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authorLinus Torvalds <torvalds@linux-foundation.org>2016-08-27 08:53:21 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2016-08-27 08:53:21 +0300
commit03cef71062cf185e95d588d90406f27bad740b3c (patch)
treea117af89e98e8c16a96b4a3212f74b8829bfd79a
parent28687b935e93a9041a485b9ecdcab0e335f8eda5 (diff)
parenta820cd3d25c2891028b5f296a8a871ce6dd92c0d (diff)
downloadlinux-03cef71062cf185e95d588d90406f27bad740b3c.tar.xz
Merge tag 'sound-4.8-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "Here are a bunch of fixes as you can see in diffstat. One core change in ASoC is about the unexpected unbinding error, and another about debugfs cleanup. The rest are wide-spread driver-specific fixes: a series of LINE6 USB fixes, a HD-audio quirk, and various ASoC fixes including OMAP boot fixes and Intel SKL fixes" * tag 'sound-4.8-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (22 commits) ALSA: hda/realtek - fix headset mic detection for MSI MS-B120 ASoC: omap-mcpdm: Fix irq resource handling ASoC: max98371: Add terminate entry for i2c_device_id tables ALSA: line6: Fix POD sysfs attributes segfault ALSA: line6: Give up on the lock while URBs are released. ALSA: line6: Remove double line6_pcm_release() after failed acquire. ASoC: omap-abe-twl6040: Correct dmic-codec device registration ASoC: core: Clean up DAPM before the card debugfs ASoC: omap-mcpdm: Drop pdmclk clock handling ASoC: atmel_ssc_dai: Don't unconditionally reset SSC on stream startup ASoC: compress: Fix leak of a widget list in soc_compr_open_fe ASoC: Intel: Skylake: Fix error return code in skl_probe() ASoC: wm2000: Fix return of uninitialised varible ASoC: Fix leak of rtd in soc_bind_dai_link ASoC: da7213: Default to 64 BCLKs per WCLK to support all formats ASoC: nau8825: fix static check error about semaphone control ASoC: nau8825: fix bug in playback when suspend ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card ASoC: simple-card-utils: add missing MODULE_xxx() ASoC: Intel: Skylake: Check list empty while getting module info ...
-rw-r--r--Documentation/devicetree/bindings/sound/omap-mcpdm.txt10
-rw-r--r--sound/pci/hda/patch_realtek.c7
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c5
-rw-r--r--sound/soc/codecs/da7213.c4
-rw-r--r--sound/soc/codecs/max98371.c1
-rw-r--r--sound/soc/codecs/nau8825.c78
-rw-r--r--sound/soc/codecs/wm2000.c2
-rw-r--r--sound/soc/generic/Makefile6
-rw-r--r--sound/soc/generic/simple-card-utils.c6
-rw-r--r--sound/soc/intel/skylake/skl-sst-utils.c5
-rw-r--r--sound/soc/intel/skylake/skl.c4
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c61
-rw-r--r--sound/soc/omap/omap-mcpdm.c22
-rw-r--r--sound/soc/samsung/s3c24xx_uda134x.c7
-rw-r--r--sound/soc/sh/rcar/src.c6
-rw-r--r--sound/soc/soc-compress.c4
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dapm.c10
-rw-r--r--sound/usb/line6/pcm.c3
-rw-r--r--sound/usb/line6/pod.c12
20 files changed, 127 insertions, 131 deletions
diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
index 6f6c2f8e908d..0741dff048dd 100644
--- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
+++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
@@ -8,8 +8,6 @@ Required properties:
- interrupts: Interrupt number for McPDM
- interrupt-parent: The parent interrupt controller
- ti,hwmods: Name of the hwmod associated to the McPDM
-- clocks: phandle for the pdmclk provider, likely <&twl6040>
-- clock-names: Must be "pdmclk"
Example:
@@ -21,11 +19,3 @@ mcpdm: mcpdm@40132000 {
interrupt-parent = <&gic>;
ti,hwmods = "mcpdm";
};
-
-In board DTS file the pdmclk needs to be added:
-
-&mcpdm {
- clocks = <&twl6040>;
- clock-names = "pdmclk";
- status = "okay";
-};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 574b1b48996f..7100f05e651a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4828,7 +4828,7 @@ enum {
ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC292_FIXUP_TPT440_DOCK,
ALC292_FIXUP_TPT440,
- ALC283_FIXUP_BXBT2807_MIC,
+ ALC283_FIXUP_HEADSET_MIC,
ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
ALC282_FIXUP_ASPIRE_V5_PINS,
ALC280_FIXUP_HP_GPIO4,
@@ -5321,7 +5321,7 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC292_FIXUP_TPT440_DOCK,
},
- [ALC283_FIXUP_BXBT2807_MIC] = {
+ [ALC283_FIXUP_HEADSET_MIC] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
{ 0x19, 0x04a110f0 },
@@ -5651,7 +5651,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN),
SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
- SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC),
+ SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 54c09acd3fed..16e459aedffe 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -299,8 +299,9 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
clk_enable(ssc_p->ssc->clk);
ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk);
- /* Reset the SSC to keep it at a clean status */
- ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+ /* Reset the SSC unless initialized to keep it in a clean state */
+ if (!ssc_p->initialized)
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dir = 0;
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index e5527bc570ae..bcf1834c5648 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1247,8 +1247,8 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
return -EINVAL;
}
- /* By default only 32 BCLK per WCLK is supported */
- dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_32;
+ /* By default only 64 BCLK per WCLK is supported */
+ dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_64;
snd_soc_write(codec, DA7213_DAI_CLK_MODE, dai_clk_mode);
snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK,
diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
index cf0a39bb631a..02352ed8961c 100644
--- a/sound/soc/codecs/max98371.c
+++ b/sound/soc/codecs/max98371.c
@@ -412,6 +412,7 @@ static int max98371_i2c_remove(struct i2c_client *client)
static const struct i2c_device_id max98371_i2c_id[] = {
{ "max98371", 0 },
+ { }
};
MODULE_DEVICE_TABLE(i2c, max98371_i2c_id);
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 5c9707ac4bbf..2e59a85e360b 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -212,31 +212,6 @@ static const unsigned short logtable[256] = {
0xfa2f, 0xfaea, 0xfba5, 0xfc60, 0xfd1a, 0xfdd4, 0xfe8e, 0xff47
};
-static struct snd_soc_dai *nau8825_get_codec_dai(struct nau8825 *nau8825)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm);
- struct snd_soc_component *component = &codec->component;
- struct snd_soc_dai *codec_dai, *_dai;
-
- list_for_each_entry_safe(codec_dai, _dai, &component->dai_list, list) {
- if (!strncmp(codec_dai->name, NUVOTON_CODEC_DAI,
- strlen(NUVOTON_CODEC_DAI)))
- return codec_dai;
- }
- return NULL;
-}
-
-static bool nau8825_dai_is_active(struct nau8825 *nau8825)
-{
- struct snd_soc_dai *codec_dai = nau8825_get_codec_dai(nau8825);
-
- if (codec_dai) {
- if (codec_dai->playback_active || codec_dai->capture_active)
- return true;
- }
- return false;
-}
-
/**
* nau8825_sema_acquire - acquire the semaphore of nau88l25
* @nau8825: component to register the codec private data with
@@ -250,19 +225,26 @@ static bool nau8825_dai_is_active(struct nau8825 *nau8825)
* Acquires the semaphore without jiffies. If no more tasks are allowed
* to acquire the semaphore, calling this function will put the task to
* sleep until the semaphore is released.
- * It returns if the semaphore was acquired.
+ * If the semaphore is not released within the specified number of jiffies,
+ * this function returns -ETIME.
+ * If the sleep is interrupted by a signal, this function will return -EINTR.
+ * It returns 0 if the semaphore was acquired successfully.
