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authorMauro Carvalho Chehab <mchehab@redhat.com>2012-10-05 09:36:26 -0300
committerMauro Carvalho Chehab <mchehab@redhat.com>2012-10-05 09:36:26 -0300
commitbd0d10498826ed150da5e4c45baf8b9c7088fb71 (patch)
treecdee4371121a355d627a655c4eef5c0047b0462a /sound
parent89de0f2cda8b784e51ebd6655fff7339e4ac552b (diff)
parent2425bb3d4016ed95ce83a90b53bd92c7f31091e4 (diff)
downloadlinux-bd0d10498826ed150da5e4c45baf8b9c7088fb71.tar.bz2
Merge branch 'staging/for_v3.7' into v4l_for_linus
* staging/for_v3.7: (2891 commits) em28xx: regression fix: use DRX-K sync firmware requests on em28xx drxk: allow loading firmware synchrousnously em28xx: Make all em28xx extensions to be initialized asynchronously [media] tda18271: properly report read errors in tda18271_get_id [media] tda18271: delay IR & RF calibration until init() if delay_cal is set [media] MAINTAINERS: add Michael Krufky as tda827x maintainer [media] MAINTAINERS: add Michael Krufky as tda8290 maintainer [media] MAINTAINERS: add Michael Krufky as cxusb maintainer [media] MAINTAINERS: add Michael Krufky as lg2160 maintainer [media] MAINTAINERS: add Michael Krufky as lgdt3305 maintainer [media] MAINTAINERS: add Michael Krufky as mxl111sf maintainer [media] MAINTAINERS: add Michael Krufky as mxl5007t maintainer [media] MAINTAINERS: add Michael Krufky as tda18271 maintainer [media] s5p-tv: Report only multi-plane capabilities in vidioc_querycap [media] s5p-mfc: Fix misplaced return statement in s5p_mfc_suspend() [media] exynos-gsc: Add missing static storage class specifiers [media] exynos-gsc: Remove <linux/version.h> header file inclusion [media] s5p-fimc: Fix incorrect condition in fimc_lite_reqbufs() [media] s5p-tv: Fix potential NULL pointer dereference error [media] s5k6aa: Fix possible NULL pointer dereference ... Conflicts: drivers/media/platform/s5p-fimc/fimc-capture.c drivers/media/platform/s5p-fimc/fimc-lite.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/compress_offload.c8
-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/mpu401/mpu401_uart.c1
-rw-r--r--sound/drivers/pcsp/pcsp.c4
-rw-r--r--sound/i2c/other/tea575x-tuner.c205
-rw-r--r--sound/isa/als100.c2
-rw-r--r--sound/isa/es1688/es1688_lib.c34
-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.c85
-rw-r--r--sound/pci/hda/hda_codec.h2
-rw-r--r--sound/pci/hda/hda_intel.c11
-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.c30
-rw-r--r--sound/pci/hda/patch_sigmatel.c37
-rw-r--r--sound/pci/hda/patch_via.c15
-rw-r--r--sound/pci/ice1712/prodigy_hifi.c3
-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/arizona.c2
-rw-r--r--sound/soc/codecs/mc13783.c10
-rw-r--r--sound/soc/codecs/sgtl5000.c3
-rw-r--r--sound/soc/codecs/stac9766.c1
-rw-r--r--sound/soc/codecs/wm2000.c2
-rw-r--r--sound/soc/codecs/wm5102.c25
-rw-r--r--sound/soc/codecs/wm5110.c12
-rw-r--r--sound/soc/codecs/wm8904.c2
-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-sgtl5000.c2
-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/am3517evm.c2
-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/dma.c8
-rw-r--r--sound/soc/samsung/pcm.c2
-rw-r--r--sound/soc/soc-core.c12
-rw-r--r--sound/soc/soc-dapm.c5
-rw-r--r--sound/soc/soc-jack.c2
-rw-r--r--sound/soc/spear/spear_pcm.c2
-rw-r--r--sound/soc/tegra/tegra_alc5632.c3
-rw-r--r--sound/soc/tegra/tegra_pcm.c4
