filter", NULL, "PLL1", is_sys_clk_from_pll },
@@ -2905,8 +3078,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "Stereo4 ADC MIXL", NULL, "Sto4 ADC MIXL" },
{ "Stereo4 ADC MIXL", NULL, "adc stereo4 filter" },
- { "adc stereo4 filter", NULL, "PLL1", is_sys_clk_from_pll },
-
{ "Stereo4 ADC MIXR", NULL, "Sto4 ADC MIXR" },
{ "Stereo4 ADC MIXR", NULL, "adc stereo4 filter" },
{ "adc stereo4 filter", NULL, "PLL1", is_sys_clk_from_pll },
@@ -3455,10 +3626,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC1 MIXL", "Stereo ADC Switch", "ADDA1 Mux" },
{ "DAC1 MIXL", "DAC1 Switch", "DAC1 Mux" },
- { "DAC1 MIXL", NULL, "dac stereo1 filter" },
{ "DAC1 MIXR", "Stereo ADC Switch", "ADDA1 Mux" },
{ "DAC1 MIXR", "DAC1 Switch", "DAC1 Mux" },
- { "DAC1 MIXR", NULL, "dac stereo1 filter" },
{ "DAC1 FS", NULL, "DAC1 MIXL" },
{ "DAC1 FS", NULL, "DAC1 MIXR" },
@@ -3525,35 +3694,46 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "Stereo DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" },
{ "Stereo DAC MIXR", "DAC1 L Switch", "DAC1 MIXL" },
{ "Stereo DAC MIXR", NULL, "dac stereo1 filter" },
+ { "dac stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll },
{ "Mono DAC MIXL", "ST L Switch", "Sidetone Mux" },
{ "Mono DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" },
{ "Mono DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" },
{ "Mono DAC MIXL", "DAC2 R Switch", "DAC2 R Mux" },
- { "Mono DAC MIXL", NULL, "dac mono left filter" },
+ { "Mono DAC MIXL", NULL, "dac mono2 left filter" },
+ { "dac mono2 left filter", NULL, "PLL1", is_sys_clk_from_pll },
{ "Mono DAC MIXR", "ST R Switch", "Sidetone Mux" },
{ "Mono DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" },
{ "Mono DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" },
{ "Mono DAC MIXR", "DAC2 L Switch", "DAC2 L Mux" },
- { "Mono DAC MIXR", NULL, "dac mono right filter" },
+ { "Mono DAC MIXR", NULL, "dac mono2 right filter" },
+ { "dac mono2 right filter", NULL, "PLL1", is_sys_clk_from_pll },
{ "DD1 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" },
{ "DD1 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" },
{ "DD1 MIXL", "DAC3 L Switch", "DAC3 L Mux" },
{ "DD1 MIXL", "DAC3 R Switch", "DAC3 R Mux" },
+ { "DD1 MIXL", NULL, "dac mono3 left filter" },
+ { "dac mono3 left filter", NULL, "PLL1", is_sys_clk_from_pll },
{ "DD1 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" },
{ "DD1 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" },
{ "DD1 MIXR", "DAC3 L Switch", "DAC3 L Mux" },
{ "DD1 MIXR", "DAC3 R Switch", "DAC3 R Mux" },
+ { "DD1 MIXR", NULL, "dac mono3 right filter" },
+ { "dac mono3 right filter", NULL, "PLL1", is_sys_clk_from_pll },
{ "DD2 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" },
{ "DD2 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" },
{ "DD2 MIXL", "DAC4 L Switch", "DAC4 L Mux" },
{ "DD2 MIXL", "DAC4 R Switch", "DAC4 R Mux" },
+ { "DD2 MIXL", NULL, "dac mono4 left filter" },
+ { "dac mono4 left filter", NULL, "PLL1", is_sys_clk_from_pll },
{ "DD2 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" },
{ "DD2 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" },
{ "DD2 MIXR", "DAC4 L Switch", "DAC4 L Mux" },
{ "DD2 MIXR", "DAC4 R Switch", "DAC4 R Mux" },
+ { "DD2 MIXR", NULL, "dac mono4 right filter" },
+ { "dac mono4 right filter", NULL, "PLL1", is_sys_clk_from_pll },
{ "Stereo DAC MIX", NULL, "Stereo DAC MIXL" },
{ "Stereo DAC MIX", NULL, "Stereo DAC MIXR" },
@@ -3575,11 +3755,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC3 SRC Mux", "DD MIX2L", "DD2 MIXL" },
{ "DAC 1", NULL, "DAC12 SRC Mux" },
- { "DAC 1", NULL, "PLL1", is_sys_clk_from_pll },
{ "DAC 2", NULL, "DAC12 SRC Mux" },
- { "DAC 2", NULL, "PLL1", is_sys_clk_from_pll },
{ "DAC 3", NULL, "DAC3 SRC Mux" },
- { "DAC 3", NULL, "PLL1", is_sys_clk_from_pll },
{ "PDM1 L Mux", "STO1 DAC MIX", "Stereo DAC MIXL" },
{ "PDM1 L Mux", "MONO DAC MIX", "Mono DAC MIXL" },
@@ -3926,7 +4103,8 @@ static int rt5677_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;
- unsigned int val = 0;
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = 0, slot_width_25 = 0;
if (rx_mask || tx_mask)
val |= (1 << 12);
@@ -3950,6 +4128,8 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
case 20:
val |= (1 << 8);
break;
+ case 25:
+ slot_width_25 = 0x8080;
case 24:
val |= (2 << 8);
break;
@@ -3963,10 +4143,16 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
switch (dai->id) {
case RT5677_AIF1:
- snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val);
+ regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, 0x1f00,
+ val);
+ regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x8000,
+ slot_width_25);
break;
case RT5677_AIF2:
- snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val);
+ regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, 0x1f00,
+ val);
+ regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x80,
+ slot_width_25);
break;
default:
break;
@@ -4751,6 +4937,11 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
RT5677_GPIO5_DIR_OUT);
}
+ if (rt5677->pdata.micbias1_vdd_3v3)
+ regmap_update_bits(rt5677->regmap, RT5677_MICBIAS,
+ RT5677_MICBIAS1_CTRL_VDD_MASK,
+ RT5677_MICBIAS1_CTRL_VDD_3_3V);
+
rt5677_init_gpio(i2c);
rt5677_init_irq(i2c);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 29cf7ce610f4..e182e6569bbd 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -155,18 +155,19 @@ struct sgtl5000_priv {
static int mic_bias_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* change mic bias resistor */
- snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+ snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
SGTL5000_BIAS_R_MASK,
sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
break;
case SND_SOC_DAPM_PRE_PMD:
- snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+ snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
SGTL5000_BIAS_R_MASK, 0);
break;
}
@@ -181,11 +182,12 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
static int power_vag_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);
const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
break;
@@ -195,9 +197,9 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
* operational to prevent inadvertently starving the
* other one of them.
*/
- if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) &
+ if ((snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER) &
mask) != mask) {
- snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
SGTL5000_VAG_POWERUP, 0);
msleep(400);
}
@@ -483,21 +485,21 @@ static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* setting i2s data format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
- i2sctl |= SGTL5000_I2S_MODE_PCM;
+ i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT;
break;
case SND_SOC_DAIFMT_DSP_B:
- i2sctl |= SGTL5000_I2S_MODE_PCM;
+ i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT;
i2sctl |= SGTL5000_I2S_LRALIGN;
break;
case SND_SOC_DAIFMT_I2S:
- i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+ i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT;
break;
case SND_SOC_DAIFMT_RIGHT_J:
- i2sctl |= SGTL5000_I2S_MODE_RJ;
+ i2sctl |= SGTL5000_I2S_MODE_RJ << SGTL5000_I2S_MODE_SHIFT;
i2sctl |= SGTL5000_I2S_LRPOL;
break;
case SND_SOC_DAIFMT_LEFT_J:
- i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+ i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT;
i2sctl |= SGTL5000_I2S_LRALIGN;
break;
default:
@@ -1462,6 +1464,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
if (ret)
return ret;
+ /* Need 8 clocks before I2C accesses */
+ udelay(1);
+
/* read chip information */
ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg);
if (ret)
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 1f451a1946eb..47b257e41809 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -233,16 +233,18 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
static int sn95031_vhs_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);
+
if (SND_SOC_DAPM_EVENT_ON(event)) {
pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n");
/* power up the rail */
- snd_soc_write(w->codec, SN95031_VHSP, 0x3D);
- snd_soc_write(w->codec, SN95031_VHSN, 0x3F);
+ snd_soc_write(codec, SN95031_VHSP, 0x3D);
+ snd_soc_write(codec, SN95031_VHSN, 0x3F);
msleep(1);
} else if (SND_SOC_DAPM_EVENT_OFF(event)) {
pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n");
- snd_soc_write(w->codec, SN95031_VHSP, 0xC4);
- snd_soc_write(w->codec, SN95031_VHSN, 0x04);
+ snd_soc_write(codec, SN95031_VHSP, 0xC4);
+ snd_soc_write(codec, SN95031_VHSN, 0x04);
}
return 0;
}
@@ -250,14 +252,16 @@ static int sn95031_vhs_event(struct snd_soc_dapm_widget *w,
static int sn95031_vihf_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);
+
if (SND_SOC_DAPM_EVENT_ON(event)) {
pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n");
/* power up the rail */
- snd_soc_write(w->codec, SN95031_VIHF, 0x27);
+ snd_soc_write(codec, SN95031_VIHF, 0x27);
msleep(1);
} else if (SND_SOC_DAPM_EVENT_OFF(event)) {
pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n");
- snd_soc_write(w->codec, SN95031_VIHF, 0x24);
+ snd_soc_write(codec, SN95031_VIHF, 0x24);
}
return 0;
}
@@ -265,6 +269,7 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w,
static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
if (SND_SOC_DAPM_EVENT_ON(event)) {
@@ -273,15 +278,16 @@ static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w,
data_dir = BIT(7);
}
/* program DMIC LDO, clock and set clock */
- snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
- snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(0), clk_dir);
- snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(7), data_dir);
+ snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
+ snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(0), clk_dir);
+ snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(7), data_dir);
return 0;
}
static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
if (SND_SOC_DAPM_EVENT_ON(event)) {
@@ -290,22 +296,23 @@ static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w,
data_dir = BIT(1);
}
/* program DMIC LDO, clock and set clock */
- snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
- snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(2), clk_dir);
- snd_soc_update_bits(w->codec, SN95031_DMICBUF45, BIT(1), data_dir);
+ snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
+ snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(2), clk_dir);
+ snd_soc_update_bits(codec, SN95031_DMICBUF45, BIT(1), data_dir);
return 0;
}
static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int ldo = 0;
if (SND_SOC_DAPM_EVENT_ON(event))
ldo = BIT(7)|BIT(6);
/* program DMIC LDO */
- snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo);
+ snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo);
return 0;
}
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
index d8e32a6262ee..d3191c983d71 100644
--- a/sound/soc/codecs/sta32x.h
+++ b/sound/soc/codecs/sta32x.h
@@ -131,7 +131,7 @@
#define STA32X_CONFF_OCFG_MASK 0x03
#define STA32X_CONFF_OCFG_SHIFT 0
#define STA32X_CONFF_IDE 0x04
-#define STA32X_CONFF_IDE_SHIFT 3
+#define STA32X_CONFF_IDE_SHIFT 2
#define STA32X_CONFF_BCLE 0x08
#define STA32X_CONFF_ECLE 0x20
#define STA32X_CONFF_PWDN 0x40
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index dc3223d6eca1..c86dd9aae157 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -349,7 +349,8 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg,
static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
unsigned int reg = AIC31XX_DACFLAG1;
unsigned int mask;
@@ -377,7 +378,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
reg = AIC31XX_ADCFLAG;
break;
default:
- dev_err(w->codec->dev, "Unknown widget '%s' calling %s\n",
+ dev_err(codec->dev, "Unknown widget '%s' calling %s\n",
w->name, __func__);
return -EINVAL;
}
@@ -388,7 +389,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
case SND_SOC_DAPM_POST_PMD:
return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);
default:
- dev_dbg(w->codec->dev,
+ dev_dbg(codec->dev,
"Unhandled dapm widget event %d from %s\n",
event, w->name);
}
@@ -433,7 +434,7 @@ static const struct snd_kcontrol_new aic31xx_dapm_spr_switch =
static int mic_bias_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
switch (event) {
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b7ebce054b4e..07603d142923 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -197,7 +197,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
static int mic_bias_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream,
delay += aic3x->tdm_delay;
/* Configure data delay */
- snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay);
+ snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
return 0;
}
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 0fe2ced5b09f..4e3e607dec13 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -423,17 +423,18 @@ exit:
static int dac33_playback_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (likely(dac33->substream)) {
- dac33_calculate_times(dac33->substream, w->codec);
- dac33_prepare_chip(dac33->substream, w->codec);
+ dac33_calculate_times(dac33->substream, codec);
+ dac33_prepare_chip(dac33->substream, codec);
}
break;
case SND_SOC_DAPM_POST_PMD:
- dac33_disable_digital(w->codec);
+ dac33_disable_digital(codec);
break;
}
return 0;
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
index 1d1205702d23..9f2dced046de 100644
--- a/sound/soc/codecs/ts3a227e.c
+++ b/sound/soc/codecs/ts3a227e.c
@@ -254,6 +254,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
struct ts3a227e *ts3a227e;
struct device *dev = &i2c->dev;
int ret;
+ unsigned int acc_reg;
ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL);
if (ts3a227e == NULL)
@@ -283,6 +284,11 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
INTB_DISABLE | ADC_COMPLETE_INT_DISABLE,
ADC_COMPLETE_INT_DISABLE);
+ /* Read jack status because chip might not trigger interrupt at boot. */
+ regmap_read(ts3a227e->regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg);
+ ts3a227e_new_jack_state(ts3a227e, acc_reg);
+ ts3a227e_jack_report(ts3a227e);
+
return 0;
}
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 44af3188afb9..d04693e9cf9f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -567,12 +567,13 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \
struct snd_kcontrol *kcontrol, int event) \
{ \
- struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); \
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); \
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); \
\
switch (event) { \
case SND_SOC_DAPM_POST_PMU: \
twl4030->pin_name##_enabled = 1; \
- twl4030_write(w->codec, reg, twl4030_read(w->codec, reg)); \
+ twl4030_write(codec, reg, twl4030_read(codec, reg)); \
break; \
case SND_SOC_DAPM_POST_PMD: \
twl4030->pin_name##_enabled = 0; \
@@ -621,12 +622,14 @@ static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1);
+ handsfree_ramp(codec, TWL4030_REG_HFL_CTL, 1);
break;
case SND_SOC_DAPM_POST_PMD:
- handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0);
+ handsfree_ramp(codec, TWL4030_REG_HFL_CTL, 0);
break;
}
return 0;
@@ -635,12 +638,14 @@ static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1);
+ handsfree_ramp(codec, TWL4030_REG_HFR_CTL, 1);
break;
case SND_SOC_DAPM_POST_PMD:
- handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0);
+ handsfree_ramp(codec, TWL4030_REG_HFR_CTL, 0);
break;
}
return 0;
@@ -649,19 +654,23 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
static int vibramux_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+ twl4030_write(codec, TWL4030_REG_VIBRA_SET, 0xff);
return 0;
}
static int apll_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- twl4030_apll_enable(w->codec, 1);
+ twl4030_apll_enable(codec, 1);
break;
case SND_SOC_DAPM_POST_PMD:
- twl4030_apll_enable(w->codec, 0);
+ twl4030_apll_enable(codec, 0);
break;
}
return 0;
@@ -670,23 +679,24 @@ static int apll_event(struct snd_soc_dapm_widget *w,
static int aif_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);
u8 audio_if;
- audio_if = twl4030_read(w->codec, TWL4030_REG_AUDIO_IF);
+ audio_if = twl4030_read(codec, TWL4030_REG_AUDIO_IF);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
/* Enable AIF */
/* enable the PLL before we use it to clock the DAI */
- twl4030_apll_enable(w->codec, 1);
+ twl4030_apll_enable(codec, 1);
- twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF,
audio_if | TWL4030_AIF_EN);
break;
case SND_SOC_DAPM_POST_PMD:
/* disable the DAI before we stop it's source PLL */
- twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF,
audio_if & ~TWL4030_AIF_EN);
- twl4030_apll_enable(w->codec, 0);
+ twl4030_apll_enable(codec, 0);
break;
}
return 0;
@@ -758,20 +768,21 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
static int headsetlpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* Do the ramp-up only once */
if (!twl4030->hsr_enabled)
- headset_ramp(w->codec, 1);
+ headset_ramp(codec, 1);
twl4030->hsl_enabled = 1;
break;
case SND_SOC_DAPM_POST_PMD:
/* Do the ramp-down only if both headsetL/R is disabled */
if (!twl4030->hsr_enabled)
- headset_ramp(w->codec, 0);
+ headset_ramp(codec, 0);
twl4030->hsl_enabled = 0;
break;
@@ -782,20 +793,21 @@ static int headsetlpga_event(struct snd_soc_dapm_widget *w,
static int headsetrpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* Do the ramp-up only once */
if (!twl4030->hsl_enabled)
- headset_ramp(w->codec, 1);
+ headset_ramp(codec, 1);
twl4030->hsr_enabled = 1;
break;
case SND_SOC_DAPM_POST_PMD:
/* Do the ramp-down only if both headsetL/R is disabled */
if (!twl4030->hsl_enabled)
- headset_ramp(w->codec, 0);
+ headset_ramp(codec, 0);
twl4030->hsr_enabled = 0;
break;
@@ -806,7 +818,8 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w,
static int digimic_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
struct twl4030_codec_data *pdata = twl4030->pdata;
if (pdata && pdata->digimic_delay)
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 90f47f988b3f..aeec27b6f1af 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -234,7 +234,7 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
u8 hslctl, hsrctl;
/*
@@ -261,7 +261,7 @@ static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
static int twl6040_ep_drv_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret = 0;
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 34ef65c52a7d..8d9de49a5052 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -683,7 +683,7 @@ static const struct snd_kcontrol_new wm2000_controls[] = {
static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
int ret;
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index b80970dc2d2f..ea09db585aa1 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -775,7 +775,8 @@ static int wm5100_out_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
- struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec);
switch (w->reg) {
case WM5100_CHANNEL_ENABLES_1:
@@ -839,7 +840,7 @@ static int wm5100_post_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec);
int ret;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index f439ae052128..6d0fe0ac95a3 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -28,6 +28,7 @@
#include <linux/mfd/arizona/core.h>
#include <linux/mfd/arizona/registers.h>
+#include <asm/unaligned.h>
#include "arizona.h"
#include "wm5102.h"
@@ -580,7 +581,7 @@ static const struct reg_default wm5102_sysclk_revb_patch[] = {
static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
@@ -617,11 +618,10 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- uint16_t data;
mutex_lock(&arizona->dac_comp_lock);
- data = cpu_to_be16(arizona->dac_comp_coeff);
- memcpy(ucontrol->value.bytes.data, &data, sizeof(data));
+ put_unaligned_be16(arizona->dac_comp_coeff,
+ ucontrol->value.bytes.data);
