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authorTakashi Iwai <tiwai@suse.de>2010-11-22 08:11:10 +0100
committerTakashi Iwai <tiwai@suse.de>2010-11-22 08:11:10 +0100
commitd2b88e4c103f37584dc81ef8a41ca584c1ce847b (patch)
tree247ac000bec902fcc580757d236caa053755ea27
parentALSA: oxygen: add HiFier Serenade support (diff)
parentALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J (diff)
downloadlinux-dev-d2b88e4c103f37584dc81ef8a41ca584c1ce847b.tar.xz
linux-dev-d2b88e4c103f37584dc81ef8a41ca584c1ce847b.zip
Merge branch 'fix/misc' into topic/misc
-rw-r--r--sound/atmel/abdac.c2
-rw-r--r--sound/core/pcm_lib.c2
-rw-r--r--sound/oss/dev_table.c6
-rw-r--r--sound/oss/midibuf.c4
-rw-r--r--sound/oss/pss.c6
-rw-r--r--sound/oss/sequencer.c4
-rw-r--r--sound/pci/asihpi/hpioctl.c2
-rw-r--r--sound/pci/azt3328.c26
-rw-r--r--sound/pci/ctxfi/ctpcm.c16
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c11
-rw-r--r--sound/pci/hda/patch_sigmatel.c14
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/pci/mixart/mixart_hwdep.h10
-rw-r--r--sound/ppc/pmac.c12
-rw-r--r--sound/soc/codecs/Kconfig3
-rw-r--r--sound/soc/codecs/tlv320dac33.c36
-rw-r--r--sound/soc/codecs/tpa6130a2.c6
-rw-r--r--sound/soc/codecs/wm8900.c6
-rw-r--r--sound/soc/codecs/wm_hubs.c2
-rw-r--r--sound/soc/pxa/tosa.c2
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/spi/at73c213.c2
23 files changed, 120 insertions, 64 deletions
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index f2f41c854221..4e47baada66f 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -420,7 +420,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
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);
goto out_put_pclk;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a1707cca9c66..b75db8e9cc0f 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -223,7 +223,7 @@ static void xrun_log(struct snd_pcm_substream *substream,
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 727bdb9ba2dc..d8cf3e58dc76 100644
--- a/sound/oss/dev_table.c
+++ b/sound/oss/dev_table.c
@@ -71,7 +71,7 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
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 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
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 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
/* 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 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
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 782b3b84dac6..ceedb1eff203 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -178,7 +178,7 @@ int MIDIbuf_open(int dev, struct file *file)
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 @@ int MIDIbuf_open(int dev, struct file *file)
}
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 e19dd5dcc2de..9b800ce5100e 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -859,7 +859,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
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 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
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 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
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 e85789e53816..5ea1098ac427 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -1646,13 +1646,13 @@ void sequencer_init(void)
{
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 62895a719fcb..22dbd91811a4 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -435,7 +435,7 @@ void __devexit asihpi_adapter_remove(struct pci_dev *pci_dev)
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 4679ed83a43b..2f3cacbd5528 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1129,10 +1129,11 @@ snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip,
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 @@ static const struct snd_pcm_hardware snd_azf3328_hardware =
.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 @@ snd_azf3328_timer_stop(struct snd_timer *timer)
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 85ab43e89212..457d21189b0d 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -129,8 +129,6 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
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 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
}
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 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
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 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
}
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 6361f752b5f3..3cfb31e77b16 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3100,6 +3100,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
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),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5f00589cb791..b7e234898fd8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10816,6 +10816,9 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
return 0;
}
+static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg);
+
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
@@ -10833,7 +10836,10 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
- err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (codec->vendor_id == 0x10ec0887)
+ err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ else
+ err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
@@ -16963,7 +16969,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
#define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c)
/* add playback controls from the parsed DAC table */
-/* Based on ALC880 version. But ALC861VD has separate,
+/* Based on ALC880 version. But ALC861VD and ALC887 have separate,
* different NIDs for mute/unmute switch and volume control */
static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
@@ -19298,6 +19304,7 @@ static const struct alc_fixup alc662_fixups[] = {
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),
{}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 93fa59cc60ef..cfd73afad882 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -389,6 +389,11 @@ static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
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),
@@ -5452,12 +5457,17 @@ again:
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 400f9ebd243e..629a5494347a 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1866,6 +1866,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
},
{