-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.c27
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.h2
-rw-r--r--sound/sound_firmware.c8
-rw-r--r--sound/usb/card.c4
-rw-r--r--sound/usb/clock.c3
-rw-r--r--sound/usb/endpoint.c32
-rw-r--r--sound/usb/endpoint.h3
-rw-r--r--sound/usb/pcm.c67
76 files changed, 696 insertions, 395 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/compress_offload.c b/sound/core/compress_offload.c
index ec2118d0e27a..eb60cb8dbb8a 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f)
int maj = imajor(inode);
int ret;
- if (f->f_flags & O_WRONLY)
+ if ((f->f_flags & O_ACCMODE) == O_WRONLY)
dirn = SND_COMPRESS_PLAYBACK;
- else if (f->f_flags & O_RDONLY)
+ else if ((f->f_flags & O_ACCMODE) == O_RDONLY)
dirn = SND_COMPRESS_CAPTURE;
- else {
- pr_err("invalid direction\n");
+ else
return -EINVAL;
- }
if (maj == snd_major)
compr = snd_lookup_minor_data(iminor(inode),
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/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 1cff331a228e..4608c2ca43f8 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -554,6 +554,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
spin_lock_init(&mpu->output_lock);
spin_lock_init(&mpu->timer_lock);
mpu->hardware = hardware;
+ mpu->irq = -1;
if (! (info_flags & MPU401_INFO_INTEGRATED)) {
int res_size = hardware == MPU401_HW_PC98II ? 4 : 2;
mpu->res = request_region(port, res_size, "MPU401 UART");
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/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index d14edb7d6484..3c6c1e3226f3 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -37,9 +37,6 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
MODULE_LICENSE("GPL");
-#define FREQ_LO ((tea->tea5759 ? 760 : 875) * 1600U)
-#define FREQ_HI ((tea->tea5759 ? 910 : 1080) * 1600U)
-
/*
* definitions
*/
@@ -50,8 +47,8 @@ MODULE_LICENSE("GPL");
#define TEA575X_BIT_BAND_MASK (3<<20)
#define TEA575X_BIT_BAND_FM (0<<20)
#define TEA575X_BIT_BAND_MW (1<<20)
-#define TEA575X_BIT_BAND_LW (1<<21)
-#define TEA575X_BIT_BAND_SW (1<<22)
+#define TEA575X_BIT_BAND_LW (2<<20)
+#define TEA575X_BIT_BAND_SW (3<<20)
#define TEA575X_BIT_PORT_0 (1<<19) /* user bit */
#define TEA575X_BIT_PORT_1 (1<<18) /* user bit */
#define TEA575X_BIT_SEARCH_MASK (3<<16) /* search level */
@@ -62,6 +59,37 @@ MODULE_LICENSE("GPL");
#define TEA575X_BIT_DUMMY (1<<15) /* buffer */
#define TEA575X_BIT_FREQ_MASK 0x7fff
+enum { BAND_FM, BAND_FM_JAPAN, BAND_AM };
+
+static const struct v4l2_frequency_band bands[] = {
+ {
+ .type = V4L2_TUNER_RADIO,
+ .index = 0,
+ .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO |
+ V4L2_TUNER_CAP_FREQ_BANDS,
+ .rangelow = 87500 * 16,
+ .rangehigh = 108000 * 16,
+ .modulation = V4L2_BAND_MODULATION_FM,
+ },
+ {
+ .type = V4L2_TUNER_RADIO,
+ .index = 0,
+ .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO |
+ V4L2_TUNER_CAP_FREQ_BANDS,
+ .rangelow = 76000 * 16,
+ .rangehigh = 91000 * 16,
+ .modulation = V4L2_BAND_MODULATION_FM,
+ },
+ {
+ .type = V4L2_TUNER_RADIO,
+ .index = 1,
+ .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_FREQ_BANDS,
+ .rangelow = 530 * 16,
+ .rangehigh = 1710 * 16,
+ .modulation = V4L2_BAND_MODULATION_AM,
+ },
+};
+
/*
* lowlevel part
*/
@@ -133,16 +161,29 @@ static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val)
if (freq == 0)
return freq;
