summaryrefslogtreecommitdiffstats
path: root/sound
diff options
context:
space:
mode:
authorBjorn Helgaas <bhelgaas@google.com>2012-09-13 16:41:01 +0200
committerBjorn Helgaas <bhelgaas@google.com>2012-09-13 16:41:01 +0200
commit78890b5989d96ddce989cde929c45ceeded0fcaf (patch)
tree4e2da81fc7c97f11aee174b1eedac110c9a68b3a /sound
parentMerge branch 'pci/stephen-const' into next (diff)
parentLinux 3.6-rc5 (diff)
downloadlinux-78890b5989d96ddce989cde929c45ceeded0fcaf.tar.xz
linux-78890b5989d96ddce989cde929c45ceeded0fcaf.zip
Merge commit 'v3.6-rc5' into next
* commit 'v3.6-rc5': (1098 commits) Linux 3.6-rc5 HID: tpkbd: work even if the new Lenovo Keyboard driver is not configured Remove user-triggerable BUG from mpol_to_str xen/pciback: Fix proper FLR steps. uml: fix compile error in deliver_alarm() dj: memory scribble in logi_dj Fix order of arguments to compat_put_time[spec|val] xen: Use correct masking in xen_swiotlb_alloc_coherent. xen: fix logical error in tlb flushing xen/p2m: Fix one-off error in checking the P2M tree directory. powerpc: Don't use __put_user() in patch_instruction powerpc: Make sure IPI handlers see data written by IPI senders powerpc: Restore correct DSCR in context switch powerpc: Fix DSCR inheritance in copy_thread() powerpc: Keep thread.dscr and thread.dscr_inherit in sync powerpc: Update DSCR on all CPUs when writing sysfs dscr_default powerpc/powernv: Always go into nap mode when CPU is offline powerpc: Give hypervisor decrementer interrupts their own handler powerpc/vphn: Fix arch_update_cpu_topology() return value ARM: gemini: fix the gemini build ... Conflicts: drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c drivers/rapidio/devices/tsi721.c
Diffstat (limited to 'sound')
-rw-r--r--sound/arm/pxa2xx-ac97.c4
-rw-r--r--sound/atmel/abdac.c3
-rw-r--r--sound/atmel/ac97c.c14
-rw-r--r--sound/core/sgbuf.c2
-rw-r--r--sound/drivers/aloop.c2
-rw-r--r--sound/drivers/dummy.c2
-rw-r--r--sound/drivers/pcsp/pcsp.c4
-rw-r--r--sound/isa/als100.c2
-rw-r--r--sound/oss/sb_audio.c4
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c2
-rw-r--r--sound/pci/ctxfi/ctatc.c4
-rw-r--r--sound/pci/emu10k1/memory.c5
-rw-r--r--sound/pci/hda/hda_auto_parser.c5
-rw-r--r--sound/pci/hda/hda_beep.c29
-rw-r--r--sound/pci/hda/hda_codec.c83
-rw-r--r--sound/pci/hda/hda_codec.h2
-rw-r--r--sound/pci/hda/hda_intel.c9
-rw-r--r--sound/pci/hda/hda_proc.c2
-rw-r--r--sound/pci/hda/patch_ca0132.c174
-rw-r--r--sound/pci/hda/patch_conexant.c6
-rw-r--r--sound/pci/hda/patch_hdmi.c12
-rw-r--r--sound/pci/hda/patch_realtek.c8
-rw-r--r--sound/pci/hda/patch_sigmatel.c13
-rw-r--r--sound/pci/hda/patch_via.c8
-rw-r--r--sound/pci/lx6464es/lx6464es.c2
-rw-r--r--sound/pci/rme9652/hdspm.c2
-rw-r--r--sound/pci/sis7019.c5
-rw-r--r--sound/ppc/powermac.c2
-rw-r--r--sound/ppc/snd_ps3.c1
-rw-r--r--sound/soc/blackfin/bf6xx-sport.c7
-rw-r--r--sound/soc/codecs/ab8500-codec.c4
-rw-r--r--sound/soc/codecs/ad1980.c1
-rw-r--r--sound/soc/codecs/mc13783.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c3
-rw-r--r--sound/soc/codecs/stac9766.c1
-rw-r--r--sound/soc/codecs/wm5102.c25
-rw-r--r--sound/soc/codecs/wm5110.c12
-rw-r--r--sound/soc/codecs/wm8962.c18
-rw-r--r--sound/soc/codecs/wm8994.c17
-rw-r--r--sound/soc/codecs/wm9712.c22
-rw-r--r--sound/soc/codecs/wm9713.c1
-rw-r--r--sound/soc/davinci/davinci-mcasp.c10
-rw-r--r--sound/soc/fsl/imx-ssi.c5
-rw-r--r--sound/soc/mxs/Kconfig2
-rw-r--r--sound/soc/mxs/mxs-saif.c24
-rw-r--r--sound/soc/omap/mcbsp.c2
-rw-r--r--sound/soc/omap/omap-mcbsp.c1
-rw-r--r--sound/soc/omap/omap-pcm.c1
-rw-r--r--sound/soc/samsung/pcm.c2
-rw-r--r--sound/soc/soc-core.c12
-rw-r--r--sound/soc/soc-jack.c2
-rw-r--r--sound/soc/tegra/tegra_alc5632.c2
-rw-r--r--sound/soc/tegra/tegra_wm8903.c10
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c2
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.c2
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.h2
-rw-r--r--sound/usb/card.c4
-rw-r--r--sound/usb/endpoint.c24
-rw-r--r--sound/usb/endpoint.h3
-rw-r--r--sound/usb/pcm.c61
60 files changed, 409 insertions, 286 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 0d7b25e81643..4e1fda75c1c9 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
.prepare = pxa2xx_ac97_pcm_prepare,
};
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int pxa2xx_ac97_do_suspend(struct snd_card *card)
{
@@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = {
.driver = {
.name = "pxa2xx-ac97",
.owner = THIS_MODULE,
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
.pm = &pxa2xx_ac97_pm_ops,
#endif
},
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index eb4ceb71123e..277ebce23a45 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
dac->regs = ioremap(regs->start, resource_size(regs));
if (!dac->regs) {
dev_dbg(&pdev->dev, "could not remap register memory\n");
+ retval = -ENOMEM;
goto out_free_card;
}
@@ -534,7 +535,7 @@ out_put_pclk:
return retval;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int atmel_abdac_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index bf47025bdf45..9052aff37f64 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream,
if (retval < 0)
return retval;
/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
- if (cpu_is_at32ap7000()) {
- if (retval < 0)
- return retval;
- /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
- if (retval == 1)
- if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
- dw_dma_cyclic_free(chip->dma.rx_chan);
- }
+ if (cpu_is_at32ap7000() && retval == 1)
+ if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
+ dw_dma_cyclic_free(chip->dma.rx_chan);
/* Set restrictions to params. */
mutex_lock(&opened_mutex);
@@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
if (!chip->regs) {
dev_dbg(&pdev->dev, "could not remap register memory\n");
+ retval = -ENOMEM;
goto err_ioremap;
}
@@ -1134,7 +1130,7 @@ err_snd_card_new:
return retval;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int atmel_ac97c_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index 4e7ec2b49873..d0f00356fc11 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -101,7 +101,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device,
chunk, &tmpb) < 0) {
if (!sgbuf->pages)
- return NULL;
+ goto _failed;
if (!res_size)
goto _failed;
size = sgbuf->pages * PAGE_SIZE;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 1128b35b2b05..5a34355e78e8 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr)
return 0;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int loopback_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index f7d3bfc6bca8..54bb6644a598 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr)
return 0;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int snd_dummy_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 6ca59fc6dcb9..ef171295f6d4 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip)
pcspkr_stop_sound();
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int pcsp_suspend(struct device *dev)
{
struct snd_pcsp *chip = dev_get_drvdata(dev);
@@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL);
