Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (41 commits)
ALSA: hda - Identify more variants for ALC269
ALSA: hda - Fix wrong ALC269 variant check
ALSA: hda - Enable jack sense for Thinkpad Edge 11
ALSA: Revert "ALSA: hda - Fix switching between dmic and mic using the same mux on IDT/STAC"
ALSA: hda - Fixed ALC887-VD initial error
ALSA: atmel - Fix the return value in error path
ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J
ALSA: snd-atmel-abdac: test wrong variable
ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer
ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup
ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of pci_get_drvdata
ALSA: sound/core/pcm_lib.c: Remove unnecessary semicolons
ALSA: sound/ppc: Use printf extension %pR for struct resource
ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and Headphone controls
ASoC: uda134x - set reg_cache_default to uda134x_reg
ASoC: Add support for MAX98089 CODEC
ASoC: davinci: fixes for multi-component
ASoC: Fix register cache setup WM8994 for multi-component
ASoC: Fix dapm_seq_compare() for multi-component
ASoC: RX1950: Fix hw_params function
...
diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c
index 9be261b..2652af1 100644
--- a/arch/arm/mach-davinci/dm355.c
+++ b/arch/arm/mach-davinci/dm355.c
@@ -359,8 +359,8 @@
CLK(NULL, "uart1", &uart1_clk),
CLK(NULL, "uart2", &uart2_clk),
CLK("i2c_davinci.1", NULL, &i2c_clk),
- CLK("davinci-asp.0", NULL, &asp0_clk),
- CLK("davinci-asp.1", NULL, &asp1_clk),
+ CLK("davinci-mcbsp.0", NULL, &asp0_clk),
+ CLK("davinci-mcbsp.1", NULL, &asp1_clk),
CLK("davinci_mmc.0", NULL, &mmcsd0_clk),
CLK("davinci_mmc.1", NULL, &mmcsd1_clk),
CLK("spi_davinci.0", NULL, &spi0_clk),
@@ -664,7 +664,7 @@
};
static struct platform_device dm355_asp1_device = {
- .name = "davinci-asp",
+ .name = "davinci-mcbsp",
.id = 1,
.num_resources = ARRAY_SIZE(dm355_asp1_resources),
.resource = dm355_asp1_resources,
diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c
index a12065e..c466d71 100644
--- a/arch/arm/mach-davinci/dm365.c
+++ b/arch/arm/mach-davinci/dm365.c
@@ -459,7 +459,7 @@
CLK(NULL, "usb", &usb_clk),
CLK("davinci_emac.1", NULL, &emac_clk),
CLK("davinci_voicecodec", NULL, &voicecodec_clk),
- CLK("davinci-asp.0", NULL, &asp0_clk),
+ CLK("davinci-mcbsp", NULL, &asp0_clk),
CLK(NULL, "rto", &rto_clk),
CLK(NULL, "mjcp", &mjcp_clk),
CLK(NULL, NULL, NULL),
@@ -922,8 +922,8 @@
};
static struct platform_device dm365_asp_device = {
- .name = "davinci-asp",
- .id = 0,
+ .name = "davinci-mcbsp",
+ .id = -1,
.num_resources = ARRAY_SIZE(dm365_asp_resources),
.resource = dm365_asp_resources,
};
diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c
index 0608dd7..9a2376b 100644
--- a/arch/arm/mach-davinci/dm644x.c
+++ b/arch/arm/mach-davinci/dm644x.c
@@ -302,7 +302,7 @@
CLK("davinci_emac.1", NULL, &emac_clk),
CLK("i2c_davinci.1", NULL, &i2c_clk),
CLK("palm_bk3710", NULL, &ide_clk),
- CLK("davinci-asp", NULL, &asp_clk),
+ CLK("davinci-mcbsp", NULL, &asp_clk),
CLK("davinci_mmc.0", NULL, &mmcsd_clk),
CLK(NULL, "spi", &spi_clk),
CLK(NULL, "gpio", &gpio_clk),
@@ -580,7 +580,7 @@
};
static struct platform_device dm644x_asp_device = {
- .name = "davinci-asp",
+ .name = "davinci-mcbsp",
.id = -1,
.num_resources = ARRAY_SIZE(dm644x_asp_resources),
.resource = dm644x_asp_resources,
diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index a95141e..bd581c6 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -522,9 +522,6 @@
#define WM8350_MCLK_SEL_PLL_32K 3
#define WM8350_MCLK_SEL_MCLK 5
-#define WM8350_MCLK_DIR_OUT 0
-#define WM8350_MCLK_DIR_IN 1
-
/* clock divider id's */
#define WM8350_ADC_CLKDIV 0
#define WM8350_DAC_CLKDIV 1
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index f2f41c8..6e24091 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -420,9 +420,9 @@
return PTR_ERR(pclk);
}
sample_clk = clk_get(&pdev->dev, "sample_clk");
- if (IS_ERR(pclk)) {
+ if (IS_ERR(sample_clk)) {
dev_dbg(&pdev->dev, "no sample clock\n");
- retval = PTR_ERR(pclk);
+ retval = PTR_ERR(sample_clk);
goto out_put_pclk;
}
clk_enable(pclk);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a1707cc..b75db8e 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -223,7 +223,7 @@
entry->jiffies = jiffies;
entry->pos = pos;
entry->period_size = runtime->period_size;
- entry->buffer_size = runtime->buffer_size;;
+ entry->buffer_size = runtime->buffer_size;
entry->old_hw_ptr = runtime->status->hw_ptr;
entry->hw_ptr_base = runtime->hw_ptr_base;
log->idx = (log->idx + 1) % XRUN_LOG_CNT;
diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c
index 727bdb9..d8cf3e5 100644
--- a/sound/oss/dev_table.c
+++ b/sound/oss/dev_table.c
@@ -71,7 +71,7 @@
if (sound_nblocks >= MAX_MEM_BLOCKS)
sound_nblocks = MAX_MEM_BLOCKS - 1;
- op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations)));
+ op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct audio_operations)));
sound_nblocks++;
if (sound_nblocks >= MAX_MEM_BLOCKS)
sound_nblocks = MAX_MEM_BLOCKS - 1;
