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-rw-r--r--sound/core/pcm_native.c2
-rw-r--r--sound/firewire/amdtp.c5
-rw-r--r--sound/firewire/bebob/bebob.c20
-rw-r--r--sound/firewire/bebob/bebob_stream.c16
-rw-r--r--sound/firewire/dice/dice-stream.c18
-rw-r--r--sound/firewire/dice/dice.c16
-rw-r--r--sound/firewire/fireworks/fireworks.c20
-rw-r--r--sound/firewire/fireworks/fireworks_stream.c19
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c6
-rw-r--r--sound/firewire/oxfw/oxfw.c21
-rw-r--r--sound/pci/hda/hda_controller.c5
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c17
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c68
-rw-r--r--sound/soc/cirrus/Kconfig2
-rw-r--r--sound/soc/codecs/Kconfig2
-rw-r--r--sound/soc/codecs/max98357a.c12
-rw-r--r--sound/soc/codecs/rt5670.c7
-rw-r--r--sound/soc/codecs/rt5677.c32
-rw-r--r--sound/soc/fsl/fsl_ssi.c11
-rw-r--r--sound/soc/intel/sst-atom-controls.h2
-rw-r--r--sound/soc/intel/sst/sst.c10
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c3
-rw-r--r--sound/soc/omap/omap-mcbsp.c11
-rw-r--r--sound/soc/omap/omap-pcm.c2
-rw-r--r--sound/soc/samsung/Kconfig10
-rw-r--r--sound/soc/sh/rcar/core.c4
27 files changed, 211 insertions, 132 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index b03a638b420c..279e24f61305 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1552,6 +1552,8 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state)
if (! snd_pcm_playback_empty(substream)) {
snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING);
snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING);
+ } else {
+ runtime->status->state = SNDRV_PCM_STATE_SETUP;
}
break;
case SNDRV_PCM_STATE_RUNNING:
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 0d580186ef1a..5cc356db5351 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -33,7 +33,7 @@
*/
#define MAX_MIDI_RX_BLOCKS 8
-#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */
+#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */
/* isochronous header parameters */
#define ISO_DATA_LENGTH_SHIFT 16
@@ -78,7 +78,7 @@ static void pcm_period_tasklet(unsigned long data);
int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
enum amdtp_stream_direction dir, enum cip_flags flags)
{
- s->unit = fw_unit_get(unit);
+ s->unit = unit;
s->direction = dir;
s->flags = flags;
s->context = ERR_PTR(-1);
@@ -102,7 +102,6 @@ void amdtp_stream_destroy(struct amdtp_stream *s)
{
WARN_ON(amdtp_stream_running(s));
mutex_destroy(&s->mutex);
- fw_unit_put(s->unit);
}
EXPORT_SYMBOL(amdtp_stream_destroy);
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index fc19c99654aa..611b7dae7ee5 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -116,11 +116,22 @@ end:
return err;
}
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
static void
bebob_card_free(struct snd_card *card)
{
struct snd_bebob *bebob = card->private_data;
+ snd_bebob_stream_destroy_duplex(bebob);
+ fw_unit_put(bebob->unit);
+
+ kfree(bebob->maudio_special_quirk);
+
if (bebob->card_index >= 0) {
mutex_lock(&devices_mutex);
clear_bit(bebob->card_index, devices_used);
@@ -205,7 +216,7 @@ bebob_probe(struct fw_unit *unit,
card->private_free = bebob_card_free;
bebob->card = card;
- bebob->unit = unit;
+ bebob->unit = fw_unit_get(unit);
bebob->spec = spec;
mutex_init(&bebob->mutex);
spin_lock_init(&bebob->lock);
@@ -306,10 +317,11 @@ static void bebob_remove(struct fw_unit *unit)
if (bebob == NULL)
return;
- kfree(bebob->maudio_special_quirk);
+ /* Awake bus-reset waiters. */
+ if (!completion_done(&bebob->bus_reset))
+ complete_all(&bebob->bus_reset);
- snd_bebob_stream_destroy_duplex(bebob);
- snd_card_disconnect(bebob->card);
+ /* No need to wait for releasing card object in this context. */
snd_card_free_when_closed(bebob->card);
}
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 0ebcabfdc7ce..98e4fc8121a1 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -410,8 +410,6 @@ break_both_connections(struct snd_bebob *bebob)
static void
destroy_both_connections(struct snd_bebob *bebob)
{
- break_both_connections(bebob);
-
cmp_connection_destroy(&bebob->in_conn);
cmp_connection_destroy(&bebob->out_conn);
}
@@ -712,22 +710,16 @@ void snd_bebob_stream_update_duplex(struct snd_bebob *bebob)
mutex_unlock(&bebob->mutex);
}