#define PCSP_PM_OPS &pcsp_pm
#else
#define PCSP_PM_OPS NULL
-#endif /* CONFIG_PM */
+#endif /* CONFIG_PM_SLEEP */
static void pcsp_shutdown(struct platform_device *dev)
{
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 2d67c78c9f4b..f7cdaf51512d 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev,
irq[dev], dma8[dev], dma16[dev]);
}
- if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
+ if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) {
snd_card_free(card);
return error;
}
diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c
index 733b014ec7d1..b2b3c014221a 100644
--- a/sound/oss/sb_audio.c
+++ b/sound/oss/sb_audio.c
@@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed)
if (speed > 0)
{
int tmp;
- int s = speed * devc->channels;
+ int s;
if (speed < 5000)
speed = 5000;
if (speed > 44100)
speed = 44100;
+ s = speed * devc->channels;
+
devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff;
tmp = 256 - devc->tconst;
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index f75f5ffdfdfb..a71d1c14a0f6 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
codec_index != CS46XX_SECONDARY_CODEC_INDEX))
- return -EINVAL;
+ return 0xffff;
chip->active_ctrl(chip, 1);
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 8e40262d4117..2f6e9c762d3f 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
atc_connect_resources(atc);
atc->timer = ct_timer_new(atc);
- if (!atc->timer)
+ if (!atc->timer) {
+ err = -ENOMEM;
goto error1;
+ }
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops);
if (err < 0)
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 4f502a2bdc3c..0a436626182b 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
for (page = blk->first_page; page <= blk->last_page; page++, idx++) {
unsigned long ofs = idx << PAGE_SHIFT;
dma_addr_t addr;
- addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+ if (ofs >= runtime->dma_bytes)
+ addr = emu->silent_page.addr;
+ else
+ addr = snd_pcm_sgbuf_get_addr(substream, ofs);
if (! is_valid_page(emu, addr)) {
printk(KERN_ERR "emu: failure page = %d\n", idx);
mutex_unlock(&hdr->block_mutex);
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 647218d69f68..4f7d2dfcef7b 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
if (cfg->dig_outs)
snd_printd(" dig-out=0x%x/0x%x\n",
cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
- snd_printd(" inputs:");
+ snd_printd(" inputs:\n");
for (i = 0; i < cfg->num_inputs; i++) {
- snd_printd(" %s=0x%x",
+ snd_printd(" %s=0x%x\n",
hda_get_autocfg_input_label(codec, cfg, i),
cfg->inputs[i].pin);
}
- snd_printd("\n");
if (cfg->dig_in_pin)
snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin);
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 0bc2315b181d..0849aac449f2 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -231,16 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
}
EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
+static bool ctl_has_mute(struct snd_kcontrol *kcontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ return query_amp_caps(codec, get_amp_nid(kcontrol),
+ get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE;
+}
+
/* get/put callbacks for beep mute mixer switches */
int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct hda_beep *beep = codec->beep;
- if (beep) {
+ if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
ucontrol->value.integer.value[0] =
- ucontrol->value.integer.value[1] =
- beep->enabled;
+ ucontrol->value.integer.value[1] = beep->enabled;
return 0;
}
return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
@@ -252,9 +258,20 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct hda_beep *beep = codec->beep;
- if (beep)
- snd_hda_enable_beep_device(codec,
- *ucontrol->value.integer.value);
+ if (beep) {
+ u8 chs = get_amp_channels(kcontrol);
+ int enable = 0;
+ long *valp = ucontrol->value.integer.value;
+ if (chs & 1) {
+ enable |= *valp;
+ valp++;
+ }
+ if (chs & 2)
+ enable |= *valp;
+ snd_hda_enable_beep_device(codec, enable);
+ }
+ if (!ctl_has_mute(kcontrol))
+ return 0;
return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88a9c20eb7a2..f25c24c743f9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec)
kfree(codec);
}
+static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec,
+ hda_nid_t fg, unsigned int power_state);
+
static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state);
@@ -1317,6 +1320,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
AC_VERB_GET_SUBSYSTEM_ID, 0);
}
+ codec->epss = snd_hda_codec_get_supported_ps(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_EPSS);
+
/* power-up all before initialization */
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
@@ -1386,6 +1393,44 @@ int snd_hda_codec_configure(struct hda_codec *codec)
}
EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
+/* update the stream-id if changed */
+static void update_pcm_stream_id(struct hda_codec *codec,
+ struct hda_cvt_setup *p, hda_nid_t nid,
+ u32 stream_tag, int channel_id)
+{
+ unsigned int oldval, newval;
+
+ if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
+ oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
+ newval = (stream_tag << 4) | channel_id;
+ if (oldval != newval)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CHANNEL_STREAMID,
+ newval);
+ p->stream_tag = stream_tag;
+ p->channel_id = channel_id;
+ }
+}
+
+/* update the format-id if changed */
+static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p,
+ hda_nid_t nid, int format)
+{
+ unsigned int oldval;
+
+ if (p->format_id != format) {
+ oldval = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_STREAM_FORMAT, 0);
+ if (oldval != format) {
+ msleep(1);
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_STREAM_FORMAT,
+ format);
+ }
+ p->format_id = format;
+ }
+}
+
/**
* snd_hda_codec_setup_stream - set up the codec for streaming
* @codec: the CODEC to set up
@@ -1400,7 +1445,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
{
struct hda_codec *c;
struct hda_cvt_setup *p;
- unsigned int oldval, newval;
int type;
int i;
@@ -1413,29 +1457,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
p = get_hda_cvt_setup(codec, nid);
if (!p)
return;
- /* update the stream-id if changed */
- if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
- oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
- newval = (stream_tag << 4) | channel_id;
- if (oldval != newval)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CHANNEL_STREAMID,
- newval);
- p->stream_tag = stream_tag;
- p->channel_id = channel_id;
- }
- /* update the format-id if changed */
- if (p->format_id != format) {
- oldval = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_STREAM_FORMAT, 0);
- if (oldval != format) {
- msleep(1);
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_STREAM_FORMAT,
- format);
- }
- p->format_id = format;
- }
+
+ if (codec->pcm_format_first)
+ update_pcm_format(codec, p, nid, format);
+ update_pcm_stream_id(codec, p, nid, stream_tag, channel_id);
+ if (!codec->pcm_format_first)
+ update_pcm_format(codec, p, nid, format);
+
p->active = 1;