@@ -81,7 +81,6 @@
sound_unload_audiodev(num);
return -(ENOMEM);
}
- memset((char *) op, 0, sizeof(struct audio_operations));
init_waitqueue_head(&op->in_sleeper);
init_waitqueue_head(&op->out_sleeper);
init_waitqueue_head(&op->poll_sleeper);
@@ -128,7 +127,7 @@
/* FIXME: This leaks a mixer_operations struct every time its called
until you unload sound! */
- op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations)));
+ op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct mixer_operations)));
sound_nblocks++;
if (sound_nblocks >= MAX_MEM_BLOCKS)
sound_nblocks = MAX_MEM_BLOCKS - 1;
@@ -137,7 +136,6 @@
printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
return -ENOMEM;
}
- memset((char *) op, 0, sizeof(struct mixer_operations));
memcpy((char *) op, (char *) driver, driver_size);
strlcpy(op->name, name, sizeof(op->name));
diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
index 782b3b8..ceedb1e 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -178,7 +178,7 @@
return err;
parms[dev].prech_timeout = MAX_SCHEDULE_TIMEOUT;
- midi_in_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+ midi_in_buf[dev] = vmalloc(sizeof(struct midi_buf));
if (midi_in_buf[dev] == NULL)
{
@@ -188,7 +188,7 @@
}
midi_in_buf[dev]->len = midi_in_buf[dev]->head = midi_in_buf[dev]->tail = 0;
- midi_out_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+ midi_out_buf[dev] = vmalloc(sizeof(struct midi_buf));
if (midi_out_buf[dev] == NULL)
{
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index e19dd5d..9b800ce 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -859,7 +859,7 @@
return 0;
case SNDCTL_COPR_LOAD:
- buf = (copr_buffer *) vmalloc(sizeof(copr_buffer));
+ buf = vmalloc(sizeof(copr_buffer));
if (buf == NULL)
return -ENOSPC;
if (copy_from_user(buf, arg, sizeof(copr_buffer))) {
@@ -871,7 +871,7 @@
return err;
case SNDCTL_COPR_SENDMSG:
- mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+ mbuf = vmalloc(sizeof(copr_msg));
if (mbuf == NULL)
return -ENOSPC;
if (copy_from_user(mbuf, arg, sizeof(copr_msg))) {
@@ -895,7 +895,7 @@
case SNDCTL_COPR_RCVMSG:
err = 0;
- mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+ mbuf = vmalloc(sizeof(copr_msg));
if (mbuf == NULL)
return -ENOSPC;
data = (unsigned short *)mbuf->data;
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index e85789e..5ea1098 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -1646,13 +1646,13 @@
{
if (sequencer_ok)
return;
- queue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * EV_SZ);
+ queue = vmalloc(SEQ_MAX_QUEUE * EV_SZ);
if (queue == NULL)
{
printk(KERN_ERR "sequencer: Can't allocate memory for sequencer output queue\n");
return;
}
- iqueue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * IEV_SZ);
+ iqueue = vmalloc(SEQ_MAX_QUEUE * IEV_SZ);
if (iqueue == NULL)
{
printk(KERN_ERR "sequencer: Can't allocate memory for sequencer input queue\n");
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 62895a7..22dbd91 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -435,7 +435,7 @@
struct hpi_message hm;
struct hpi_response hr;
struct hpi_adapter *pa;
- pa = (struct hpi_adapter *)pci_get_drvdata(pci_dev);
+ pa = pci_get_drvdata(pci_dev);
hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM,
HPI_SUBSYS_DELETE_ADAPTER);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 4679ed8..2f3cacb 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1129,10 +1129,11 @@
count_areas = size/2;
addr_area2 = addr+count_areas;
- count_areas--; /* max. index */
snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n",
addr, count_areas, addr_area2, count_areas);
+ count_areas--; /* max. index */
+
/* build combined I/O buffer length word */
lengths = (count_areas << 16) | (count_areas);
spin_lock_irqsave(&chip->reg_lock, flags);
@@ -1740,11 +1741,15 @@
.rate_max = AZF_FREQ_66200,
.channels_min = 1,
.channels_max = 2,
- .buffer_bytes_max = 65536,
- .period_bytes_min = 64,
- .period_bytes_max = 65536,
- .periods_min = 1,
- .periods_max = 1024,
+ .buffer_bytes_max = (64*1024),
+ .period_bytes_min = 1024,
+ .period_bytes_max = (32*1024),
+ /* We simply have two DMA areas (instead of a list of descriptors
+ such as other cards); I believe that this is a fixed hardware
+ attribute and there isn't much driver magic to be done to expand it.
+ Thus indicate that we have at least and at most 2 periods. */
+ .periods_min = 2,
+ .periods_max = 2,
/* FIXME: maybe that card actually has a FIFO?
* Hmm, it seems newer revisions do have one, but we still don't know
* its size... */
@@ -1980,8 +1985,13 @@
chip = snd_timer_chip(timer);
spin_lock_irqsave(&chip->reg_lock, flags);
/* disable timer countdown and interrupt */
- /* FIXME: should we write TIMER_IRQ_ACK here? */
- snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0);
+ /* Hmm, should we write TIMER_IRQ_ACK here?
+ YES indeed, otherwise a rogue timer operation - which prompts
+ ALSA(?) to call repeated stop() in vain, but NOT start() -
+ will never end (value 0x03 is kept shown in control byte).