mutex_unlock(&arizona->dac_comp_lock);
return 0;
@@ -1272,19 +1272,24 @@ SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 4456b38a3ef5..fbaeddb3e903 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -134,7 +134,7 @@ static const struct reg_default wm5110_sysclk_revd_patch[] = {
static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
@@ -905,22 +905,28 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT3R", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 574579b98872..c81a9eab3e3e 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -259,7 +259,7 @@ static void wm8350_pga_work(struct work_struct *work)
static int pga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
struct wm8350_output *out;
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 8ee446987aa9..b0d84e552fca 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -324,6 +324,7 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
static int outmixer_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 soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
u32 reg_shift = mc->shift;
@@ -332,7 +333,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
switch (reg_shift) {
case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) :
- reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1);
+ reg = snd_soc_read(codec, WM8400_OUTPUT_MIXER1);
if (reg & WM8400_LDLO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 1 LDLO Set\n");
@@ -340,7 +341,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
}
break;
case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8):
- reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2);
+ reg = snd_soc_read(codec, WM8400_OUTPUT_MIXER2);
if (reg & WM8400_RDRO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 2 RDRO Set\n");
@@ -348,7 +349,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
}
break;
case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8):
- reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER);
+ reg = snd_soc_read(codec, WM8400_SPEAKER_MIXER);
if (reg & WM8400_LDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer LDSPK Set\n");
@@ -356,7 +357,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
}
break;
case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8):
- reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER);
+ reg = snd_soc_read(codec, WM8400_SPEAKER_MIXER);
if (reg & WM8400_RDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer RDSPK Set\n");
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index b9211b42f6e9..098c143f44d6 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -217,7 +217,8 @@ SND_SOC_DAPM_INPUT("LLINEIN"),
static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
return wm8731->sysclk_type == WM8731_SYSCLK_XTAL;
}
@@ -717,6 +718,8 @@ static int wm8731_i2c_probe(struct i2c_client *i2c,
if (wm8731 == NULL)
return -ENOMEM;
+ mutex_init(&wm8731->lock);
+
wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap);
if (IS_ERR(wm8731->regmap)) {
ret = PTR_ERR(wm8731->regmap);
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index f6847fdd6ddd..eb0a1644ba11 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -323,7 +323,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("ROUT2"),
SND_SOC_DAPM_OUTPUT("MONO1"),
SND_SOC_DAPM_OUTPUT("OUT3"),
- SND_SOC_DAPM_OUTPUT("VREF"),
+ SND_SOC_DAPM_VMID("VREF"),
SND_SOC_DAPM_INPUT("LINPUT1"),
SND_SOC_DAPM_INPUT("LINPUT2"),
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 180e7a098726..53e977da2f86 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -308,9 +308,7 @@ static const struct snd_soc_dapm_route wm8770_intercon[] = {
static int vout12supply_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec;
-
- codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -327,9 +325,7 @@ static int vout12supply_event(struct snd_soc_dapm_widget *w,
static int vout34supply_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec;
-
- codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3a0d4b7d692f..2eb986c19b88 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -224,7 +224,7 @@ static void wm8900_reset(struct snd_soc_codec *codec)
static int wm8900_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
u16 hpctl1 = snd_soc_read(codec, WM8900_REG_HPCTL1);
switch (event) {
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index cc6b0ef98a34..dde462c082be 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -260,7 +260,7 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
static int wm8903_dcs_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
switch (event) {
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 4d2d2b1380d5..c5eaa0198ef0 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -673,7 +673,7 @@ static int cp_event(struct snd_soc_dapm_widget *w,
static int sysclk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -711,7 +711,7 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
static int out_pga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int reg, val;
int dcs_mask;
@@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
{ "Right Capture PGA", NULL, "Right Capture Mux" },
{ "Right Capture PGA", NULL, "Right Capture Inverting Mux" },
- { "AIFOUTL", "Left", "ADCL" },
- { "AIFOUTL", "Right", "ADCR" },
- { "AIFOUTR", "Left", "ADCL" },
- { "AIFOUTR", "Right", "ADCR" },
+ { "AIFOUTL Mux", "Left", "ADCL" },
+ { "AIFOUTL Mux", "Right", "ADCR" },
+ { "AIFOUTR Mux", "Left", "ADCL" },
+ { "AIFOUTR Mux", "Right", "ADCR" },
+
+ { "AIFOUTL", NULL, "AIFOUTL Mux" },
+ { "AIFOUTR", NULL, "AIFOUTR Mux" },
{ "ADCL", NULL, "CLK_DSP" },
{ "ADCL", NULL, "Left Capture PGA" },
@@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
};
static const struct snd_soc_dapm_route dac_intercon[] = {
- { "DACL", "Right", "AIFINR" },
- { "DACL", "Left", "AIFINL" },
+ { "DACL Mux", "Left", "AIFINL" },
+ { "DACL Mux", "Right", "AIFINR" },
+
+ { "DACR Mux", "Left", "AIFINL" },
+ { "DACR Mux", "Right", "AIFINR" },
+
+ { "DACL", NULL, "DACL Mux" },
{ "DACL", NULL, "CLK_DSP" },
- { "DACR", "Right", "AIFINR" },
- { "DACR", "Left", "AIFINL" },
+ { "DACR", NULL, "DACR Mux" },
{ "DACR", NULL, "CLK_DSP" },
{ "Charge pump", NULL, "SYSCLK" },
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 1173f7fef5a7..1ab2d462afad 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -333,7 +333,7 @@ static int wm8955_configure_clocking(struct snd_soc_codec *codec)
static int wm8955_sysclk(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
int ret = 0;
/* Always disable the clocks - if we're doing reconfiguration this
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 3cbc82b33292..c799cca5abeb 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -418,7 +418,7 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start)
int wm8958_aif_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
int i;
switch (event) {
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 031a1ae71d94..a96eb497a379 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -556,7 +556,7 @@ static struct {
{ 22050, 2 },
{ 24000, 2 },
{ 16000, 3 },
- { 11250, 4 },
+ { 11025, 4 },
{ 12000, 4 },
{ 8000, 5 },
};
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index eeffd05384b4..95e2c1bfc809 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -194,7 +194,7 @@ static bool wm8961_readable(struct device *dev, unsigned int reg)
static int wm8961_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
u16 hp_reg = snd_soc_read(codec, WM8961_ANALOGUE_HP_0);
u16 cp_reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_1);
u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2);
@@ -286,7 +286,7 @@ static int wm8961_hp_event(struct snd_soc_dapm_widget *w,
static int wm8961_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2);
u16 spk_reg = snd_soc_read(codec, WM8961_CLASS_D_CONTROL_1);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index d32d554f5b34..118b0034ba23 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1866,7 +1866,7 @@ static int cp_event(struct snd_soc_dapm_widget *w,
static int hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
int timeout;
int reg;
int expected = (WM8962_DCS_STARTUP_DONE_HP1L |
@@ -1960,7 +1960,7 @@ static int hp_event(struct snd_soc_dapm_widget *w,
static int out_pga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
int reg;
switch (w->shift) {
@@ -1993,7 +1993,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
static int dsp2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
switch (event) {
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index e418199155a8..baff2cc222a6 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -244,7 +244,7 @@ SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V,
static int wm8988_lrc_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
u16 adctl2 = snd_soc_read(codec, WM8988_ADCTL2);
/* Use the DAC to gate LRC if active, otherwise use ADC */
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 8a584229310a..c93bffcb3cfb 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -374,13 +374,14 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
static int outmixer_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);
u32 reg_shift = kcontrol->private_value & 0xfff;
int ret = 0;
u16 reg;
switch (reg_shift) {
case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) :
- reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER1);
+ reg = snd_soc_read(codec, WM8990_OUTPUT_MIXER1);
if (reg & WM8990_LDLO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 1 LDLO Set\n");
@@ -388,7 +389,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
}
break;
case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8):
- reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER2);
+ reg = snd_soc_read(codec, WM8990_OUTPUT_MIXER2);
if (reg & WM8990_RDRO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 2 RDRO Set\n");
@@ -396,7 +397,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
}
break;
case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8):
- reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER);
+ reg = snd_soc_read(codec, WM8990_SPEAKER_MIXER);
if (reg & WM8990_LDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer LDSPK Set\n");
@@ -404,7 +405,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
}
break;
case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8):
- reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER);
+ reg = snd_soc_read(codec, WM8990_SPEAKER_MIXER);
if (reg & WM8990_RDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer RDSPK Set\n");
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index b0ac2c3e31b9..49df0dc607e6 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -382,13 +382,14 @@ static const struct snd_kcontrol_new wm8991_snd_controls[] = {
static int outmixer_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);
u32 reg_shift = kcontrol->private_value & 0xfff;
int ret = 0;
u16 reg;
switch (reg_shift) {
case WM8991_SPEAKER_MIXER | (WM8991_LDSPK_BIT << 8):
- reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER1);
+ reg = snd_soc_read(codec, WM8991_OUTPUT_MIXER1);
if (reg & WM8991_LDLO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 1 LDLO Set\n");
@@ -397,7 +398,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
break;
case WM8991_SPEAKER_MIXER | (WM8991_RDSPK_BIT << 8):
- reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER2);
+ reg = snd_soc_read(codec, WM8991_OUTPUT_MIXER2);
if (reg & WM8991_RDRO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 2 RDRO Set\n");
@@ -406,7 +407,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
break;
case WM8991_OUTPUT_MIXER1 | (WM8991_LDLO_BIT << 8):
- reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER);
+ reg = snd_soc_read(codec, WM8991_SPEAKER_MIXER);
if (reg & WM8991_LDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer LDSPK Set\n");
@@ -415,7 +416,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
break;
case WM8991_OUTPUT_MIXER2 | (WM8991_RDRO_BIT << 8):
- reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER);
+ reg = snd_soc_read(codec, WM8991_SPEAKER_MIXER);
if (reg & WM8991_RDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer RDSPK Set\n");
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 53c6fe359496..2e70a270eb28 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -810,7 +810,7 @@ SOC_SINGLE_TLV("EQ5 Volume", WM8993_EQ6, 0, 24, 0, eq_tlv),
static int clk_sys_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1b97de2e4e67..4fbc7689339a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -249,7 +249,8 @@ static int configure_clock(struct snd_soc_codec *codec)
static int check_clk_sys(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- int reg = snd_soc_read(source->codec, WM8994_CLOCKING_1);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ int reg = snd_soc_read(codec, WM8994_CLOCKING_1);
const char *clk;
/* Check what we're currently using for CLK_SYS */
@@ -806,7 +807,7 @@ static void active_dereference(struct snd_soc_codec *codec)
static int clk_sys_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -981,7 +982,7 @@ static void vmid_dereference(struct snd_soc_codec *codec)
static int vmid_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -1037,7 +1038,7 @@ static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec)
static int aif1clk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
@@ -1135,7 +1136,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
static int aif2clk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
int i;
int dac;
int adc;
@@ -1220,7 +1221,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w,
static int aif1clk_late_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -1238,7 +1239,7 @@ static int aif1clk_late_ev(struct snd_soc_dapm_widget *w,
static int aif2clk_late_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -1256,7 +1257,7 @@ static int aif2clk_late_ev(struct snd_soc_dapm_widget *w,
static int late_enable_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -1289,7 +1290,7 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w,
static int late_disable_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -1331,7 +1332,7 @@ static int micbias_ev(struct snd_soc_dapm_widget *w,
static int dac_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int mask = 1 << w->shift;
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
@@ -1372,7 +1373,7 @@ SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 0, 1, 0),
static int post_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
dev_dbg(codec->dev, "SRC status: %x\n",
snd_soc_read(codec,
WM8994_RATE_STATUS));
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index c280f0a3a424..79e1aead5131 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -534,10 +534,11 @@ static void wm8995_update_class_w(struct snd_soc_codec *codec)
static int check_clk_sys(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
unsigned int reg;
const char *clk;
- reg = snd_soc_read(source->codec, WM8995_CLOCKING_1);
+ reg = snd_soc_read(codec, WM8995_CLOCKING_1);
/* Check what we're currently using for CLK_SYS */
if (reg & WM8995_SYSCLK_SRC)
clk = "AIF2CLK";
@@ -560,9 +561,7 @@ static int wm8995_put_class_w(struct snd_kcontrol *kcontrol,
static int hp_supply_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec;
-
- codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -611,10 +610,9 @@ static void dc_servo_cmd(struct snd_soc_codec *codec,
static int hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int reg;
- codec = w->codec;
reg = snd_soc_read(codec, WM8995_ANALOGUE_HP_1);
switch (event) {
@@ -761,9 +759,7 @@ static int configure_clock(struct snd_soc_codec *codec)
static int clk_sys_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec;
-
- codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index b1dcc11c1b23..dc92d5e4e942 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -599,7 +599,7 @@ static void wm8996_bg_disable(struct snd_soc_codec *codec)
static int bg_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
int ret = 0;
switch (event) {
@@ -634,7 +634,8 @@ static int cp_event(struct snd_soc_dapm_widget *w,
static int rmv_short_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
/* Record which outputs we enabled */
switch (event) {
@@ -758,7 +759,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm,
static int dcs_start(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index 7e8bfe27566b..a4d11770630c 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -84,7 +84,7 @@ static const struct reg_default wm8997_sysclk_reva_patch[] = {
static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
@@ -610,13 +610,16 @@ SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index b1d946facd57..13a3f335ea5b 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -734,7 +734,7 @@ static int configure_clock(struct snd_soc_codec *codec)
static int clk_sys_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
/* This should be done on init() for bypass paths */
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 6ffe8dc4f3fa..60d243c904f5 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -254,7 +254,7 @@ SOC_SINGLE_TLV("MIXOUTR IN2B Volume", WM9090_OUTPUT_MIXER4, 0, 3, 1,
static int hp_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int reg = snd_soc_read(codec, WM9090_ANALOGUE_HP_0);
switch (event) {
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 3eddb18fefd1..5cc457ef8894 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -344,23 +344,27 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec)
struct snd_ac97 *ac97;
int ret = 0;
- ac97 = snd_soc_new_ac97_codec(codec);
+ ac97 = snd_soc_alloc_ac97_codec(codec);
if (IS_ERR(ac97)) {
ret = PTR_ERR(ac97);
dev_err(codec->dev, "Failed to register AC97 codec\n");
return ret;
}
- snd_soc_codec_set_drvdata(codec, ac97);
-
ret = wm9705_reset(codec);
if (ret)
- goto reset_err;
+ goto err_put_device;
+
+ ret = device_add(&ac97->dev);
+ if (ret)
+ goto err_put_device;
+
+ snd_soc_codec_set_drvdata(codec, ac97);
return 0;
-reset_err:
- snd_soc_free_ac97_codec(ac97);
+err_put_device:
+ put_device(&ac97->dev);
return ret;
}
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index e04643d2bb24..9517571e820d 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -666,7 +666,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec)
struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
- wm9712->ac97 = snd_soc_new_ac97_codec(codec);
+ wm9712->ac97 = snd_soc_alloc_ac97_codec(codec);
if (IS_ERR(wm9712->ac97)) {
ret = PTR_ERR(wm9712->ac97);
dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret);
@@ -675,15 +675,19 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec)
ret = wm9712_reset(codec, 0);
if (ret < 0)
- goto reset_err;
+ goto err_put_device;
+
+ ret = device_add(&wm9712->ac97->dev);
+ if (ret)
+ goto err_put_device;
/* set alc mux to none */
ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
return 0;
-reset_err:
- snd_soc_free_ac97_codec(wm9712->ac97);
+err_put_device:
+ put_device(&wm9712->ac97->dev);
return ret;
}
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 71b9d5b0734d..68222917b396 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -217,7 +217,7 @@ SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
u16 status, rate;
if (WARN_ON(event != SND_SOC_DAPM_PRE_PMD))
@@ -1225,7 +1225,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec)
struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
int ret = 0, reg;
- wm9713->ac97 = snd_soc_new_ac97_codec(codec);
+ wm9713->ac97 = snd_soc_alloc_ac97_codec(codec);
if (IS_ERR(wm9713->ac97))
return PTR_ERR(wm9713->ac97);
@@ -1234,7 +1234,11 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec)
wm9713_reset(codec, 0);
ret = wm9713_reset(codec, 1);
if (ret < 0)
- goto reset_err;
+ goto err_put_device;
+
+ ret = device_add(&wm9713->ac97->dev);
+ if (ret)
