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-rw-r--r--arch/arm/plat-pxa/ssp.c1
-rw-r--r--include/linux/pxa2xx_ssp.h2
-rw-r--r--include/linux/spi/pxa2xx_spi.h2
-rw-r--r--sound/pci/hda/hda_codec.c46
-rw-r--r--sound/pci/hda/hda_codec.h2
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/soc/codecs/wm8904.c26
-rw-r--r--sound/soc/codecs/wm8996.c8
-rw-r--r--sound/soc/pxa/pxa-ssp.c38
-rw-r--r--sound/soc/tegra/tegra_wm8903.c13
-rw-r--r--sound/usb/6fire/firmware.c2
-rw-r--r--sound/usb/pcm.c21
12 files changed, 97 insertions, 66 deletions
diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c
index 58b79809d20c..584c9bf8ed2d 100644
--- a/arch/arm/plat-pxa/ssp.c
+++ b/arch/arm/plat-pxa/ssp.c
@@ -193,6 +193,7 @@ static const struct platform_device_id ssp_id_table[] = {
{ "pxa25x-nssp", PXA25x_NSSP },
{ "pxa27x-ssp", PXA27x_SSP },
{ "pxa168-ssp", PXA168_SSP },
+ { "pxa910-ssp", PXA910_SSP },
{ },
};
diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h
index 44835fb39793..f36632061c66 100644
--- a/include/linux/pxa2xx_ssp.h
+++ b/include/linux/pxa2xx_ssp.h
@@ -160,7 +160,9 @@ enum pxa_ssp_type {
PXA25x_SSP, /* pxa 210, 250, 255, 26x */
PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */
PXA27x_SSP,
+ PXA3xx_SSP,
PXA168_SSP,
+ PXA910_SSP,
CE4100_SSP,
};
diff --git a/include/linux/spi/pxa2xx_spi.h b/include/linux/spi/pxa2xx_spi.h
index d3e1075f7b60..c73d1445c77e 100644
--- a/include/linux/spi/pxa2xx_spi.h
+++ b/include/linux/spi/pxa2xx_spi.h
@@ -43,7 +43,7 @@ struct pxa2xx_spi_chip {
void (*cs_control)(u32 command);
};
-#ifdef CONFIG_ARCH_PXA
+#if defined(CONFIG_ARCH_PXA) || defined(CONFIG_ARCH_MMP)
#include <linux/clk.h>
#include <mach/dma.h>
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 41ca803a1fff..7504e62188d6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -4393,20 +4393,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec)
codec->power_jiffies += delta;
}
-/**
- * snd_hda_power_up - Power-up the codec
- * @codec: HD-audio codec
- *
- * Increment the power-up counter and power up the hardware really when
- * not turned on yet.
- */
-void snd_hda_power_up(struct hda_codec *codec)
+/* Transition to powered up, if wait_power_down then wait for a pending
+ * transition to D3 to complete. A pending D3 transition is indicated
+ * with power_transition == -1. */
+static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
{
struct hda_bus *bus = codec->bus;
spin_lock(&codec->power_lock);
codec->power_count++;
- if (codec->power_on || codec->power_transition > 0) {
+ /* Return if power_on or transitioning to power_on, unless currently
+ * powering down. */
+ if ((codec->power_on || codec->power_transition > 0) &&
+ !(wait_power_down && codec->power_transition < 0)) {
spin_unlock(&codec->power_lock);
return;
}
@@ -4430,8 +4429,37 @@ void snd_hda_power_up(struct hda_codec *codec)
codec->power_transition = 0;
spin_unlock(&codec->power_lock);
}
+
+/**
+ * snd_hda_power_up - Power-up the codec
+ * @codec: HD-audio codec
+ *
+ * Increment the power-up counter and power up the hardware really when
+ * not turned on yet.
+ */
+void snd_hda_power_up(struct hda_codec *codec)
+{
+ __snd_hda_power_up(codec, false);
+}
EXPORT_SYMBOL_HDA(snd_hda_power_up);
+/**
+ * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending
+ * D3 transition to complete. This differs from snd_hda_power_up() when
+ * power_transition == -1. snd_hda_power_up sees this case as a nop,
+ * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers
+ * back up.
+ * @codec: HD-audio codec
+ *
+ * Cancel any power down operation hapenning on the work queue, then power up.