+ Simply manually poking 0x04 _once_ immediately successfully stops
+ the hardware/ALSA interrupt activity. */
+ snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x04);
spin_unlock_irqrestore(&chip->reg_lock, flags);
snd_azf3328_dbgcallleave();
return 0;
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 85ab43e..457d211 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -129,8 +129,6 @@
apcm->substream = substream;
apcm->interrupt = ct_atc_pcm_interrupt;
- runtime->private_data = apcm;
- runtime->private_free = ct_atc_pcm_free_substream;
if (IEC958 == substream->pcm->device) {
runtime->hw = ct_spdif_passthru_playback_hw;
atc->spdif_out_passthru(atc, 1);
@@ -155,8 +153,12 @@
}
apcm->timer = ct_timer_instance_new(atc->timer, apcm);
- if (!apcm->timer)
+ if (!apcm->timer) {
+ kfree(apcm);
return -ENOMEM;
+ }
+ runtime->private_data = apcm;
+ runtime->private_free = ct_atc_pcm_free_substream;
return 0;
}
@@ -278,8 +280,6 @@
apcm->started = 0;
apcm->substream = substream;
apcm->interrupt = ct_atc_pcm_interrupt;
- runtime->private_data = apcm;
- runtime->private_free = ct_atc_pcm_free_substream;
runtime->hw = ct_pcm_capture_hw;
runtime->hw.rate_max = atc->rsr * atc->msr;
@@ -298,8 +298,12 @@
}
apcm->timer = ct_timer_instance_new(atc->timer, apcm);
- if (!apcm->timer)
+ if (!apcm->timer) {
+ kfree(apcm);
return -ENOMEM;
+ }
+ runtime->private_data = apcm;
+ runtime->private_free = ct_atc_pcm_free_substream;
return 0;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6361f75..846d1ea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3100,6 +3100,7 @@
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3110,6 +3111,7 @@
SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5f00589..0ac6aed 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14623,7 +14623,10 @@
/* different alc269-variants */
enum {
ALC269_TYPE_NORMAL,
+ ALC269_TYPE_ALC258,
ALC269_TYPE_ALC259,
+ ALC269_TYPE_ALC269VB,
+ ALC269_TYPE_ALC270,
ALC269_TYPE_ALC271X,
};
@@ -15023,7 +15026,7 @@
static int patch_alc269(struct hda_codec *codec)
{
struct alc_spec *spec;
- int board_config;
+ int board_config, coef;
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -15034,14 +15037,23 @@
alc_auto_parse_customize_define(codec);
- if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){
+ coef = alc_read_coef_idx(codec, 0);
+ if ((coef & 0x00f0) == 0x0010) {
if (codec->bus->pci->subsystem_vendor == 0x1025 &&
spec->cdefine.platform_type == 1) {
alc_codec_rename(codec, "ALC271X");
spec->codec_variant = ALC269_TYPE_ALC271X;
- } else {
+ } else if ((coef & 0xf000) == 0x1000) {
+ spec->codec_variant = ALC269_TYPE_ALC270;
+ } else if ((coef & 0xf000) == 0x2000) {
alc_codec_rename(codec, "ALC259");
spec->codec_variant = ALC269_TYPE_ALC259;
+ } else if ((coef & 0xf000) == 0x3000) {
+ alc_codec_rename(codec, "ALC258");
+ spec->codec_variant = ALC269_TYPE_ALC258;
+ } else {
+ alc_codec_rename(codec, "ALC269VB");
+ spec->codec_variant = ALC269_TYPE_ALC269VB;
}
} else
alc_fix_pll_init(codec, 0x20, 0x04, 15);
@@ -15104,7 +15116,7 @@
spec->stream_digital_capture = &alc269_pcm_digital_capture;
if (!spec->adc_nids) { /* wasn't filled automatically? use default */
- if (spec->codec_variant != ALC269_TYPE_NORMAL) {
+ if (spec->codec_variant == ALC269_TYPE_NORMAL) {
spec->adc_nids = alc269_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
spec->capsrc_nids = alc269_capsrc_nids;
@@ -19298,6 +19310,7 @@
static struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
+ SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
{}
@@ -19419,7 +19432,10 @@
{
if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){
kfree(codec->chip_name);
- codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
+ if (codec->vendor_id == 0x10ec0887)
+ codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL);
+ else
+ codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
if (!codec->chip_name) {
alc_free(codec);
return -ENOMEM;
@@ -19909,7 +19925,7 @@
{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
.patch = patch_alc882 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
- { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
+ { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
.patch = patch_alc882 },
{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 93fa59c..5c71080 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -389,6 +389,11 @@
0x11, 0x20, 0
};
+#define STAC92HD87B_NUM_DMICS 1
+static hda_nid_t stac92hd87b_dmic_nids[STAC92HD87B_NUM_DMICS + 1] = {
+ 0x11, 0
+};
+
#define STAC92HD83XXX_NUM_CAPS 2
static unsigned long stac92hd83xxx_capvols[] = {
HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
@@ -3486,10 +3491,8 @@
return err;
}
- if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) {
+ if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1)
snd_hda_add_imux_item(imux, label, index, NULL);
- spec->num_analog_muxes++;
- }
}
return 0;
@@ -5452,12 +5455,17 @@
stac92hd83xxx_brd_tbl[spec->board_config]);
switch (codec->vendor_id) {
+ case 0x111d76d1:
+ case 0x111d76d9:
+ spec->dmic_nids = stac92hd87b_dmic_nids;
+ spec->num_dmics = stac92xx_connected_ports(codec,
+ stac92hd87b_dmic_nids,
+ STAC92HD87B_NUM_DMICS);
+ /* Fall through */
case 0x111d7666:
case 0x111d7667:
case 0x111d7668:
case 0x111d7669:
- case 0x111d76d1:
- case 0x111d76d9:
spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids);
spec->pin_nids = stac92hd88xxx_pin_nids;
spec->mono_nid = 0;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 400f9eb..629a549 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1866,6 +1866,12 @@
},
{
.subvendor = 0x1028,
+ .subdevice = 0x0182,