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob)
{
- mutex_lock(&bebob->mutex);
-
- amdtp_stream_pcm_abort(&bebob->rx_stream);
- amdtp_stream_pcm_abort(&bebob->tx_stream);
-
- amdtp_stream_stop(&bebob->rx_stream);
- amdtp_stream_stop(&bebob->tx_stream);
-
amdtp_stream_destroy(&bebob->rx_stream);
amdtp_stream_destroy(&bebob->tx_stream);
destroy_both_connections(bebob);
-
- mutex_unlock(&bebob->mutex);
}
/*
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index fa9cf761b610..07dbd01d7a6b 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -311,14 +311,21 @@ end:
return err;
}
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
static void destroy_stream(struct snd_dice *dice, struct amdtp_stream *stream)
{
- amdtp_stream_destroy(stream);
+ struct fw_iso_resources *resources;
if (stream == &dice->tx_stream)
- fw_iso_resources_destroy(&dice->tx_resources);
+ resources = &dice->tx_resources;
else
- fw_iso_resources_destroy(&dice->rx_resources);
+ resources = &dice->rx_resources;
+
+ amdtp_stream_destroy(stream);
+ fw_iso_resources_destroy(resources);
}
int snd_dice_stream_init_duplex(struct snd_dice *dice)
@@ -332,6 +339,8 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice)
goto end;
err = init_stream(dice, &dice->rx_stream);
+ if (err < 0)
+ destroy_stream(dice, &dice->tx_stream);
end:
return err;
}
@@ -340,10 +349,7 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice)
{
snd_dice_transaction_clear_enable(dice);
- stop_stream(dice, &dice->tx_stream);
destroy_stream(dice, &dice->tx_stream);
-
- stop_stream(dice, &dice->rx_stream);
destroy_stream(dice, &dice->rx_stream);
dice->substreams_counter = 0;
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 90d8f40ff727..70a111d7f428 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -226,11 +226,20 @@ static void dice_card_strings(struct snd_dice *dice)
strcpy(card->mixername, "DICE");
}
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
static void dice_card_free(struct snd_card *card)
{
struct snd_dice *dice = card->private_data;
+ snd_dice_stream_destroy_duplex(dice);
snd_dice_transaction_destroy(dice);
+ fw_unit_put(dice->unit);
+
mutex_destroy(&dice->mutex);
}
@@ -251,7 +260,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
dice = card->private_data;
dice->card = card;
- dice->unit = unit;
+ dice->unit = fw_unit_get(unit);
card->private_free = dice_card_free;
spin_lock_init(&dice->lock);
@@ -305,10 +314,7 @@ static void dice_remove(struct fw_unit *unit)
{
struct snd_dice *dice = dev_get_drvdata(&unit->device);
- snd_card_disconnect(dice->card);
-
- snd_dice_stream_destroy_duplex(dice);
-
+ /* No need to wait for releasing card object in this context. */
snd_card_free_when_closed(dice->card);
}
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 3e2ed8e82cbc..2682e7e3e5c9 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -173,11 +173,23 @@ end:
return err;
}
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
static void
efw_card_free(struct snd_card *card)
{
struct snd_efw *efw = card->private_data;
+ snd_efw_stream_destroy_duplex(efw);
+ snd_efw_transaction_remove_instance(efw);
+ fw_unit_put(efw->unit);
+
+ kfree(efw->resp_buf);
+
if (efw->card_index >= 0) {
mutex_lock(&devices_mutex);
clear_bit(efw->card_index, devices_used);
@@ -185,7 +197,6 @@ efw_card_free(struct snd_card *card)
}
mutex_destroy(&efw->mutex);
- kfree(efw->resp_buf);
}
static int
@@ -218,7 +229,7 @@ efw_probe(struct fw_unit *unit,
card->private_free = efw_card_free;
efw->card = card;
- efw->unit = unit;
+ efw->unit = fw_unit_get(unit);
mutex_init(&efw->mutex);
spin_lock_init(&efw->lock);
init_waitqueue_head(&efw->hwdep_wait);
@@ -289,10 +300,7 @@ static void efw_remove(struct fw_unit *unit)
{
struct snd_efw *efw = dev_get_drvdata(&unit->device);
- snd_efw_stream_destroy_duplex(efw);
- snd_efw_transaction_remove_instance(efw);
-
- snd_card_disconnect(efw->card);
+ /* No need to wait for releasing card object in this context. */
snd_card_free_when_closed(efw->card);
}
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 4f440e163667..c55db1bddc80 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -100,17 +100,22 @@ end:
return err;
}
+/*
+ * This function should be called before starting the stream or after stopping
+ * the streams.