+ goto err_put_device;
/* unmute the adc - move to kcontrol */
reg = ac97_read(codec, AC97_CD) & 0x7fff;
@@ -1242,8 +1246,8 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec)
return 0;
-reset_err:
- snd_soc_free_ac97_codec(wm9713->ac97);
+err_put_device:
+ put_device(&wm9713->ac97->dev);
return ret;
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 720d6e852986..ff67b334065b 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1373,7 +1373,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
struct wm_adsp_alg_region *alg_region;
@@ -1605,7 +1605,7 @@ err:
int wm_adsp2_early_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
@@ -1626,7 +1626,7 @@ EXPORT_SYMBOL_GPL(wm_adsp2_early_event);
int wm_adsp2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
struct wm_adsp_alg_region *alg_region;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 374537d5e179..8366e19657a7 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -500,7 +500,7 @@ SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1,
static int hp_supply_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -542,7 +542,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w,
static int hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int reg = snd_soc_read(codec, WM8993_ANALOGUE_HP_0);
switch (event) {
@@ -594,7 +594,7 @@ static int hp_event(struct snd_soc_dapm_widget *w,
static int earpiece_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *control, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
u16 reg = snd_soc_read(codec, WM8993_ANTIPOP1) & ~WM8993_HPOUT2_IN_ENA;
switch (event) {
@@ -619,7 +619,7 @@ static int earpiece_event(struct snd_soc_dapm_widget *w,
static int lineout_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *control, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
bool *flag;
@@ -649,7 +649,7 @@ static int lineout_event(struct snd_soc_dapm_widget *w,
static int micbias_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
switch (w->shift) {
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 8e948c63f3d9..2b81ca418d2a 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -58,13 +58,12 @@ choice
depends on MACH_DAVINCI_DM365_EVM
config SND_DM365_AIC3X_CODEC
- bool "Audio Codec - AIC3101"
+ tristate "Audio Codec - AIC3101"
help
Say Y if you want to add support for AIC3101 audio codec
config SND_DM365_VOICE_CODEC
tristate "Voice Codec - CQ93VC"
- depends on SND_DAVINCI_SOC
select MFD_DAVINCI_VOICECODEC
select SND_DAVINCI_SOC_VCIF
select SND_SOC_CQ0093VC
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 158cb3d1db70..b6bb5947a8a8 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -14,7 +14,6 @@
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
-#include <linux/platform_data/edma.h>
#include <linux/i2c.h>
#include <linux/of_platform.h>
#include <linux/clk.h>
@@ -25,11 +24,6 @@
#include <asm/dma.h>
#include <asm/mach-types.h>
-#include <linux/edma.h>
-
-#include "davinci-pcm.h"
-#include "davinci-i2s.h"
-
struct snd_soc_card_drvdata_davinci {
struct clk *mclk;
unsigned sysclk;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 30b94d4f9c5d..de3b155a5011 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -364,6 +364,20 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data)
return IRQ_RETVAL(handled_mask);
}
+static irqreturn_t davinci_mcasp_common_irq_handler(int irq, void *data)
+{
+ struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+ irqreturn_t ret = IRQ_NONE;
+
+ if (mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK])
+ ret = davinci_mcasp_tx_irq_handler(irq, data);
+
+ if (mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE])
+ ret |= davinci_mcasp_rx_irq_handler(irq, data);
+
+ return ret;
+}
+
static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
@@ -1313,16 +1327,19 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of(
pdata->tx_dma_channel = dma_spec.args[0];
- ret = of_property_match_string(np, "dma-names", "rx");
- if (ret < 0)
- goto nodata;
+ /* RX is not valid in DIT mode */
+ if (pdata->op_mode != DAVINCI_MCASP_DIT_MODE) {
+ ret = of_property_match_string(np, "dma-names", "rx");
+ if (ret < 0)
+ goto nodata;
- ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret,
- &dma_spec);
- if (ret < 0)
- goto nodata;
+ ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret,
+ &dma_spec);
+ if (ret < 0)
+ goto nodata;
- pdata->rx_dma_channel = dma_spec.args[0];
+ pdata->rx_dma_channel = dma_spec.args[0];
+ }
ret = of_property_read_u32(np, "tx-num-evt", &val);
if (ret >= 0)
@@ -1441,6 +1458,23 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
mcasp->dev = &pdev->dev;
+ irq = platform_get_irq_byname(pdev, "common");
+ if (irq >= 0) {
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n",
+ dev_name(&pdev->dev));
+ ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+ davinci_mcasp_common_irq_handler,
+ IRQF_ONESHOT | IRQF_SHARED,
+ irq_name, mcasp);
+ if (ret) {
+ dev_err(&pdev->dev, "common IRQ request failed\n");
+ goto err;
+ }
+
+ mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN;
+ mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN;
+ }
+
irq = platform_get_irq_byname(pdev, "rx");
if (irq >= 0) {
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
@@ -1501,19 +1535,34 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
else
dma_data->filter_data = &dma_params->channel;
- dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
- dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
- dma_params->asp_chan_q = pdata->asp_chan_q;
- dma_params->ram_chan_q = pdata->ram_chan_q;
- dma_params->sram_pool = pdata->sram_pool;
- dma_params->sram_size = pdata->sram_size_capture;
- if (dat)
- dma_params->dma_addr = dat->start;
- else
- dma_params->dma_addr = mem->start + pdata->rx_dma_offset;
-
- /* Unconditional dmaengine stuff */
- dma_data->addr = dma_params->dma_addr;
+ /* RX is not valid in DIT mode */
+ if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
+ dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+ dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+ dma_params->asp_chan_q = pdata->asp_chan_q;
+ dma_params->ram_chan_q = pdata->ram_chan_q;
+ dma_params->sram_pool = pdata->sram_pool;
+ dma_params->sram_size = pdata->sram_size_capture;
+ if (dat)
+ dma_params->dma_addr = dat->start;
+ else
+ dma_params->dma_addr = mem->start + pdata->rx_dma_offset;
+
+ /* Unconditional dmaengine stuff */
+ dma_data->addr = dma_params->dma_addr;
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (res)
+ dma_params->channel = res->start;
+ else
+ dma_params->channel = pdata->rx_dma_channel;
+
+ /* dmaengine filter data for DT and non-DT boot */
+ if (pdev->dev.of_node)
+ dma_data->filter_data = "rx";
+ else
+ dma_data->filter_data = &dma_params->channel;
+ }
if (mcasp->version < MCASP_VERSION_3) {
mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
@@ -1523,18 +1572,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
}
- res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (res)
- dma_params->channel = res->start;
- else
- dma_params->channel = pdata->rx_dma_channel;
-
- /* dmaengine filter data for DT and non-DT boot */
- if (pdev->dev.of_node)
- dma_data->filter_data = "rx";
- else
- dma_data->filter_data = &dma_params->channel;
-
dev_set_drvdata(&pdev->dev, mcasp);
mcasp_reparent_fck(pdev);
diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig
index e334900cf0b8..d50e08517dce 100644
--- a/sound/soc/dwc/Kconfig
+++ b/sound/soc/dwc/Kconfig
@@ -1,6 +1,7 @@
config SND_DESIGNWARE_I2S
tristate "Synopsys I2S Device Driver"
depends on CLKDEV_LOOKUP
+ select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M if you want to add support for I2S driver for
Synopsys desigwnware I2S device. The device supports upto
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index b93168d4f648..a3e97b46b64e 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -22,6 +22,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
/* common register for all channel */
#define IER 0x000
@@ -54,9 +55,39 @@
#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;
@@ -65,8 +96,8 @@ struct dw_i2s_dev {
struct device *dev;
/* data related to DMA transfers b/w i2s and DMAC */
- struct i2s_dma_data play_dma_data;
- struct i2s_dma_data capture_dma_data;
+ 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);
};
@@ -153,7 +184,7 @@ static int dw_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
- struct i2s_dma_data *dma_data = NULL;
+ union dw_i2s_snd_dma_data *dma_data = NULL;
if (!(dev->capability & DWC_I2S_RECORD) &&
(substream->stream == SNDRV_PCM_STREAM_CAPTURE))
@@ -209,16 +240,9 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
switch (config->chan_nr) {
case EIGHT_CHANNEL_SUPPORT:
- ch_reg = 3;
- break;
case SIX_CHANNEL_SUPPORT:
- ch_reg = 2;
- break;
case FOUR_CHANNEL_SUPPORT:
- ch_reg = 1;
- break;
case TWO_CHANNEL_SUPPORT:
- ch_reg = 0;
break;
default:
dev_err(dev->dev, "channel not supported\n");
@@ -227,31 +251,43 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
i2s_disable_channels(dev, substream->stream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution);
- i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
- 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), xfer_resolution);
- i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
- 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);
+ for (ch_reg = 0; ch_reg < (config->chan_nr / 2); ch_reg++) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ i2s_write_reg(dev->i2s_base, TCR(ch_reg),
+ xfer_resolution);
+ i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
+ 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),
+ xfer_resolution);
+ i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
+ 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);
+ }
}
i2s_write_reg(dev->i2s_base, CCR, ccr);
config->sample_rate = params_rate(params);
- if (!dev->i2s_clk_cfg)
- return -EINVAL;
+ if (dev->i2s_clk_cfg) {
+ ret = dev->i2s_clk_cfg(config);
+ if (ret < 0) {
+ dev_err(dev->dev, "runtime audio clk config fail\n");
+ return ret;
+ }
+ } else {
+ u32 bitclk = config->sample_rate * config->data_width * 2;
- ret = dev->i2s_clk_cfg(config);
- if (ret < 0) {
- dev_err(dev->dev, "runtime audio clk config fail\n");
- return ret;
+ ret = clk_set_rate(dev->clk, bitclk);
+ if (ret) {
+ dev_err(dev->dev, "Can't set I2S clock rate: %d\n",
+ ret);
+ return ret;
+ }
}
return 0;
@@ -263,6 +299,19 @@ static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
snd_soc_dai_set_dma_data(dai, substream, NULL);
}
+static int dw_i2s_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ i2s_write_reg(dev->i2s_base, TXFFR, 1);
+ else
+ i2s_write_reg(dev->i2s_base, RXFFR, 1);
+
+ return 0;
+}
+
static int dw_i2s_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
@@ -294,6 +343,7 @@ static struct snd_soc_dai_ops dw_i2s_dai_ops = {
.startup = dw_i2s_startup,
.shutdown = dw_i2s_shutdown,
.hw_params = dw_i2s_hw_params,
+ .prepare = dw_i2s_prepare,
.trigger = dw_i2s_trigger,
};
@@ -324,20 +374,162 @@ static int dw_i2s_resume(struct snd_soc_dai *dai)
#define dw_i2s_resume NULL
#endif
+/*
+ * The following tables allow a direct lookup of various parameters
+ * defined in the I2S block's configuration in terms of sound system
+ * parameters. Each table is sized to the number of entries possible
+ * according to the number of configuration bits describing an I2S
+ * block parameter.
+ */
+
+/* Maximum bit resolution of a channel - not uniformly spaced */
+static const u32 fifo_width[COMP_MAX_WORDSIZE] = {
+ 12, 16, 20, 24, 32, 0, 0, 0
+};
+
+/* Width of (DMA) bus */
+static const u32 bus_widths[COMP_MAX_DATA_WIDTH] = {
+ DMA_SLAVE_BUSWIDTH_1_BYTE,
+ DMA_SLAVE_BUSWIDTH_2_BYTES,
+ DMA_SLAVE_BUSWIDTH_4_BYTES,
+ DMA_SLAVE_BUSWIDTH_UNDEFINED
+};
+
+/* PCM format to support channel resolution */
+static const u32 formats[COMP_MAX_WORDSIZE] = {
+ SNDRV_PCM_FMTBIT_S16_LE,
+ SNDRV_PCM_FMTBIT_S16_LE,
+ SNDRV_PCM_FMTBIT_S24_LE,
+ SNDRV_PCM_FMTBIT_S24_LE,
+ SNDRV_PCM_FMTBIT_S32_LE,
+ 0,
+ 0,
+ 0
+};
+
+static int dw_configure_dai(struct dw_i2s_dev *dev,
+ struct snd_soc_dai_driver *dw_i2s_dai,
+ unsigned int rates)
+{
+ /*
+ * Read component parameter registers to extract
+ * the I2S block's configuration.
+ */
+ u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1);
+ u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2);
+ u32 idx;
+
+ if (COMP1_TX_ENABLED(comp1)) {
+ dev_dbg(dev->dev, " designware: play supported\n");
+ idx = COMP1_TX_WORDSIZE_0(comp1);
+ if (WARN_ON(idx >= ARRAY_SIZE(formats)))
+ return -EINVAL;
+ dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM;
+ dw_i2s_dai->playback.channels_max =
+ 1 << (COMP1_TX_CHANNELS(comp1) + 1);
+ dw_i2s_dai->playback.formats = formats[idx];
+ dw_i2s_dai->playback.rates = rates;
+ }
+
+ if (COMP1_RX_ENABLED(comp1)) {
+ dev_dbg(dev->dev, "designware: record supported\n");
+ idx = COMP2_RX_WORDSIZE_0(comp2);
+ if (WARN_ON(idx >= ARRAY_SIZE(formats)))
+ return -EINVAL;
+ dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM;
+ dw_i2s_dai->capture.channels_max =
+ 1 << (COMP1_RX_CHANNELS(comp1) + 1);
+ dw_i2s_dai->capture.formats = formats[idx];
+ dw_i2s_dai->capture.rates = rates;
+ }
+
+ return 0;
+}
+
+static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev,
+ struct snd_soc_dai_driver *dw_i2s_dai,
+ struct resource *res,
+ const struct i2s_platform_data *pdata)
+{
+ u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1);
+ u32 idx = COMP1_APB_DATA_WIDTH(comp1);
+ int ret;
+
+ if (WARN_ON(idx >= ARRAY_SIZE(bus_widths)))
+ return -EINVAL;
+
+ ret = dw_configure_dai(dev, dw_i2s_dai, pdata->snd_rates);
+ if (ret < 0)
+ return ret;
+
+ /* Set DMA slaves info */
+ dev->play_dma_data.pd.data = pdata->play_dma_data;
+ dev->capture_dma_data.pd.data = pdata->capture_dma_data;
+ dev->play_dma_data.pd.addr = res->start + I2S_TXDMA;
+ dev->capture_dma_data.pd.addr = res->start + I2S_RXDMA;
+ dev->play_dma_data.pd.max_burst = 16;
+ dev->capture_dma_data.pd.max_burst = 16;
+ dev->play_dma_data.pd.addr_width = bus_widths[idx];
+ dev->capture_dma_data.pd.addr_width = bus_widths[idx];
+ dev->play_dma_data.pd.filter = pdata->filter;
+ dev->capture_dma_data.pd.filter = pdata->filter;
+
+ return 0;
+}
+
+static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev,
+ struct snd_soc_dai_driver *dw_i2s_dai,
+ struct resource *res)
+{
+ u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1);
+ u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2);
+ u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1));
+ u32 idx = COMP1_APB_DATA_WIDTH(comp1);
+ u32 idx2;
+ int ret;
+
+ if (WARN_ON(idx >= ARRAY_SIZE(bus_widths)))
+ return -EINVAL;
+
+ ret = dw_configure_dai(dev, dw_i2s_dai, SNDRV_PCM_RATE_8000_192000);
+ if (ret < 0)
+ return ret;
+
+ if (COMP1_TX_ENABLED(comp1)) {
+ idx2 = COMP1_TX_WORDSIZE_0(comp1);
+
+ dev->capability |= DWC_I2S_PLAY;
+ dev->play_dma_data.dt.addr = res->start + I2S_TXDMA;
+ dev->play_dma_data.dt.addr_width = bus_widths[idx];
+ dev->play_dma_data.dt.chan_name = "TX";
+ dev->play_dma_data.dt.fifo_size = fifo_depth *
+ (fifo_width[idx2]) >> 8;
+ dev->play_dma_data.dt.maxburst = 16;
+ }
+ if (COMP1_RX_ENABLED(comp1)) {
+ idx2 = COMP2_RX_WORDSIZE_0(comp2);
+
+ dev->capability |= DWC_I2S_RECORD;
+ dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA;
+ dev->capture_dma_data.dt.addr_width = bus_widths[idx];
+ dev->capture_dma_data.dt.chan_name = "RX";
+ dev->capture_dma_data.dt.fifo_size = fifo_depth *
+ (fifo_width[idx2] >> 8);
+ dev->capture_dma_data.dt.maxburst = 16;
+ }
+
+ return 0;
+
+}
+
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;
- unsigned int cap;
struct snd_soc_dai_driver *dw_i2s_dai;
- if (!pdata) {
- dev_err(&pdev->dev, "Invalid platform data\n");
- return -EINVAL;
- }
-
dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
if (!dev) {
dev_warn(&pdev->dev, "kzalloc fail\n");
@@ -345,83 +537,67 @@ static int dw_i2s_probe(struct platform_device *pdev)
}
dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL);
- if (!dw_i2s_dai) {
- dev_err(&pdev->dev, "mem allocation failed for dai driver\n");
+ if (!dw_i2s_dai)
return -ENOMEM;
- }
dw_i2s_dai->ops = &dw_i2s_dai_ops;
dw_i2s_dai->suspend = dw_i2s_suspend;
dw_i2s_dai->resume = dw_i2s_resume;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- dev_err(&pdev->dev, "no i2s resource defined\n");
- return -ENODEV;
- }
-
dev->i2s_base = devm_ioremap_resource(&pdev->dev, res);
- if (IS_ERR(dev->i2s_base)) {
- dev_err(&pdev->dev, "ioremap fail for i2s_region\n");
+ if (IS_ERR(dev->i2s_base))
return PTR_ERR(dev->i2s_base);
- }
- cap = pdata->cap;
- dev->capability = cap;
- dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
+ dev->dev = &pdev->dev;
+ if (pdata) {
+ ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
+ if (ret < 0)
+ return ret;
+
+ dev->capability = pdata->cap;
+ dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
+ if (!dev->i2s_clk_cfg) {
+ dev_err(&pdev->dev, "no clock configure method\n");
+ return -ENODEV;
+ }
- /* Set DMA slaves info */
+ dev->clk = devm_clk_get(&pdev->dev, NULL);
+ } else {
+ ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res);
+ if (ret < 0)
+ return ret;
- dev->play_dma_data.data = pdata->play_dma_data;
- dev->capture_dma_data.data = pdata->capture_dma_data;
- dev->play_dma_data.addr = res->start + I2S_TXDMA;
- dev->capture_dma_data.addr = res->start + I2S_RXDMA;
- dev->play_dma_data.max_burst = 16;
- dev->capture_dma_data.max_burst = 16;
- dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
- dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
- dev->play_dma_data.filter = pdata->filter;
- dev->capture_dma_data.filter = pdata->filter;
-
- dev->clk = clk_get(&pdev->dev, NULL);
+ dev->clk = devm_clk_get(&pdev->dev, "i2sclk");
+ }
if (IS_ERR(dev->clk))
- return PTR_ERR(dev->clk);
+ return PTR_ERR(dev->clk);
- ret = clk_enable(dev->clk);
+ ret = clk_prepare_enable(dev->clk);
if (ret < 0)
- goto err_clk_put;
-
- if (cap & DWC_I2S_PLAY) {
- dev_dbg(&pdev->dev, " designware: play supported\n");
- dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM;
- dw_i2s_dai->playback.channels_max = pdata->channel;
- dw_i2s_dai->playback.formats = pdata->snd_fmts;
- dw_i2s_dai->playback.rates = pdata->snd_rates;
- }
-
- if (cap & DWC_I2S_RECORD) {
- dev_dbg(&pdev->dev, "designware: record supported\n");
- dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM;
- dw_i2s_dai->capture.channels_max = pdata->channel;
- dw_i2s_dai->capture.formats = pdata->snd_fmts;
- dw_i2s_dai->capture.rates = pdata->snd_rates;
- }
+ return ret;
- dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
- ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component,
+ ret = devm_snd_soc_register_component(&pdev->dev, &dw_i2s_component,
dw_i2s_dai, 1);
if (ret != 0) {
dev_err(&pdev->dev, "not able to register dai\n");
goto err_clk_disable;
}
+ if (!pdata) {
+ ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "Could not register PCM: %d\n", ret);
+ goto err_clk_disable;
+ }
+ }
+
return 0;
err_clk_disable:
- clk_disable(dev->clk);
-err_clk_put:
- clk_put(dev->clk);
+ clk_disable_unprepare(dev->clk);
return ret;
}
@@ -429,18 +605,26 @@ static int dw_i2s_remove(struct platform_device *pdev)
{
struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_component(&pdev->dev);
-
- clk_put(dev->clk);
+ clk_disable_unprepare(dev->clk);
return 0;
}
+#ifdef CONFIG_OF
+static const struct of_device_id dw_i2s_of_match[] = {
+ { .compatible = "snps,designware-i2s", },
+ {},
+};
+
+MODULE_DEVICE_TABLE(of, dw_i2s_of_match);
+#endif
+
static struct platform_driver dw_i2s_driver = {
.probe = dw_i2s_probe,
.remove = dw_i2s_remove,
.driver = {
.name = "designware-i2s",
+ .of_match_table = of_match_ptr(dw_i2s_of_match),
},
};
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 9ce70fc67b09..8c9e9006dd84 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -42,25 +42,6 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- /* fsl_ssi lacks the set_fmt ops. */