+ .name = "Dell Latitude D610", /* STAC9750/51 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
.subdevice = 0x0186,
.name = "Dell Latitude D810", /* cf. Malone #41015 */
.type = AC97_TUNE_HP_MUTE_LED
diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h
index a46f508..812e288 100644
--- a/sound/pci/mixart/mixart_hwdep.h
+++ b/sound/pci/mixart/mixart_hwdep.h
@@ -25,11 +25,21 @@
#include <sound/hwdep.h>
+#ifndef readl_be
#define readl_be(x) be32_to_cpu(__raw_readl(x))
-#define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr)
+#endif
+#ifndef writel_be
+#define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr)
+#endif
+
+#ifndef readl_le
#define readl_le(x) le32_to_cpu(__raw_readl(x))
+#endif
+
+#ifndef writel_le
#define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr)
+#endif
#define MIXART_MEM(mgr,x) ((mgr)->mem[0].virt + (x))
#define MIXART_REG(mgr,x) ((mgr)->mem[1].virt + (x))
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index 8508117..b47cfd4 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -1228,10 +1228,8 @@
chip->rsrc[i].start + 1,
rnames[i]) == NULL) {
printk(KERN_ERR "snd: can't request rsrc "
- " %d (%s: 0x%016llx:%016llx)\n",
- i, rnames[i],
- (unsigned long long)chip->rsrc[i].start,
- (unsigned long long)chip->rsrc[i].end);
+ " %d (%s: %pR)\n",
+ i, rnames[i], &chip->rsrc[i]);
err = -ENODEV;
goto __error;
}
@@ -1256,10 +1254,8 @@
chip->rsrc[i].start + 1,
rnames[i]) == NULL) {
printk(KERN_ERR "snd: can't request rsrc "
- " %d (%s: 0x%016llx:%016llx)\n",
- i, rnames[i],
- (unsigned long long)chip->rsrc[i].start,
- (unsigned long long)chip->rsrc[i].end);
+ " %d (%s: %pR)\n",
+ i, rnames[i], &chip->rsrc[i]);
err = -ENODEV;
goto __error;
}
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index e720d5e..bee3c94 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -16,7 +16,8 @@
config SND_AT91_SOC_SAM9G20_WM8731
tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
- depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+ depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC && \
+ AT91_PROGRAMMABLE_CLOCKS
select SND_ATMEL_SOC_SSC
select SND_SOC_WM8731
help
@@ -25,7 +26,7 @@
config SND_AT32_SOC_PLAYPAQ
tristate "SoC Audio support for PlayPaq with WM8510"
- depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+ depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
select SND_ATMEL_SOC_SSC
select SND_SOC_WM8510
help
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index bc22ee9..470cb93 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -28,6 +28,11 @@
#include <sound/max98088.h>
#include "max98088.h"
+enum max98088_type {
+ MAX98088,
+ MAX98089,
+};
+
struct max98088_cdata {
unsigned int rate;
unsigned int fmt;
@@ -36,6 +41,7 @@
struct max98088_priv {
u8 reg_cache[M98088_REG_CNT];
+ enum max98088_type devtype;
void *control_data;
struct max98088_pdata *pdata;
unsigned int sysclk;
@@ -2040,6 +2046,8 @@
if (max98088 == NULL)
return -ENOMEM;
+ max98088->devtype = id->driver_data;
+
i2c_set_clientdata(i2c, max98088);
max98088->control_data = i2c;
max98088->pdata = i2c->dev.platform_data;
@@ -2059,7 +2067,8 @@
}
static const struct i2c_device_id max98088_i2c_id[] = {
- { "max98088", 0 },
+ { "max98088", MAX98088 },
+ { "max98089", MAX98089 },
{ }
};
MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 7540a50..464f0cf 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -597,6 +597,7 @@
.resume = uda134x_soc_resume,
.reg_cache_size = sizeof(uda134x_reg),
.reg_word_size = sizeof(u8),
+ .reg_cache_default = uda134x_reg,
.reg_cache_step = 1,
.read = uda134x_read_reg_cache,
.write = uda134x_write,
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f4f1fba..7611add 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -831,7 +831,7 @@
}
/* MCLK direction */
- if (dir == WM8350_MCLK_DIR_OUT)
+ if (dir == SND_SOC_CLOCK_OUT)
wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
WM8350_MCLK_DIR);
else
@@ -1586,6 +1586,13 @@
wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+ /* Make sure AIF tristating is disabled by default */
+ wm8350_clear_bits(wm8350, WM8350_AI_FORMATING, WM8350_AIF_TRI);
+
+ /* Make sure we've got a sane companding setup too */
+ wm8350_clear_bits(wm8350, WM8350_ADC_DAC_COMP,
+ WM8350_DAC_COMP | WM8350_LOOPBACK);
+
/* Make sure jack detect is disabled to start off with */
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
WM8350_JDL_ENA | WM8350_JDR_ENA);
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 04182c4..0132a27 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -34,7 +34,6 @@
/* codec private data */
struct wm8776_priv {
enum snd_soc_control_type control_type;
- u16 reg_cache[WM8776_CACHEREGNUM];
int sysclk[2];
};
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 894d0cd..e809274 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3500,8 +3500,11 @@
{
struct wm8962_priv *wm8962 = dev_get_drvdata(dev);
long int time;
+ int ret;
- strict_strtol(buf, 10, &time);
+ ret = strict_strtol(buf, 10, &time);
+ if (ret != 0)
+ return ret;
input_event(wm8962->beep, EV_SND, SND_TONE, time);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0db59c3..830dfdd66 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3903,6 +3903,8 @@
return -ENOMEM;
snd_soc_codec_set_drvdata(codec, wm8994);
+ codec->reg_cache = &wm8994->reg_cache;
+
wm8994->pdata = dev_get_platdata(codec->dev->parent);
wm8994->codec = codec;
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 2b07b17..bc9e6b0b 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -157,12 +157,23 @@
}
/* davinci-evm digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link evm_dai = {