*/
-static void nau8825_sema_acquire(struct nau8825 *nau8825, long timeout)
+static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout)
{
int ret;
- if (timeout)
+ if (timeout) {
ret = down_timeout(&nau8825->xtalk_sem, timeout);
- else
+ if (ret < 0)
+ dev_warn(nau8825->dev, "Acquire semaphone timeout\n");
+ } else {
ret = down_interruptible(&nau8825->xtalk_sem);
+ if (ret < 0)
+ dev_warn(nau8825->dev, "Acquire semaphone fail\n");
+ }
- if (ret < 0)
- dev_warn(nau8825->dev, "Acquire semaphone fail\n");
+ return ret;
}
/**
@@ -1205,6 +1187,8 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream,
struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
unsigned int val_len = 0;
+ nau8825_sema_acquire(nau8825, 2 * HZ);
+
switch (params_width(params)) {
case 16:
val_len |= NAU8825_I2S_DL_16;
@@ -1225,6 +1209,9 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream,
regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1,
NAU8825_I2S_DL_MASK, val_len);
+ /* Release the semaphone. */
+ nau8825_sema_release(nau8825);
+
return 0;
}
@@ -1234,6 +1221,8 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
unsigned int ctrl1_val = 0, ctrl2_val = 0;
+ nau8825_sema_acquire(nau8825, 2 * HZ);
+
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
ctrl2_val |= NAU8825_I2S_MS_MASTER;
@@ -1282,6 +1271,9 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2,
NAU8825_I2S_MS_MASK, ctrl2_val);
+ /* Release the semaphone. */
+ nau8825_sema_release(nau8825);
+
return 0;
}
@@ -1611,8 +1603,11 @@ static irqreturn_t nau8825_interrupt(int irq, void *data)
* cess and restore changes if process
* is ongoing when ejection.
*/
+ int ret;
nau8825->xtalk_protect = true;
- nau8825_sema_acquire(nau8825, 0);
+ ret = nau8825_sema_acquire(nau8825, 0);
+ if (ret < 0)
+ nau8825->xtalk_protect = false;
}
/* Startup cross talk detection process */
nau8825->xtalk_state = NAU8825_XTALK_PREPARE;
@@ -2238,23 +2233,14 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec)
static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec)
{
struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+ int ret;
regcache_cache_only(nau8825->regmap, false);
regcache_sync(nau8825->regmap);
- if (nau8825_is_jack_inserted(nau8825->regmap)) {
- /* If the jack is inserted, we need to check whether the play-
- * back is active before suspend. If active, the driver has to
- * raise the protection for cross talk function to avoid the
- * playback recovers before cross talk process finish. Other-
- * wise, the playback will be interfered by cross talk func-
- * tion. It is better to apply hardware related parameters
- * before starting playback or record.
- */
- if (nau8825_dai_is_active(nau8825)) {
- nau8825->xtalk_protect = true;
- nau8825_sema_acquire(nau8825, 0);
- }
- }
+ nau8825->xtalk_protect = true;
+ ret = nau8825_sema_acquire(nau8825, 0);
+ if (ret < 0)
+ nau8825->xtalk_protect = false;
enable_irq(nau8825->irq);
return 0;
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index a67ea10f41a1..f2664396be6f 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -581,7 +581,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000,
if (anc_transitions[i].dest == ANC_OFF)
clk_disable_unprepare(wm2000->mclk);
- return ret;
+ return 0;
}
static int wm2000_anc_set_mode(struct wm2000_priv *wm2000)
diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile
index 45602ca8536e..2d53c8d70705 100644
--- a/sound/soc/generic/Makefile
+++ b/sound/soc/generic/Makefile
@@ -1,5 +1,5 @@
-obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) := simple-card-utils.o
-
+snd-soc-simple-card-utils-objs := simple-card-utils.o
snd-soc-simple-card-objs := simple-card.o
-obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o
+obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o
+obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index d89a9a1b2471..9599de69a880 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -7,6 +7,7 @@
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
+#include <linux/module.h>