- /* freq *= 12.5 */
- freq *= 125;
- freq /= 10;
- /* crystal fixup */
- if (tea->tea5759)
- freq += TEA575X_FMIF;
- else
+ switch (tea->band) {
+ case BAND_FM:
+ /* freq *= 12.5 */
+ freq *= 125;
+ freq /= 10;
+ /* crystal fixup */
freq -= TEA575X_FMIF;
+ break;
+ case BAND_FM_JAPAN:
+ /* freq *= 12.5 */
+ freq *= 125;
+ freq /= 10;
+ /* crystal fixup */
+ freq += TEA575X_FMIF;
+ break;
+ case BAND_AM:
+ /* crystal fixup */
+ freq -= TEA575X_AMIF;
+ break;
+ }
- return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */
+ return clamp(freq * 16, bands[tea->band].rangelow,
+ bands[tea->band].rangehigh); /* from kHz */
}
static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
@@ -150,21 +191,37 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea));
}
-static void snd_tea575x_set_freq(struct snd_tea575x *tea)
+void snd_tea575x_set_freq(struct snd_tea575x *tea)
{
- u32 freq = tea->freq;
+ u32 freq = tea->freq / 16; /* to kHz */
+ u32 band = 0;
- freq /= 16; /* to kHz */
- /* crystal fixup */
- if (tea->tea5759)
- freq -= TEA575X_FMIF;
- else
+ switch (tea->band) {
+ case BAND_FM:
+ band = TEA575X_BIT_BAND_FM;
+ /* crystal fixup */
freq += TEA575X_FMIF;
- /* freq /= 12.5 */
- freq *= 10;
- freq /= 125;
+ /* freq /= 12.5 */
+ freq *= 10;
+ freq /= 125;
+ break;
+ case BAND_FM_JAPAN:
+ band = TEA575X_BIT_BAND_FM;
+ /* crystal fixup */
+ freq -= TEA575X_FMIF;
+ /* freq /= 12.5 */
+ freq *= 10;
+ freq /= 125;
+ break;
+ case BAND_AM:
+ band = TEA575X_BIT_BAND_MW;
+ /* crystal fixup */
+ freq += TEA575X_AMIF;
+ break;
+ }
- tea->val &= ~TEA575X_BIT_FREQ_MASK;
+ tea->val &= ~(TEA575X_BIT_FREQ_MASK | TEA575X_BIT_BAND_MASK);
+ tea->val |= band;
tea->val |= freq & TEA575X_BIT_FREQ_MASK;
snd_tea575x_write(tea, tea->val);
tea->freq = snd_tea575x_val_to_freq(tea, tea->val);
@@ -190,23 +247,57 @@ static int vidioc_querycap(struct file *file, void *priv,
return 0;
}
+static int vidioc_enum_freq_bands(struct file *file, void *priv,
+ struct v4l2_frequency_band *band)
+{
+ struct snd_tea575x *tea = video_drvdata(file);
+ int index;
+
+ if (band->tuner != 0)
+ return -EINVAL;
+
+ switch (band->index) {
+ case 0:
+ if (tea->tea5759)
+ index = BAND_FM_JAPAN;
+ else
+ index = BAND_FM;
+ break;
+ case 1:
+ if (tea->has_am) {
+ index = BAND_AM;
+ break;
+ }
+ /* Fall through */
+ default:
+ return -EINVAL;
+ }
+
+ *band = bands[index];
+ if (!tea->cannot_read_data)
+ band->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED;
+
+ return 0;
+}
+
static int vidioc_g_tuner(struct file *file, void *priv,
struct v4l2_tuner *v)
{
struct snd_tea575x *tea = video_drvdata(file);
+ struct v4l2_frequency_band band_fm = { 0, };
if (v->index > 0)
return -EINVAL;
snd_tea575x_read(tea);
+ vidioc_enum_freq_bands(file, priv, &band_fm);
- strcpy(v->name, "FM");
+ memset(v, 0, sizeof(*v));
+ strlcpy(v->name, tea->has_am ? "FM/AM" : "FM", sizeof(v->name));
v->type = V4L2_TUNER_RADIO;
- v->capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO;
- if (!tea->cannot_read_data)
- v->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED;
- v->rangelow = FREQ_LO;
- v->rangehigh = FREQ_HI;
+ v->capability = band_fm.capability;
+ v->rangelow = tea->has_am ? bands[BAND_AM].rangelow : band_fm.rangelow;
+ v->rangehigh = band_fm.rangehigh;
v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO;
v->audmode = (tea->val & TEA575X_BIT_MONO) ?