+static struct snd_soc_dai_link dm6446_evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
- .cpu_dai_name = "davinci-mcasp.0",
+ .cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "tlv320aic3x-hifi",
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-001b",
+ .platform_name = "davinci-pcm-audio",
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link dm355_evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai_name = "davinci-mcbsp.1",
+ .codec_dai_name = "tlv320aic3x-hifi",
+ .codec_name = "tlv320aic3x-codec.1-001b",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
@@ -172,10 +183,10 @@
#ifdef CONFIG_SND_DM365_AIC3X_CODEC
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
- .cpu_dai_name = "davinci-i2s",
+ .cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "tlv320aic3x-hifi",
.init = evm_aic3x_init,
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-0018",
.ops = &evm_ops,
#elif defined(CONFIG_SND_DM365_VOICE_CODEC)
.name = "Voice Codec - CQ93VC",
@@ -219,10 +230,17 @@
.ops = &evm_ops,
};
-/* davinci dm6446, dm355 evm audio machine driver */
-static struct snd_soc_card snd_soc_card_evm = {
- .name = "DaVinci EVM",
- .dai_link = &evm_dai,
+/* davinci dm6446 evm audio machine driver */
+static struct snd_soc_card dm6446_snd_soc_card_evm = {
+ .name = "DaVinci DM6446 EVM",
+ .dai_link = &dm6446_evm_dai,
+ .num_links = 1,
+};
+
+/* davinci dm355 evm audio machine driver */
+static struct snd_soc_card dm355_snd_soc_card_evm = {
+ .name = "DaVinci DM355 EVM",
+ .dai_link = &dm355_evm_dai,
.num_links = 1,
};
@@ -261,10 +279,10 @@
int ret;
if (machine_is_davinci_evm()) {
- evm_snd_dev_data = &snd_soc_card_evm;
+ evm_snd_dev_data = &dm6446_snd_soc_card_evm;
index = 0;
} else if (machine_is_davinci_dm355_evm()) {
- evm_snd_dev_data = &snd_soc_card_evm;
+ evm_snd_dev_data = &dm355_snd_soc_card_evm;
index = 1;
} else if (machine_is_davinci_dm365_evm()) {
evm_snd_dev_data = &dm365_snd_soc_card_evm;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index d46b545..9e0e565 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -426,9 +426,6 @@
snd_pcm_format_t fmt;
unsigned element_cnt = 1;
- dai->capture_dma_data = dev->dma_params;
- dai->playback_dma_data = dev->dma_params;
-
/* general line settings */
spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -601,6 +598,15 @@
return ret;
}
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+ return 0;
+}
+
static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -612,6 +618,7 @@
#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+ .startup = davinci_i2s_startup,
.shutdown = davinci_i2s_shutdown,
.prepare = davinci_i2s_prepare,
.trigger = davinci_i2s_trigger,
@@ -749,7 +756,7 @@
.probe = davinci_i2s_probe,
.remove = davinci_i2s_remove,
.driver = {
- .name = "davinci-i2s",
+ .name = "davinci-mcbsp",
.owner = THIS_MODULE,
},
};
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 86918ee..fb55d2c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -715,9 +715,6 @@
int word_length;
u8 fifo_level;
- cpu_dai->capture_dma_data = dev->dma_params;
- cpu_dai->playback_dma_data = dev->dma_params;
-
davinci_hw_common_param(dev, substream->stream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
fifo_level = dev->txnumevt;
@@ -799,7 +796,17 @@
return ret;
}
+static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+ return 0;
+}
+
static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+ .startup = davinci_mcasp_startup,
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 009b652..6c6666a 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -84,7 +84,7 @@
static struct snd_soc_dai_link sffsdr_dai = {
.name = "PCM3008", /* Codec name */
.stream_name = "PCM3008 HiFi",
- .cpu_dai_name = "davinci-asp.0",
+ .cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "pcm3008-hifi",
.codec_name = "pcm3008-codec",
.platform_name = "davinci-pcm-audio",
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index ea232f6..fb4cc1e 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -97,9 +97,6 @@
&davinci_vcif_dev->dma_params[substream->stream];
u32 w;
- dai->capture_dma_data = davinci_vcif_dev->dma_params;
- dai->playback_dma_data = davinci_vcif_dev->dma_params;
-
/* Restart the codec before setup */
davinci_vcif_stop(substream);
davinci_vcif_start(substream);
@@ -174,9 +171,19 @@
return ret;
}
+static int davinci_vcif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+ return 0;
+}
+
#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000
static struct snd_soc_dai_ops davinci_vcif_dai_ops = {
+ .startup = davinci_vcif_startup,
.trigger = davinci_vcif_trigger,
.hw_params = davinci_vcif_hw_params,
};
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 74ffed4..9018fa5 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -160,7 +160,7 @@
rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
if (rc != 0) {
pr_err("Failed to register DAI\n");
- return 0;
+ return rc;
}
psc_dma = dev_get_drvdata(&op->dev);
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index b596752..dd4fffd 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -34,8 +34,8 @@
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret;
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
@@ -79,10 +79,10 @@
static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
- .codec_dai = "tlv320aic23-hifi",
+ .codec_dai_name = "tlv320aic23-hifi",
.platform_name = "imx-pcm-audio.0",
.codec_name = "tlv320aic23-codec.0-001a",
- .cpu_dai = "imx-ssi.0",
+ .cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
};
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index fd493ee..671ef8d 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -20,6 +20,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