#include <linux/of.h>
#include <sound/simple_card_utils.h>
@@ -95,3 +96,8 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
return 0;
}
EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name);
+
+/* Module information */
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_DESCRIPTION("ALSA SoC Simple Card Utils");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c
index 25fcb796bd86..ddcb52a51854 100644
--- a/sound/soc/intel/skylake/skl-sst-utils.c
+++ b/sound/soc/intel/skylake/skl-sst-utils.c
@@ -123,6 +123,11 @@ int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid,
uuid_mod = (uuid_le *)uuid;
+ if (list_empty(&ctx->uuid_list)) {
+ dev_err(ctx->dev, "Module list is empty\n");
+ return -EINVAL;
+ }
+
list_for_each_entry(module, &ctx->uuid_list, list) {
if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) {
dfw_config->module_id = module->id;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index cd59536a761d..e3e764167765 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -672,8 +672,10 @@ static int skl_probe(struct pci_dev *pci,
skl->nhlt = skl_nhlt_init(bus->dev);
- if (skl->nhlt == NULL)
+ if (skl->nhlt == NULL) {
+ err = -ENODEV;
goto out_free;
+ }
skl_nhlt_update_topology_bin(skl);
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 0843a68f277c..f61b3b58083b 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -38,10 +38,10 @@
struct abe_twl6040 {
int jack_detection; /* board can detect jack events */
int mclk_freq; /* MCLK frequency speed for twl6040 */
-
- struct platform_device *dmic_codec_dev;
};
+struct platform_device *dmic_codec_dev;
+
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -258,8 +258,6 @@ static int omap_abe_probe(struct platform_device *pdev)
if (priv == NULL)
return -ENOMEM;
- priv->dmic_codec_dev = ERR_PTR(-EINVAL);
-
if (snd_soc_of_parse_card_name(card, "ti,model")) {
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
@@ -284,13 +282,6 @@ static int omap_abe_probe(struct platform_device *pdev)
num_links = 2;
abe_twl6040_dai_links[1].cpu_of_node = dai_node;
abe_twl6040_dai_links[1].platform_of_node = dai_node;
-
- priv->dmic_codec_dev = platform_device_register_simple(
- "dmic-codec", -1, NULL, 0);
- if (IS_ERR(priv->dmic_codec_dev)) {
- dev_err(&pdev->dev, "Can't instantiate dmic-codec\n");
- return PTR_ERR(priv->dmic_codec_dev);
- }
} else {
num_links = 1;
}
@@ -299,16 +290,14 @@ static int omap_abe_probe(struct platform_device *pdev)
of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq);
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency not provided\n");
- ret = -EINVAL;
- goto err_unregister;
+ return -EINVAL;
}
card->fully_routed = 1;
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
- ret = -ENODEV;
- goto err_unregister;
+ return -ENODEV;
}
card->dai_link = abe_twl6040_dai_links;
@@ -317,17 +306,9 @@ static int omap_abe_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(card, priv);
ret = snd_soc_register_card(card);
- if (ret) {
+ if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
- goto err_unregister;
- }
-
- return 0;
-
-err_unregister:
- if (!IS_ERR(priv->dmic_codec_dev))
- platform_device_unregister(priv->dmic_codec_dev);
return ret;
}
@@ -335,13 +316,9 @@ err_unregister:
static int omap_abe_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
- struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
snd_soc_unregister_card(card);
- if (!IS_ERR(priv->dmic_codec_dev))
- platform_device_unregister(priv->dmic_codec_dev);
-
return 0;
}
@@ -361,7 +338,33 @@ static struct platform_driver omap_abe_driver = {
.remove = omap_abe_remove,
};
-module_platform_driver(omap_abe_driver);
+static int __init omap_abe_init(void)
+{
+ int ret;
+
+ dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL,
+ 0);
+ if (IS_ERR(dmic_codec_dev)) {
+ pr_err("%s: dmic-codec device registration failed\n", __func__);
+ return PTR_ERR(dmic_codec_dev);
+ }
+
+ ret = platform_driver_register(&omap_abe_driver);
+ if (ret) {
+ pr_err("%s: platform driver registration failed\n", __func__);