p->dirty = 0;
@@ -3497,7 +3525,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg
{
int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE);
- if (sup < 0)
+ if (sup == -1)
return false;
if (sup & power_state)
return true;
@@ -3522,8 +3550,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
/* this delay seems necessary to avoid click noise at power-down */
if (power_state == AC_PWRST_D3) {
/* transition time less than 10ms for power down */
- bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS);
- msleep(epss ? 10 : 100);
+ msleep(codec->epss ? 10 : 100);
}
/* repeat power states setting at most 10 times*/
@@ -4433,6 +4460,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
* then there is no need to go through power up here.
*/
if (codec->power_on) {
+ if (codec->power_transition < 0)
+ codec->power_transition = 0;
spin_unlock(&codec->power_lock);
return;
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index c422d330ca54..e5a7e19a8071 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -861,6 +861,8 @@ struct hda_codec {
unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
unsigned int no_jack_detect:1; /* Machine has no jack-detection */
+ unsigned int pcm_format_first:1; /* PCM format must be set first */
+ unsigned int epss:1; /* supporting EPSS? */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
int power_transition; /* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8aced182fd1..60882c62f180 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, CPT},"
"{Intel, PPT},"
"{Intel, LPT},"
+ "{Intel, LPT_LP},"
"{Intel, HPT},"
"{Intel, PBG},"
"{Intel, SCH},"
@@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ PCI_DEVICE(0x8086, 0x8c20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+ /* Lynx Point-LP */
+ { PCI_DEVICE(0x8086, 0x9c20),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+ /* Lynx Point-LP */
+ { PCI_DEVICE(0x8086, 0x9c21),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
/* Haswell */
{ PCI_DEVICE(0x8086, 0x0c0c),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 7e46258fc700..6894ec66258c 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
if (digi1 & AC_DIG1_EMPHASIS)
snd_iprintf(buffer, " Preemphasis");
if (digi1 & AC_DIG1_COPYRIGHT)
- snd_iprintf(buffer, " Copyright");
+ snd_iprintf(buffer, " Non-Copyright");
if (digi1 & AC_DIG1_NONAUDIO)
snd_iprintf(buffer, " Non-Audio");
if (digi1 & AC_DIG1_PROFESSIONAL)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index d0d3540e39e7..49750a96d649 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
}
- if (dac)
+ if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP))
snd_hda_codec_write(codec, dac, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
}
@@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
}
- if (adc)
+ if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP))
snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
}
@@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+ if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) {
+ snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid);
+ return 0;
+ }
sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
@@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+ if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) {
+ snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid);
+ return 0;
+ }
sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
@@ -464,50 +472,17 @@ exit:
}
/*
- * PCM stuffs
+ * PCM callbacks
*/
-static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid,
- u32 stream_tag,
- int channel_id, int format)
+static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
{
- unsigned int oldval, newval;
-
- if (!nid)
- return;
-
- snd_printdd("ca0132_setup_stream: "
- "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
- nid, stream_tag, channel_id, format);
-
- /* update the format-id if changed */
- oldval = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_STREAM_FORMAT,
- 0);
- if (oldval != format) {
- msleep(20);
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_STREAM_FORMAT,
- format);
- }
-
- oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
- newval = (stream_tag << 4) | channel_id;
- if (oldval != newval) {
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CHANNEL_STREAMID,
- newval);
- }
-}
-
-static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
-{
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+ struct ca0132_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
-/*
- * PCM callbacks
- */
static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
@@ -515,10 +490,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
-
- return 0;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
}
static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
@@ -526,92 +499,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->dacs[0]);
-
- return 0;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
}
/*
* Digital out
*/
-static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format);
-
- return 0;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
-static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->dig_out);
-
- return 0;
-}
-
-/*
- * Analog capture
- */
-static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_setup_stream(codec, spec->adcs[substream->number],
- stream_tag, 0, format);
-
- return 0;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
}
-static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->adcs[substream->number]);
-
- return 0;
+ return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
}
-/*
- * Digital capture
- */
-static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format);
-
- return 0;
-}
-
-static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->dig_in);
-
- return 0;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
/*
@@ -621,6 +547,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
.channels_min = 2,
.channels_max = 2,
.ops = {
+ .open = ca0132_playback_pcm_open,
.prepare = ca0132_playback_pcm_prepare,
.cleanup = ca0132_playback_pcm_cleanup
},
@@ -630,10 +557,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
- .ops = {
- .prepare = ca0132_capture_pcm_prepare,
- .cleanup = ca0132_capture_pcm_cleanup
- },
};
static struct hda_pcm_stream ca0132_pcm_digital_playback = {
@@ -641,6 +564,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = {
.channels_min = 2,
.channels_max = 2,
.ops = {
+ .open = ca0132_dig_playback_pcm_open,
+ .close = ca0132_dig_playback_pcm_close,
.prepare = ca0132_dig_playback_pcm_prepare,
.cleanup = ca0132_dig_playback_pcm_cleanup
},
@@ -650,10 +575,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
- .ops = {
- .prepare = ca0132_dig_capture_pcm_prepare,
- .cleanup = ca0132_dig_capture_pcm_cleanup
- },
};
static int ca0132_build_pcms(struct hda_codec *codec)
@@ -928,18 +849,16 @@ static int ca0132_build_controls(struct hda_codec *codec)
spec->dig_out);
if (err < 0)
return err;
- err = add_out_volume(codec, spec->dig_out, "IEC958");
+ err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
if (err < 0)
return err;