- if (ret && ret != -ENOTSUPP) {
- dev_err(cpu_dai->dev,
- "Failed to set the cpu dai format.\n");
- return ret;
- }
-
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret) {
- dev_err(cpu_dai->dev,
- "Failed to set the codec format.\n");
- return ret;
- }
-
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
CODEC_CLOCK, SND_SOC_CLOCK_OUT);
if (ret) {
@@ -91,6 +72,8 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai_name = "tlv320aic23-hifi",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &eukrea_tlv320_snd_ops,
};
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 026a80117540..c068494bae30 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -818,7 +818,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
return -ENOMEM;
asrc_priv->pdev = pdev;
- strncpy(asrc_priv->name, np->name, sizeof(asrc_priv->name) - 1);
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -837,12 +836,12 @@ static int fsl_asrc_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0,
- asrc_priv->name, asrc_priv);
+ dev_name(&pdev->dev), asrc_priv);
if (ret) {
dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret);
return ret;
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
index a3f211f53c23..4aed63c4b431 100644
--- a/sound/soc/fsl/fsl_asrc.h
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -433,7 +433,6 @@ struct fsl_asrc_pair {
* @channel_avail: non-occupied channel numbers
* @asrc_rate: default sample rate for ASoC Back-Ends
* @asrc_width: default sample width for ASoC Back-Ends
- * @name: driver name
*/
struct fsl_asrc {
struct snd_dmaengine_dai_dma_data dma_params_rx;
@@ -452,8 +451,6 @@ struct fsl_asrc {
int asrc_rate;
int asrc_width;
-
- char name[32];
};
extern struct snd_soc_platform_driver fsl_asrc_platform;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 1c08ab13637c..5c7597191e3f 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -774,7 +774,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 91a550f4a10d..5e793bbb6b02 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -302,7 +302,7 @@
#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT)
#define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK)
#define ESAI_xCCR_xDC_SHIFT 9
-#define ESAI_xCCR_xDC_WIDTH 4
+#define ESAI_xCCR_xDC_WIDTH 5
#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT)
#define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK)
#define ESAI_xCCR_xPSR_SHIFT 8
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 032d2d33619c..ec79c3d5e65e 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -612,7 +612,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index af0429421fc8..75870c0ea2c9 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -90,7 +90,6 @@ struct spdif_mixer_control {
* @sysclk: system clock for rx clock rate measurement
* @dma_params_tx: DMA parameters for transmit channel
* @dma_params_rx: DMA parameters for receive channel
- * @name: driver name
*/
struct fsl_spdif_priv {
struct spdif_mixer_control fsl_spdif_control;
@@ -109,12 +108,8 @@ struct fsl_spdif_priv {
struct clk *sysclk;
struct snd_dmaengine_dai_dma_data dma_params_tx;
struct snd_dmaengine_dai_dma_data dma_params_rx;
-
- /* The name space will be allocated dynamically */
- char name[0];
};
-
/* DPLL locked and lock loss interrupt handler */
static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
{
@@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev)
if (!np)
return -ENODEV;
- spdif_priv = devm_kzalloc(&pdev->dev,
- sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1,
- GFP_KERNEL);
+ spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL);
if (!spdif_priv)
return -ENOMEM;
- strcpy(spdif_priv->name, np->name);
-
spdif_priv->pdev = pdev;
/* Initialize this copy of the CPU DAI driver structure */
memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
- spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+ spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev);
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -1198,12 +1189,12 @@ static int fsl_spdif_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
return irq;
}
ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
- spdif_priv->name, spdif_priv);
+ dev_name(&pdev->dev), spdif_priv);
if (ret) {
dev_err(&pdev->dev, "could not claim irq %u\n", irq);
return ret;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index a65f17d57ffb..46549de60e50 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -160,7 +160,7 @@ struct fsl_ssi_soc_data {
*/
struct fsl_ssi_private {
struct regmap *regs;
- unsigned int irq;
+ int irq;
struct snd_soc_dai_driver cpu_dai_drv;
unsigned int dai_fmt;
@@ -1363,8 +1363,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
ssi_private->irq = platform_get_irq(pdev, 0);
if (!ssi_private->irq) {
- dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
- return -ENXIO;
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+ return ssi_private->irq;
}
/* Are the RX and the TX clocks locked? */
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index e94704f1b9ee..33da26a12457 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
data->card.dev = &pdev->dev;
data->card.dai_link = &data->dai;
data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
ret = snd_soc_of_parse_card_name(&data->card, "model");
if (ret)
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 4caacb05a623..cd146d4fa805 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev)
if (ret)
goto clk_fail;
data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
data->card.dai_link = &data->dai;
data->card.dapm_widgets = imx_wm8962_dapm_widgets;
data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index b1ced7b8d80c..198eeb3f3f7a 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -55,16 +55,6 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
- u32 dai_format;
-
- dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM;
-
- /* set codec DAI configuration */
- snd_soc_dai_set_fmt(codec_dai, dai_format);
-
- /* set cpu DAI configuration */
- snd_soc_dai_set_fmt(cpu_dai, dai_format);
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
25000000, SND_SOC_CLOCK_OUT);
@@ -164,6 +154,8 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
.platform_name = "imx-ssi.0",
.codec_name = "tlv320aic32x4.0-0018",
.cpu_dai_name = "imx-ssi.0",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &mx27vis_aic32x4_snd_ops,
};
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index 804749a6c61e..d072bd13db09 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -87,7 +87,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
snd_pcm_format_t format = params_format(params);
unsigned int rate = params_rate(params);
unsigned int channels = params_channels(params);
- u32 dai_format;
/* find the correct audio parameters */
for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
@@ -104,15 +103,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
/* codec FLL input is 14.75 MHz from MCLK */
snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
- dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM;
-
- /* set codec DAI configuration */
- snd_soc_dai_set_fmt(codec_dai, dai_format);
-
- /* set cpu DAI configuration */
- snd_soc_dai_set_fmt(cpu_dai, dai_format);
-
/* TODO: The SSI driver should figure this out for us */
switch (channels) {
case 2:
@@ -244,6 +234,8 @@ static struct snd_soc_dai_link wm1133_ev1_dai = {
.init = wm1133_ev1_init,
.ops = &wm1133_ev1_ops,
.symmetric_rates = 1,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
};
static struct snd_soc_card wm1133_ev1 = {
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index fb9240fdc9b7..7fe3009b1c43 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node,
}
/* Decrease the reference count of the device nodes */
-static int asoc_simple_card_unref(struct platform_device *pdev)
+static int asoc_simple_card_unref(struct snd_soc_card *card)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
struct snd_soc_dai_link *dai_link;
int num_links;
@@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
return ret;
err:
- asoc_simple_card_unref(pdev);
+ asoc_simple_card_unref(&priv->snd_card);
return ret;
}
@@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
snd_soc_jack_free_gpios(&simple_card_mic_jack, 1,
&simple_card_mic_jack_gpio);
- return asoc_simple_card_unref(pdev);
+ return asoc_simple_card_unref(card);
}
static const struct of_device_id asoc_simple_of_match[] = {
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index e989ecf046c9..ee03dbdda235 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -46,7 +46,7 @@ config SND_SOC_INTEL_BAYTRAIL
config SND_SOC_INTEL_HASWELL_MACH
tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
- depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \\
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \
I2C_DESIGNWARE_PLATFORM
select SND_SOC_INTEL_HASWELL
select SND_SOC_RT5640
@@ -76,7 +76,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH
config SND_SOC_INTEL_BROADWELL_MACH
tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
- depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \\
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \
I2C_DESIGNWARE_PLATFORM
select SND_SOC_INTEL_HASWELL
select SND_COMPRESS_OFFLOAD
@@ -89,7 +89,7 @@ config SND_SOC_INTEL_BROADWELL_MACH
config SND_SOC_INTEL_BYTCR_RT5640_MACH
tristate "ASoC Audio DSP Support for MID BYT Platform"
- depends on X86
+ depends on X86 && I2C
select SND_SOC_RT5640
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
@@ -101,7 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH
config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
- depends on X86_INTEL_LPSS
+ depends on X86_INTEL_LPSS && I2C
select SND_SOC_RT5670
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
@@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
platforms with RT5672 audio codec.
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 codec"
+ depends on X86_INTEL_LPSS
+ select SND_SOC_RT5645
+ select SND_SST_MFLD_PLATFORM
+ select SND_SST_IPC_ACPI
+ help
+ This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
+ platforms with RT5645 audio codec.
+ If unsure select "N".
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index e928ec385300..a8e53c45c6b6 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
snd-soc-sst-broadwell-objs := broadwell.o
snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o
snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
+snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.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
@@ -35,6 +36,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
# DSP driver
obj-$(CONFIG_SND_SST_IPC) += sst/
diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c
index 7cf95d5d5d80..9cf7d01479ad 100644
--- a/sound/soc/intel/broadwell.c
+++ b/sound/soc/intel/broadwell.c
@@ -140,8 +140,6 @@ static struct snd_soc_ops broadwell_rt286_ops = {
static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
struct sst_hsw *broadwell = pdata->dsp;
int ret;
@@ -155,14 +153,6 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
- /* always connected - check HP for jack detect */
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Speaker");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "DMIC1");
- snd_soc_dapm_enable_pin(dapm, "DMIC2");
-
return 0;
}
diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c
index 0cba7830c5e9..354eaad886e1 100644
--- a/sound/soc/intel/byt-rt5640.c
+++ b/sound/soc/intel/byt-rt5640.c
@@ -132,7 +132,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
struct snd_soc_codec *codec = runtime->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_card *card = runtime->card;
const struct snd_soc_dapm_route *custom_map;
int num_routes;
@@ -161,7 +160,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map);
}
- ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes);
+ ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes);
if (ret)
return ret;
@@ -171,13 +170,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
return ret;
}
- snd_soc_dapm_ignore_suspend(dapm, "HPOL");
- snd_soc_dapm_ignore_suspend(dapm, "HPOR");
-
- snd_soc_dapm_ignore_suspend(dapm, "SPOLP");
- snd_soc_dapm_ignore_suspend(dapm, "SPOLN");
- snd_soc_dapm_ignore_suspend(dapm, "SPORP");
- snd_soc_dapm_ignore_suspend(dapm, "SPORN");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
return ret;
}
diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c
index f5d0fc1ab10c..59308629043e 100644
--- a/sound/soc/intel/bytcr_dpcm_rt5640.c
+++ b/sound/soc/intel/bytcr_dpcm_rt5640.c
@@ -215,7 +215,6 @@ static int snd_byt_mc_probe(struct platform_device *pdev)
static struct platform_driver snd_byt_mc_driver = {
.driver = {
- .owner = THIS_MODULE,
.name = "bytt100_rt5640",
.pm = &snd_soc_pm_ops,
},
@@ -227,4 +226,4 @@ module_platform_driver(snd_byt_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
MODULE_AUTHOR("Subhransu S. Prusty <subhransu.s.prusty@intel.com>");
MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:bytrt5640-audio");
+MODULE_ALIAS("platform:bytt100_rt5640");
diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c
new file mode 100644
index 000000000000..bd29617a9ab9
--- /dev/null
+++ b/sound/soc/intel/cht_bsw_rt5645.c
@@ -0,0 +1,326 @@
+/*
+ * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
+ * Cherrytrail and Braswell, with RT5645 codec.
+ *
+ * Copyright (C) 2015 Intel Corp
+ * Author: Fang, Yang A <yang.a.fang@intel.com>
+ * N,Harshapriya <harshapriya.n@intel.com>
+ * This file is modified from cht_bsw_rt5672.c
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * 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 of the License.
+ *
+ * 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 <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/rt5645.h"
+#include "sst-atom-controls.h"
+
+#define CHT_PLAT_CLK_3_HZ 19200000
+#define CHT_CODEC_DAI "rt5645-aif1"
+
+struct cht_mc_private {
+ struct snd_soc_jack hp_jack;
+ struct snd_soc_jack mic_jack;
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+ int i;
+
+ for (i = 0; i < card->num_rtd; i++) {
+ struct snd_soc_pcm_runtime *rtd;
+
+ rtd = card->rtd + i;
+ if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+ strlen(CHT_CODEC_DAI)))
+ return rtd->codec_dai;
+ }
+ return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct snd_soc_dai *codec_dai;
+ int ret;
+
+ codec_dai = cht_get_codec_dai(card);
+ if (!codec_dai) {
+ dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+ return -EIO;
+ }
+
+ if (!SND_SOC_DAPM_EVENT_OFF(event))
+ return 0;
+
+ /* Set codec sysclk source to its internal clock because codec PLL will
+ * be off when idle and MCLK will also be off by ACPI when codec is
+ * runtime suspended. Codec needs clock for jack detection and button
+ * press.
+ */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+ platform_clock_control, SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+ {"IN1P", NULL, "Headset Mic"},
+ {"IN1N", NULL, "Headset Mic"},
+ {"DMIC L1", NULL, "Int Mic"},
+ {"DMIC R1", NULL, "Int Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Ext Spk", NULL, "SPOL"},
+ {"Ext Spk", NULL, "SPOR"},
+ {"AIF1 Playback", NULL, "ssp2 Tx"},
+ {"ssp2 Tx", NULL, "codec_out0"},
+ {"ssp2 Tx", NULL, "codec_out1"},
+ {"codec_in0", NULL, "ssp2 Rx" },
+ {"codec_in1", NULL, "ssp2 Rx" },
+ {"ssp2 Rx", NULL, "AIF1 Capture"},
+ {"Headphone", NULL, "Platform Clock"},
+ {"Headset Mic", NULL, "Platform Clock"},
+ {"Int Mic", NULL, "Platform Clock"},
+ {"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_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;
+
+ /* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
+ CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
+ params_rate(params) * 512, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_codec *codec = runtime->codec;
+ struct snd_soc_dai *codec_dai = runtime->codec_dai;
+ struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
+
+ /* Select clk_i2s1_asrc as ASRC clock source */
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_DA_STEREO_FILTER |
+ RT5645_DA_MONO_L_FILTER |
+ RT5645_DA_MONO_R_FILTER |
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S1_ASRC);
+
+ /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+ if (ret < 0) {
+ dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_jack_new(codec, "Headphone Jack",
+ SND_JACK_HEADPHONE,
+ &ctx->hp_jack);
+ if (ret) {
+ dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_jack_new(codec, "Mic Jack",
+ SND_JACK_MICROPHONE,
+ &ctx->mic_jack);
+ if (ret) {
+ dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
+ return ret;
+ }
+
+ rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack);
+
+ return ret;
+}
+
+static int cht_codec_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);
+
+ /* The DSP will covert the FE rate to 48k, stereo, 24bits */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP2 to 24-bit */
+ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S24_LE);
+ return 0;
+}
+
+static unsigned int rates_48000[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+ .count = ARRAY_SIZE(rates_48000),
+ .list = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+ .startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+ .hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+ [MERR_DPCM_AUDIO] = {
+ .name = "Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "media-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ .ignore_suspend = 1,
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_aif1_ops,
+ },
+ [MERR_DPCM_COMPR] = {
+ .name = "Compressed Port",
+ .stream_name = "Compress",
+ .cpu_dai_name = "compress-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ },
+ /* CODEC<->CODEC link */
+ /* back ends */
+ {
+ .name = "SSP2-Codec",
+ .be_id = 1,
+ .cpu_dai_name = "ssp2-port",
+ .platform_name = "sst-mfld-platform",
+ .no_pcm = 1,
+ .codec_dai_name = "rt5645-aif1",
+ .codec_name = "i2c-10EC5645:00",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .init = cht_codec_init,
+ .be_hw_params_fixup = cht_codec_fixup,
+ .ignore_suspend = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_be_ssp2_ops,
+ },
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+ .name = "chtrt5645",
+ .dai_link = cht_dailink,
+ .num_links = ARRAY_SIZE(cht_dailink),
+ .dapm_widgets = cht_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+ .dapm_routes = cht_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+ .controls = cht_mc_controls,
+ .num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+ struct cht_mc_private *drv;
+
+ drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
+ if (!drv)
+ return -ENOMEM;
+
+ snd_soc_card_cht.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+ if (ret_val) {
+ dev_err(&pdev->dev,
+ "snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+ platform_set_drvdata(pdev, &snd_soc_card_cht);
+ return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+ .driver = {
+ .name = "cht-bsw-rt5645",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
+MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5645");
diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c
index 9b8b561171b7..ff016621583a 100644
--- a/sound/soc/intel/cht_bsw_rt5672.c
+++ b/sound/soc/intel/cht_bsw_rt5672.c
@@ -140,6 +140,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
struct snd_soc_dai *codec_dai = runtime->codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
@@ -148,6 +149,19 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
return ret;
}
+ /* Select codec ASRC clock source to track I2S1 clock, because codec
+ * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot
+ * be supported by RT5672. Otherwise, ASRC will be disabled and cause
+ * noise.