+ */
static void
destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream)
{
- stop_stream(efw, stream);
-
- amdtp_stream_destroy(stream);
+ struct cmp_connection *conn;
if (stream == &efw->tx_stream)
- cmp_connection_destroy(&efw->out_conn);
+ conn = &efw->out_conn;
else
- cmp_connection_destroy(&efw->in_conn);
+ conn = &efw->in_conn;
+
+ amdtp_stream_destroy(stream);
+ cmp_connection_destroy(&efw->out_conn);
}
static int
@@ -319,12 +324,8 @@ void snd_efw_stream_update_duplex(struct snd_efw *efw)
void snd_efw_stream_destroy_duplex(struct snd_efw *efw)
{
- mutex_lock(&efw->mutex);
-
destroy_stream(efw, &efw->rx_stream);
destroy_stream(efw, &efw->tx_stream);
-
- mutex_unlock(&efw->mutex);
}
void snd_efw_stream_lock_changed(struct snd_efw *efw)
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index bda845afb470..29ccb3637164 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -337,6 +337,10 @@ void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw,
stop_stream(oxfw, stream);
}
+/*
+ * This function should be called before starting the stream or after stopping
+ * the streams.
+ */
void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw,
struct amdtp_stream *stream)
{
@@ -347,8 +351,6 @@ void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw,
else
conn = &oxfw->in_conn;
- stop_stream(oxfw, stream);
-
amdtp_stream_destroy(stream);
cmp_connection_destroy(conn);
}
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 60e5cad0531a..8c6ce019f437 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -104,11 +104,23 @@ end:
return err;
}
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
static void oxfw_card_free(struct snd_card *card)
{
struct snd_oxfw *oxfw = card->private_data;
unsigned int i;
+ snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
+ if (oxfw->has_output)
+ snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
+
+ fw_unit_put(oxfw->unit);
+
for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
kfree(oxfw->tx_stream_formats[i]);
kfree(oxfw->rx_stream_formats[i]);
@@ -136,7 +148,7 @@ static int oxfw_probe(struct fw_unit *unit,
oxfw = card->private_data;
oxfw->card = card;
mutex_init(&oxfw->mutex);
- oxfw->unit = unit;
+ oxfw->unit = fw_unit_get(unit);
oxfw->device_info = (const struct device_info *)id->driver_data;
spin_lock_init(&oxfw->lock);
init_waitqueue_head(&oxfw->hwdep_wait);
@@ -212,12 +224,7 @@ static void oxfw_remove(struct fw_unit *unit)
{
struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device);
- snd_card_disconnect(oxfw->card);
-
- snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
- if (oxfw->has_output)
- snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
-
+ /* No need to wait for releasing card object in this context. */
snd_card_free_when_closed(oxfw->card);
}
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index dfcb5e929f9f..a2ce773bdc62 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -961,7 +961,6 @@ static int azx_alloc_cmd_io(struct azx *chip)
dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n");
return err;
}
-EXPORT_SYMBOL_GPL(azx_alloc_cmd_io);
static void azx_init_cmd_io(struct azx *chip)
{
@@ -1026,7 +1025,6 @@ static void azx_init_cmd_io(struct azx *chip)
azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN);
spin_unlock_irq(&chip->reg_lock);
}
-EXPORT_SYMBOL_GPL(azx_init_cmd_io);
static void azx_free_cmd_io(struct azx *chip)
{
@@ -1036,7 +1034,6 @@ static void azx_free_cmd_io(struct azx *chip)
azx_writeb(chip, CORBCTL, 0);
spin_unlock_irq(&chip->reg_lock);
}
-EXPORT_SYMBOL_GPL(azx_free_cmd_io);
static unsigned int azx_command_addr(u32 cmd)
{
@@ -1316,7 +1313,6 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
else
return azx_corb_send_cmd(bus, val);
}
-EXPORT_SYMBOL_GPL(azx_send_cmd);
/* get a response */
static unsigned int azx_get_response(struct hda_bus *bus,
@@ -1330,7 +1326,6 @@ static unsigned int azx_get_response(struct hda_bus *bus,
else
return azx_rirb_get_response(bus, addr);
}
-EXPORT_SYMBOL_GPL(azx_get_response);
#ifdef CONFIG_SND_HDA_DSP_LOADER
/*
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 36d2f20db7a4..4ca3d5d02436 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1966,7 +1966,7 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
/* Panther Point */
{ PCI_DEVICE(0x8086, 0x1e20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
/* Lynx Point */
{ PCI_DEVICE(0x8086, 0x8c20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6d36c5b78805..87eff3173ce9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -79,6 +79,7 @@ enum {
STAC_ALIENWARE_M17X,
STAC_92HD89XX_HP_FRONT_JACK,
STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK,
+ STAC_92HD73XX_ASUS_MOBO,
STAC_92HD73XX_MODELS
};
@@ -1911,7 +1912,18 @@ static const struct hda_fixup stac92hd73xx_fixups[] = {
[STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK] = {
.type = HDA_FIXUP_PINS,
.v.pins = stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs,
- }
+ },
+ [STAC_92HD73XX_ASUS_MOBO] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ /* enable 5.1 and SPDIF out */
+ { 0x0c, 0x01014411 },
+ { 0x0d, 0x01014410 },
+ { 0x0e, 0x01014412 },
+ { 0x22, 0x014b1180 },
+ { }
+ }
+ },
};
static const struct hda_model_fixup stac92hd73xx_models[] = {
@@ -1923,6 +1935,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = {
{ .id = STAC_DELL_M6_BOTH, .name = "dell-m6" },
{ .id = STAC_DELL_EQ, .name = "dell-eq" },
{ .id = STAC_ALIENWARE_M17X, .name = "alienware" },
+ { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" },
{}
};
@@ -1975,6 +1988,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = {
"HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17,
"unknown HP", STAC_92HD89XX_HP_FRONT_JACK),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_ASUSTEK, 0x83f8, "ASUS AT4NM10",
+ STAC_92HD73XX_ASUS_MOBO),
{} /* terminator */
};
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f5ad214663f9..8de836165cf2 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -46,8 +46,6 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <asm/mach-types.h>
-
#include "../codecs/wm8731.h"
#include "atmel-pcm.h"
#include "atmel_ssc_dai.h"
@@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
int ret;
if (!np) {
- if (!(machine_is_at91sam9g20ek() ||
- machine_is_at91sam9g20ek_2mmc()))
- return -ENODEV;
+ return -ENODEV;
}
ret = atmel_ssc_set_audio(0);
@@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
/* Parse device node info */
- if (np) {
- ret = snd_soc_of_parse_card_name(card, "atmel,model");
- if (ret)
- goto err;
-
- ret = snd_soc_of_parse_audio_routing(card,
- "atmel,audio-routing");
- if (ret)
- goto err;
-
- /* Parse codec info */
- at91sam9g20ek_dai.codec_name = NULL;
- codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
- if (!codec_np) {
- dev_err(&pdev->dev, "codec info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.codec_of_node = codec_np;
-
- /* Parse dai and platform info */
- at91sam9g20ek_dai.cpu_dai_name = NULL;
- at91sam9g20ek_dai.platform_name = NULL;
- cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
- if (!cpu_np) {
- dev_err(&pdev->dev, "dai and pcm info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.cpu_of_node = cpu_np;
- at91sam9g20ek_dai.platform_of_node = cpu_np;
-
- of_node_put(codec_np);
- of_node_put(cpu_np);
+ ret = snd_soc_of_parse_card_name(card, "atmel,model");
+ if (ret)
+ goto err;
+
+ ret = snd_soc_of_parse_audio_routing(card,
+ "atmel,audio-routing");
+ if (ret)
+ goto err;
+
+ /* Parse codec info */
+ at91sam9g20ek_dai.codec_name = NULL;
+ codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "codec info missing\n");
+ return -EINVAL;
+ }
+ at91sam9g20ek_dai.codec_of_node = codec_np;
+
+ /* Parse dai and platform info */
+ at91sam9g20ek_dai.cpu_dai_name = NULL;
+ at91sam9g20ek_dai.platform_name = NULL;
+ cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "dai and pcm info missing\n");
+ return -EINVAL;
}
+ at91sam9g20ek_dai.cpu_of_node = cpu_np;
+ at91sam9g20ek_dai.platform_of_node = cpu_np;
+
+ of_node_put(codec_np);
+ of_node_put(cpu_np);
ret = snd_soc_register_card(card);
if (ret) {
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 7b7fbcd49e5e..c7cd60f009e9 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97
config SND_EP93XX_SOC_SNAPPERCL15
tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
- depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
+ depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C
select SND_EP93XX_SOC_I2S
select SND_SOC_TLV320AIC23_I2C
help
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 064e6c18e109..ea9f0e31f9d4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,7 +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_MAX98357A if GPIOLIB
select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 1806333ea29e..e9e6efbc21dd 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -12,9 +12,19 @@
* max98357a.c -- MAX98357A ALSA SoC Codec driver
*/
-#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/err.h>
#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/kernel.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
#define DRV_NAME "max98357a"
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index e1a4a45c57e2..fd102613d20d 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg)
case RT5670_ADC_EQ_CTRL1:
case RT5670_EQ_CTRL1:
case RT5670_ALC_CTRL_1:
- case RT5670_IRQ_CTRL1:
case RT5670_IRQ_CTRL2:
case RT5670_INT_IRQ_ST:
case RT5670_IL_CMD:
@@ -2703,6 +2702,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
regmap_write(rt5670->regmap, RT5670_RESET, 0);
+ regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val);
+ if (val >= 4)
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980);
+ else
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00);
+
ret = regmap_register_patch(rt5670->regmap, init_list,
ARRAY_SIZE(init_list));
if (ret != 0)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5d0bb8748dd1..fb9c20eace3f 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB45 Bypass Mux", "Bypass", "IB45 Mux" },
{ "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" },
- { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "IB6 Mux", "SLB DAC 6", "SLB DAC6" },
{ "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" },
{ "IB6 Mux", "IF4 DAC L", "IF4 DAC L" },
@@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" },
{ "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" },
- { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "IB7 Mux", "SLB DAC 7", "SLB DAC7" },
{ "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" },
{ "IB7 Mux", "IF4 DAC R", "IF4 DAC R" },
@@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC1 FS", NULL, "DAC1 MIXL" },
{ "DAC1 FS", NULL, "DAC1 MIXR" },
- { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" },
- { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" },
+ { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" },
+ { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" },
{ "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" },
{ "DAC2 L Mux", "OB 2", "OutBound2" },
- { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" },
- { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" },
+ { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" },
+ { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" },
{ "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" },
@@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC2 R Mux", "Haptic Generator", "Haptic Generator" },
{ "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" },
- { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" },
- { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" },
+ { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" },
+ { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" },
{ "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" },
{ "DAC3 L Mux", "OB 4", "OutBound4" },
- { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" },
- { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" },
+ { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" },
+ { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" },
{ "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" },
{ "DAC3 R Mux", "OB 5", "OutBound5" },
- { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" },
{ "DAC4 L Mux", "OB 6", "OutBound6" },
- { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" },
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2595611e8a6d..b9fabbf69db6 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
factor = (div2 + 1) * (7 * psr + 1) * 2;
for (i = 0; i < 255; i++) {
- /* The bclk rate must be smaller than 1/5 sysclk rate */
- if (factor * (i + 1) < 5)
- continue;
-
tmprate = freq * factor * (i + 2);
if (baudclk_is_used)
@@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+ continue;
+
clkrate /= factor;
afreq = clkrate / (i + 1);
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index dfebfdd5eb2a..daecc58f28af 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -150,7 +150,7 @@ enum sst_cmd_type {
enum sst_task {
SST_TASK_SBA = 1,
- SST_TASK_MMX,
+ SST_TASK_MMX = 3,
};
enum sst_type {
diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 8a8d56a146e7..11c578651c1c 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx,
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
- shim_regs->imrx = sst_shim_read64(shim, SST_IMRX),
+ shim_regs->imrx = sst_shim_read64(shim, SST_IMRX);
+ shim_regs->csr = sst_shim_read64(shim, SST_CSR);
+
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
@@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx,
*/
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
sst_shim_write64(shim, SST_IMRX, shim_regs->imrx),
+ sst_shim_write64(shim, SST_CSR, shim_regs->csr),
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
@@ -379,6 +382,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx)
* initially active. So change the state to active before
* enabling the pm
*/
+
+ if (!acpi_disabled)
+ pm_runtime_set_active(ctx->dev);
+
pm_runtime_enable(ctx->dev);
if (acpi_disabled)
@@ -409,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev)
synchronize_irq(ctx->irq_num);
flush_workqueue(ctx->post_msg_wq);
+ ctx->ops->reset(ctx);
/* save the shim registers because PMC doesn't save state */
sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64);
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index ccfb41c22e53..f7eb42aa3f38 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
return ret;
card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
card->name = devm_kasprintf(dev, GFP_KERNEL,
"HDMI %s", dev_name(ad->dssdev));
card->owner = THIS_MODULE;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index c7eb9dd67f60..fd99d89de6a8 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case OMAP_MCBSP_SYSCLK_CLKX_EXT:
regs->srgr2 |= CLKSM;
+ regs->pcr0 |= SCLKME;
+ /*
+ * If McBSP is master but yet the CLKX/CLKR pin drives the SRG,
+ * disable output on those pins. This enables to inject the
+ * reference clock through CLKX/CLKR. For this to work
+ * set_dai_sysclk() _needs_ to be called after set_dai_fmt().
+ */
+ regs->pcr0 &= ~CLKXM;
+ break;
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
+ /* Disable ouput on CLKR pin in master mode */
+ regs->pcr0 &= ~CLKRM;
break;
default:
err = -ENODEV;
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index f4b05bc23e4b..1343ecbf0bd5 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64));
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
if (ret)
return ret;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 3cebf6ca03df..0632a36852c8 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM
config SND_SOC_SPEYSIDE
tristate "Audio support for Wolfson Speyside"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
@@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY
config SND_SOC_BELLS
tristate "Audio support for Wolfson Bells"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM5102
select SND_SOC_WM5110
@@ -206,7 +206,7 @@ config SND_SOC_LOWLAND
config SND_SOC_LITTLEMILL
tristate "Audio support for Wolfson Littlemill"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C
select SND_SAMSUNG_I2S
select MFD_WM8994
select SND_SOC_WM8994
@@ -223,7 +223,7 @@ config SND_SOC_SNOW
config SND_SOC_ODROIDX2
tristate "Audio support for Odroid-X2 and Odroid-U3"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SOC_MAX98090
select SND_SAMSUNG_I2S
help
@@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2
config SND_SOC_ARNDALE_RT5631_ALC5631
tristate "Audio support for RT5631(ALC5631) on Arndale Board"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SAMSUNG_I2S
select SND_SOC_RT5631
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 1b53605f7154..110577c52317 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1252,6 +1252,8 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_probe;
}
+ dev_set_drvdata(dev, priv);
+
/*
* asoc register
*/
@@ -1268,8 +1270,6 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_soc;
}
- dev_set_drvdata(dev, priv);
-
pm_runtime_enable(dev);
dev_info(dev, "probed\n");