+ /* spec->multiout.share_spdif = 1; */
}
if (spec->dig_in) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
if (err < 0)
return err;
- err = add_in_volume(codec, spec->dig_in, "IEC958");
- if (err < 0)
- return err;
}
return 0;
}
@@ -961,6 +880,9 @@ static void ca0132_config(struct hda_codec *codec)
struct ca0132_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
+ codec->pcm_format_first = 1;
+ codec->no_sticky_stream = 1;
+
/* line-outs */
cfg->line_outs = 1;
cfg->line_out_pins[0] = 0x0b; /* front */
@@ -988,14 +910,24 @@ static void ca0132_config(struct hda_codec *codec)
/* Mic-in */
spec->input_pins[0] = 0x12;
- spec->input_labels[0] = "Mic-In";
+ spec->input_labels[0] = "Mic";
spec->adcs[0] = 0x07;
/* Line-In */
spec->input_pins[1] = 0x11;
- spec->input_labels[1] = "Line-In";
+ spec->input_labels[1] = "Line";
spec->adcs[1] = 0x08;
spec->num_inputs = 2;
+
+ /* SPDIF I/O */
+ spec->dig_out = 0x05;
+ spec->multiout.dig_out_nid = spec->dig_out;
+ cfg->dig_out_pins[0] = 0x0c;
+ cfg->dig_outs = 1;
+ cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+ spec->dig_in = 0x09;
+ cfg->dig_in_pin = 0x0e;
+ cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
}
static void ca0132_init_chip(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 14361184ae1e..5e22a8f43d2e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2967,12 +2967,10 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
};
static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO),
SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
- SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
@@ -2988,14 +2986,10 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
- SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
- SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
{}
};
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 69b928449789..8f23374fa642 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -877,8 +877,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
struct hdmi_eld *eld;
struct hdmi_spec_per_cvt *per_cvt = NULL;
- hinfo->nid = 0; /* clear the leftover value */
-
/* Validate hinfo */
pin_idx = hinfo_to_pin_index(spec, hinfo);
if (snd_BUG_ON(pin_idx < 0))
@@ -1163,6 +1161,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
}
+static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+ return 0;
+}
+
static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
@@ -1202,6 +1208,7 @@ static const struct hda_pcm_ops generic_ops = {
.open = hdmi_pcm_open,
.close = hdmi_pcm_close,
.prepare = generic_hdmi_playback_pcm_prepare,
+ .cleanup = generic_hdmi_playback_pcm_cleanup,
};
static int generic_hdmi_build_pcms(struct hda_codec *codec)
@@ -1220,7 +1227,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
pstr->substreams = 1;
pstr->ops = generic_ops;
- pstr->nid = 1; /* FIXME: just for avoiding a debug WARNING */
/* other pstr fields are set in open */
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 344b221d2102..4f81dd44c837 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6099,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = {
[ALC269_FIXUP_PCM_44K] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc269_fixup_pcm_44k,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_QUANTA_MUTE
},
[ALC269_FIXUP_STEREO_DMIC] = {
.type = ALC_FIXUP_FUNC,
@@ -6206,9 +6208,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
- SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
- SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
+ SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
#if 0
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 94040ccf8e8f..6f806d3e56bb 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec)
unsigned int gpio;
int i;
- snd_hda_sequence_write(codec, spec->init);
+ if (spec->init)
+ snd_hda_sequence_write(codec, spec->init);
/* power down adcs initially */
if (spec->powerdown_adcs)
@@ -4542,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec,
struct auto_pin_cfg *cfg = &spec->autocfg;
int i;
+ if (cfg->speaker_outs == 0)
+ return;
+
for (i = 0; i < cfg->line_outs; i++) {
if (presence)
break;
@@ -5530,6 +5534,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e);
}
+ codec->epss = 0; /* longer delay needed for D3 */
codec->no_trigger_sense = 1;
codec->spec = spec;
@@ -5748,7 +5753,6 @@ again:
/* fallthru */
case 0x111d76b4: /* 6 Port without Analog Mixer */
case 0x111d76b5:
- spec->init = stac92hd71bxx_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd71bxx_dmic_nids,
@@ -5773,7 +5777,6 @@ again:
spec->stream_delay = 40; /* 40 milliseconds */
/* disable VSW */
- spec->init = stac92hd71bxx_core_init;
unmute_init++;
snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
@@ -5788,7 +5791,6 @@ again:
/* fallthru */
default:
- spec->init = stac92hd71bxx_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd71bxx_dmic_nids,
@@ -5796,6 +5798,9 @@ again:
break;
}
+ if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB)
+ spec->init = stac92hd71bxx_core_init;
+
if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
snd_hda_sequence_write_cache(codec, unmute_init);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 80d90cb42853..430771776915 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
vt1708_stop_hp_work(spec);
+
+ if (spec->codec_type == VT1802) {
+ /* Fix pop noise on headphones */
+ int i;
+ for (i = 0; i < spec->autocfg.hp_outs; i++)
+ snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0);
+ }
+
return 0;
}
#endif
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d1ab43706735..5579b08bb35b 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip)
/* hardcoded device name & channel count */
err = snd_pcm_new(chip->card, (char *)card_name, 0,
1, 1, &pcm);
+ if (err < 0)
+ return err;
pcm->private_data = chip;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index b8ac8710f47f..b12308b5ba2a 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
snd_printk(KERN_ERR "HDSPM: "
"unable to kmalloc Mixer memory of %d Bytes\n",
(int)sizeof(struct hdspm_mixer));
- return err;
+ return -ENOMEM;
}
hdspm->port_names_in = NULL;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 512434efcc31..805ab6e9a78f 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card,
if (rc)
goto error_out_cleanup;
- if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
- sis)) {
+ rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
+ sis);
+ if (rc) {
dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq);
goto error_out_cleanup;
}
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index f5ceb6f282de..210cafe04890 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr)
return 0;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int snd_pmac_driver_suspend(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 1aa52eff526a..9b18b5243a56 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
GFP_KERNEL);
if (!the_card.null_buffer_start_vaddr) {
pr_info("%s: nullbuffer alloc failed\n", __func__);
+ ret = -ENOMEM;
goto clean_preallocate;
}
pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__,