+ */
+ rt5670_sel_asrc_clk_src(codec,
+ RT5670_DA_STEREO_FILTER
+ | RT5670_DA_MONO_L_FILTER
+ | RT5670_DA_MONO_R_FILTER
+ | RT5670_AD_STEREO_FILTER
+ | RT5670_AD_MONO_L_FILTER
+ | RT5670_AD_MONO_R_FILTER,
+ RT5670_CLK_SEL_I2S1_ASRC);
return 0;
}
@@ -270,7 +284,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
static struct platform_driver snd_cht_mc_driver = {
.driver = {
- .owner = THIS_MODULE,
.name = "cht-bsw-rt5672",
.pm = &snd_soc_pm_ops,
},
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 3bb6288d8b4d..224c49c9f135 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -320,11 +320,6 @@ static struct snd_pcm_ops sst_byt_pcm_ops = {
.mmap = sst_byt_pcm_mmap,
};
-static void sst_byt_pcm_free(struct snd_pcm *pcm)
-{
- snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm *pcm = rtd->pcm;
@@ -403,7 +398,6 @@ static struct snd_soc_platform_driver byt_soc_platform = {
.remove = sst_byt_pcm_remove,
.ops = &sst_byt_pcm_ops,
.pcm_new = sst_byt_pcm_new,
- .pcm_free = sst_byt_pcm_free,
};
static const struct snd_soc_component_driver byt_dai_component = {
diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c
index 86e410845670..64e94212d2d2 100644
--- a/sound/soc/intel/sst-dsp.c
+++ b/sound/soc/intel/sst-dsp.c
@@ -410,8 +410,7 @@ void sst_dsp_free(struct sst_dsp *sst)
if (sst->ops->free)
sst->ops->free(sst);
- if (sst->dma)
- sst_dma_free(sst->dma);
+ sst_dma_free(sst->dma);
}
EXPORT_SYMBOL_GPL(sst_dsp_free);
diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c
index 4a5bde9c686b..5f71ef607a57 100644
--- a/sound/soc/intel/sst-firmware.c
+++ b/sound/soc/intel/sst-firmware.c
@@ -497,6 +497,7 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw,
sst_module->sst_fw = sst_fw;
sst_module->scratch_size = template->scratch_size;
sst_module->persistent_size = template->persistent_size;
+ sst_module->entry = template->entry;
INIT_LIST_HEAD(&sst_module->block_list);
INIT_LIST_HEAD(&sst_module->runtime_list);
@@ -706,6 +707,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
struct list_head *block_list)
{
struct sst_mem_block *block, *tmp;
+ struct sst_block_allocator ba_tmp = *ba;
u32 end = ba->offset + ba->size, block_end;
int err;
@@ -730,9 +732,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
if (ba->offset >= block->offset && ba->offset < block_end) {
/* align ba to block boundary */
- ba->size -= block_end - ba->offset;
- ba->offset = block_end;
- err = block_alloc_contiguous(dsp, ba, block_list);
+ ba_tmp.size -= block_end - ba->offset;
+ ba_tmp.offset = block_end;
+ err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
if (err < 0)
return -ENOMEM;
@@ -763,10 +765,14 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
/* does block span more than 1 section */
if (ba->offset >= block->offset && ba->offset < block_end) {
+ /* add block */
+ list_move(&block->list, &dsp->used_block_list);
+ list_add(&block->module_list, block_list);
/* align ba to block boundary */
- ba->offset = block->offset;
+ ba_tmp.size -= block_end - ba->offset;
+ ba_tmp.offset = block_end;
- err = block_alloc_contiguous(dsp, ba, block_list);
+ err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
if (err < 0)
return -ENOMEM;
@@ -785,6 +791,7 @@ int sst_module_alloc_blocks(struct sst_module *module)
struct sst_block_allocator ba;
int ret;
+ memset(&ba, 0, sizeof(ba));
ba.size = module->size;
ba.type = module->type;
ba.offset = module->offset;
@@ -858,6 +865,7 @@ int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime,
if (module->persistent_size == 0)
return 0;
+ memset(&ba, 0, sizeof(ba));
ba.size = module->persistent_size;
ba.type = SST_MEM_DRAM;
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index 57039b00efc2..c42ffae5fe9f 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -306,7 +306,7 @@ static void hsw_reset(struct sst_dsp *sst)
static int hsw_set_dsp_D0(struct sst_dsp *sst)
{
int tries = 10;
- u32 reg;
+ u32 reg, fw_dump_bit;
/* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */
reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2);
@@ -368,7 +368,9 @@ finish:
can't be accessed, please enable each block before accessing. */
reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK;
- writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0);
+ /* for D0, always enable the block(DSRAM[0]) used for FW dump */
+ fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT;
+ writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0);
/* disable DMA finish function for SSP0 & SSP1 */
@@ -491,6 +493,7 @@ static const struct sst_sram_shift sram_shift[] = {
{SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */
{SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */
};
+
static u32 hsw_block_get_bit(struct sst_mem_block *block)
{
u32 bit = 0, shift = 0, index;
@@ -587,7 +590,9 @@ static int hsw_block_disable(struct sst_mem_block *block)
val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
bit = hsw_block_get_bit(block);
- writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0);
+ /* don't disable DSRAM[0], keep it always enable for FW dump*/
+ if (bit != (1 << SST_VDRTCL0_DSRAMPGE_SHIFT))
+ writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0);
/* wait 18 DSP clock ticks */
udelay(10);
@@ -612,7 +617,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
const struct sst_adsp_memregion *region;
struct device *dev;
int ret = -ENODEV, i, j, region_count;
- u32 offset, size;
+ u32 offset, size, fw_dump_bit;
dev = sst->dma_dev;
@@ -669,9 +674,11 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
}
}
+ /* always enable the block(DSRAM[0]) used for FW dump */
+ fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT;
/* set default power gating control, enable power gating control for all blocks. that is,
can't be accessed, please enable each block before accessing. */
- writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0);
+ writel(0xffffffff & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0);
return 0;
}
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 3f8c48231364..0ab1309ef274 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -94,6 +94,8 @@
/* Mailbox */
#define IPC_MAX_MAILBOX_BYTES 256
+#define INVALID_STREAM_HW_ID 0xffffffff
+
/* Global Message - Types and Replies */
enum ipc_glb_type {
IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */
@@ -275,7 +277,6 @@ struct sst_hsw {
/* FW config */
struct sst_hsw_ipc_fw_ready fw_ready;
struct sst_hsw_ipc_fw_version version;
- struct sst_module *scratch;
bool fw_done;
struct sst_fw *sst_fw;
@@ -337,12 +338,6 @@ static inline u32 msg_get_stage_type(u32 msg)
return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT;
}
-static inline u32 msg_set_stage_type(u32 msg, u32 type)
-{
- return (msg & ~IPC_STG_TYPE_MASK) +
- (type << IPC_STG_TYPE_SHIFT);
-}
-
static inline u32 msg_get_stream_id(u32 msg)
{
return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT;
@@ -651,11 +646,11 @@ static void hsw_notification_work(struct work_struct *work)
}
/* tell DSP that notification has been handled */
- sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD,
+ sst_dsp_shim_update_bits(hsw->dsp, SST_IPCD,
SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE);
/* unmask busy interrupt */
- sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0);
+ sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0);
}
static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header)
@@ -969,45 +964,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw,
}
/* Mixer Controls */
-int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
- u32 stage_id, u32 channel)
-{
- int ret;
-
- ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel,
- &stream->mute_volume[channel]);
- if (ret < 0)
- return ret;
-
- ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0);
- if (ret < 0) {
- dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
- stream->reply.stream_hw_id, channel);
- return ret;
- }
-
- stream->mute[channel] = 1;
- return 0;
-}
-
-int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
- u32 stage_id, u32 channel)
-
-{
- int ret;
-
- stream->mute[channel] = 0;
- ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel,
- stream->mute_volume[channel]);
- if (ret < 0) {
- dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
- stream->reply.stream_hw_id, channel);
- return ret;
- }
-
- return 0;
-}
-
int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
u32 stage_id, u32 channel, u32 *volume)
{
@@ -1021,17 +977,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream
return 0;
}
-int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u64 curve_duration,
- enum sst_hsw_volume_curve curve)
-{
- /* curve duration in steps of 100ns */
- stream->vol_req.curve_duration = curve_duration;
- stream->vol_req.curve_type = curve;
-
- return 0;
-}
-
/* stream volume */
int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume)
@@ -1083,42 +1028,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
return 0;
}
-int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
-{
- int ret;
-
- ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel,
- &hsw->mute_volume[channel]);
- if (ret < 0)
- return ret;
-
- ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0);
- if (ret < 0) {
- dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
- channel);
- return ret;
- }
-
- hsw->mute[channel] = 1;
- return 0;
-}
-
-int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
-{
- int ret;
-
- ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel,
- hsw->mixer_info.volume_register_address[channel]);
- if (ret < 0) {
- dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
- channel);
- return ret;
- }
-
- hsw->mute[channel] = 0;
- return 0;
-}
-
int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
u32 *volume)
{
@@ -1132,16 +1041,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
return 0;
}
-int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
- u64 curve_duration, enum sst_hsw_volume_curve curve)
-{
- /* curve duration in steps of 100ns */
- hsw->curve_duration = curve_duration;
- hsw->curve_type = curve;
-
- return 0;
-}
-
/* global mixer volume */
int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
u32 volume)
@@ -1208,6 +1107,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
return NULL;
spin_lock_irqsave(&sst->spinlock, flags);
+ stream->reply.stream_hw_id = INVALID_STREAM_HW_ID;
list_add(&stream->node, &hsw->stream_list);
stream->notify_position = notify_position;
stream->pdata = data;
@@ -1228,6 +1128,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
struct sst_dsp *sst = hsw->dsp;
unsigned long flags;
+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n");
+ return 0;
+ }
+
/* dont free DSP streams that are not commited */
if (!stream->commited)
goto out;
@@ -1415,6 +1320,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
u32 header;
int ret;
+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n");
+ return 0;
+ }
+
+ if (stream->commited) {
+ dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n");
+ return 0;
+ }
+
trace_ipc_request("stream alloc", stream->host_id);
header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
@@ -1434,48 +1349,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
/* Stream Information - these calls could be inline but we want the IPC
ABI to be opaque to client PCM drivers to cope with any future ABI changes */
-int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream)
-{
- return stream->reply.stream_hw_id;
-}
-
-int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream)
-{
- return stream->reply.mixer_hw_id;
-}
-
-u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream)
-{
- return stream->reply.read_position_register_address;
-}
-
-u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream)
-{
- return stream->reply.presentation_position_register_address;
-}
-
-u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u32 channel)
-{
- if (channel >= 2)
- return 0;
-
- return stream->reply.peak_meter_register_address[channel];
-}
-
-u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u32 channel)
-{
- if (channel >= 2)
- return 0;
-
- return stream->reply.volume_register_address[channel];
-}
-
int sst_hsw_mixer_get_info(struct sst_hsw *hsw)
{
struct sst_hsw_ipc_stream_info_reply *reply;
@@ -1519,6 +1392,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
{
int ret;
+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n");
+ return 0;
+ }
+
trace_ipc_request("stream pause", stream->reply.stream_hw_id);
ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE,
@@ -1535,6 +1413,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
{
int ret;
+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n");
+ return 0;
+ }
+
trace_ipc_request("stream resume", stream->reply.stream_hw_id);
ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME,
@@ -1550,6 +1433,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
{
int ret, tries = 10;
+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n");
+ return 0;
+ }
+
/* dont reset streams that are not commited */
if (!stream->commited)
return 0;
@@ -1598,30 +1486,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
return ppos;
}
-int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u32 stage_id, u32 position)
-{
- u32 header;
- int ret;
-
- trace_stream_write_position(stream->reply.stream_hw_id, position);
-
- header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
- IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
- header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
- header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT);
- header |= (stage_id << IPC_STG_ID_SHIFT);
- stream->wpos.position = position;
-
- ret = ipc_tx_message_nowait(hsw, header, &stream->wpos,
- sizeof(stream->wpos));
- if (ret < 0)
- dev_err(hsw->dev, "error: stream %d set position %d failed\n",
- stream->reply.stream_hw_id, position);
-
- return ret;
-}
-
/* physical BE config */
int sst_hsw_device_set_config(struct sst_hsw *hsw,
enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
@@ -2102,7 +1966,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata)
dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE,
hsw->dx_context, hsw->dx_context_paddr);
sst_dsp_free(hsw->dsp);
- kfree(hsw->scratch);
kthread_stop(hsw->tx_thread);
kfree(hsw->msg);
}
diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h
index 138e894ab413..c1ad901342f2 100644
--- a/sound/soc/intel/sst-haswell-ipc.h
+++ b/sound/soc/intel/sst-haswell-ipc.h
@@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw,
u32 create_channel_map(enum sst_hsw_channel_config config);
/* Stream Mixer Controls - */
-int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
- u32 stage_id, u32 channel);
-int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
- u32 stage_id, u32 channel);
-
int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume);
int sst_hsw_stream_get_volume(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume);
-int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u64 curve_duration,
- enum sst_hsw_volume_curve curve);
-
/* Global Mixer Controls - */
-int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
-int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
-
int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
u32 volume);
int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
u32 *volume);
-int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
- u64 curve_duration, enum sst_hsw_volume_curve curve);
-
/* Stream API */
struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data),
@@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 offset, u32 size);
int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 offset, u32 size);
-int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream);
-int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream);
-u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u32 channel);
-u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u32 channel);
int sst_hsw_mixer_get_info(struct sst_hsw *hsw);
/* Stream ALSA trigger operations */
@@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 *position);
int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw,
struct sst_hsw_stream *stream, u32 *position);
-int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
- struct sst_hsw_stream *stream, u32 stage_id, u32 position);
u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw,
struct sst_hsw_stream *stream);
u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
@@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
/* DX Config */
int sst_hsw_dx_set_state(struct sst_hsw *hsw,
enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx);
-int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
- u32 *offset, u32 *size, u32 *source);
/* init */
int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index 619525200705..78fa01be57f2 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -78,7 +78,6 @@ static const u32 volume_map[] = {
#define HSW_PCM_DAI_ID_OFFLOAD0 1
#define HSW_PCM_DAI_ID_OFFLOAD1 2
#define HSW_PCM_DAI_ID_LOOPBACK 3
-#define HSW_PCM_DAI_ID_CAPTURE 4
static const struct snd_pcm_hardware hsw_pcm_hardware = {
@@ -99,6 +98,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = {
struct hsw_pcm_module_map {
int dai_id;
+ int stream;
enum sst_hsw_module_id mod_id;
};
@@ -119,8 +119,9 @@ struct hsw_pcm_data {
};
enum hsw_pm_state {
- HSW_PM_STATE_D3 = 0,
- HSW_PM_STATE_D0 = 1,
+ HSW_PM_STATE_D0 = 0,
+ HSW_PM_STATE_RTD3 = 1,
+ HSW_PM_STATE_D3 = 2,
};
/* private data for the driver */
@@ -135,7 +136,17 @@ struct hsw_priv_data {
struct snd_dma_buffer dmab[HSW_PCM_COUNT][2];
/* DAI data */
- struct hsw_pcm_data pcm[HSW_PCM_COUNT];
+ struct hsw_pcm_data pcm[HSW_PCM_COUNT][2];
+};
+
+
+/* static mappings between PCMs and modules - may be dynamic in future */
+static struct hsw_pcm_module_map mod_map[] = {
+ {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM},
+ {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM},
+ {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM},
+ {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE},
+ {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE},
};
static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data);
@@ -168,9 +179,14 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol,
(struct soc_mixer_control *)kcontrol->private_value;
struct hsw_priv_data *pdata =
snd_soc_platform_get_drvdata(platform);
- struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
+ struct hsw_pcm_data *pcm_data;
struct sst_hsw *hsw = pdata->hsw;
u32 volume;
+ int dai, stream;
+
+ dai = mod_map[mc->reg].dai_id;
+ stream = mod_map[mc->reg].stream;
+ pcm_data = &pdata->pcm[dai][stream];
mutex_lock(&pcm_data->mutex);
pm_runtime_get_sync(pdata->dev);
@@ -212,9 +228,14 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol,
(struct soc_mixer_control *)kcontrol->private_value;
struct hsw_priv_data *pdata =
snd_soc_platform_get_drvdata(platform);
- struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
+ struct hsw_pcm_data *pcm_data;
struct sst_hsw *hsw = pdata->hsw;
u32 volume;
+ int dai, stream;
+
+ dai = mod_map[mc->reg].dai_id;
+ stream = mod_map[mc->reg].stream;
+ pcm_data = &pdata->pcm[dai][stream];
mutex_lock(&pcm_data->mutex);
pm_runtime_get_sync(pdata->dev);
@@ -309,7 +330,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = {
ARRAY_SIZE(volume_map) - 1, 0,
hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
/* Mic Capture volume */
- SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8,
+ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8,
ARRAY_SIZE(volume_map) - 1, 0,
hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
};
@@ -353,7 +374,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_runtime *runtime = substream->runtime;
struct hsw_priv_data *pdata =
