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-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/atmel_ssc_dai.c24
-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.c2
-rw-r--r--sound/soc/codecs/rt286.c46
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/codecs/rt5670.c38
-rw-r--r--sound/soc/codecs/rt5677.c222
-rw-r--r--sound/soc/codecs/sgtl5000.c13
-rw-r--r--sound/soc/codecs/tlv320aic3x.c2
-rw-r--r--sound/soc/codecs/ts3a227e.c6
-rw-r--r--sound/soc/codecs/wm8731.c2
-rw-r--r--sound/soc/codecs/wm8750.c2
-rw-r--r--sound/soc/codecs/wm8904.c23
-rw-r--r--sound/soc/codecs/wm8960.c2
-rw-r--r--sound/soc/codecs/wm9705.c16
-rw-r--r--sound/soc/codecs/wm9712.c12
-rw-r--r--sound/soc/codecs/wm9713.c12
-rw-r--r--sound/soc/dwc/designware_i2s.c135
-rw-r--r--sound/soc/fsl/fsl_esai.h2
-rw-r--r--sound/soc/fsl/fsl_ssi.c4
-rw-r--r--sound/soc/fsl/imx-wm8962.c1
-rw-r--r--sound/soc/generic/simple-card.c7
-rw-r--r--sound/soc/intel/Kconfig8
-rw-r--r--sound/soc/intel/bytcr_dpcm_rt5640.c3
-rw-r--r--sound/soc/intel/cht_bsw_rt5672.c1
-rw-r--r--sound/soc/intel/sst-firmware.c16
-rw-r--r--sound/soc/intel/sst-haswell-ipc.c34
-rw-r--r--sound/soc/intel/sst/sst.h3
-rw-r--r--sound/soc/intel/sst/sst_acpi.c11
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c1
-rw-r--r--sound/soc/omap/omap-mcbsp.c2
-rw-r--r--sound/soc/pxa/spitz.c1
-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/soc-ac97.c36
-rw-r--r--sound/soc/soc-compress.c9
-rw-r--r--sound/soc/soc-core.c17
-rw-r--r--sound/soc/soc-dapm.c105
-rw-r--r--sound/soc/soc-pcm.c7
-rw-r--r--sound/usb/caiaq/audio.c2
-rw-r--r--sound/usb/mixer.c1
55 files changed, 625 insertions, 383 deletions
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/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 99ff35e2a25d..35e44e463cfe 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -348,7 +348,6 @@ 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;
@@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
* 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);
-
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 +468,14 @@ 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 +485,7 @@ 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));
@@ -512,7 +502,7 @@ 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 +517,7 @@ 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);
@@ -556,7 +546,7 @@ 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);
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..30c673cdc12e 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -188,8 +188,8 @@ 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",
};
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 2cd4fe463102..f14d335b07b1 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -305,6 +305,8 @@ 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;
@@ -417,6 +419,8 @@ 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,
@@ -538,32 +542,6 @@ 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)
{
@@ -667,12 +645,10 @@ 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 +837,8 @@ 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;
}
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index c3f2decd643c..1ff726c29249 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2124,6 +2124,7 @@ 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/rt5670.c b/sound/soc/codecs/rt5670.c
index 8a0833de1665..0a027bc94399 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>
@@ -2188,6 +2190,13 @@ 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 +2558,17 @@ 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 +2588,12 @@ 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 +2635,10 @@ 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 +2746,26 @@ 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/rt5677.c b/sound/soc/codecs/rt5677.c
index 81fe1464d268..d27630accf03 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];
@@ -921,6 +922,97 @@ 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)
+{
+ 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;
+ }
+ }
+
+ val = (snd_soc_read(source->codec, reg) >> 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,
@@ -2082,10 +2174,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w,
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;
}
@@ -2100,10 +2196,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w,
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;
}
@@ -2211,9 +2311,50 @@ 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 +2786,18 @@ 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 +2870,31 @@ 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 +3025,6 @@ 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 +3045,6 @@ 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 +3059,6 @@ 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 +3073,6 @@ 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 +3621,8 @@ 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 +3689,46 @@ 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 +3750,8 @@ 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" },
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 29cf7ce610f4..aa98be32bb60 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -483,21 +483,21 @@ 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 +1462,9 @@ 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/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b7ebce054b4e..dd222b10ce13 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -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/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/wm8731.c b/sound/soc/codecs/wm8731.c
index b9211b42f6e9..b115ed815db9 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -717,6 +717,8 @@ 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/wm8904.c b/sound/soc/codecs/wm8904.c
index 4d2d2b1380d5..75b87c5c0f04 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -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/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/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..6ab1122a3872 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -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/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index b93168d4f648..06d3a34ac90a 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -209,16 +209,9 @@ 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");
@@ -227,18 +220,22 @@ 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);
@@ -263,6 +260,19 @@ 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 +304,7 @@ 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,
};
@@ -324,13 +335,47 @@ static int dw_i2s_resume(struct snd_soc_dai *dai)
#define dw_i2s_resume NULL
#endif
+static void 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)
+{
+ /* Set DMA slaves info */
+
+ 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;
+
+ if (pdata->cap & DWC_I2S_PLAY) {
+ dev_dbg(dev->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 (pdata->cap & DWC_I2S_RECORD) {
+ dev_dbg(dev->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;
+ }
+}
+
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) {
@@ -345,44 +390,23 @@ 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;
- /* Set DMA slaves info */
-
- 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->dev = &pdev->dev;
+ dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
+ dev->capability = pdata->cap;
+ dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
dev->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(dev->clk))
return PTR_ERR(dev->clk);
@@ -391,23 +415,6 @@ static int dw_i2s_probe(struct platform_device *pdev)
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;
- }
-
- dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component,
dw_i2s_dai, 1);
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_ssi.c b/sound/soc/fsl/fsl_ssi.c
index a65f17d57ffb..059496ed9ad7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1362,9 +1362,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}
ssi_private->irq = platform_get_irq(pdev, 0);
- if (!ssi_private->irq) {
+ if (ssi_private->irq < 0) {
dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
- return -ENXIO;
+ return ssi_private->irq;
}
/* Are the RX and the TX clocks locked? */
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/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..c0813f546d1f 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
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_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c
index 9b8b561171b7..a406c6104897 100644
--- a/sound/soc/intel/cht_bsw_rt5672.c
+++ b/sound/soc/intel/cht_bsw_rt5672.c
@@ -270,7 +270,6 @@ 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-firmware.c b/sound/soc/intel/sst-firmware.c
index 4a5bde9c686b..a2ae2c5f2e9f 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;
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 3f8c48231364..8156cc1accb7 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -651,11 +651,11 @@ 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)
@@ -1228,6 +1228,11 @@ 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 +1420,16 @@ 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);
@@ -1519,6 +1534,11 @@ 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 +1555,11 @@ 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 +1575,11 @@ 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;
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..e541d0e69ea2 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 +343,14 @@ 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 },
{},
};
@@ -366,7 +366,6 @@ 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/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/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d7d5fb20ea6f..a6d680acd907 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -352,7 +352,6 @@ 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/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/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..c024962ba500 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1626,9 +1626,6 @@ 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);
@@ -3230,7 +3227,7 @@ 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 +3245,7 @@ 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 +3253,9 @@ 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 +3263,7 @@ 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 +3272,7 @@ 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..ea496842ee83 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2279,6 +2279,9 @@ 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 +2294,9 @@ 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 +3091,24 @@ 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;
@@ -3809,93 +3823,6 @@ 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
*
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/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 */