+#include <linux/dmaengine.h>
#include <sound/core.h>
#include <sound/initval.h>
@@ -27,165 +28,146 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <mach/dma-mx1-mx2.h>
+#include <mach/dma.h>
#include "imx-ssi.h"
struct imx_pcm_runtime_data {
- int sg_count;
- struct scatterlist *sg_list;
- int period;
+ int period_bytes;
int periods;
- unsigned long dma_addr;
int dma;
- struct snd_pcm_substream *substream;
unsigned long offset;
unsigned long size;
- unsigned long period_cnt;
void *buf;
int period_time;
+ struct dma_async_tx_descriptor *desc;
+ struct dma_chan *dma_chan;
+ struct imx_dma_data dma_data;
};
-/* Called by the DMA framework when a period has elapsed */
-static void imx_ssi_dma_progression(int channel, void *data,
- struct scatterlist *sg)
+static void audio_dma_irq(void *data)
{
- struct snd_pcm_substream *substream = data;
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- if (!sg)
- return;
+ iprtd->offset += iprtd->period_bytes;
+ iprtd->offset %= iprtd->period_bytes * iprtd->periods;
- runtime = iprtd->substream->runtime;
-
- iprtd->offset = sg->dma_address - runtime->dma_addr;
-
- snd_pcm_period_elapsed(iprtd->substream);
+ snd_pcm_period_elapsed(substream);
}
-static void imx_ssi_dma_callback(int channel, void *data)
+static bool filter(struct dma_chan *chan, void *param)
{
- pr_err("%s shouldn't be called\n", __func__);
+ struct imx_pcm_runtime_data *iprtd = param;
+
+ if (!imx_dma_is_general_purpose(chan))
+ return false;
+
+ chan->private = &iprtd->dma_data;
+
+ return true;
}
-static void snd_imx_dma_err_callback(int channel, void *data, int err)
-{
- struct snd_pcm_substream *substream = data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct imx_pcm_dma_params *dma_params =
- snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- int ret;
-
- pr_err("DMA timeout on channel %d -%s%s%s%s\n",
- channel,
- err & IMX_DMA_ERR_BURST ? " burst" : "",
- err & IMX_DMA_ERR_REQUEST ? " request" : "",
- err & IMX_DMA_ERR_TRANSFER ? " transfer" : "",
- err & IMX_DMA_ERR_BUFFER ? " buffer" : "");
-
- imx_dma_disable(iprtd->dma);
- ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
- IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
- substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- DMA_MODE_WRITE : DMA_MODE_READ);
- if (!ret)
- imx_dma_enable(iprtd->dma);
-}
-
-static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
+static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct imx_pcm_dma_params *dma_params;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+ struct dma_slave_config slave_config;
+ dma_cap_mask_t mask;
+ enum dma_slave_buswidth buswidth;
int ret;
dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
- if (iprtd->dma < 0) {
- pr_err("Failed to claim the audio DMA\n");
- return -ENODEV;
+ iprtd->dma_data.peripheral_type = IMX_DMATYPE_SSI;
+ iprtd->dma_data.priority = DMA_PRIO_HIGH;
+ iprtd->dma_data.dma_request = dma_params->dma;
+
+ /* Try to grab a DMA channel */
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+ if (!iprtd->dma_chan)
+ return -EINVAL;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ case SNDRV_PCM_FORMAT_S24_LE:
+ buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ break;
+ default:
+ return 0;
}
- ret = imx_dma_setup_handlers(iprtd->dma,
- imx_ssi_dma_callback,
- snd_imx_dma_err_callback, substream);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ slave_config.direction = DMA_TO_DEVICE;
+ slave_config.dst_addr = dma_params->dma_addr;
+ slave_config.dst_addr_width = buswidth;
+ slave_config.dst_maxburst = dma_params->burstsize;
+ } else {
+ slave_config.direction = DMA_FROM_DEVICE;
+ slave_config.src_addr = dma_params->dma_addr;
+ slave_config.src_addr_width = buswidth;
+ slave_config.src_maxburst = dma_params->burstsize;
+ }
+
+ ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config);
if (ret)
- goto out;
-
- ret = imx_dma_setup_progression_handler(iprtd->dma,
- imx_ssi_dma_progression);
- if (ret) {
- pr_err("Failed to setup the DMA handler\n");
- goto out;
- }
-
- ret = imx_dma_config_channel(iprtd->dma,
- IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
- IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
- dma_params->dma, 1);
- if (ret < 0) {
- pr_err("Cannot configure DMA channel: %d\n", ret);
- goto out;
- }
-
- imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2);
+ return ret;
return 0;
-out:
- imx_dma_free(iprtd->dma);
- return ret;
}
static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- int i;
unsigned long dma_addr;
+ struct dma_chan *chan;
+ struct imx_pcm_dma_params *dma_params;
+ int ret;
- imx_ssi_dma_alloc(substream);
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ ret = imx_ssi_dma_alloc(substream, params);
+ if (ret)
+ return ret;
+ chan = iprtd->dma_chan;
iprtd->size = params_buffer_bytes(params);
iprtd->periods = params_periods(params);
- iprtd->period = params_period_bytes(params);
+ iprtd->period_bytes = params_period_bytes(params);
iprtd->offset = 0;
iprtd->period_time = HZ / (params_rate(params) /
params_period_size(params));
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- if (iprtd->sg_count != iprtd->periods) {
- kfree(iprtd->sg_list);
-
- iprtd->sg_list = kcalloc(iprtd->periods + 1,
- sizeof(struct scatterlist), GFP_KERNEL);
- if (!iprtd->sg_list)
- return -ENOMEM;
- iprtd->sg_count = iprtd->periods + 1;
- }
-
- sg_init_table(iprtd->sg_list, iprtd->sg_count);
dma_addr = runtime->dma_addr;
- for (i = 0; i < iprtd->periods; i++) {
- iprtd->sg_list[i].page_link = 0;