snd_soc_platform_get_drvdata(rtd->platform);
- struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct hsw_pcm_data *pcm_data;
struct sst_hsw *hsw = pdata->hsw;
struct sst_module *module_data;
struct sst_dsp *dsp;
@@ -362,7 +383,10 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
enum sst_hsw_stream_path_id path_id;
u32 rate, bits, map, pages, module_id;
u8 channels;
- int ret;
+ int ret, dai;
+
+ dai = mod_map[rtd->cpu_dai->id].dai_id;
+ pcm_data = &pdata->pcm[dai][substream->stream];
/* check if we are being called a subsequent time */
if (pcm_data->allocated) {
@@ -552,8 +576,12 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct hsw_priv_data *pdata =
snd_soc_platform_get_drvdata(rtd->platform);
- struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct hsw_pcm_data *pcm_data;
struct sst_hsw *hsw = pdata->hsw;
+ int dai;
+
+ dai = mod_map[rtd->cpu_dai->id].dai_id;
+ pcm_data = &pdata->pcm[dai][substream->stream];
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -597,11 +625,16 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct hsw_priv_data *pdata =
snd_soc_platform_get_drvdata(rtd->platform);
- struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct hsw_pcm_data *pcm_data;
struct sst_hsw *hsw = pdata->hsw;
snd_pcm_uframes_t offset;
uint64_t ppos;
- u32 position = sst_hsw_get_dsp_position(hsw, pcm_data->stream);
+ u32 position;
+ int dai;
+
+ dai = mod_map[rtd->cpu_dai->id].dai_id;
+ pcm_data = &pdata->pcm[dai][substream->stream];
+ position = sst_hsw_get_dsp_position(hsw, pcm_data->stream);
offset = bytes_to_frames(runtime, position);
ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream);
@@ -618,8 +651,10 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream)
snd_soc_platform_get_drvdata(rtd->platform);
struct hsw_pcm_data *pcm_data;
struct sst_hsw *hsw = pdata->hsw;
+ int dai;
- pcm_data = &pdata->pcm[rtd->cpu_dai->id];
+ dai = mod_map[rtd->cpu_dai->id].dai_id;
+ pcm_data = &pdata->pcm[dai][substream->stream];
mutex_lock(&pcm_data->mutex);
pm_runtime_get_sync(pdata->dev);
@@ -648,9 +683,12 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct hsw_priv_data *pdata =
snd_soc_platform_get_drvdata(rtd->platform);
- struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct hsw_pcm_data *pcm_data;
struct sst_hsw *hsw = pdata->hsw;
- int ret;
+ int ret, dai;
+
+ dai = mod_map[rtd->cpu_dai->id].dai_id;
+ pcm_data = &pdata->pcm[dai][substream->stream];
mutex_lock(&pcm_data->mutex);
ret = sst_hsw_stream_reset(hsw, pcm_data->stream);
@@ -685,15 +723,6 @@ static struct snd_pcm_ops hsw_pcm_ops = {
.page = snd_pcm_sgbuf_ops_page,
};
-/* static mappings between PCMs and modules - may be dynamic in future */
-static struct hsw_pcm_module_map mod_map[] = {
- {HSW_PCM_DAI_ID_SYSTEM, SST_HSW_MODULE_PCM_SYSTEM},
- {HSW_PCM_DAI_ID_OFFLOAD0, SST_HSW_MODULE_PCM},
- {HSW_PCM_DAI_ID_OFFLOAD1, SST_HSW_MODULE_PCM},
- {HSW_PCM_DAI_ID_LOOPBACK, SST_HSW_MODULE_PCM_REFERENCE},
- {HSW_PCM_DAI_ID_CAPTURE, SST_HSW_MODULE_PCM_CAPTURE},
-};
-
static int hsw_pcm_create_modules(struct hsw_priv_data *pdata)
{
struct sst_hsw *hsw = pdata->hsw;
@@ -701,7 +730,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata)
int i;
for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
- pcm_data = &pdata->pcm[i];
+ pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
/* create new runtime module, use same offset if recreated */
pcm_data->runtime = sst_hsw_runtime_module_create(hsw,
@@ -716,7 +745,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata)
err:
for (--i; i >= 0; i--) {
- pcm_data = &pdata->pcm[i];
+ pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
sst_hsw_runtime_module_free(pcm_data->runtime);
}
@@ -729,17 +758,12 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata)
int i;
for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
- pcm_data = &pdata->pcm[i];
+ pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
sst_hsw_runtime_module_free(pcm_data->runtime);
}
}
-static void hsw_pcm_free(struct snd_pcm *pcm)
-{
- snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm *pcm = rtd->pcm;
@@ -762,7 +786,10 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
return ret;
}
}
- priv_data->pcm[rtd->cpu_dai->id].hsw_pcm = pcm;
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream)
+ priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm;
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream)
+ priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm;
return ret;
}
@@ -871,10 +898,9 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform)
/* allocate DSP buffer page tables */
for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) {
- mutex_init(&priv_data->pcm[i].mutex);
-
/* playback */
if (hsw_dais[i].playback.channels_min) {
+ mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_PLAYBACK].mutex);
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev,
PAGE_SIZE, &priv_data->dmab[i][0]);
if (ret < 0)
@@ -883,6 +909,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform)
/* capture */
if (hsw_dais[i].capture.channels_min) {
+ mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_CAPTURE].mutex);
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev,
PAGE_SIZE, &priv_data->dmab[i][1]);
if (ret < 0)
@@ -936,7 +963,6 @@ static struct snd_soc_platform_driver hsw_soc_platform = {
.remove = hsw_pcm_remove,
.ops = &hsw_pcm_ops,
.pcm_new = hsw_pcm_new,
- .pcm_free = hsw_pcm_free,
};
static const struct snd_soc_component_driver hsw_dai_component = {
@@ -1010,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev)
struct hsw_priv_data *pdata = dev_get_drvdata(dev);
struct sst_hsw *hsw = pdata->hsw;
- if (pdata->pm_state == HSW_PM_STATE_D3)
+ if (pdata->pm_state >= HSW_PM_STATE_RTD3)
return 0;
sst_hsw_dsp_runtime_suspend(hsw);
sst_hsw_dsp_runtime_sleep(hsw);
- pdata->pm_state = HSW_PM_STATE_D3;
+ pdata->pm_state = HSW_PM_STATE_RTD3;
return 0;
}
@@ -1026,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev)
struct sst_hsw *hsw = pdata->hsw;
int ret;
- if (pdata->pm_state == HSW_PM_STATE_D0)
+ if (pdata->pm_state != HSW_PM_STATE_RTD3)
return 0;
ret = sst_hsw_dsp_load(hsw);
@@ -1066,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev)
struct hsw_pcm_data *pcm_data;
int i, err;
- if (pdata->pm_state == HSW_PM_STATE_D0)
+ if (pdata->pm_state != HSW_PM_STATE_D3)
return;
err = sst_hsw_dsp_load(hsw);
@@ -1081,8 +1107,8 @@ static void hsw_pcm_complete(struct device *dev)
return;
}
- for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) {
- pcm_data = &pdata->pcm[i];
+ for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
+ pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
if (!pcm_data->substream)
continue;
@@ -1114,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev)
if (pdata->pm_state == HSW_PM_STATE_D3)
return 0;
- /* suspend all active streams */
- for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) {
- pcm_data = &pdata->pcm[i];
+ else if (pdata->pm_state == HSW_PM_STATE_D0) {
+ /* suspend all active streams */
+ for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
+ pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
+
+ if (!pcm_data->substream)
+ continue;
+ dev_dbg(dev, "suspending pcm %d\n", i);
+ snd_pcm_suspend_all(pcm_data->hsw_pcm);
+
+ /* We need to wait until the DSP FW stops the streams */
+ msleep(2);
+ }
- if (!pcm_data->substream)
- continue;
- dev_dbg(dev, "suspending pcm %d\n", i);
- snd_pcm_suspend_all(pcm_data->hsw_pcm);
+ /* preserve persistent memory */
+ for (i = 0; i < ARRAY_SIZE(mod_map); i++) {
+ pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
+
+ if (!pcm_data->substream)
+ continue;
- /* We need to wait until the DSP FW stops the streams */
- msleep(2);
+ dev_dbg(dev, "saving context pcm %d\n", i);
+ err = sst_module_runtime_save(pcm_data->runtime,
+ &pcm_data->context);
+ if (err < 0)
+ dev_err(dev, "failed to save context for PCM %d\n", i);
+ }
+ /* enter D3 state and stall */
+ sst_hsw_dsp_runtime_suspend(hsw);
+ /* put the DSP to sleep */
+ sst_hsw_dsp_runtime_sleep(hsw);
}
snd_soc_suspend(pdata->soc_card->dev);
snd_soc_poweroff(pdata->soc_card->dev);
- /* enter D3 state and stall */
- sst_hsw_dsp_runtime_suspend(hsw);
-
- /* preserve persistent memory */
- for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) {
- pcm_data = &pdata->pcm[i];
-
- if (!pcm_data->substream)
- continue;
-
- dev_dbg(dev, "saving context pcm %d\n", i);
- err = sst_module_runtime_save(pcm_data->runtime,
- &pcm_data->context);
- if (err < 0)
- dev_err(dev, "failed to save context for PCM %d\n", i);
- }
-
- /* put the DSP to sleep */
- sst_hsw_dsp_runtime_sleep(hsw);
pdata->pm_state = HSW_PM_STATE_D3;
return 0;
diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index a1a8d9d91539..7523cbef8780 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -643,12 +643,6 @@ static struct snd_pcm_ops sst_platform_ops = {
.pointer = sst_platform_pcm_pointer,
};
-static void sst_pcm_free(struct snd_pcm *pcm)
-{
- dev_dbg(pcm->dev, "sst_pcm_free called\n");
- snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *dai = rtd->cpu_dai;
@@ -679,7 +673,6 @@ static struct snd_soc_platform_driver sst_soc_platform_drv = {
.ops = &sst_platform_ops,
.compr_ops = &sst_platform_compr_ops,
.pcm_new = sst_pcm_new,
- .pcm_free = sst_pcm_free,
};
static const struct snd_soc_component_driver sst_component = {
diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h
index 7f4bbfcbc6f5..562bc483d6b7 100644
--- a/sound/soc/intel/sst/sst.h
+++ b/sound/soc/intel/sst/sst.h
@@ -58,6 +58,7 @@ enum sst_algo_ops {
#define SST_BLOCK_TIMEOUT 1000
#define FW_SIGNATURE_SIZE 4
+#define FW_NAME_SIZE 32
/* stream states */
enum sst_stream_states {
@@ -426,7 +427,7 @@ struct intel_sst_drv {
* Holder for firmware name. Due to async call it needs to be
* persistent till worker thread gets called
*/
- char firmware_name[20];
+ char firmware_name[FW_NAME_SIZE];
};
/* misc definitions */
diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c
index 3abc29e8a928..b782dfdcdbba 100644
--- a/sound/soc/intel/sst/sst_acpi.c
+++ b/sound/soc/intel/sst/sst_acpi.c
@@ -47,7 +47,7 @@ struct sst_machines {
char board[32];
char machine[32];
void (*machine_quirk)(void);
- char firmware[32];
+ char firmware[FW_NAME_SIZE];
struct sst_platform_info *pdata;
};
@@ -245,7 +245,7 @@ static struct sst_machines *sst_acpi_find_machine(
return NULL;
}
-int sst_acpi_probe(struct platform_device *pdev)
+static int sst_acpi_probe(struct platform_device *pdev)
{
struct device *dev = &pdev->dev;
int ret = 0;
@@ -332,7 +332,7 @@ do_sst_cleanup:
* This function is called by OS when a device is unloaded
* This frees the interrupt etc
*/
-int sst_acpi_remove(struct platform_device *pdev)
+static int sst_acpi_remove(struct platform_device *pdev)
{
struct intel_sst_drv *ctx;
@@ -343,14 +343,16 @@ int sst_acpi_remove(struct platform_device *pdev)
}
static struct sst_machines sst_acpi_bytcr[] = {
- {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin",
+ {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin",
&byt_rvp_platform_data },
{},
};
/* Cherryview-based platforms: CherryTrail and Braswell */
static struct sst_machines sst_acpi_chv[] = {
- {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin",
+ {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin",
+ &chv_platform_data },
+ {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin",
&chv_platform_data },
{},
};
@@ -366,7 +368,6 @@ MODULE_DEVICE_TABLE(acpi, sst_acpi_ids);
static struct platform_driver sst_acpi_driver = {
.driver = {
.name = "intel_sst_acpi",
- .owner = THIS_MODULE,
.acpi_match_table = ACPI_PTR(sst_acpi_ids),
.pm = &intel_sst_pm,
},
diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c
index b580f96e25e5..7888cd707853 100644
--- a/sound/soc/intel/sst/sst_loader.c
+++ b/sound/soc/intel/sst/sst_loader.c
@@ -324,8 +324,7 @@ void sst_firmware_load_cb(const struct firmware *fw, void *context)
if (ctx->sst_state != SST_RESET ||
ctx->fw_in_mem != NULL) {
- if (fw != NULL)
- release_firmware(fw);
+ release_firmware(fw);
mutex_unlock(&ctx->sst_lock);
return;
}
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index d3d45c6f064f..07f77815a586 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -14,6 +14,8 @@
#include <linux/init.h>
#include <linux/io.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/platform_device.h>
@@ -83,6 +85,8 @@
#define JZ_AIC_I2S_STATUS_BUSY BIT(2)
#define JZ_AIC_CLK_DIV_MASK 0xf
+#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
struct jz4740_i2s {
struct resource *mem;
@@ -237,10 +241,14 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
{
struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
unsigned int sample_size;
- uint32_t ctrl;
+ uint32_t ctrl, div_reg;
+ int div;
ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
+ div_reg = jz4740_i2s_read(i2s, JZ_REG_AIC_CLK_DIV);
+ div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
+
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
sample_size = 0;
@@ -264,7 +272,10 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
}
+ div_reg &= ~I2SDIV_DV_MASK;
+ div_reg |= (div - 1) << I2SDIV_DV_SHIFT;
jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
+ jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div_reg);
return 0;
}
@@ -415,6 +426,13 @@ static const struct snd_soc_component_driver jz4740_i2s_component = {
.name = "jz4740-i2s",
};
+#ifdef CONFIG_OF
+static const struct of_device_id jz4740_of_matches[] = {
+ { .compatible = "ingenic,jz4740-i2s" },
+ { /* sentinel */ }
+};
+#endif
+
static int jz4740_i2s_dev_probe(struct platform_device *pdev)
{
struct jz4740_i2s *i2s;
@@ -455,6 +473,7 @@ static struct platform_driver jz4740_i2s_driver = {
.probe = jz4740_i2s_dev_probe,
.driver = {
.name = "jz4740-i2s",
+ .of_match_table = of_match_ptr(jz4740_of_matches)
},
};
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index d9865082160c..c866ade28ad0 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -710,7 +710,7 @@ static int mxs_saif_probe(struct platform_device *pdev)
struct device_node *np = pdev->dev.of_node;
struct resource *iores;
struct mxs_saif *saif;
- int ret = 0;
+ int irq, ret = 0;
struct device_node *master;
if (!np)
@@ -763,16 +763,16 @@ static int mxs_saif_probe(struct platform_device *pdev)
if (IS_ERR(saif->base))
return PTR_ERR(saif->base);
- saif->irq = platform_get_irq(pdev, 0);
- if (saif->irq < 0) {
- ret = saif->irq;
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0) {
+ ret = irq;
dev_err(&pdev->dev, "failed to get irq resource: %d\n",
ret);
return ret;
}
saif->dev = &pdev->dev;
- ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0,
+ ret = devm_request_irq(&pdev->dev, irq, mxs_saif_irq, 0,
dev_name(&pdev->dev), saif);
if (ret) {
dev_err(&pdev->dev, "failed to request irq\n");
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
index fbaf7badfdfb..9a4c0b291b9e 100644
--- a/sound/soc/mxs/mxs-saif.h
+++ b/sound/soc/mxs/mxs-saif.h
@@ -116,7 +116,6 @@ struct mxs_saif {
unsigned int mclk;
unsigned int mclk_in_use;
void __iomem *base;
- int irq;
unsigned int id;
unsigned int master_id;
unsigned int cur_rate;
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 6f1916b71815..6e6fce6a14ba 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -36,7 +36,7 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int rate = params_rate(params);
- u32 dai_format, mclk;
+ u32 mclk;
int ret;
/* sgtl5000 does not support 512*rate when in 96000 fs */
@@ -65,26 +65,6 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
return ret;
}
- /* set codec to slave mode */
- dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS;
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
- if (ret) {
- dev_err(codec_dai->dev, "Failed to set dai format to %08x\n",
- dai_format);
- return ret;
- }
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
- if (ret) {
- dev_err(cpu_dai->dev, "Failed to set dai format to %08x\n",
- dai_format);
- return ret;
- }
-
return 0;
}
@@ -92,17 +72,22 @@ static struct snd_soc_ops mxs_sgtl5000_hifi_ops = {
.hw_params = mxs_sgtl5000_hw_params,
};
+#define MXS_SGTL5000_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBS_CFS)
+
static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
{
.name = "HiFi Tx",
.stream_name = "HiFi Playback",
.codec_dai_name = "sgtl5000",
+ .dai_fmt = MXS_SGTL5000_DAI_FMT,
.ops = &mxs_sgtl5000_hifi_ops,
.playback_only = true,
}, {
.name = "HiFi Rx",
.stream_name = "HiFi Capture",
.codec_dai_name = "sgtl5000",
+ .dai_fmt = MXS_SGTL5000_DAI_FMT,
.ops = &mxs_sgtl5000_hifi_ops,
.capture_only = true,
},
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index b779a3d9b5dd..b809fa909e4d 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -306,11 +306,6 @@ static struct snd_pcm_ops nuc900_dma_ops = {
.mmap = nuc900_dma_mmap,
};
-static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm)
-{
- snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
@@ -330,7 +325,6 @@ static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_platform_driver nuc900_soc_platform = {
.ops = &nuc900_dma_ops,
.pcm_new = nuc900_dma_new,
- .pcm_free = nuc900_dma_free_dma_buffers,
};
static int nuc900_soc_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 4c6afb75eea6..706613077c15 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -412,21 +412,7 @@ static struct tty_ldisc_ops cx81801_ops = {
* over the modem port.
*/
-static int ams_delta_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
-
- /* Set cpu DAI configuration */
- return snd_soc_dai_set_fmt(rtd->cpu_dai,
- SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
-}
-
-static struct snd_soc_ops ams_delta_ops = {
- .hw_params = ams_delta_hw_params,
-};
+static struct snd_soc_ops ams_delta_ops;
/* Digital mute implemented using modem/CPU multiplexer.
@@ -546,6 +532,8 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
.platform_name = "omap-mcbsp.1",
.codec_name = "cx20442-codec",
.ops = &ams_delta_ops,
+ .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
};
/* Audio card driver */
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index 3f9ac7dbdc80..ccfb41c22e53 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -393,7 +393,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev)
static struct platform_driver hdmi_audio_driver = {
.driver = {
.name = DRV_NAME,
- .owner = THIS_MODULE,
},
.probe = omap_hdmi_audio_probe,
.remove = omap_hdmi_audio_remove,
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8b79cafab1e2..c7eb9dd67f60 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
case SND_SOC_DAIFMT_CBM_CFS:
/* McBSP slave. FS clock as output */
regs->srgr2 |= FSGM;
- regs->pcr0 |= FSXM;
+ regs->pcr0 |= FSXM | FSRM;
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* McBSP slave */
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index 5e551c762b7a..fb1f6bb87cd4 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -53,11 +53,7 @@ static int omap_twl4030_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;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_card *card = rtd->card;
unsigned int fmt;
- int ret;
switch (params_channels(params)) {
case 2: /* Stereo I2S mode */
@@ -74,21 +70,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- /* Set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, fmt);
- if (ret < 0) {
- dev_err(card->dev, "can't set codec DAI configuration\n");
- return ret;
- }
-
- /* Set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
- if (ret < 0) {
- dev_err(card->dev, "can't set cpu DAI configuration\n");
- return ret;
- }
-
- return 0;
+ return snd_soc_runtime_set_dai_fmt(rtd, fmt);
}
static struct snd_soc_ops omap_twl4030_ops = {
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 2434b6d61675..39cea80846c3 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -140,7 +140,7 @@ config SND_PXA910_SOC
Marvell PXA910 reference platform.