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index 318c5ba5360f..dfb744381c42 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create);
void sport_delete(struct sport_device *sport)
{
+ if (sport->tx_desc)
+ dma_free_coherent(NULL, sport->tx_desc_size,
+ sport->tx_desc, 0);
+ if (sport->rx_desc)
+ dma_free_coherent(NULL, sport->rx_desc_size,
+ sport->rx_desc, 0);
sport_free_resource(sport);
+ kfree(sport);
}
EXPORT_SYMBOL(sport_delete);
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 3c795921c5f6..23b40186f9b8 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
/* Setup AB8500 according to board-settings */
pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent);
+
+ /* Inform SoC Core that we have our own I/O arrangements. */
+ codec->control_data = (void *)true;
+
status = ab8500_audio_setup_mics(codec, &pdata->codec->amics);
if (status < 0) {
pr_err("%s: Failed to setup mics (%d)!\n", __func__, status);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 8c39dddd7d00..11b1b714b8b5 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec)
printk(KERN_INFO "AD1980 SoC Audio Codec\n");
+ codec->control_data = codec; /* we don't use regmap! */
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0) {
printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 6276e352125f..8f726c063f42 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec)
{
struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+ codec->control_data = priv->mc13xxx;
+
mc13xxx_lock(priv->mc13xxx);
/* these are the reset values */
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 8af6a5245b18..df2f99d1d428 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = {
{"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */
{"LO", NULL, "DAC"}, /* dac --> line_out */
+ {"LINE_IN", NULL, "VAG_POWER"},
{"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */
{"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */
@@ -1357,8 +1358,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
if (ret)
goto err;
- snd_soc_dapm_new_widgets(&codec->dapm);
-
return 0;
err:
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 982e437799a8..33c0f3d39c87 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec)
printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
+ codec->control_data = codec; /* we don't use regmap! */
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0)
goto codec_err;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 6537f16d383e..e33d327396ad 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
-SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
- ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
@@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
@@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
NULL, 0),
SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
NULL, 0),
-SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
- NULL, 0),
-SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
- NULL, 0),
SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
NULL, 0),
@@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
-ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
-ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
@@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
{ name, "EQ4", "EQ4" }, \
{ name, "DRC1L", "DRC1L" }, \
{ name, "DRC1R", "DRC1R" }, \
- { name, "DRC2L", "DRC2L" }, \
- { name, "DRC2R", "DRC2R" }, \
{ name, "LHPF1", "LHPF1" }, \
{ name, "LHPF2", "LHPF2" }, \
{ name, "LHPF3", "LHPF3" }, \
@@ -639,6 +625,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
{ "AIF2 Capture", NULL, "SYSCLK" },
{ "AIF3 Capture", NULL, "SYSCLK" },
+ { "IN1L PGA", NULL, "IN1L" },
+ { "IN1R PGA", NULL, "IN1R" },
+
+ { "IN2L PGA", NULL, "IN2L" },
+ { "IN2R PGA", NULL, "IN2R" },
+
+ { "IN3L PGA", NULL, "IN3L" },
+ { "IN3R PGA", NULL, "IN3R" },
+
ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
@@ -675,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
- ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
- ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 8033f7065189..01ebbcc5c6a4 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
{ "AIF2 Capture", NULL, "SYSCLK" },
{ "AIF3 Capture", NULL, "SYSCLK" },
+ { "IN1L PGA", NULL, "IN1L" },
+ { "IN1R PGA", NULL, "IN1R" },
+
+ { "IN2L PGA", NULL, "IN2L" },
+ { "IN2R PGA", NULL, "IN2R" },
+
+ { "IN3L PGA", NULL, "IN3L" },
+ { "IN3R PGA", NULL, "IN3R" },
+
+ { "IN4L PGA", NULL, "IN4L" },
+ { "IN4R PGA", NULL, "IN4R" },
+
ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index eaf65863ec21..ce6720073798 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
/* VMID 2*250k */
snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
WM8962_VMID_SEL_MASK, 0x100);
+
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ msleep(100);
break;
case SND_SOC_BIAS_OFF:
@@ -3730,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev)
regcache_sync(wm8962->regmap);
- regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP,
- WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA,
- WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA);
-
- /* Bias enable at 2*50k for ramp */
- regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
- WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA,
- WM8962_BIAS_ENA | 0x180);
-
- msleep(5);
-
- /* VMID back to 2x250k for standby */
- regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
- WM8962_VMID_SEL_MASK, 0x100);
-
return 0;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index bb62f4b3d563..6c9eeca85b95 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- bclk_rate = params_rate(params) * 2;
+ bclk_rate = params_rate(params) * 4;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
bclk_rate *= 16;
@@ -3253,10 +3253,13 @@ static void wm8994_mic_work(struct work_struct *work)
int ret;
int report;
+ pm_runtime_get_sync(dev);
+
ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, &reg);
if (ret < 0) {
dev_err(dev, "Failed to read microphone status: %d\n",
ret);
+ pm_runtime_put(dev);
return;
}
@@ -3299,6 +3302,8 @@ static void wm8994_mic_work(struct work_struct *work)
snd_soc_jack_report(priv->micdet[1].jack, report,
SND_JACK_HEADSET | SND_JACK_BTN_0);
+
+ pm_runtime_put(dev);
}
static irqreturn_t wm8994_mic_irq(int irq, void *data)
@@ -3421,12 +3426,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
int reg;
bool present;
+ pm_runtime_get_sync(codec->dev);
+
mutex_lock(&wm8994->accdet_lock);
reg = snd_soc_read(codec, WM1811_JACKDET_CTRL);
if (reg < 0) {
dev_err(codec->dev, "Failed to read jack status: %d\n", reg);
mutex_unlock(&wm8994->accdet_lock);
+ pm_runtime_put(codec->dev);
return IRQ_NONE;
}
@@ -3491,6 +3499,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
SND_JACK_MECHANICAL | SND_JACK_HEADSET |
wm8994->btn_mask);
+ pm_runtime_put(codec->dev);
return IRQ_HANDLED;
}
@@ -3602,6 +3611,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA))
return IRQ_HANDLED;
+ pm_runtime_get_sync(codec->dev);
+
/* We may occasionally read a detection without an impedence
* range being provided - if that happens loop again.
*/
@@ -3612,6 +3623,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
dev_err(codec->dev,
"Failed to read mic detect status: %d\n",
reg);
+ pm_runtime_put(codec->dev);
return IRQ_NONE;
}
@@ -3639,6 +3651,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
dev_warn(codec->dev, "Accessory detection with no callback\n");
out:
+ pm_runtime_put(codec->dev);
return IRQ_HANDLED;
}
@@ -4025,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
break;
case WM8958:
if (wm8994->revision < 1) {
+ snd_soc_dapm_add_routes(dapm, wm8994_intercon,
+ ARRAY_SIZE(wm8994_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
ARRAY_SIZE(wm8994_revd_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 099e6ec32125..c6d2076a796b 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
-SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv),
@@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]);
/* Mic select */
static const struct snd_kcontrol_new wm9712_mic_src_controls =
-SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
/* diff select */
static const struct snd_kcontrol_new wm9712_diff_sel_controls =
@@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectl_controls),
SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectr_controls),
-SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
&wm9712_mic_src_controls),
SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
&wm9712_diff_sel_controls),
@@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
SND_SOC_DAPM_OUTPUT("MONOOUT"),
SND_SOC_DAPM_OUTPUT("HPOUTL"),
@@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = {
{"Mic PGA", NULL, "MIC1"},
{"Mic PGA", NULL, "MIC2"},
+ /* microphones */
+ {"Differential Mic", NULL, "MIC1"},
+ {"Differential Mic", NULL, "MIC2"},
+ {"Left Mic Select Source", "Mic 1", "MIC1"},
+ {"Left Mic Select Source", "Mic 2", "MIC2"},
+ {"Left Mic Select Source", "Stereo", "MIC1"},
+ {"Left Mic Select Source", "Differential", "Differential Mic"},
+ {"Right Mic Select Source", "Mic 1", "MIC1"},
+ {"Right Mic Select Source", "Mic 2", "MIC2"},
+ {"Right Mic Select Source", "Stereo", "MIC2"},
+ {"Right Mic Select Source", "Differential", "Differential Mic"},
+
/* left capture selector */
{"Left Capture Select", "Mic", "MIC1"},
{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
@@ -619,6 +634,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec)
{
int ret = 0;
+ codec->control_data = codec; /* we don't use regmap! */
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0) {
printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 3eb19fb71d17..d0b8a3287a85 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec)
if (wm9713 == NULL)
return -ENOMEM;
snd_soc_codec_set_drvdata(codec, wm9713);
+ codec->control_data = wm9713; /* we don't use regmap! */
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 95441bfc8190..ce5e5cd254dd 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
{
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (dev->txnumevt) /* enable FIFO */
+ if (dev->txnumevt) { /* enable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
FIFO_ENABLE);
+ }
mcasp_start_tx(dev);
} else {
- if (dev->rxnumevt) /* enable FIFO */
+ if (dev->rxnumevt) { /* enable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
FIFO_ENABLE);
+ }
mcasp_start_rx(dev);
}
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 28dd76c7cb1c..81d7728cf67f 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
static struct snd_soc_dai_driver imx_ssi_dai = {
.probe = imx_ssi_dai_probe,
.playback = {
- .channels_min = 1,
+ /* The SSI does not support monaural audio. */
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
index 99a997f19bb9..b6fa77678d97 100644
--- a/sound/soc/mxs/Kconfig
+++ b/sound/soc/mxs/Kconfig
@@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC
if SND_MXS_SOC
config SND_SOC_MXS_SGTL5000
- tristate "SoC Audio support for i.MX boards with sgtl5000"
+ tristate "SoC Audio support for MXS boards with sgtl5000"
depends on I2C
select SND_SOC_SGTL5000
help
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index aba71bfa33b1..b3030718c228 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+ struct mxs_saif *master_saif;
u32 scr, stat;
int ret;
+ master_saif = mxs_saif_get_master(saif);
+ if (!master_saif)
+ return -EINVAL;
+
/* mclk should already be set */
if (!saif->mclk && saif->mclk_in_use) {
dev_err(cpu_dai->dev, "set mclk first\n");
@@ -420,6 +425,25 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+ /* prepare clk in hw_param, enable in trigger */
+ clk_prepare(saif->clk);
+ if (saif != master_saif) {
+ /*
+ * Set an initial clock rate for the saif internal logic to work
+ * properly. This is important when working in EXTMASTER mode
+ * that uses the other saif's BITCLK&LRCLK but it still needs a
+ * basic clock which should be fast enough for the internal
+ * logic.