- iprtd->sg_list[i].offset = 0;
- iprtd->sg_list[i].dma_address = dma_addr;
- iprtd->sg_list[i].length = iprtd->period;
- dma_addr += iprtd->period;
+ iprtd->buf = (unsigned int *)substream->dma_buffer.area;
+
+ iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+ iprtd->period_bytes * iprtd->periods,
+ iprtd->period_bytes,
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ if (!iprtd->desc) {
+ dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+ return -EINVAL;
}
- /* close the loop */
- iprtd->sg_list[iprtd->sg_count - 1].offset = 0;
- iprtd->sg_list[iprtd->sg_count - 1].length = 0;
- iprtd->sg_list[iprtd->sg_count - 1].page_link =
- ((unsigned long) iprtd->sg_list | 0x01) & ~0x02;
+ iprtd->desc->callback = audio_dma_irq;
+ iprtd->desc->callback_param = substream;
+
return 0;
}
@@ -194,41 +176,21 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- if (iprtd->dma >= 0) {
- imx_dma_free(iprtd->dma);
- iprtd->dma = -EINVAL;
+ if (iprtd->dma_chan) {
+ dma_release_channel(iprtd->dma_chan);
+ iprtd->dma_chan = NULL;
}
- kfree(iprtd->sg_list);
- iprtd->sg_list = NULL;
-
return 0;
}
static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct imx_pcm_dma_params *dma_params;
- struct imx_pcm_runtime_data *iprtd = runtime->private_data;
- int err;
dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- iprtd->substream = substream;
- iprtd->buf = (unsigned int *)substream->dma_buffer.area;
- iprtd->period_cnt = 0;
-
- pr_debug("%s: buf: %p period: %d periods: %d\n",
- __func__, iprtd->buf, iprtd->period, iprtd->periods);
-
- err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
- IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
- substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- DMA_MODE_WRITE : DMA_MODE_READ);
- if (err)
- return err;
-
return 0;
}
@@ -241,14 +203,14 @@
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- imx_dma_enable(iprtd->dma);
+ dmaengine_submit(iprtd->desc);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- imx_dma_disable(iprtd->dma);
+ dmaengine_terminate_all(iprtd->dma_chan);
break;
default:
@@ -263,6 +225,9 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+ pr_debug("%s: %ld %ld\n", __func__, iprtd->offset,
+ bytes_to_frames(substream->runtime, iprtd->offset));
+
return bytes_to_frames(substream->runtime, iprtd->offset);
}
@@ -279,7 +244,7 @@
.channels_max = 2,
.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
.period_bytes_min = 128,
- .period_bytes_max = 16 * 1024,
+ .period_bytes_max = 65535, /* Limited by SDMA engine */
.periods_min = 2,
.periods_max = 255,
.fifo_size = 0,
@@ -304,11 +269,23 @@
}
snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
+
+ return 0;
+}
+
+static int snd_imx_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+ kfree(iprtd);
+
return 0;
}
static struct snd_pcm_ops imx_pcm_ops = {
.open = snd_imx_open,
+ .close = snd_imx_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_imx_pcm_hw_params,
.hw_free = snd_imx_pcm_hw_free,
@@ -340,7 +317,6 @@
.name = "imx-pcm-audio",
.owner = THIS_MODULE,
},
-
.probe = imx_soc_platform_probe,
.remove = __devexit_p(imx_soc_platform_remove),
};
@@ -356,4 +332,3 @@
platform_driver_unregister(&imx_pcm_driver);
}
module_exit(snd_imx_pcm_exit);
-
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index d4bd345..d2d98c7 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -439,7 +439,22 @@
}
EXPORT_SYMBOL_GPL(imx_pcm_free);
+static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
+ uint32_t val;
+
+ snd_soc_dai_set_drvdata(dai, ssi);
+
+ val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
+ SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
+ writel(val, ssi->base + SSI_SFCSR);
+
+ return 0;
+}
+
static struct snd_soc_dai_driver imx_ssi_dai = {
+ .probe = imx_ssi_dai_probe,
.playback = {
.channels_min = 2,
.channels_max = 2,
@@ -455,20 +470,6 @@
.ops = &imx_ssi_pcm_dai_ops,
};
-static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
-{
- struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
- uint32_t val;
-
- snd_soc_dai_set_drvdata(dai, ssi);
-
- val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
- SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
- writel(val, ssi->base + SSI_SFCSR);
-
- return 0;
-}
-
static struct snd_soc_dai_driver imx_ac97_dai = {
.probe = imx_ssi_dai_probe,
.ac97_control = 1,
@@ -677,7 +678,17 @@
goto failed_register;
}
- ssi->soc_platform_pdev = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
+ ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
+ if (!ssi->soc_platform_pdev_fiq)
+ goto failed_pdev_fiq_alloc;
+ platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi);
+ ret = platform_device_add(ssi->soc_platform_pdev_fiq);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to add platform device\n");
+ goto failed_pdev_fiq_add;
+ }
+
+ ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id);
if (!ssi->soc_platform_pdev)
goto failed_pdev_alloc;
platform_set_drvdata(ssi->soc_platform_pdev, ssi);
@@ -692,6 +703,9 @@
failed_pdev_add:
platform_device_put(ssi->soc_platform_pdev);
failed_pdev_alloc:
+failed_pdev_fiq_add:
+ platform_device_put(ssi->soc_platform_pdev_fiq);
+failed_pdev_fiq_alloc:
snd_soc_unregister_dai(&pdev->dev);
failed_register:
failed_ac97:
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index 53b780d..a4406a1 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -185,6 +185,9 @@
#define DRV_NAME "imx-ssi"
+#include <linux/dmaengine.h>
+#include <mach/dma.h>
+
struct imx_pcm_dma_params {
int dma;
unsigned long dma_addr;
@@ -212,6 +215,7 @@
int enabled;
struct platform_device *soc_platform_pdev;
+ struct platform_device *soc_platform_pdev_fiq;
};
struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev,
diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c
index 6a65dd7..39f2373 100644
--- a/sound/soc/imx/phycore-ac97.c
+++ b/sound/soc/imx/phycore-ac97.c
@@ -20,9 +20,6 @@
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include "../codecs/wm9712.h"
-#include "imx-ssi.h"
-
static struct snd_soc_card imx_phycore;
static struct snd_soc_ops imx_phycore_hifi_ops = {
@@ -41,7 +38,7 @@
};
static struct snd_soc_card imx_phycore = {
- .name = "PhyCORE-audio",
+ .name = "PhyCORE-ac97-audio",
.dai_link = imx_phycore_dai_ac97,
.num_links = ARRAY_SIZE(imx_phycore_dai_ac97),
};
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d211c9f..7e84f24 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -644,15 +644,23 @@
case OMAP_MCBSP_CLKR_SRC_CLKR:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR);
break;
case OMAP_MCBSP_CLKR_SRC_CLKX:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX);
break;
case OMAP_MCBSP_FSR_SRC_FSR:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR);
break;
case OMAP_MCBSP_FSR_SRC_FSX:
+ if (cpu_class_is_omap1())
+ break;
omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX);
break;
default:
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 97e9423..f451acd 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -100,8 +100,13 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
corgi_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index b8207ce..5ef0526 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -72,9 +72,13 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
magician_ext_control(codec);
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index af84ee9..84edd03 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -77,8 +77,13 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
poodle_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index f470f36..0b30d7d 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -108,8 +108,13 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
spitz_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 73d0edd..7b983f9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -81,8 +81,13 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
+ mutex_lock(&codec->mutex);
+
/* check the jack status at stream startup */
tosa_ext_control(codec);
+
+ mutex_unlock(&codec->mutex);
+
return 0;
}
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 8a6b53c..d85bf8a 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -2,6 +2,7 @@
tristate "SoC Audio for the Samsung S3CXXXX chips"
depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210
select S3C64XX_DMA if ARCH_S3C64XX
+ select S3C2410_DMA if ARCH_S3C2410
help
Say Y or M if you want to add support for codecs attached to
the S3C24XX AC97 or I2S interfaces. You will also need to
diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c
index ffd5cf2..468cc11 100644
--- a/sound/soc/s3c24xx/rx1950_uda1380.c
+++ b/sound/soc/s3c24xx/rx1950_uda1380.c
@@ -50,7 +50,6 @@
16000,
44100,
48000,
- 88200,
};
static struct snd_pcm_hw_constraint_list hw_rates = {
@@ -130,7 +129,6 @@
};
static struct platform_device *s3c24xx_snd_device;
-static struct clk *xtal;
static int rx1950_startup(struct snd_pcm_substream *substream)
{
@@ -179,10 +177,8 @@
case 44100:
case 88200:
clk_source = S3C24XX_CLKSRC_MPLL;
- fs_mode = S3C2410_IISMOD_256FS;
- div = clk_get_rate(xtal) / (256 * rate);
- if (clk_get_rate(xtal) % (256 * rate) > (128 * rate))
- div++;
+ fs_mode = S3C2410_IISMOD_384FS;
+ div = 1;
break;
default:
printk(KERN_ERR "%s: rate %d is not supported\n",
@@ -210,7 +206,7 @@
/* set MCLK division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
- S3C2410_IISMOD_384FS);
+ fs_mode);
if (ret < 0)
return ret;
@@ -295,17 +291,8 @@
goto err_plat_add;
}
- xtal = clk_get(&s3c24xx_snd_device->dev, "xtal");
-
- if (IS_ERR(xtal)) {
- ret = PTR_ERR(xtal);
- platform_device_unregister(s3c24xx_snd_device);
- goto err_clk;
- }
-
return 0;
-err_clk:
err_plat_add:
err_plat_alloc:
err_gpio_conf:
@@ -320,7 +307,6 @@
platform_device_unregister(s3c24xx_snd_device);
snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
hp_jack_gpios);
- clk_put(xtal);
gpio_free(S3C2410_GPA(1));
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 614a8b3..441285a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3043,8 +3043,10 @@
for (i = 0; i < count; i++) {
dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL);
- if (dai == NULL)
- return -ENOMEM;
+ if (dai == NULL) {
+ ret = -ENOMEM;
+ goto err;
+ }
/* create DAI component name */
dai->name = fmt_multiple_name(dev, &dai_drv[i]);
@@ -3263,9 +3265,6 @@
return 0;
error:
- for (i--; i >= 0; i--)
- snd_soc_unregister_dai(dev);
-
if (codec->reg_cache)
kfree(codec->reg_cache);
kfree(codec->name);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7d85c64..75ed649 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -683,12 +683,12 @@
struct snd_soc_dapm_widget *b,
int sort[])
{
- if (a->codec != b->codec)
- return (unsigned long)a - (unsigned long)b;
if (sort[a->id] != sort[b->id])
return sort[a->id] - sort[b->id];
if (a->reg != b->reg)
return a->reg - b->reg;
+ if (a->codec != b->codec)
+ return (unsigned long)a->codec - (unsigned long)b->codec;
return 0;
}
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 1bc56b2..337a002 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -155,7 +155,7 @@
if (max_tries < 1)
max_tries = 1;
- /* ssc_div must be a power of 2. */
+ /* ssc_div must be even. */
ssc_div = (ssc_div + 1) & ~1UL;
if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) {