config SND_SOC_TTC_DKB
- bool "SoC Audio support for TTC DKB"
+ tristate "SoC Audio support for TTC DKB"
depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y
select PXA_SSP
select SND_PXA_SOC_SSP
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index b7cd0a71fd70..3580d10c9f28 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -259,20 +259,6 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
corgi_set_spk),
};
-/*
- * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
- */
-static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_nc_pin(dapm, "LLINEIN");
- snd_soc_dapm_nc_pin(dapm, "RLINEIN");
-
- return 0;
-}
-
/* corgi digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
@@ -281,7 +267,6 @@ static struct snd_soc_dai_link corgi_dai = {
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8731.0-001b",
- .init = corgi_wm8731_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ops = &corgi_ops,
@@ -300,6 +285,7 @@ static struct snd_soc_card corgi = {
.num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
.dapm_routes = corgi_audio_map,
.num_dapm_routes = ARRAY_SIZE(corgi_audio_map),
+ .fully_routed = true,
};
static int corgi_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 7c691aae8af2..d72e124a3676 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -88,24 +88,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Mic Amp", NULL, "Mic (Internal)"},
};
-static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_nc_pin(dapm, "HPOUTL");
- snd_soc_dapm_nc_pin(dapm, "HPOUTR");
- snd_soc_dapm_nc_pin(dapm, "PHONE");
- snd_soc_dapm_nc_pin(dapm, "LINEINL");
- snd_soc_dapm_nc_pin(dapm, "LINEINR");
- snd_soc_dapm_nc_pin(dapm, "CDINL");
- snd_soc_dapm_nc_pin(dapm, "CDINR");
- snd_soc_dapm_nc_pin(dapm, "PCBEEP");
- snd_soc_dapm_nc_pin(dapm, "MIC2");
-
- return 0;
-}
-
static struct snd_soc_dai_link e740_dai[] = {
{
.name = "AC97",
@@ -114,7 +96,6 @@ static struct snd_soc_dai_link e740_dai[] = {
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
- .init = e740_ac97_init,
},
{
.name = "AC97 Aux",
@@ -136,6 +117,7 @@ static struct snd_soc_card e740 = {
.num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
};
static struct gpio e740_audio_gpios[] = {
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 30544b65b5a8..48f2d7c2e68c 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -70,24 +70,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"MIC1", NULL, "Mic (Internal)"},
};
-static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_nc_pin(dapm, "LOUT");
- snd_soc_dapm_nc_pin(dapm, "ROUT");
- snd_soc_dapm_nc_pin(dapm, "PHONE");
- snd_soc_dapm_nc_pin(dapm, "LINEINL");
- snd_soc_dapm_nc_pin(dapm, "LINEINR");
- snd_soc_dapm_nc_pin(dapm, "CDINL");
- snd_soc_dapm_nc_pin(dapm, "CDINR");
- snd_soc_dapm_nc_pin(dapm, "PCBEEP");
- snd_soc_dapm_nc_pin(dapm, "MIC2");
-
- return 0;
-}
-
static struct snd_soc_dai_link e750_dai[] = {
{
.name = "AC97",
@@ -96,7 +78,6 @@ static struct snd_soc_dai_link e750_dai[] = {
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
- .init = e750_ac97_init,
/* use ops to check startup state */
},
{
@@ -119,6 +100,7 @@ static struct snd_soc_card e750 = {
.num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
};
static struct gpio e750_audio_gpios[] = {
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index ce26551052a3..73eb5ddf9753 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -127,15 +127,8 @@ static const struct snd_soc_dapm_route hx4700_audio_map[] = {
static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
- /* NC codec pins */
- /* FIXME: is anything connected here? */
- snd_soc_dapm_nc_pin(dapm, "MOUT1");
- snd_soc_dapm_nc_pin(dapm, "MICEXT");
- snd_soc_dapm_nc_pin(dapm, "AUX");
-
/* Jack detection API stuff */
err = snd_soc_jack_new(codec, "Headphone Jack",
SND_JACK_HEADPHONE, &hs_jack);
@@ -184,6 +177,7 @@ static struct snd_soc_card snd_soc_card_hx4700 = {
.num_dapm_widgets = ARRAY_SIZE(hx4700_dapm_widgets),
.dapm_routes = hx4700_audio_map,
.num_dapm_routes = ARRAY_SIZE(hx4700_audio_map),
+ .fully_routed = true,
};
static struct gpio hx4700_audio_gpios[] = {
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 259e048681c0..241d0be42d7a 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -391,25 +391,6 @@ static const struct snd_kcontrol_new uda1380_magician_controls[] = {
magician_get_input, magician_set_input),
};
-/*
- * Logic for a uda1380 as connected on a HTC Magician
- */
-static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* NC codec pins */
- snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
- snd_soc_dapm_nc_pin(dapm, "VOUTRHP");
-
- /* FIXME: is anything connected here? */
- snd_soc_dapm_nc_pin(dapm, "VINL");
- snd_soc_dapm_nc_pin(dapm, "VINR");
-
- return 0;
-}
-
/* magician digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link magician_dai[] = {
{
@@ -419,7 +400,6 @@ static struct snd_soc_dai_link magician_dai[] = {
.codec_dai_name = "uda1380-hifi-playback",
.platform_name = "pxa-pcm-audio",
.codec_name = "uda1380-codec.0-0018",
- .init = magician_uda1380_init,
.ops = &magician_playback_ops,
},
{
@@ -446,6 +426,7 @@ static struct snd_soc_card snd_soc_card_magician = {
.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
};
static struct platform_device *magician_snd_device;
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 396dbd51a64f..a9615a574546 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -81,7 +81,7 @@ static int rear_amp_power(struct snd_soc_codec *codec, int power)
static int rear_amp_event(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kctl, int event)
{
- struct snd_soc_codec *codec = widget->codec;
+ struct snd_soc_codec *codec = widget->dapm->card->rtd[0].codec;
return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event));
}
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 1eebca2f0a97..910336c5ebeb 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -68,7 +68,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Ext. Speaker", NULL, "ROUT2"},
/* mic connected to MIC1 */
- {"Ext. Microphone", NULL, "MIC1"},
+ {"MIC1", NULL, "Ext. Microphone"},
};
static struct snd_soc_card palm27x_asoc;
@@ -76,18 +76,8 @@ static struct snd_soc_card palm27x_asoc;
static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
- /* not connected pins */
- snd_soc_dapm_nc_pin(dapm, "OUT3");
- snd_soc_dapm_nc_pin(dapm, "MONOOUT");
- snd_soc_dapm_nc_pin(dapm, "LINEINL");
- snd_soc_dapm_nc_pin(dapm, "LINEINR");
- snd_soc_dapm_nc_pin(dapm, "PCBEEP");
- snd_soc_dapm_nc_pin(dapm, "PHONE");
- snd_soc_dapm_nc_pin(dapm, "MIC2");
-
/* Jack detection API stuff */
err = snd_soc_jack_new(codec, "Headphone Jack",
SND_JACK_HEADPHONE, &hs_jack);
@@ -133,7 +123,8 @@ static struct snd_soc_card palm27x_asoc = {
.dapm_widgets = palm27x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(palm27x_dapm_widgets),
.dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map)
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
};
static int palm27x_asoc_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index 083706595495..552b763005ed 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -88,7 +88,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- unsigned int fmt, clk = 0;
+ unsigned int clk = 0;
int ret = 0;
switch (params_rate(params)) {
@@ -112,15 +112,6 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS;
-
- /* setup the CODEC DAI */
- ret = snd_soc_dai_set_fmt(codec_dai, fmt);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0);
if (ret < 0)
return ret;
@@ -130,10 +121,6 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
if (ret < 0)
return ret;
@@ -169,9 +156,8 @@ static int raumfeld_ak4104_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;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int fmt, ret = 0, clk = 0;
+ int ret = 0, clk = 0;
switch (params_rate(params)) {
case 44100:
@@ -194,22 +180,11 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF;
-
- /* setup the CODEC DAI */
- ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
/* setup the CPU DAI */
ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
if (ret < 0)
return ret;
@@ -233,6 +208,9 @@ static struct snd_soc_ops raumfeld_ak4104_ops = {
.platform_name = "pxa-pcm-audio", \
.codec_dai_name = "cs4270-hifi", \
.codec_name = "cs4270.0-0048", \
+ .dai_fmt = SND_SOC_DAIFMT_I2S | \
+ SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBS_CFS, \
.ops = &raumfeld_cs4270_ops, \
}
@@ -243,6 +221,9 @@ static struct snd_soc_ops raumfeld_ak4104_ops = {
.cpu_dai_name = "pxa-ssp-dai.1", \
.codec_dai_name = "ak4104-hifi", \
.platform_name = "pxa-pcm-audio", \
+ .dai_fmt = SND_SOC_DAIFMT_I2S | \
+ SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBS_CFS, \
.ops = &raumfeld_ak4104_ops, \
.codec_name = "spi0.0", \
}
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d7d5fb20ea6f..461123ad5ff2 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -256,26 +256,6 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
spitz_set_spk),
};
-/*
- * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device
- */
-static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* NC codec pins */
- snd_soc_dapm_nc_pin(dapm, "RINPUT1");
- snd_soc_dapm_nc_pin(dapm, "LINPUT2");
- snd_soc_dapm_nc_pin(dapm, "RINPUT2");
- snd_soc_dapm_nc_pin(dapm, "LINPUT3");
- snd_soc_dapm_nc_pin(dapm, "RINPUT3");
- snd_soc_dapm_nc_pin(dapm, "OUT3");
- snd_soc_dapm_nc_pin(dapm, "MONO1");
-
- return 0;
-}
-
/* spitz digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link spitz_dai = {
.name = "wm8750",
@@ -284,7 +264,6 @@ static struct snd_soc_dai_link spitz_dai = {
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8750.0-001b",
- .init = spitz_wm8750_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ops = &spitz_ops,
@@ -303,6 +282,7 @@ static struct snd_soc_card snd_soc_spitz = {
.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
.dapm_routes = spitz_audio_map,
.num_dapm_routes = ARRAY_SIZE(spitz_audio_map),
+ .fully_routed = true,
};
static int spitz_probe(struct platform_device *pdev)
@@ -352,7 +332,6 @@ static int spitz_remove(struct platform_device *pdev)
static struct platform_driver spitz_driver = {
.driver = {
.name = "spitz-audio",
- .owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
},
.probe = spitz_probe,
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index e3d7257ad09c..5001dbb9b257 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -76,10 +76,6 @@ static const struct snd_soc_dapm_route ttc_audio_map[] = {
static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
- snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
/* Headset jack detection */
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 23bf991e95d5..8f301c72ee5e 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -130,16 +130,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
return 0;
}
@@ -172,6 +162,8 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.platform_name = "pxa-pcm-audio",
.cpu_dai_name = "pxa-ssp-dai.2",
.codec_dai_name = "wm9713-voice",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &zylonite_voice_ops,
},
};
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 26ec5117b35c..acb5be53bfb4 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -247,6 +247,10 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val);
regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val);
+ regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK,
+ I2S_DMACR_TDL(16));
+ regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK,
+ I2S_DMACR_RDL(16));
return 0;
}
@@ -335,6 +339,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
SNDRV_PCM_FMTBIT_S24_LE),
},
.ops = &rockchip_i2s_dai_ops,
+ .symmetric_rates = 1,
};
static const struct snd_soc_component_driver rockchip_i2s_component = {
@@ -454,11 +459,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
i2s->playback_dma_data.addr = res->start + I2S_TXDR;
i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
- i2s->playback_dma_data.maxburst = 16;
+ i2s->playback_dma_data.maxburst = 4;
i2s->capture_dma_data.addr = res->start + I2S_RXDR;
i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
- i2s->capture_dma_data.maxburst = 16;
+ i2s->capture_dma_data.maxburst = 4;
i2s->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, i2s);
diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h
index 89a5d8bc6ee7..93f456f518a9 100644
--- a/sound/soc/rockchip/rockchip_i2s.h
+++ b/sound/soc/rockchip/rockchip_i2s.h
@@ -127,7 +127,7 @@
#define I2S_DMACR_TDE_DISABLE (0 << I2S_DMACR_TDE_SHIFT)
#define I2S_DMACR_TDE_ENABLE (1 << I2S_DMACR_TDE_SHIFT)
#define I2S_DMACR_TDL_SHIFT 0
-#define I2S_DMACR_TDL(x) ((x - 1) << I2S_DMACR_TDL_SHIFT)
+#define I2S_DMACR_TDL(x) ((x) << I2S_DMACR_TDL_SHIFT)
#define I2S_DMACR_TDL_MASK (0x1f << I2S_DMACR_TDL_SHIFT)
/*
diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c
index 1e2b61ca8db2..8bf2e2c4bafb 100644
--- a/sound/soc/samsung/arndale_rt5631.c
+++ b/sound/soc/samsung/arndale_rt5631.c
@@ -135,7 +135,6 @@ MODULE_DEVICE_TABLE(of, samsung_arndale_rt5631_of_match);
static struct platform_driver arndale_audio_driver = {
.driver = {
.name = "arndale-audio",
- .owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
},
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index 3b527dcfc0aa..fad56b9e7369 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -136,22 +136,9 @@ static int goni_hifi_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int pll_out = 24000000;
int ret = 0;
- /* set the cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
/* set the codec FLL */
ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out,
params_rate(params) * 256);
@@ -182,12 +169,6 @@ static int goni_voice_hw_params(struct snd_pcm_substream *substream,
if (params_rate(params) != 8000)
return -EINVAL;
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J |
- SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
/* set the codec FLL */
ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out,
params_rate(params) * 256);
@@ -234,6 +215,8 @@ static struct snd_soc_dai_link goni_dai[] = {
.codec_dai_name = "wm8994-aif1",
.platform_name = "samsung-i2s.0",
.codec_name = "wm8994-codec.0-001a",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.init = goni_wm8994_init,
.ops = &goni_hifi_ops,
}, {
@@ -242,6 +225,8 @@ static struct snd_soc_dai_link goni_dai[] = {
.cpu_dai_name = "goni-voice-dai",
.codec_dai_name = "wm8994-aif2",
.codec_name = "wm8994-codec.0-001a",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &goni_voice_ops,
},
};
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index f2d7980d7ddc..59b044255b78 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -76,7 +76,6 @@ static int h1940_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
int div;
int ret;
unsigned int rate = params_rate(params);
@@ -95,18 +94,6 @@ static int h1940_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
/* select clock source */
ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
SND_SOC_CLOCK_OUT);
@@ -207,6 +194,8 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = {
.init = h1940_uda1380_init,
.platform_name = "s3c24xx-iis",
.codec_name = "uda1380-codec.0-001a",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &h1940_ops,
},
};
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index b5f6abd9d221..6c3b359bb4c1 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -61,20 +61,6 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
s3c_i2sv2_get_clock(cpu_dai));
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
SND_SOC_CLOCK_IN);
@@ -121,6 +107,8 @@ static struct snd_soc_dai_link jive_dai = {
.platform_name = "s3c2412-i2s",
.codec_name = "wm8750.0-001a",
.init = jive_wm8750_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &jive_ops,
};
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 9b4a09f14b6c..65602b935377 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -70,20 +70,6 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
break;
}
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
SND_SOC_CLOCK_IN);
@@ -151,13 +137,6 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
- /* todo: gg check mode (DSP_B) against CSR datasheet */
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
SND_SOC_CLOCK_IN);
@@ -300,6 +279,8 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
.codec_name = "wm8753.0-001a",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -309,6 +290,8 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.cpu_dai_name = "bt-sco-pcm",
.codec_dai_name = "wm8753-voice",
.codec_name = "wm8753.0-001a",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 37688ebbb2b4..873f2cb4bebe 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -89,6 +89,8 @@ static struct snd_soc_dai_link rx1950_uda1380_dai[] = {
.init = rx1950_uda1380_init,
.platform_name = "s3c24xx-iis",
.codec_name = "uda1380-codec.0-001a",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &rx1950_ops,
},
};
@@ -154,7 +156,6 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
int div;
int ret;
unsigned int rate = params_rate(params);
@@ -181,18 +182,6 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
/* select clock source */
ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source, rate,
SND_SOC_CLOCK_OUT);
diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c
index 2c015f62ead6..dcc008d1e1ab 100644
--- a/sound/soc/samsung/s3c24xx_simtec.c
+++ b/sound/soc/samsung/s3c24xx_simtec.c
@@ -169,24 +169,6 @@ static int simtec_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
- /* Set the CODEC as the bus clock master, I2S */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret) {
- pr_err("%s: failed set cpu dai format\n", __func__);
- return ret;
- }
-
- /* Set the CODEC as the bus clock master */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret) {
- pr_err("%s: failed set codec dai format\n", __func__);
- return ret;
- }
-
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
CODEC_CLOCK, SND_SOC_CLOCK_IN);
if (ret) {
@@ -320,6 +302,8 @@ int simtec_audio_core_probe(struct platform_device *pdev,
int ret;
card->dai_link->ops = &simtec_snd_ops;
+ card->dai_link->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
pdata = pdev->dev.platform_data;
if (!pdata) {
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index 9c6f7db56f60..50849e137fc0 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -173,16 +173,6 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
@@ -223,6 +213,8 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
.codec_name = "uda134x-codec",
.codec_dai_name = "uda134x-hifi",
.cpu_dai_name = "s3c24xx-iis",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &s3c24xx_uda134x_ops,
.platform_name = "s3c24xx-iis",
};
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index 9b0ffacab790..8291d2a5f152 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -56,20 +56,6 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream,
break;
}
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
/* Use PCLK for I2S signal generation */
ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0,
0, SND_SOC_CLOCK_IN);
@@ -199,6 +185,8 @@ static struct snd_soc_dai_link smartq_dai[] = {
.platform_name = "samsung-i2s.0",
.codec_name = "wm8750.0-0x1a",
.init = smartq_wm8987_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &smartq_hifi_ops,
},
};
diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c
index b1a519f83b29..17a2f717ec02 100644
--- a/sound/soc/samsung/smdk_wm8580.c
+++ b/sound/soc/samsung/smdk_wm8580.c
@@ -32,7 +32,6 @@ static int smdk_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 *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
unsigned int pll_out;
int bfs, rfs, ret;
@@ -77,20 +76,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream,
}
pll_out = params_rate(params) * rfs;
- /* Set the Codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* Set the AP DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
- | SND_SOC_DAIFMT_NB_NF
- | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
/* Set WM8580 to drive MCLK from its PLLA */
ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK,
WM8580_CLKSRC_PLLA);
@@ -168,6 +153,9 @@ enum {
SEC_PLAYBACK,
};
+#define SMDK_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBM_CFM)
+
static struct snd_soc_dai_link smdk_dai[] = {
[PRI_PLAYBACK] = { /* Primary Playback i/f */
.name = "WM8580 PAIF RX",
@@ -176,6 +164,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
.codec_dai_name = "wm8580-hifi-playback",
.platform_name = "samsung-i2s.0",
.codec_name = "wm8580.0-001b",
+ .dai_fmt = SMDK_DAI_FMT,
.ops = &smdk_ops,
},
[PRI_CAPTURE] = { /* Primary Capture i/f */
@@ -185,6 +174,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
.codec_dai_name = "wm8580-hifi-capture",
.platform_name = "samsung-i2s.0",
.codec_name = "wm8580.0-001b",
+ .dai_fmt = SMDK_DAI_FMT,
.init = smdk_wm8580_init_paiftx,
.ops = &smdk_ops,
},
@@ -195,6 +185,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
.codec_dai_name = "wm8580-hifi-playback",
.platform_name = "samsung-i2s-sec",
.codec_name = "wm8580.0-001b",
+ .dai_fmt = SMDK_DAI_FMT,
.ops = &smdk_ops,
},
};
diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c
index 05c609c62de9..6deec5234c92 100644
--- a/sound/soc/samsung/smdk_wm8580pcm.c
+++ b/sound/soc/samsung/smdk_wm8580pcm.c
@@ -62,20 +62,6 @@ static int smdk_wm8580_pcm_hw_params(struct snd_pcm_substream *substream,
rfs = mclk_freq / params_rate(params) / 2;
- /* Set the codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B
- | SND_SOC_DAIFMT_IB_NF
- | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /* Set the cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B
- | SND_SOC_DAIFMT_IB_NF
- | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
if (mclk_freq == xtal_freq) {
ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_MCLK,
mclk_freq, SND_SOC_CLOCK_IN);
@@ -121,6 +107,9 @@ static struct snd_soc_ops smdk_wm8580_pcm_ops = {
.hw_params = smdk_wm8580_pcm_hw_params,
};
+#define SMDK_DAI_FMT (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | \
+ SND_SOC_DAIFMT_CBS_CFS)
+
static struct snd_soc_dai_link smdk_dai[] = {
{
.name = "WM8580 PAIF PCM RX",
@@ -129,6 +118,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
.codec_dai_name = "wm8580-hifi-playback",
.platform_name = "samsung-audio",
.codec_name = "wm8580.0-001b",
+ .dai_fmt = SMDK_DAI_FMT,
.ops = &smdk_wm8580_pcm_ops,
}, {
.name = "WM8580 PAIF PCM TX",
@@ -137,6 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
.codec_dai_name = "wm8580-hifi-capture",
.platform_name = "samsung-pcm.0",
.codec_name = "wm8580.0-001b",
+ .dai_fmt = SMDK_DAI_FMT,
.ops = &smdk_wm8580_pcm_ops,
},
};
diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c
index c470e8eed6e1..b1c89ec2d999 100644
--- a/sound/soc/samsung/smdk_wm8994pcm.c
+++ b/sound/soc/samsung/smdk_wm8994pcm.c
@@ -68,20 +68,6 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream,
mclk_freq = params_rate(params) * rfs;
- /* Set the codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B
- | SND_SOC_DAIFMT_IB_NF
- | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /* Set the cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B
- | SND_SOC_DAIFMT_IB_NF
- | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
mclk_freq, SND_SOC_CLOCK_IN);
if (ret < 0)
@@ -118,6 +104,8 @@ static struct snd_soc_dai_link smdk_dai[] = {
.codec_dai_name = "wm8994-aif1",
.platform_name = "samsung-pcm.0",
.codec_name = "wm8994-codec",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &smdk_wm8994_pcm_ops,
},
};
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 8869971d7884..d49f25f9efd3 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -820,12 +820,9 @@ static int fsi_clk_enable(struct device *dev,
return ret;
}
- if (clock->xck)
- clk_enable(clock->xck);
- if (clock->ick)
- clk_enable(clock->ick);
- if (clock->div)
- clk_enable(clock->div);
+ clk_enable(clock->xck);
+ clk_enable(clock->ick);
+ clk_enable(clock->div);
clock->count++;
}
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index c58c2529f103..82f582344fe7 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -63,16 +63,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF |
- SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai, SND_SOC_DAIFMT_NB_IF |
- SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
codec_freq = rate * 512;
/*
* This propagates the parent frequency change to children and
@@ -144,6 +134,8 @@ static struct snd_soc_dai_link migor_dai = {
.codec_dai_name = "wm8978-hifi",
.platform_name = "siu-pcm-audio",
.codec_name = "wm8978.0-001a",
+ .dai_fmt = SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_CBS_CFS,
.ops = &migor_dai_ops,
};
diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c
index 2e10e9a38376..08d7259bbaab 100644
--- a/sound/soc/soc-ac97.c
+++ b/sound/soc/soc-ac97.c
@@ -48,15 +48,18 @@ static void soc_ac97_device_release(struct device *dev)
}
/**
- * snd_soc_new_ac97_codec - initailise AC97 device
- * @codec: audio codec
+ * snd_soc_alloc_ac97_codec() - Allocate new a AC'97 device
+ * @codec: The CODEC for which to create the AC'97 device
*
- * Initialises AC97 codec resources for use by ad-hoc devices only.