+ */
+ clk_enable(saif->clk);
+ ret = clk_set_rate(saif->clk, 24000000);
+ clk_disable(saif->clk);
+ if (ret)
+ return ret;
+
+ clk_prepare(master_saif->clk);
+ }
+
scr = __raw_readl(saif->base + SAIF_CTRL);
scr &= ~BM_SAIF_CTRL_WORD_LENGTH;
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 34835e8a9160..d33c48baaf71 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux)
{
const char *signal, *src;
- if (mcbsp->pdata->mux_signal)
+ if (!mcbsp->pdata->mux_signal)
return -EINVAL;
switch (mux) {
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 1046083e90a0..acdd3ef14e08 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver);
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
MODULE_DESCRIPTION("OMAP I2S SoC Interface");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-mcbsp");
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 5a649da9122a..f0feb06615f8 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver);
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-pcm-audio");
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index b7b2a1f91425..89b064650f14 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -20,7 +20,7 @@
#include <sound/pcm_params.h>
#include <plat/audio.h>
-#include <plat/dma.h>
+#include <mach/dma.h>
#include "dma.h"
#include "pcm.h"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f219b2f7ee68..c501af6d8dbe 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
}
if (!rtd->cpu_dai) {
- dev_dbg(card->dev, "CPU DAI %s not registered\n",
+ dev_err(card->dev, "CPU DAI %s not registered\n",
dai_link->cpu_dai_name);
return -EPROBE_DEFER;
}
@@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
}
if (!rtd->codec_dai) {
- dev_dbg(card->dev, "CODEC DAI %s not registered\n",
+ dev_err(card->dev, "CODEC DAI %s not registered\n",
dai_link->codec_dai_name);
return -EPROBE_DEFER;
}
}
if (!rtd->codec) {
- dev_dbg(card->dev, "CODEC %s not registered\n",
+ dev_err(card->dev, "CODEC %s not registered\n",
dai_link->codec_name);
return -EPROBE_DEFER;
}
@@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
rtd->platform = platform;
}
if (!rtd->platform) {
- dev_dbg(card->dev, "platform %s not registered\n",
+ dev_err(card->dev, "platform %s not registered\n",
dai_link->platform_name);
return -EPROBE_DEFER;
}
@@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card,
}
/* If the driver didn't set I/O up try regmap */
- if (!codec->control_data)
+ if (!codec->write && dev_get_regmap(codec->dev, NULL))
snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
if (driver->controls)
@@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num)
return 0;
}
+ dev_err(card->dev, "%s not registered\n", aux_dev->codec_name);
+
return -EPROBE_DEFER;
}
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7f8b3b7428bb..0c172938b82a 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
}
/* Report before the DAPM sync to help users updating micbias status */
- blocking_notifier_call_chain(&jack->notifier, status, jack);
+ blocking_notifier_call_chain(&jack->notifier, jack->status, jack);
snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index d684df294c0c..e463529b38bb 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -177,7 +177,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
}
alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
- if (alc5632->gpio_hp_det == -ENODEV)
+ if (alc5632->gpio_hp_det == -EPROBE_DEFER)
return -EPROBE_DEFER;
ret = snd_soc_of_parse_card_name(card, "nvidia,model");
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0c5bb33d258e..d4f14e492341 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
} else if (np) {
pdata->gpio_spkr_en = of_get_named_gpio(np,
"nvidia,spkr-en-gpios", 0);
- if (pdata->gpio_spkr_en == -ENODEV)
+ if (pdata->gpio_spkr_en == -EPROBE_DEFER)
return -EPROBE_DEFER;
pdata->gpio_hp_mute = of_get_named_gpio(np,
"nvidia,hp-mute-gpios", 0);
- if (pdata->gpio_hp_mute == -ENODEV)
+ if (pdata->gpio_hp_mute == -EPROBE_DEFER)
return -EPROBE_DEFER;
pdata->gpio_hp_det = of_get_named_gpio(np,
"nvidia,hp-det-gpios", 0);
- if (pdata->gpio_hp_det == -ENODEV)
+ if (pdata->gpio_hp_det == -EPROBE_DEFER)
return -EPROBE_DEFER;
pdata->gpio_int_mic_en = of_get_named_gpio(np,
"nvidia,int-mic-en-gpios", 0);
- if (pdata->gpio_int_mic_en == -ENODEV)
+ if (pdata->gpio_int_mic_en == -EPROBE_DEFER)
return -EPROBE_DEFER;
pdata->gpio_ext_mic_en = of_get_named_gpio(np,
"nvidia,ext-mic-en-gpios", 0);
- if (pdata->gpio_ext_mic_en == -ENODEV)
+ if (pdata->gpio_ext_mic_en == -EPROBE_DEFER)
return -EPROBE_DEFER;
}
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index 62ac0285bfaf..057e28ef770e 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -21,7 +21,7 @@
#include <linux/mfd/dbx500-prcmu.h>
#include <mach/hardware.h>
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
#include <sound/soc.h>
#include <sound/soc-dai.h>
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index ee14d2dac2f5..5c472f335a64 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -19,7 +19,7 @@
#include <linux/slab.h>
#include <mach/hardware.h>
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
#include <sound/soc.h>
diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h
index 7f71b4a0d4bc..2d9136da9865 100644
--- a/sound/soc/ux500/ux500_msp_i2s.h
+++ b/sound/soc/ux500/ux500_msp_i2s.h
@@ -17,7 +17,7 @@
#include <linux/platform_device.h>
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
#define MSP_INPUT_FREQ_APB 48000000
diff --git a/sound/usb/card.c b/sound/usb/card.c
index d5b5c3388e28..4a469f0cb6d4 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev,
struct snd_usb_audio *chip)
{
struct snd_card *card;
- struct list_head *p;
+ struct list_head *p, *n;
if (chip == (void *)-1L)
return;
@@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev,
snd_usb_stream_disconnect(p);
}
/* release the endpoint resources */
- list_for_each(p, &chip->ep_list) {
+ list_for_each_safe(p, n, &chip->ep_list) {
snd_usb_endpoint_free(p);
}
/* release the midi resources */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 0f647d22cb4a..d6e2bb49c59c 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -141,7 +141,7 @@ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep)
*
* For implicit feedback, next_packet_size() is unused.
*/
-static int next_packet_size(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
{
unsigned long flags;
int ret;
@@ -177,15 +177,6 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep,
ep->retire_data_urb(ep->data_subs, urb);
}
-static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep,
- struct snd_urb_ctx *ctx)
-{
- int i;
-
- for (i = 0; i < ctx->packets; ++i)
- ctx->packet_size[i] = next_packet_size(ep);
-}
-
/*
* Prepare a PLAYBACK urb for submission to the bus.
*/
@@ -370,7 +361,6 @@ static void snd_complete_urb(struct urb *urb)
goto exit_clear;
}
- prepare_outbound_urb_sizes(ep, ctx);
prepare_outbound_urb(ep, ctx);
} else {
retire_inbound_urb(ep, ctx);
@@ -799,7 +789,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
/**
* snd_usb_endpoint_start: start an snd_usb_endpoint
*
- * @ep: the endpoint to start
+ * @ep: the endpoint to start
+ * @can_sleep: flag indicating whether the operation is executed in
+ * non-atomic context
*
* A call to this function will increment the use count of the endpoint.
* In case it is not already running, the URBs for this endpoint will be
@@ -809,7 +801,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
*
* Returns an error if the URB submission failed, 0 in all other cases.
*/
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep)
{
int err;
unsigned int i;
@@ -822,8 +814,9 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
return 0;
/* just to be sure */
- deactivate_urbs(ep, 0, 1);
- wait_clear_urbs(ep);
+ deactivate_urbs(ep, 0, can_sleep);
+ if (can_sleep)
+ wait_clear_urbs(ep);
ep->active_mask = 0;
ep->unlink_mask = 0;
@@ -854,7 +847,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
goto __error;
if (usb_pipeout(ep->pipe)) {
- prepare_outbound_urb_sizes(ep, urb->context);
prepare_outbound_urb(ep, urb->context);
} else {
prepare_inbound_urb(ep, urb->context);
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index ee2723fb174f..cbbbdf226d66 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -13,7 +13,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
struct audioformat *fmt,
struct snd_usb_endpoint *sync_ep);
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep);
void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
int force, int can_sleep, int wait);
int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
@@ -21,6 +21,7 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep);
void snd_usb_endpoint_free(struct list_head *head);
int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep);
void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
struct snd_usb_endpoint *sender,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a1298f379428..fd5e982fc98c 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -212,7 +212,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
}
}
-static int start_endpoints(struct snd_usb_substream *subs)
+static int start_endpoints(struct snd_usb_substream *subs, int can_sleep)
{
int err;
@@ -225,7 +225,7 @@ static int start_endpoints(struct snd_usb_substream *subs)
snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep);
ep->data_subs = subs;
- err = snd_usb_endpoint_start(ep);
+ err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
return err;
@@ -236,10 +236,25 @@ static int start_endpoints(struct snd_usb_substream *subs)
!test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) {
struct snd_usb_endpoint *ep = subs->sync_endpoint;
+ if (subs->data_endpoint->iface != subs->sync_endpoint->iface ||
+ subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) {
+ err = usb_set_interface(subs->dev,
+ subs->sync_endpoint->iface,
+ subs->sync_endpoint->alt_idx);
+ if (err < 0) {
+ snd_printk(KERN_ERR
+ "%d:%d:%d: cannot set interface (%d)\n",
+ subs->dev->devnum,
+ subs->sync_endpoint->iface,
+ subs->sync_endpoint->alt_idx, err);
+ return -EIO;
+ }
+ }
+
snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep);
ep->sync_slave = subs->data_endpoint;
- err = snd_usb_endpoint_start(ep);
+ err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
return err;
@@ -547,7 +562,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
/* for playback, submit the URBs now; otherwise, the first hwptr_done
* updates for all URBs would happen at the same time when starting */
if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
- return start_endpoints(subs);
+ return start_endpoints(subs, 1);
return 0;
}
@@ -1029,6 +1044,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
struct urb *urb)
{
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ struct snd_usb_endpoint *ep = subs->data_endpoint;
struct snd_urb_ctx *ctx = urb->context;
unsigned int counts, frames, bytes;
int i, stride, period_elapsed = 0;
@@ -1040,7 +1056,11 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
urb->number_of_packets = 0;
spin_lock_irqsave(&subs->lock, flags);
for (i = 0; i < ctx->packets; i++) {
- counts = ctx->packet_size[i];
+ if (ctx->packet_size[i])
+ counts = ctx->packet_size[i];
+ else
+ counts = snd_usb_endpoint_next_packet_size(ep);
+
/* set up descriptor */
urb->iso_frame_desc[i].offset = frames * stride;
urb->iso_frame_desc[i].length = counts * stride;
@@ -1091,7 +1111,16 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
subs->hwptr_done += bytes;
if (subs->hwptr_done >= runtime->buffer_size * stride)
subs->hwptr_done -= runtime->buffer_size * stride;
+
+ /* update delay with exact number of samples queued */
+ runtime->delay = subs->last_delay;
runtime->delay += frames;
+ subs->last_delay = runtime->delay;
+
+ /* realign last_frame_number */
+ subs->last_frame_number = usb_get_current_frame_number(subs->dev);
+ subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
+
spin_unlock_irqrestore(&subs->lock, flags);
urb->transfer_buffer_length = bytes;
if (period_elapsed)
@@ -1109,12 +1138,26 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
int stride = runtime->frame_bits >> 3;
int processed = urb->transfer_buffer_length / stride;
+ int est_delay;
spin_lock_irqsave(&subs->lock, flags);
- if (processed > runtime->delay)
- runtime->delay = 0;
+ est_delay = snd_usb_pcm_delay(subs, runtime->rate);
+ /* update delay with exact number of samples played */
+ if (processed > subs->last_delay)
+ subs->last_delay = 0;
else
- runtime->delay -= processed;
+ subs->last_delay -= processed;
+ runtime->delay = subs->last_delay;
+
+ /*
+ * Report when delay estimate is off by more than 2ms.
+ * The error should be lower than 2ms since the estimate relies
+ * on two reads of a counter updated every ms.
+ */
+ if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
+ snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
+ est_delay, subs->last_delay);
+
spin_unlock_irqrestore(&subs->lock, flags);
}
@@ -1172,7 +1215,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- err = start_endpoints(subs);
+ err = start_endpoints(subs, 0);
if (err < 0)
return err;