.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 a46f5083db99..812e288ef2e7 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))
+#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 85081172403f..b47cfd45b3b9 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -1228,10 +1228,8 @@ int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
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 @@ int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
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/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 94a9d06b9027..3b5690d28b8b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -25,8 +25,9 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS42L51 if I2C
select SND_SOC_CS4270 if I2C
+ select SND_SOC_CX20442
select SND_SOC_DA7210 if I2C
- select SND_SOC_JZ4740 if SOC_JZ4740
+ select SND_SOC_JZ4740_CODEC if SOC_JZ4740
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index d251ff54a2d3..c5ab8c805771 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -58,7 +58,7 @@
(1000000000 / ((rate * 1000) / samples))
#define US_TO_SAMPLES(rate, us) \
- (rate / (1000000 / us))
+ (rate / (1000000 / (us < 1000000 ? us : 1000000)))
#define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate)))
@@ -200,7 +200,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
u8 *value)
{
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
- int val;
+ int val, ret = 0;
*value = reg & 0xff;
@@ -210,6 +210,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
if (val < 0) {
dev_err(codec->dev, "Read failed (%d)\n", val);
value[0] = dac33_read_reg_cache(codec, reg);
+ ret = val;
} else {
value[0] = val;
dac33_write_reg_cache(codec, reg, val);
@@ -218,7 +219,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
value[0] = dac33_read_reg_cache(codec, reg);
}
- return 0;
+ return ret;
}
static int dac33_write(struct snd_soc_codec *codec, unsigned int reg,
@@ -329,13 +330,18 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
}
-static inline void dac33_read_id(struct snd_soc_codec *codec)
+static inline int dac33_read_id(struct snd_soc_codec *codec)
{
+ int i, ret = 0;
u8 reg;
- dac33_read(codec, DAC33_DEVICE_ID_MSB, &reg);
- dac33_read(codec, DAC33_DEVICE_ID_LSB, &reg);
- dac33_read(codec, DAC33_DEVICE_REV_ID, &reg);
+ for (i = 0; i < 3; i++) {
+ ret = dac33_read(codec, DAC33_DEVICE_ID_MSB + i, &reg);
+ if (ret < 0)
+ break;
+ }
+
+ return ret;
}
static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
@@ -1076,6 +1082,9 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
/* Number of samples under i2c latency */
dac33->alarm_threshold = US_TO_SAMPLES(rate,
dac33->mode1_latency);
+ nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
+ dac33->alarm_threshold;
+
if (dac33->auto_fifo_config) {
if (period_size <= dac33->alarm_threshold)
/*
@@ -1086,6 +1095,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
((dac33->alarm_threshold / period_size) +
(dac33->alarm_threshold % period_size ?
1 : 0));
+ else if (period_size > nsample_limit)
+ dac33->nsample = nsample_limit;
else
dac33->nsample = period_size;
} else {
@@ -1097,8 +1108,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
*/
dac33->nsample_max = substream->runtime->buffer_size -
period_size;
- nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
- dac33->alarm_threshold;
+
if (dac33->nsample_max > nsample_limit)
dac33->nsample_max = nsample_limit;
@@ -1414,9 +1424,15 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
dev_err(codec->dev, "Failed to power up codec: %d\n", ret);
goto err_power;
}
- dac33_read_id(codec);
+ ret = dac33_read_id(codec);
dac33_hard_power(codec, 0);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to read chip ID: %d\n", ret);
+ ret = -ENODEV;
+ goto err_power;
+ }
+
/* Check if the IRQ number is valid and request it */
if (dac33->irq >= 0) {
ret = request_irq(dac33->irq, dac33_interrupt_handler,
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 329acc1a2074..ee4fb201de60 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -119,13 +119,13 @@ static int tpa6130a2_power(int power)
{
struct tpa6130a2_data *data;
u8 val;
- int ret;
+ int ret = 0;
BUG_ON(tpa6130a2_client == NULL);
data = i2c_get_clientdata(tpa6130a2_client);
mutex_lock(&data->mutex);
- if (power) {
+ if (power && !data->power_state) {
/* Power on */
if (data->power_gpio >= 0)
gpio_set_value(data->power_gpio, 1);
@@ -153,7 +153,7 @@ static int tpa6130a2_power(int power)
val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
val &= ~TPA6130A2_SWS;
tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
- } else {
+ } else if (!power && data->power_state) {
/* set SWS */
val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
val |= TPA6130A2_SWS;
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index b4f11724a63f..aca4b1ea10bb 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg)
{
switch (reg) {
case WM8900_REG_ID:
- case WM8900_REG_POWER1:
return 1;
default:
return 0;
@@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec)
return -ENODEV;
}
- /* Read back from the chip */
- reg = snd_soc_read(codec, WM8900_REG_POWER1);
- reg = (reg >> 12) & 0xf;
- dev_info(codec->dev, "WM8900 revision %d\n", reg);
-
wm8900_reset(codec);
/* Turn the chip on */
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 2cb81538cd91..19ca782ac970 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -123,7 +123,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
break;
default:
- WARN(1, "Unknown DCS readback method");
+ WARN(1, "Unknown DCS readback method\n");
break;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index a3bfb2e8b70f..73d0edd8ded9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
static int tosa_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->card->codec;
+ struct snd_soc_codec *codec = rtd->codec;
/* check the jack status at stream startup */
tosa_ext_control(codec);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1c8f3f507f54..614a8b30d87b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev,
{
struct snd_soc_pcm_runtime *rtd =
container_of(dev, struct snd_soc_pcm_runtime, dev);
+ int ret;
- strict_strtol(buf, 10, &rtd->pmdown_time);
+ ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+ if (ret)
+ return ret;
return count;
}
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 1bc56b2b94e2..337a00241a1f 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -155,7 +155,7 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip)
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) {