+ * Allocated a new snd_ac97 device and intializes it, but does not yet register
+ * it. The caller is responsible to either call device_add(&ac97->dev) to
+ * register the device, or to call put_device(&ac97->dev) to free the device.
+ *
+ * Returns: A snd_ac97 device or a PTR_ERR in case of an error.
*/
-struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec)
+struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97;
- int ret;
ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
if (ac97 == NULL)
@@ -73,7 +76,28 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec)
codec->component.card->snd_card->number, 0,
codec->component.name);
- ret = device_register(&ac97->dev);
+ device_initialize(&ac97->dev);
+
+ return ac97;
+}
+EXPORT_SYMBOL(snd_soc_alloc_ac97_codec);
+
+/**
+ * snd_soc_new_ac97_codec - initailise AC97 device
+ * @codec: audio codec
+ *
+ * Initialises AC97 codec resources for use by ad-hoc devices only.
+ */
+struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec)
+{
+ struct snd_ac97 *ac97;
+ int ret;
+
+ ac97 = snd_soc_alloc_ac97_codec(codec);
+ if (IS_ERR(ac97))
+ return ac97;
+
+ ret = device_add(&ac97->dev);
if (ret) {
put_device(&ac97->dev);
return ERR_PTR(ret);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 590a82f01d0b..025c38fbe3c0 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
rtd->dai_link->stream_name);
ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
- 1, 0, &be_pcm);
+ rtd->dai_link->dpcm_playback,
+ rtd->dai_link->dpcm_capture, &be_pcm);
if (ret < 0) {
dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n",
rtd->dai_link->name);
@@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
rtd->pcm = be_pcm;
rtd->fe_compr = 1;
- be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
- be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
+ if (rtd->dai_link->dpcm_playback)
+ be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
+ else if (rtd->dai_link->dpcm_capture)
+ be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops));
} else
memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops));
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 985052b3fbed..678823d2e14a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -191,6 +191,39 @@ static ssize_t pmdown_time_set(struct device *dev,
static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set);
+static struct attribute *soc_dev_attrs[] = {
+ &dev_attr_codec_reg.attr,
+ &dev_attr_pmdown_time.attr,
+ NULL
+};
+
+static umode_t soc_dev_attr_is_visible(struct kobject *kobj,
+ struct attribute *attr, int idx)
+{
+ struct device *dev = kobj_to_dev(kobj);
+ struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
+
+ if (attr == &dev_attr_pmdown_time.attr)
+ return attr->mode; /* always visible */
+ return rtd->codec ? attr->mode : 0; /* enabled only with codec */
+}
+
+static const struct attribute_group soc_dapm_dev_group = {
+ .attrs = soc_dapm_dev_attrs,
+ .is_visible = soc_dev_attr_is_visible,
+};
+
+static const struct attribute_group soc_dev_roup = {
+ .attrs = soc_dev_attrs,
+ .is_visible = soc_dev_attr_is_visible,
+};
+
+static const struct attribute_group *soc_dev_attr_groups[] = {
+ &soc_dapm_dev_group,
+ &soc_dev_roup,
+ NULL
+};
+
#ifdef CONFIG_DEBUG_FS
static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
size_t count, loff_t *ppos)
@@ -949,8 +982,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
/* unregister the rtd device */
if (rtd->dev_registered) {
- device_remove_file(rtd->dev, &dev_attr_pmdown_time);
- device_remove_file(rtd->dev, &dev_attr_codec_reg);
device_unregister(rtd->dev);
rtd->dev_registered = 0;
}
@@ -1120,6 +1151,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
device_initialize(rtd->dev);
rtd->dev->parent = rtd->card->dev;
rtd->dev->release = rtd_release;
+ rtd->dev->groups = soc_dev_attr_groups;
dev_set_name(rtd->dev, "%s", name);
dev_set_drvdata(rtd->dev, rtd);
mutex_init(&rtd->pcm_mutex);
@@ -1136,23 +1168,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
return ret;
}
rtd->dev_registered = 1;
-
- if (rtd->codec) {
- /* add DAPM sysfs entries for this codec */
- ret = snd_soc_dapm_sys_add(rtd->dev);
- if (ret < 0)
- dev_err(rtd->dev,
- "ASoC: failed to add codec dapm sysfs entries: %d\n",
- ret);
-
- /* add codec sysfs entries */
- ret = device_create_file(rtd->dev, &dev_attr_codec_reg);
- if (ret < 0)
- dev_err(rtd->dev,
- "ASoC: failed to add codec sysfs files: %d\n",
- ret);
- }
-
return 0;
}
@@ -1308,11 +1323,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
}
#endif
- ret = device_create_file(rtd->dev, &dev_attr_pmdown_time);
- if (ret < 0)
- dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n",
- ret);
-
if (cpu_dai->driver->compress_dai) {
/*create compress_device"*/
ret = soc_new_compress(rtd, num);
@@ -1427,11 +1437,76 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec)
return 0;
}
+/**
+ * snd_soc_runtime_set_dai_fmt() - Change DAI link format for a ASoC runtime
+ * @rtd: The runtime for which the DAI link format should be changed
+ * @dai_fmt: The new DAI link format
+ *
+ * This function updates the DAI link format for all DAIs connected to the DAI
+ * link for the specified runtime.
+ *
+ * Note: For setups with a static format set the dai_fmt field in the
+ * corresponding snd_dai_link struct instead of using this function.
+ *
+ * Returns 0 on success, otherwise a negative error code.
+ */
+int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
+ unsigned int dai_fmt)
+{
+ struct snd_soc_dai **codec_dais = rtd->codec_dais;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int i;
+ int ret;
+
+ for (i = 0; i < rtd->num_codecs; i++) {
+ struct snd_soc_dai *codec_dai = codec_dais[i];
+
+ ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
+ if (ret != 0 && ret != -ENOTSUPP) {
+ dev_warn(codec_dai->dev,
+ "ASoC: Failed to set DAI format: %d\n", ret);
+ return ret;
+ }
+ }
+
+ /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */
+ if (cpu_dai->codec) {
+ unsigned int inv_dai_fmt;
+
+ inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK;
+ switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ break;
+ }
+
+ dai_fmt = inv_dai_fmt;
+ }
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt);
+ if (ret != 0 && ret != -ENOTSUPP) {
+ dev_warn(cpu_dai->dev,
+ "ASoC: Failed to set DAI format: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt);
+
static int snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct snd_soc_codec *codec;
- struct snd_soc_dai_link *dai_link;
- int ret, i, order, dai_fmt;
+ int ret, i, order;
mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
@@ -1542,60 +1617,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
card->num_dapm_routes);
for (i = 0; i < card->num_links; i++) {
- struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
- dai_link = &card->dai_link[i];
- dai_fmt = dai_link->dai_fmt;
-
- if (dai_fmt) {
- struct snd_soc_dai **codec_dais = rtd->codec_dais;
- int j;
-
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = codec_dais[j];
-
- ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
- if (ret != 0 && ret != -ENOTSUPP)
- dev_warn(codec_dai->dev,
- "ASoC: Failed to set DAI format: %d\n",
- ret);
- }
- }
-
- /* If this is a regular CPU link there will be a platform */
- if (dai_fmt &&
- (dai_link->platform_name || dai_link->platform_of_node)) {
- ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai,
- dai_fmt);
- if (ret != 0 && ret != -ENOTSUPP)
- dev_warn(card->rtd[i].cpu_dai->dev,
- "ASoC: Failed to set DAI format: %d\n",
- ret);
- } else if (dai_fmt) {
- /* Flip the polarity for the "CPU" end */
- dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
- switch (dai_link->dai_fmt &
- SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBM_CFM:
- dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
- break;
- case SND_SOC_DAIFMT_CBM_CFS:
- dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
- break;
- case SND_SOC_DAIFMT_CBS_CFM:
- dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
- break;
- case SND_SOC_DAIFMT_CBS_CFS:
- dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
- break;
- }
-
- ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai,
- dai_fmt);
- if (ret != 0 && ret != -ENOTSUPP)
- dev_warn(card->rtd[i].cpu_dai->dev,
- "ASoC: Failed to set DAI format: %d\n",
- ret);
- }
+ if (card->dai_link[i].dai_fmt)
+ snd_soc_runtime_set_dai_fmt(&card->rtd[i],
+ card->dai_link[i].dai_fmt);
}
snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname),
@@ -1626,9 +1650,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
}
- if (card->fully_routed)
- snd_soc_dapm_auto_nc_pins(card);
-
snd_soc_dapm_new_widgets(card);
ret = snd_card_register(card->snd_card);
@@ -2386,8 +2407,8 @@ int snd_soc_unregister_card(struct snd_soc_card *card)
card->instantiated = false;
snd_soc_dapm_shutdown(card);
soc_cleanup_card_resources(card);
+ dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
}
- dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
return 0;
}
@@ -3230,7 +3251,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
const char *propname)
{
struct device_node *np = card->dev->of_node;
- int num_routes, old_routes;
+ int num_routes;
struct snd_soc_dapm_route *routes;
int i, ret;
@@ -3248,9 +3269,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
return -EINVAL;
}
- old_routes = card->num_dapm_routes;
- routes = devm_kzalloc(card->dev,
- (old_routes + num_routes) * sizeof(*routes),
+ routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes),
GFP_KERNEL);
if (!routes) {
dev_err(card->dev,
@@ -3258,11 +3277,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
return -EINVAL;
}
- memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes));
-
for (i = 0; i < num_routes; i++) {
ret = of_property_read_string_index(np, propname,
- 2 * i, &routes[old_routes + i].sink);
+ 2 * i, &routes[i].sink);
if (ret) {
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
@@ -3270,7 +3287,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
return -EINVAL;
}
ret = of_property_read_string_index(np, propname,
- (2 * i) + 1, &routes[old_routes + i].source);
+ (2 * i) + 1, &routes[i].source);
if (ret) {
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
@@ -3279,7 +3296,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
}
}
- card->num_dapm_routes += num_routes;
+ card->num_dapm_routes = num_routes;
card->dapm_routes = routes;
return 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c5136bb1f982..b6f88202b8c9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -517,8 +517,8 @@ static int soc_dapm_update_bits(struct snd_soc_dapm_context *dapm,
{
if (!dapm->component)
return -EIO;
- return snd_soc_component_update_bits_async(dapm->component, reg,
- mask, value);
+ return snd_soc_component_update_bits(dapm->component, reg,
+ mask, value);
}
static int soc_dapm_test_bits(struct snd_soc_dapm_context *dapm,
@@ -2127,15 +2127,10 @@ static ssize_t dapm_widget_show(struct device *dev,
static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
-int snd_soc_dapm_sys_add(struct device *dev)
-{
- return device_create_file(dev, &dev_attr_dapm_widget);
-}
-
-static void snd_soc_dapm_sys_remove(struct device *dev)
-{
- device_remove_file(dev, &dev_attr_dapm_widget);
-}
+struct attribute *soc_dapm_dev_attrs[] = {
+ &dev_attr_dapm_widget.attr,
+ NULL
+};
static void dapm_free_path(struct snd_soc_dapm_path *path)
{
@@ -2279,6 +2274,9 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w)
switch (w->id) {
case snd_soc_dapm_input:
+ /* On a fully routed card a input is never a source */
+ if (w->dapm->card->fully_routed)
+ break;
w->is_source = 1;
list_for_each_entry(p, &w->sources, list_sink) {
if (p->source->id == snd_soc_dapm_micbias ||
@@ -2291,6 +2289,9 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w)
}
break;
case snd_soc_dapm_output:
+ /* On a fully routed card a output is never a sink */
+ if (w->dapm->card->fully_routed)
+ break;
w->is_sink = 1;
list_for_each_entry(p, &w->sinks, list_source) {
if (p->sink->id == snd_soc_dapm_spk ||
@@ -3085,16 +3086,24 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
switch (w->id) {
case snd_soc_dapm_mic:
- case snd_soc_dapm_input:
w->is_source = 1;
w->power_check = dapm_generic_check_power;
break;
+ case snd_soc_dapm_input:
+ if (!dapm->card->fully_routed)
+ w->is_source = 1;
+ w->power_check = dapm_generic_check_power;
+ break;
case snd_soc_dapm_spk:
case snd_soc_dapm_hp:
- case snd_soc_dapm_output:
w->is_sink = 1;
w->power_check = dapm_generic_check_power;
break;
+ case snd_soc_dapm_output:
+ if (!dapm->card->fully_routed)
+ w->is_sink = 1;
+ w->power_check = dapm_generic_check_power;
+ break;
case snd_soc_dapm_vmid:
case snd_soc_dapm_siggen:
w->is_source = 1;
@@ -3130,8 +3139,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
}
w->dapm = dapm;
- if (dapm->component)
- w->codec = dapm->component->codec;
INIT_LIST_HEAD(&w->sources);
INIT_LIST_HEAD(&w->sinks);
INIT_LIST_HEAD(&w->list);
@@ -3809,93 +3816,6 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
/**
- * dapm_is_external_path() - Checks if a path is a external path
- * @card: The card the path belongs to
- * @path: The path to check
- *
- * Returns true if the path is either between two different DAPM contexts or
- * between two external pins of the same DAPM context. Otherwise returns
- * false.
- */
-static bool dapm_is_external_path(struct snd_soc_card *card,
- struct snd_soc_dapm_path *path)
-{
- dev_dbg(card->dev,
- "... Path %s(id:%d dapm:%p) - %s(id:%d dapm:%p)\n",
- path->source->name, path->source->id, path->source->dapm,
- path->sink->name, path->sink->id, path->sink->dapm);
-
- /* Connection between two different DAPM contexts */
- if (path->source->dapm != path->sink->dapm)
- return true;
-
- /* Loopback connection from external pin to external pin */
- if (path->sink->id == snd_soc_dapm_input) {
- switch (path->source->id) {
- case snd_soc_dapm_output:
- case snd_soc_dapm_micbias:
- return true;
- default:
- break;
- }
- }
-
- return false;
-}
-
-static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card,
- struct snd_soc_dapm_widget *w)
-{
- struct snd_soc_dapm_path *p;
-
- list_for_each_entry(p, &w->sources, list_sink) {
- if (dapm_is_external_path(card, p))
- return true;
- }
-
- list_for_each_entry(p, &w->sinks, list_source) {
- if (dapm_is_external_path(card, p))
- return true;
- }
-
- return false;
-}
-
-/**
- * snd_soc_dapm_auto_nc_pins - call snd_soc_dapm_nc_pin for unused pins
- * @card: The card whose pins should be processed
- *
- * Automatically call snd_soc_dapm_nc_pin() for any external pins in the card
- * which are unused. Pins are used if they are connected externally to a
- * component, whether that be to some other device, or a loop-back connection to
- * the component itself.
- */
-void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card)
-{
- struct snd_soc_dapm_widget *w;
-
- dev_dbg(card->dev, "ASoC: Auto NC: DAPMs: card:%p\n", &card->dapm);
-
- list_for_each_entry(w, &card->widgets, list) {
- switch (w->id) {
- case snd_soc_dapm_input:
- case snd_soc_dapm_output:
- case snd_soc_dapm_micbias:
- dev_dbg(card->dev, "ASoC: Auto NC: Checking widget %s\n",
- w->name);
- if (!snd_soc_dapm_widget_in_card_paths(card, w)) {
- dev_dbg(card->dev,
- "... Not in map; disabling\n");
- snd_soc_dapm_nc_pin(w->dapm, w->name);
- }
- break;
- default:
- break;
- }
- }
-}
-
-/**
* snd_soc_dapm_free - free dapm resources
* @dapm: DAPM context
*
@@ -3903,7 +3823,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card)
*/
void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm)
{
- snd_soc_dapm_sys_remove(dapm->dev);
dapm_debugfs_cleanup(dapm);
dapm_free_widgets(dapm);
list_del(&dapm->list);
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index 057e5ef7dcce..a57921eeee81 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res)
/**
* devm_snd_soc_register_platform - resource managed platform registration
* @dev: Device used to manage platform
- * @platform: platform to register
+ * @platform_drv: platform to register
*
* Register a platform driver with automatic unregistration when the device is
* unregistered.
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index b329b84bc5af..4864392bfcba 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -200,11 +200,6 @@ static int dmaengine_pcm_open(struct snd_pcm_substream *substream)
return snd_dmaengine_pcm_open(substream, chan);
}
-static void dmaengine_pcm_free(struct snd_pcm *pcm)
-{
- snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
static struct dma_chan *dmaengine_pcm_compat_request_channel(
struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_substream *substream)
@@ -283,8 +278,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
if (!pcm->chan[i]) {
dev_err(rtd->platform->dev,
"Missing dma channel for stream: %d\n", i);
- ret = -EINVAL;
- goto err_free;
+ return -EINVAL;
}
ret = snd_pcm_lib_preallocate_pages(substream,
@@ -293,7 +287,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
prealloc_buffer_size,
max_buffer_size);
if (ret)
- goto err_free;
+ return ret;
/*
* This will only return false if we know for sure that at least
@@ -307,10 +301,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
}
return 0;
-
-err_free:
- dmaengine_pcm_free(rtd->pcm);
- return ret;
}
static snd_pcm_uframes_t dmaengine_pcm_pointer(
@@ -341,7 +331,6 @@ static const struct snd_soc_platform_driver dmaengine_pcm_platform = {
},
.ops = &dmaengine_pcm_ops,
.pcm_new = dmaengine_pcm_new,
- .pcm_free = dmaengine_pcm_free,
};
static const char * const dmaengine_pcm_dma_channel_names[] = {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index eb87d96e2cf0..0ae0e2a9eed7 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -746,7 +746,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
codec_dai);
if (ret < 0) {
dev_err(codec_dai->dev,
- "ASoC: DAI prepare error: %d\n", ret);
+ "ASoC: codec DAI prepare error: %d\n",
+ ret);
goto out;
}
}
@@ -755,8 +756,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) {
ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
if (ret < 0) {
- dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n",
- ret);
+ dev_err(cpu_dai->dev,
+ "ASoC: cpu DAI prepare error: %d\n", ret);
goto out;
}
}
diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c
index be4f1ac7cd5e..aa65370db82a 100644
--- a/sound/soc/ux500/mop500_ab8500.c
+++ b/sound/soc/ux500/mop500_ab8500.c
@@ -290,21 +290,9 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream,
SND_SOC_DAIFMT_GATED;
}
- ret = snd_soc_dai_set_fmt(codec_dai, fmt);
- if (ret < 0) {
- dev_err(dev,
- "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n",
- __func__, ret);
- return ret;
- }
-
- ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
- if (ret < 0) {
- dev_err(dev,
- "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n",
- __func__, ret);
+ ret = snd_soc_runtime_set_dai_fmt(rtd, fmt);
+ if (ret)
return ret;
- }
/* Setup TDM-slots */
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 272844746135..327f8642ca80 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -816,7 +816,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev)
return -EINVAL;
}
- if (cdev->n_streams < 2) {
+ if (cdev->n_streams < 1) {
dev_err(dev, "bogus number of streams: %d\n", cdev->n_streams);
return -EINVAL;
}
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 41650d5b93b7..3e2ef61c627b 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -913,6 +913,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
case USB_ID(0x046d, 0x0807): /* Logitech Webcam C500 */
case USB_ID(0x046d, 0x0808):
case USB_ID(0x046d, 0x0809):
+ case USB_ID(0x046d, 0x0819): /* Logitech Webcam C210 */
case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */
case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */