summaryrefslogtreecommitdiff
path: root/sound
diff options
context:
space:
mode:
authorMarcel Ziswiler <marcel.ziswiler@toradex.com>2019-03-28 16:27:49 +0100
committerMarcel Ziswiler <marcel.ziswiler@toradex.com>2019-03-28 16:27:49 +0100
commitd899927728beca8357a5b4120b690cb3c1d80844 (patch)
treeccb170439cc8638d71f6120ae08a6faded46db98 /sound
parent8d60367808c45e33c0a9127621f4e5fc34914f6b (diff)
parent0a8ab17689e628c84a666195bfc6ab85d11cf057 (diff)
Diffstat (limited to 'sound')
-rw-r--r--sound/core/compress_offload.c3
-rw-r--r--sound/core/pcm.c2
-rw-r--r--sound/core/pcm_native.c14
-rw-r--r--sound/firewire/Kconfig1
-rw-r--r--sound/firewire/bebob/bebob.c16
-rw-r--r--sound/firewire/oxfw/oxfw.c8
-rw-r--r--sound/pci/cs46xx/dsp_spos.c3
-rw-r--r--sound/pci/emu10k1/emufx.c5
-rw-r--r--sound/pci/hda/hda_bind.c3
-rw-r--r--sound/pci/hda/hda_codec.c57
-rw-r--r--sound/pci/hda/hda_codec.h1
-rw-r--r--sound/pci/hda/hda_intel.c6
-rw-r--r--sound/pci/hda/hda_tegra.c2
-rw-r--r--sound/pci/hda/patch_conexant.c2
-rw-r--r--sound/pci/hda/patch_realtek.c25
-rw-r--r--sound/pci/rme9652/hdsp.c10
-rw-r--r--sound/soc/codecs/rt5514-spi.c2
-rw-r--r--sound/soc/fsl/Kconfig2
-rw-r--r--sound/soc/fsl/imx-audmux.c24
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c8
-rw-r--r--sound/soc/intel/atom/sst/sst_loader.c8
-rw-r--r--sound/soc/intel/boards/broadwell.c2
-rw-r--r--sound/soc/intel/boards/haswell.c2
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c67
-rw-r--r--sound/soc/omap/omap-dmic.c9
-rw-r--r--sound/soc/omap/omap-mcpdm.c43
-rw-r--r--sound/soc/soc-core.c1
-rw-r--r--sound/soc/soc-dapm.c10
-rw-r--r--sound/soc/soc-topology.c8
-rw-r--r--sound/synth/emux/emux_hwdep.c7
-rw-r--r--sound/usb/card.c5
-rw-r--r--sound/usb/mixer.c10
-rw-r--r--sound/usb/pcm.c9
-rw-r--r--sound/usb/quirks-table.h3
34 files changed, 290 insertions, 88 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 4490a699030b..555df64d46ff 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -529,7 +529,8 @@ static int snd_compress_check_input(struct snd_compr_params *params)
{
/* first let's check the buffer parameter's */
if (params->buffer.fragment_size == 0 ||
- params->buffer.fragments > INT_MAX / params->buffer.fragment_size)
+ params->buffer.fragments > INT_MAX / params->buffer.fragment_size ||
+ params->buffer.fragments == 0)
return -EINVAL;
/* now codec parameters */
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 6bda8f6c5f84..cdff5f976480 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -25,6 +25,7 @@
#include <linux/time.h>
#include <linux/mutex.h>
#include <linux/device.h>
+#include <linux/nospec.h>
#include <sound/core.h>
#include <sound/minors.h>
#include <sound/pcm.h>
@@ -125,6 +126,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card,
return -EFAULT;
if (stream < 0 || stream > 1)
return -EINVAL;
+ stream = array_index_nospec(stream, 2);
if (get_user(subdevice, &info->subdevice))
return -EFAULT;
mutex_lock(&register_mutex);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 79018697b477..3586ab41dec4 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -35,6 +35,7 @@
#include <sound/timer.h>
#include <sound/minors.h>
#include <linux/uio.h>
+#include <linux/delay.h>
/*
* Compatibility
@@ -78,12 +79,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem);
* and this may lead to a deadlock when the code path takes read sem
* twice (e.g. one in snd_pcm_action_nonatomic() and another in
* snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to
- * spin until it gets the lock.
+ * sleep until all the readers are completed without blocking by writer.