+ platform_device_unregister(dmic_codec_dev);
+ }
+
+ return ret;
+}
+module_init(omap_abe_init);
+
+static void __exit omap_abe_exit(void)
+{
+ platform_driver_unregister(&omap_abe_driver);
+ platform_device_unregister(dmic_codec_dev);
+}
+module_exit(omap_abe_exit);
MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index e7cdc51fd806..64609c77a79d 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -31,7 +31,6 @@
#include <linux/err.h>
#include <linux/io.h>
#include <linux/irq.h>
-#include <linux/clk.h>
#include <linux/slab.h>
#include <linux/pm_runtime.h>
#include <linux/of_device.h>
@@ -55,7 +54,6 @@ struct omap_mcpdm {
unsigned long phys_base;
void __iomem *io_base;
int irq;
- struct clk *pdmclk;
struct mutex mutex;
@@ -390,15 +388,14 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
int ret;
- clk_prepare_enable(mcpdm->pdmclk);
pm_runtime_enable(mcpdm->dev);
/* Disable lines while request is ongoing */
pm_runtime_get_sync(mcpdm->dev);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00);
- ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler,
- 0, "McPDM", (void *)mcpdm);
+ ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM",
+ (void *)mcpdm);
pm_runtime_put_sync(mcpdm->dev);
@@ -423,9 +420,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+ free_irq(mcpdm->irq, (void *)mcpdm);
pm_runtime_disable(mcpdm->dev);
- clk_disable_unprepare(mcpdm->pdmclk);
return 0;
}
@@ -445,8 +442,6 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai)
mcpdm->pm_active_count++;
}
- clk_disable_unprepare(mcpdm->pdmclk);
-
return 0;
}
@@ -454,8 +449,6 @@ static int omap_mcpdm_resume(struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
- clk_prepare_enable(mcpdm->pdmclk);
-
if (mcpdm->pm_active_count) {
while (mcpdm->pm_active_count--)
pm_runtime_get_sync(mcpdm->dev);
@@ -549,15 +542,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
mcpdm->dev = &pdev->dev;
- mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk");
- if (IS_ERR(mcpdm->pdmclk)) {
- if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER)
- return -EPROBE_DEFER;
- dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n",
- PTR_ERR(mcpdm->pdmclk));
- mcpdm->pdmclk = NULL;
- }
-
ret = devm_snd_soc_register_component(&pdev->dev,
&omap_mcpdm_component,
&omap_mcpdm_dai, 1);
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index 50849e137fc0..92e88bca386e 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -58,10 +58,12 @@ static struct platform_device *s3c24xx_uda134x_snd_device;
static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
{
- int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
#ifdef ENFORCE_RATES
struct snd_pcm_runtime *runtime = substream->runtime;
#endif
+ int ret = 0;
mutex_lock(&clk_lock);
pr_debug("%s %d\n", __func__, clk_users);
@@ -71,8 +73,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
printk(KERN_ERR "%s cannot get xtal\n", __func__);
ret = PTR_ERR(xtal);
} else {
- pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
- "pclk");
+ pclk = clk_get(cpu_dai->dev, "iis");
if (IS_ERR(pclk)) {
printk(KERN_ERR "%s cannot get pclk\n",
__func__);
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index e39f916d0f2f..969a5169de25 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -226,8 +226,12 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
ifscr = 0;
fsrate = 0;
if (fin != fout) {
+ u64 n;
+
ifscr = 1;
- fsrate = 0x0400000 / fout * fin;
+ n = (u64)0x0400000 * fin;
+ do_div(n, fout);
+ fsrate = n;
}
/*
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index d2df46c14c68..bf7b52fce597 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -121,7 +121,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
dpcm_be_disconnect(fe, stream);
fe->dpcm[stream].runtime = NULL;
- goto fe_err;
+ goto path_err;
}
dpcm_clear_pending_state(fe, stream);
@@ -136,6 +136,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
return 0;
+path_err:
+ dpcm_path_put(&list);
fe_err:
if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown)