V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO;
@@ -218,13 +309,17 @@ static int vidioc_s_tuner(struct file *file, void *priv,
struct v4l2_tuner *v)
{
struct snd_tea575x *tea = video_drvdata(file);
+ u32 orig_val = tea->val;
if (v->index)
return -EINVAL;
tea->val &= ~TEA575X_BIT_MONO;
if (v->audmode == V4L2_TUNER_MODE_MONO)
tea->val |= TEA575X_BIT_MONO;
- snd_tea575x_write(tea, tea->val);
+ /* Only apply changes if currently tuning FM */
+ if (tea->band != BAND_AM && tea->val != orig_val)
+ snd_tea575x_set_freq(tea);
+
return 0;
}
@@ -248,24 +343,56 @@ static int vidioc_s_frequency(struct file *file, void *priv,
if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO)
return -EINVAL;
- tea->val &= ~TEA575X_BIT_SEARCH;
- tea->freq = clamp(f->frequency, FREQ_LO, FREQ_HI);
+ if (tea->has_am && f->frequency < (20000 * 16))
+ tea->band = BAND_AM;
+ else if (tea->tea5759)
+ tea->band = BAND_FM_JAPAN;
+ else
+ tea->band = BAND_FM;
+
+ tea->freq = clamp(f->frequency, bands[tea->band].rangelow,
+ bands[tea->band].rangehigh);
snd_tea575x_set_freq(tea);
return 0;
}
static int vidioc_s_hw_freq_seek(struct file *file, void *fh,
- struct v4l2_hw_freq_seek *a)
+ const struct v4l2_hw_freq_seek *a)
{
struct snd_tea575x *tea = video_drvdata(file);
unsigned long timeout;
- int i;
+ int i, spacing;
if (tea->cannot_read_data)
return -ENOTTY;
if (a->tuner || a->wrap_around)
return -EINVAL;
+ if (file->f_flags & O_NONBLOCK)
+ return -EWOULDBLOCK;
+
+ if (a->rangelow || a->rangehigh) {
+ for (i = 0; i < ARRAY_SIZE(bands); i++) {
+ if ((i == BAND_FM && tea->tea5759) ||
+ (i == BAND_FM_JAPAN && !tea->tea5759) ||
+ (i == BAND_AM && !tea->has_am))
+ continue;
+ if (bands[i].rangelow == a->rangelow &&
+ bands[i].rangehigh == a->rangehigh)
+ break;
+ }
+ if (i == ARRAY_SIZE(bands))
+ return -EINVAL; /* No matching band found */
+ if (i != tea->band) {
+ tea->band = i;
+ tea->freq = clamp(tea->freq, bands[i].rangelow,
+ bands[i].rangehigh);
+ snd_tea575x_set_freq(tea);
+ }
+ }
+
+ spacing = (tea->band == BAND_AM) ? 5 : 50; /* kHz */
+
/* clear the frequency, HW will fill it in */
tea->val &= ~TEA575X_BIT_FREQ_MASK;
tea->val |= TEA575X_BIT_SEARCH;
@@ -297,10 +424,10 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh,
if (freq == 0) /* shouldn't happen */
break;
/*
- * if we moved by less than 50 kHz, or in the wrong
- * direction, continue seeking
+ * if we moved by less than the spacing, or in the
+ * wrong direction, continue seeking
*/
- if (abs(tea->freq - freq) < 16 * 50 ||
+ if (abs(tea->freq - freq) < 16 * spacing ||
(a->seek_upward && freq < tea->freq) ||
(!a->seek_upward && freq > tea->freq)) {
snd_tea575x_write(tea, tea->val);
@@ -344,6 +471,7 @@ static const struct v4l2_ioctl_ops tea575x_ioctl_ops = {
.vidioc_g_frequency = vidioc_g_frequency,
.vidioc_s_frequency = vidioc_s_frequency,
.vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek,
+ .vidioc_enum_freq_bands = vidioc_enum_freq_bands,
.vidioc_log_status = v4l2_ctrl_log_status,
.vidioc_subscribe_event = v4l2_ctrl_subscribe_event,
.vidioc_unsubscribe_event = v4l2_event_unsubscribe,
@@ -446,3 +574,4 @@ module_exit(alsa_tea575x_module_exit)
EXPORT_SYMBOL(snd_tea575x_init);
EXPORT_SYMBOL(snd_tea575x_exit);
+EXPORT_SYMBOL(snd_tea575x_set_freq);
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/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index 1d47be8170b5..b3b4f15e45ba 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -612,10 +612,10 @@ static int snd_es1688_capture_close(struct snd_pcm_substream *substream)
static int snd_es1688_free(struct snd_es1688 *chip)
{
- if (chip->res_port) {
+ if (chip->hardware != ES1688_HW_UNDEF)
snd_es1688_init(chip, 0);
+ if (chip->res_port)
release_and_free_resource(chip->res_port);
- }
if (chip->irq >= 0)
free_irq(chip->irq, (void *) chip);
if (chip->dma8 >= 0) {
@@ -657,19 +657,27 @@ int snd_es1688_create(struct snd_card *card,
return -ENOMEM;
chip->irq = -1;
chip->dma8 = -1;
+ chip->hardware = ES1688_HW_UNDEF;
- if ((chip->res_port = request_region(port + 4, 12, "ES1688")) == NULL) {
+ chip->res_port = request_region(port + 4, 12, "ES1688");
+ if (chip->res_port == NULL) {
snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4);
- return -EBUSY;
+ err = -EBUSY;
+ goto exit;
}
- if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) {
+
+ err = request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip);
+ if (err < 0) {
snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq);
- return -EBUSY;
+ goto exit;
}
+
chip->irq = irq;
- if (request_dma(dma8, "ES1688")) {
+ err = request_dma(dma8, "ES1688");
+
+ if (err < 0) {
snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8);
- return -EBUSY;
+ goto exit;
}
chip->dma8 = dma8;
@@ -685,14 +693,18 @@ int snd_es1688_create(struct snd_card *card,
err = snd_es1688_probe(chip);
if (err < 0)
- return err;
+ goto exit;
err = snd_es1688_init(chip, 1);
if (err < 0)
- return err;
+ goto exit;
/* Register device */
- return snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+exit:
+ if (err)
+ snd_es1688_free(chip);
+ return err;
}
static struct snd_pcm_ops snd_es1688_playback_ops = {
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..1c65cc5e3a31 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;
@@ -2325,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
}
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
+ memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
snd_hda_jack_tbl_clear(codec);
codec->proc_widget_hook = NULL;
codec->spec = NULL;
@@ -2340,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec)
codec->num_pcms = 0;
codec->pcm_info = NULL;
codec->preset = NULL;
- memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
codec->slave_dig_outs = NULL;
codec->spdif_status_reset = 0;
module_put(codec->owner);
@@ -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..c4763c52eaf6 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},"
@@ -2700,6 +2701,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF),
+ SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
@@ -3270,6 +3273,14 @@ 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 641408dc28c0..8f23374fa642 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1165,14 +1165,20 @@ 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)
+{
struct hdmi_spec *spec = codec->spec;
int cvt_idx, pin_idx;
struct hdmi_spec_per_cvt *per_cvt;
struct hdmi_spec_per_pin *per_pin;
int pinctl;
- snd_hda_codec_cleanup_stream(codec, hinfo->nid);
-
if (hinfo->nid) {
cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid);
if (snd_BUG_ON(cvt_idx < 0))
@@ -1195,12 +1201,12 @@ static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
pinctl & ~PIN_OUT);
snd_hda_spdif_ctls_unassign(codec, pin_idx);
}
-
return 0;
}
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,
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f141395dfee6..4f81dd44c837 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -203,6 +203,7 @@ struct alc_spec {
unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
unsigned int inv_dmic_fixup:1; /* has inverted digital-mic workaround */
unsigned int inv_dmic_muted:1; /* R-ch of inv d-mic is muted? */
+ unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
/* auto-mute control */
int automute_mode;
@@ -4323,7 +4324,8 @@ static int alc_parse_auto_config(struct hda_codec *codec,
return 0; /* can't find valid BIOS pin config */
}
- if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
+ if (!spec->no_primary_hp &&
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
cfg->line_outs <= cfg->hp_outs) {
/* use HP as primary out */
cfg->speaker_outs = cfg->line_outs;
@@ -5050,6 +5052,7 @@ enum {
ALC889_FIXUP_MBP_VREF,
ALC889_FIXUP_IMAC91_VREF,
ALC882_FIXUP_INV_DMIC,
+ ALC882_FIXUP_NO_PRIMARY_HP,
};
static void alc889_fixup_coef(struct hda_codec *codec,
@@ -5171,6 +5174,17 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec,
spec->keep_vref_in_automute = 1;
}
+/* Don't take HP output as primary
+ * strangely, the speaker output doesn't work on VAIO Z through DAC 0x05
+ */
+static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action == ALC_FIXUP_ACT_PRE_PROBE)
+ spec->no_primary_hp = 1;
+}
+
static const struct alc_fixup alc882_fixups[] = {
[ALC882_FIXUP_ABIT_AW9D_MAX] = {
.type = ALC_FIXUP_PINS,
@@ -5357,6 +5371,10 @@ static const struct alc_fixup alc882_fixups[] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc_fixup_inv_dmic_0x12,
},
+ [ALC882_FIXUP_NO_PRIMARY_HP] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc882_fixup_no_primary_hp,
+ },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -5391,6 +5409,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
+ SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
/* All Apple entries are in codec SSIDs */
SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF),
@@ -5432,6 +5451,7 @@ static const struct alc_model_fixup alc882_fixup_models[] = {
{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
{.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"},
+ {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"},
{}
};
@@ -6079,6 +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,
@@ -6186,9 +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 a1596a3b171c..3d4722f0a1ca 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -101,6 +101,8 @@ enum {
STAC_92HD83XXX_HP_cNB11_INTQUAD,
STAC_HP_DV7_4000,
STAC_HP_ZEPHYR,
+ STAC_92HD83XXX_HP_LED,
+ STAC_92HD83XXX_HP_INV_LED,
STAC_92HD83XXX_MODELS
};
@@ -1073,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = {
static const char * const slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
- "Headphone", "Speaker", "IEC958",
+ "Headphone", "Speaker", "IEC958", "PCM",
NULL
};
@@ -1675,6 +1677,8 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
[STAC_HP_DV7_4000] = "hp-dv7-4000",
[STAC_HP_ZEPHYR] = "hp-zephyr",
+ [STAC_92HD83XXX_HP_LED] = "hp-led",
+ [STAC_92HD83XXX_HP_INV_LED] = "hp-inv-led",
};
static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1729,6 +1733,8 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
"HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
"HP", STAC_HP_ZEPHYR),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3660,
+ "HP Mini", STAC_92HD83XXX_HP_LED),
{} /* terminator */
};
@@ -4266,7 +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)
@@ -4414,7 +4421,12 @@ static int stac92xx_init(struct hda_codec *codec)
snd_hda_jack_report_sync(codec);
/* sync mute LED */
- snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+ if (spec->gpio_led) {
+ if (spec->vmaster_mute.hook)
+ snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+ else /* the very first init call doesn't have vmaster yet */
+ stac92xx_update_led_status(codec, false);
+ }
/* sync the power-map */
if (spec->num_pwrs)
@@ -4531,6 +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;
@@ -5507,6 +5522,7 @@ static void stac92hd8x_fill_auto_spec(struct hda_codec *codec)
static int patch_stac92hd83xxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
+ int default_polarity = -1; /* no default cfg */
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5518,6 +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;
@@ -5555,9 +5572,15 @@ again:
case STAC_HP_ZEPHYR:
spec->init = stac92hd83xxx_hp_zephyr_init;
break;
+ case STAC_92HD83XXX_HP_LED:
+ default_polarity = 0;
+ break;
+ case STAC_92HD83XXX_HP_INV_LED:
+ default_polarity = 1;
+ break;
}
- if (find_mute_led_cfg(codec, -1/*no default cfg*/))
+ if (find_mute_led_cfg(codec, default_polarity))
snd_printd("mute LED gpio %d polarity %d\n",
spec->gpio_led,
spec->gpio_led_polarity);
@@ -5730,7 +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,
@@ -5755,7 +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);
@@ -5770,7 +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,
@@ -5778,6 +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 90645560ed39..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
@@ -3226,7 +3234,7 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
int imux_is_smixer;
- unsigned int parm;
+ unsigned int parm, parm2;
/* MUX6 (1eh) = stereo mixer */
imux_is_smixer =
snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
@@ -3249,7 +3257,7 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x27, &parm);
update_power_state(codec, 0x1a, parm);
- update_power_state(codec, 0xb, parm);
+ parm2 = parm; /* for pin 0x0b */
/* PW2 (26h), AOW2 (ah) */
parm = AC_PWRST_D3;
@@ -3264,6 +3272,9 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
if (!spec->hp_independent_mode) /* check for redirected HP */
set_pin_power_state(codec, 0x28, &parm);
update_power_state(codec, 0x8, parm);
+ if (!spec->hp_independent_mode && parm2 != AC_PWRST_D3)
+ parm = parm2;
+ update_power_state(codec, 0xb, parm);
/* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm);
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 764cc93dbca4..075d5aa1fee0 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
}
static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
{
@@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
.info = ak4396_dac_vol_info,
.get = ak4396_dac_vol_get,
.put = ak4396_dac_vol_put,
- .tlv = { .p = db_scale_wm_dac },
+ .tlv = { .p = ak4396_db_scale },
},
};
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/arizona.c b/sound/soc/codecs/arizona.c
index 5c9cacaf2d52..1cf7a32d1b21 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = {
940800,
1411200,
1881600,
- 2882400,
+ 2822400,
3763200,
5644800,
7526400,
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 6276e352125f..115a40301810 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 */
@@ -657,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = {
.id = MC13783_ID_STEREO_DAC,
.playback = {
.stream_name = "Playback",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = MC13783_FORMATS,
@@ -668,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = {
.id = MC13783_ID_STEREO_CODEC,
.capture = {
.stream_name = "Capture",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = MC13783_RATES_RECORD,
.formats = MC13783_FORMATS,
@@ -690,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = {
.id = MC13783_ID_SYNC,
.playback = {
.stream_name = "Playback",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = MC13783_FORMATS,
},
.capture = {
.stream_name = "Capture",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = MC13783_RATES_RECORD,
.formats = MC13783_FORMATS,
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/wm2000.c b/sound/soc/codecs/wm2000.c
index 3fd5b29dc933..a3acb7a85f6a 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -702,7 +702,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg)
}
static const struct regmap_config wm2000_regmap = {
- .reg_bits = 8,
+ .reg_bits = 16,
.val_bits = 8,
.max_register = WM2000_REG_IF_CTL,
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/wm8904.c b/sound/soc/codecs/wm8904.c
index 0013afe48e66..dc4262eea4b7 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = {
{ 14, 0x0000 }, /* R14 - Power Management 2 */
{ 15, 0x0000 }, /* R15 - Power Management 3 */
{ 18, 0x0000 }, /* R18 - Power Management 6 */
- { 19, 0x945E }, /* R20 - Clock Rates 0 */
+ { 20, 0x945E }, /* R20 - Clock Rates 0 */
{ 21, 0x0C05 }, /* R21 - Clock Rates 1 */
{ 22, 0x0006 }, /* R22 - Clock Rates 2 */
{ 24, 0x0050 }, /* R24 - Audio Interface 0 */
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-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index fb21b17f17f5..199408ec4261 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -94,7 +94,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "audmux internal port setup failed\n");
return ret;
}
- imx_audmux_v2_configure_port(ext_port,
+ ret = imx_audmux_v2_configure_port(ext_port,
IMX_AUDMUX_V2_PTCR_SYN,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
if (ret) {
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/am3517evm.c b/sound/soc/omap/am3517evm.c
index 009533ab8d18..df65f98211ec 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -59,7 +59,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream,
return ret;
}
- snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
+ ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n");
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/dma.c b/sound/soc/samsung/dma.c
index f3ebc38c10fe..b70964ea448c 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME,
+ SNDRV_PCM_INFO_MMAP_VALID,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_LE |
SNDRV_PCM_FMTBIT_U8 |
@@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->state |= ST_RUNNING;
prtd->params->ops->trigger(prtd->params->ch);
break;
case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
prtd->state &= ~ST_RUNNING;
prtd->params->ops->stop(prtd->params->ch);
break;
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-dapm.c b/sound/soc/soc-dapm.c
index dd7c49fafd75..f90139b5f50d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -291,8 +291,11 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
if (dapm->codec->driver->set_bias_level)
ret = dapm->codec->driver->set_bias_level(dapm->codec,
level);
- } else
+ else
+ dapm->bias_level = level;
+ } else if (!card || dapm != &card->dapm) {
dapm->bias_level = level;
+ }
if (ret != 0)
goto out;
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/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 97c2cac8e92c..8c7f23729446 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -138,7 +138,7 @@ static void spear_pcm_free(struct snd_pcm *pcm)
continue;
buf = &substream->dma_buffer;
- if (!buf && !buf->area)
+ if (!buf || !buf->area)
continue;
dma_free_writecombine(pcm->card->dev, buf->bytes,
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index d684df294c0c..76cb1b363b71 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = {
.name = "Headset detection",
.report = SND_JACK_HEADSET,
.debounce_time = 150,
- .invert = 1,
};
static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = {
@@ -177,7 +176,7 @@ 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_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 5658bcec1931..8d6900c1ee47 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
slave_config.dst_addr = dmap->addr;
- slave_config.src_maxburst = 0;
+ slave_config.dst_maxburst = 4;
} else {
slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