+ */
+void snd_hda_power_up_d3wait(struct hda_codec *codec)
+{
+ /* This will cancel and wait for pending power_work to complete. */
+ __snd_hda_power_up(codec, true);
+}
+EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait);
+
#define power_save(codec) \
((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 4fc3960c8591..2fdaadbb4326 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -1056,10 +1056,12 @@ const char *snd_hda_get_jack_location(u32 cfg);
*/
#ifdef CONFIG_SND_HDA_POWER_SAVE
void snd_hda_power_up(struct hda_codec *codec);
+void snd_hda_power_up_d3wait(struct hda_codec *codec);
void snd_hda_power_down(struct hda_codec *codec);
void snd_hda_update_power_acct(struct hda_codec *codec);
#else
static inline void snd_hda_power_up(struct hda_codec *codec) {}
+static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {}
static inline void snd_hda_power_down(struct hda_codec *codec) {}
#endif
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 02763827dde0..7757536b9d5f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1766,7 +1766,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
buff_step);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
buff_step);
- snd_hda_power_up(apcm->codec);
+ snd_hda_power_up_d3wait(apcm->codec);
err = hinfo->ops.open(hinfo, apcm->codec, substream);
if (err < 0) {
azx_release_device(azx_dev);
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 65d525d74c54..812acd83fb48 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
+ regcache_cache_only(wm8904->regmap, false);
regcache_sync(wm8904->regmap);
/* Enable bias */
@@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
WM8904_BIAS_ENA, 0);
-#ifdef CONFIG_REGULATOR
- /* Post 2.6.34 we will be able to get a callback when
- * the regulators are disabled which we can use but
- * for now just assume that the power will be cut if
- * the regulator API is in use.
- */
- codec->cache_sync = 1;
-#endif
+ regcache_cache_only(wm8904->regmap, true);
+ regcache_mark_dirty(wm8904->regmap);
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
@@ -2084,10 +2079,8 @@ static int wm8904_probe(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
- u16 *reg_cache = codec->reg_cache;
int ret, i;
- codec->cache_sync = 1;
codec->control_data = wm8904->regmap;
switch (wm8904->devtype) {
@@ -2150,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec)
goto err_enable;
}
+ regcache_cache_only(wm8904->regmap, true);
/* Change some default settings - latch VU and enable ZC */
snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
WM8904_ADC_VU, WM8904_ADC_VU);
@@ -2180,14 +2174,18 @@ static int wm8904_probe(struct snd_soc_codec *codec)
if (!pdata->gpio_cfg[i])
continue;
- reg_cache[WM8904_GPIO_CONTROL_1 + i]
- = pdata->gpio_cfg[i] & 0xffff;
+ regmap_update_bits(wm8904->regmap,
+ WM8904_GPIO_CONTROL_1 + i,
+ 0xffff,
+ pdata->gpio_cfg[i]);
}
/* Zero is the default value for these anyway */
for (i = 0; i < WM8904_MIC_REGS; i++)
- reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i]
- = pdata->mic_cfg[i];
+ regmap_update_bits(wm8904->regmap,
+ WM8904_MIC_BIAS_CONTROL_0 + i,
+ 0xffff,
+ pdata->mic_cfg[i]);
}
/* Set Class W by default - this will be managed by the Class
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 8af422e38fd0..dc9b42b7fc4d 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2837,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
}
}
- regcache_cache_only(codec->control_data, true);
-
/* Apply platform data settings */
snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
@@ -3051,7 +3049,6 @@ static int wm8996_remove(struct snd_soc_codec *codec)
for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
regulator_unregister_notifier(wm8996->supplies[i].consumer,
&wm8996->disable_nb[i]);
- regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
return 0;
}
@@ -3206,14 +3203,15 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
dev_info(&i2c->dev, "revision %c\n",
(reg & WM8996_CHIP_REV_MASK) + 'A');
- regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
-
ret = wm8996_reset(wm8996);
if (ret < 0) {
dev_err(&i2c->dev, "Failed to issue reset\n");
goto err_regmap;
}
+ regcache_cache_only(wm8996->regmap, true);
+ regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+
wm8996_init_gpio(wm8996);
ret = snd_soc_register_codec(&i2c->dev,
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 1c2aa7fab3fd..4da5fc55c7ee 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -33,7 +33,6 @@
#include <mach/hardware.h>
#include <mach/dma.h>
-#include <mach/audio.h>
#include "../../arm/pxa2xx-pcm.h"
#include "pxa-ssp.h"
@@ -194,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
{
u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
- if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+ if (ssp->type == PXA25x_SSP) {
sscr0 &= ~0x0000ff00;
sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
} else {
@@ -212,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
u32 div;
- if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+ if (ssp->type == PXA25x_SSP)
div = ((sscr0 >> 8) & 0xff) * 2 + 2;
else
div = ((sscr0 >> 8) & 0xfff) + 1;
@@ -242,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
break;
case PXA_SSP_CLK_PLL:
/* Internal PLL is fixed */
- if (cpu_is_pxa25x())
+ if (ssp->type == PXA25x_SSP)
priv->sysclk = 1843200;
else
priv->sysclk = 13000000;
@@ -266,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
/* The SSP clock must be disabled when changing SSP clock mode
* on PXA2xx. On PXA3xx it must be enabled when doing so. */
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
clk_disable(ssp->clk);
val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
pxa_ssp_write_reg(ssp, SSCR0, val);
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
clk_enable(ssp->clk);
return 0;
@@ -294,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
case PXA_SSP_AUDIO_DIV_SCDB:
val = pxa_ssp_read_reg(ssp, SSACD);
val &= ~SSACD_SCDB;
-#if defined(CONFIG_PXA3xx)
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
val &= ~SSACD_SCDX8;
-#endif
switch (div) {
case PXA_SSP_CLK_SCDB_1:
val |= SSACD_SCDB;
break;
case PXA_SSP_CLK_SCDB_4:
break;
-#if defined(CONFIG_PXA3xx)
case PXA_SSP_CLK_SCDB_8:
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
val |= SSACD_SCDX8;
else
return -EINVAL;
break;
-#endif
default:
return -EINVAL;
}
@@ -337,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
struct ssp_device *ssp = priv->ssp;
u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
-#if defined(CONFIG_PXA3xx)
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
pxa_ssp_write_reg(ssp, SSACDD, 0);
-#endif
switch (freq_out) {
case 5622000:
@@ -365,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
break;
default:
-#ifdef CONFIG_PXA3xx
/* PXA3xx has a clock ditherer which can be used to generate
* a wider range of frequencies - calculate a value for it.
*/
- if (cpu_is_pxa3xx()) {
+ if (ssp->type == PXA3xx_SSP) {
u32 val;
u64 tmp = 19968;
tmp *= 1000000;
@@ -386,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
val, freq_out);
break;
}
-#endif
return -EINVAL;
}
@@ -590,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
-#ifdef CONFIG_PXA3xx
- if (cpu_is_pxa3xx())
+ if (ssp->type == PXA3xx_SSP)
sscr0 |= SSCR0_FPCKE;
-#endif
sscr0 |= SSCR0_DataSize(16);
break;
case SNDRV_PCM_FORMAT_S24_LE:
@@ -618,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
* trying and failing a lot; some of the registers
* needed for that mode are only available on PXA3xx.
*/
-
-#ifdef CONFIG_PXA3xx
- if (!cpu_is_pxa3xx())
+ if (ssp->type != PXA3xx_SSP)
return -EINVAL;
sspsp |= SSPSP_SFRMWDTH(width * 2);
@@ -628,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
sspsp |= SSPSP_EDMYSTOP(3);
sspsp |= SSPSP_DMYSTOP(3);
sspsp |= SSPSP_DMYSTRT(1);
-#else
- return -EINVAL;
-#endif
} else {
/* The frame width is the width the LRCLK is
* asserted for; the delay is expressed in
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0b0df49d9d33..3b6da91188a9 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -346,6 +346,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
+static int tegra_wm8903_remove(struct snd_soc_card *card)
+{
+ struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]);
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ wm8903_mic_detect(codec, NULL, 0, 0);
+
+ return 0;
+}
+
static struct snd_soc_dai_link tegra_wm8903_dai = {
.name = "WM8903",
.stream_name = "WM8903 PCM",
@@ -363,6 +374,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = {
.dai_link = &tegra_wm8903_dai,
.num_links = 1,
+ .remove = tegra_wm8903_remove,
+
.controls = tegra_wm8903_controls,
.num_controls = ARRAY_SIZE(tegra_wm8903_controls),
.dapm_widgets = tegra_wm8903_dapm_widgets,
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 6f9715ab32fe..56ad923bf6b5 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload(
int ret;
u8 data;
struct usb_device *device = interface_to_usbdev(intf);
- const struct firmware *fw = 0;
+ const struct firmware *fw = NULL;
struct ihex_record *rec = kmalloc(sizeof(struct ihex_record),
GFP_KERNEL);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index cdf8b7601973..54607f8c4f66 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
(get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
get_endpoint(alts, 1)->bSynchAddress != 0 &&
!implicit_fb)) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
- dev->devnum, fmt->iface, fmt->altsetting);
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ get_endpoint(alts, 1)->bmAttributes,
+ get_endpoint(alts, 1)->bLength,
+ get_endpoint(alts, 1)->bSynchAddress);
return -EINVAL;
}
ep = get_endpoint(alts, 1)->bEndpointAddress;
- if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ if (!implicit_fb &&
+ get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
(( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
- (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) ||
- ( is_playback && !implicit_fb))) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
- dev->devnum, fmt->iface, fmt->altsetting);
+ (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
return -EINVAL;
}
@@ -1147,7 +1151,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
return -EINVAL;
}
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
{
int err;
struct snd_usb_substream *subs = substream->runtime->private_data;
generated by cgit v1.2.3-58-ga151 (git 2.39.0) at 2024-12-24 04:18:15 +0000