fe->dai_link->compr_ops->shutdown(cstream);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 16369cad4803..4afa8dba5e98 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1056,7 +1056,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
if (!rtd->platform) {
dev_err(card->dev, "ASoC: platform %s not registered\n",
dai_link->platform_name);
- return -EPROBE_DEFER;
+ goto _err_defer;
}
soc_add_pcm_runtime(card, rtd);
@@ -2083,14 +2083,13 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card)
/* remove auxiliary devices */
soc_remove_aux_devices(card);
+ snd_soc_dapm_free(&card->dapm);
soc_cleanup_card_debugfs(card);
/* remove the card */
if (card->remove)
card->remove(card);
- snd_soc_dapm_free(&card->dapm);
-
snd_card_free(card->snd_card);
return 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8698c26773b3..d908ff8f9755 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3493,6 +3493,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
const struct snd_soc_pcm_stream *config = w->params + w->params_select;
struct snd_pcm_substream substream;
struct snd_pcm_hw_params *params = NULL;
+ struct snd_pcm_runtime *runtime = NULL;
u64 fmt;
int ret;
@@ -3541,6 +3542,14 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
memset(&substream, 0, sizeof(substream));
+ /* Allocate a dummy snd_pcm_runtime for startup() and other ops() */
+ runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
+ if (!runtime) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ substream.runtime = runtime;
+
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
substream.stream = SNDRV_PCM_STREAM_CAPTURE;
@@ -3606,6 +3615,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
}
out:
+ kfree(runtime);
kfree(params);
return ret;
}
diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c
index 204cc074adb9..41aa3355e920 100644
--- a/sound/usb/line6/pcm.c
+++ b/sound/usb/line6/pcm.c
@@ -55,7 +55,6 @@ static int snd_line6_impulse_volume_put(struct snd_kcontrol *kcontrol,
err = line6_pcm_acquire(line6pcm, LINE6_STREAM_IMPULSE);
if (err < 0) {
line6pcm->impulse_volume = 0;
- line6_pcm_release(line6pcm, LINE6_STREAM_IMPULSE);
return err;
}
} else {
@@ -211,7 +210,9 @@ static void line6_stream_stop(struct snd_line6_pcm *line6pcm, int direction,
spin_lock_irqsave(&pstr->lock, flags);
clear_bit(type, &pstr->running);
if (!pstr->running) {
+ spin_unlock_irqrestore(&pstr->lock, flags);
line6_unlink_audio_urbs(line6pcm, pstr);
+ spin_lock_irqsave(&pstr->lock, flags);
if (direction == SNDRV_PCM_STREAM_CAPTURE) {
line6pcm->prev_fbuf = NULL;
line6pcm->prev_fsize = 0;
diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c
index daf81d169a42..45dd34874f43 100644
--- a/sound/usb/line6/pod.c
+++ b/sound/usb/line6/pod.c
@@ -244,8 +244,8 @@ static int pod_set_system_param_int(struct usb_line6_pod *pod, int value,
static ssize_t serial_number_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
- struct usb_interface *interface = to_usb_interface(dev);
- struct usb_line6_pod *pod = usb_get_intfdata(interface);
+ struct snd_card *card = dev_to_snd_card(dev);
+ struct usb_line6_pod *pod = card->private_data;
return sprintf(buf, "%u\n", pod->serial_number);
}
@@ -256,8 +256,8 @@ static ssize_t serial_number_show(struct device *dev,
static ssize_t firmware_version_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
- struct usb_interface *interface = to_usb_interface(dev);
- struct usb_line6_pod *pod = usb_get_intfdata(interface);
+ struct snd_card *card = dev_to_snd_card(dev);
+ struct usb_line6_pod *pod = card->private_data;
return sprintf(buf, "%d.%02d\n", pod->firmware_version / 100,
pod->firmware_version % 100);
@@ -269,8 +269,8 @@ static ssize_t firmware_version_show(struct device *dev,
static ssize_t device_id_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
- struct usb_interface *interface = to_usb_interface(dev);
- struct usb_line6_pod *pod = usb_get_intfdata(interface);
+ struct snd_card *card = dev_to_snd_card(dev);
+ struct usb_line6_pod *pod = card->private_data;
return sprintf(buf, "%d\n", pod->device_id);
}