slave_config.src_addr = dmap->addr;
- slave_config.dst_maxburst = 0;
+ slave_config.src_maxburst = 4;
}
slave_config.slave_id = dmap->req_sel;
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..eb85113d472a 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>
@@ -663,7 +663,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
struct ux500_msp **msp_p,
struct msp_i2s_platform_data *platform_data)
{
- int ret = 0;
struct resource *res = NULL;
struct i2s_controller *i2s_cont;
struct ux500_msp *msp;
@@ -685,15 +684,14 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
if (res == NULL) {
dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n",
__func__);
- ret = -ENOMEM;
- goto err_res;
+ return -ENOMEM;
}
- msp->registers = ioremap(res->start, (res->end - res->start + 1));
+ msp->registers = devm_ioremap(&pdev->dev, res->start,
+ resource_size(res));
if (msp->registers == NULL) {
dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__);
- ret = -ENOMEM;
- goto err_res;
+ return -ENOMEM;
}
msp->msp_state = MSP_STATE_IDLE;
@@ -705,7 +703,7 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
dev_err(&pdev->dev,
"%s: ERROR: Failed to allocate I2S-controller!\n",
__func__);
- goto err_i2s_cont;
+ return -ENOMEM;
}
i2s_cont->dev.parent = &pdev->dev;
i2s_cont->data = (void *)msp;
@@ -716,14 +714,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
msp->i2s_cont = i2s_cont;
return 0;
-
-err_i2s_cont:
- iounmap(msp->registers);
-
-err_res:
- devm_kfree(&pdev->dev, msp);
-
- return ret;
}
void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
@@ -732,11 +722,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id);
device_unregister(&msp->i2s_cont->dev);
- devm_kfree(&pdev->dev, msp->i2s_cont);
-
- iounmap(msp->registers);
-
- devm_kfree(&pdev->dev, msp);
}
MODULE_LICENSE("GPL v2");
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/sound_firmware.c b/sound/sound_firmware.c
index 7e96249536b4..37711a5d0d6b 100644
--- a/sound/sound_firmware.c
+++ b/sound/sound_firmware.c
@@ -23,14 +23,14 @@ static int do_mod_firmware_load(const char *fn, char **fp)
if (l <= 0 || l > 131072)
{
printk(KERN_INFO "Invalid firmware '%s'\n", fn);
- filp_close(filp, current->files);
+ filp_close(filp, NULL);
return 0;
}
dp = vmalloc(l);
if (dp == NULL)
{
printk(KERN_INFO "Out of memory loading '%s'.\n", fn);
- filp_close(filp, current->files);
+ filp_close(filp, NULL);
return 0;
}
pos = 0;
@@ -38,10 +38,10 @@ static int do_mod_firmware_load(const char *fn, char **fp)
{
printk(KERN_INFO "Failed to read '%s'.\n", fn);
vfree(dp);
- filp_close(filp, current->files);
+ filp_close(filp, NULL);
return 0;
}
- filp_close(filp, current->files);
+ filp_close(filp, NULL);
*fp = dp;
return (int) l;
}
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/clock.c b/sound/usb/clock.c
index 379baad3d5ad..5e634a2eb282 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -111,7 +111,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
return 0;
/* If a clock source can't tell us whether it's valid, we assume it is */
- if (!uac2_control_is_readable(cs_desc->bmControls, UAC2_CS_CONTROL_CLOCK_VALID))
+ if (!uac2_control_is_readable(cs_desc->bmControls,
+ UAC2_CS_CONTROL_CLOCK_VALID - 1))
return 1;
err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 0f647d22cb4a..060dccb9ec75 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.
*/
@@ -206,7 +197,13 @@ static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
/* no data provider, so send silence */
unsigned int offs = 0;
for (i = 0; i < ctx->packets; ++i) {
- int counts = ctx->packet_size[i];
+ int counts;
+
+ if (ctx->packet_size[i])
+ counts = ctx->packet_size[i];
+ else
+ counts = snd_usb_endpoint_next_packet_size(ep);
+
urb->iso_frame_desc[i].offset = offs * ep->stride;
urb->iso_frame_desc[i].length = counts * ep->stride;
offs += counts;
@@ -370,7 +367,6 @@ 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 +795,9 @@ 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 +807,7 @@ 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 +820,9 @@ 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 +853,6 @@ 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..f782ce19bf5a 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 +1138,32 @@ 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;
+
+ /* ignore the delay accounting when procssed=0 is given, i.e.
+ * silent payloads are procssed before handling the actual data
+ */
+ if (!processed)
+ return;
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 +1221,7 @@ 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;