*/
-static inline void down_write_nonblock(struct rw_semaphore *lock)
+static inline void down_write_nonfifo(struct rw_semaphore *lock)
{
while (!down_write_trylock(lock))
- cond_resched();
+ msleep(1);
}
/**
@@ -1825,7 +1826,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
res = -ENOMEM;
goto _nolock;
}
- down_write_nonblock(&snd_pcm_link_rwsem);
+ down_write_nonfifo(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
substream->runtime->status->state != substream1->runtime->status->state ||
@@ -1872,7 +1873,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
struct snd_pcm_substream *s;
int res = 0;
- down_write_nonblock(&snd_pcm_link_rwsem);
+ down_write_nonfifo(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (!snd_pcm_stream_linked(substream)) {
res = -EALREADY;
@@ -2224,7 +2225,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
static void pcm_release_private(struct snd_pcm_substream *substream)
{
- snd_pcm_unlink(substream);
+ if (snd_pcm_stream_linked(substream))
+ snd_pcm_unlink(substream);
}
void snd_pcm_release_substream(struct snd_pcm_substream *substream)
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index ab894ed1ff67..8557e54d2659 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -40,6 +40,7 @@ config SND_OXFW
* Mackie(Loud) U.420/U.420d
* TASCAM FireOne
* Stanton Controllers & Systems 1 Deck/Mixer
+ * APOGEE duet FireWire
To compile this driver as a module, choose M here: the module
will be called snd-oxfw.
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index d0dfa822266b..a205b93fd9ac 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -434,7 +434,7 @@ static const struct ieee1394_device_id bebob_id_table[] = {
/* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */
SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal),
/* Apogee Electronics, Ensemble */
- SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal),
+ SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal),
/* ESI, Quatafire610 */
SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal),
/* AcousticReality, eARMasterOne */
@@ -474,7 +474,19 @@ static const struct ieee1394_device_id bebob_id_table[] = {
/* Focusrite, SaffirePro 26 I/O */
SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec),
/* Focusrite, SaffirePro 10 I/O */
- SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec),
+ {
+ // The combination of vendor_id and model_id is the same as the
+ // same as the one of Liquid Saffire 56.
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = VEN_FOCUSRITE,
+ .model_id = 0x000006,
+ .specifier_id = 0x00a02d,
+ .version = 0x010001,
+ .driver_data = (kernel_ulong_t)&saffirepro_10_spec,
+ },
/* Focusrite, Saffire(no label and LE) */
SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH,
&saffire_spec),
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 696b6cf35003..b0395c4209ab 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -20,6 +20,7 @@
#define VENDOR_LACIE 0x00d04b
#define VENDOR_TASCAM 0x00022e
#define OUI_STANTON 0x001260
+#define OUI_APOGEE 0x0003db
#define MODEL_SATELLITE 0x00200f
@@ -441,6 +442,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = {
.vendor_id = OUI_STANTON,
.model_id = 0x002000,
},
+ // APOGEE, duet FireWire
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_APOGEE,
+ .model_id = 0x01dddd,
+ },
{ }
};
MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table);
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index 4a0cbd2241d8..3191666ac129 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -899,6 +899,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip)
struct dsp_spos_instance * ins = chip->dsp_spos_instance;
int i;
+ if (!ins)
+ return 0;
+
snd_info_free_entry(ins->proc_sym_info_entry);
ins->proc_sym_info_entry = NULL;
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 50b216fc369f..5d422d65e62b 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -36,6 +36,7 @@
#include <linux/init.h>
#include <linux/mutex.h>
#include <linux/moduleparam.h>
+#include <linux/nospec.h>
#include <sound/core.h>
#include <sound/tlv.h>
@@ -1000,6 +1001,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu,
if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
return -EINVAL;
+ ipcm->substream = array_index_nospec(ipcm->substream,
+ EMU10K1_FX8010_PCM_COUNT);
if (ipcm->channels > 32)
return -EINVAL;
pcm = &emu->fx8010.pcm[ipcm->substream];
@@ -1046,6 +1049,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu,
if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
return -EINVAL;
+ ipcm->substream = array_index_nospec(ipcm->substream,
+ EMU10K1_FX8010_PCM_COUNT);
pcm = &emu->fx8010.pcm[ipcm->substream];
mutex_lock(&emu->fx8010.lock);
spin_lock_irq(&emu->reg_lock);
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 6efadbfb3fe3..7ea201c05e5d 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -109,7 +109,8 @@ static int hda_codec_driver_probe(struct device *dev)
err = snd_hda_codec_build_controls(codec);
if (err < 0)
goto error_module;
- if (codec->card->registered) {
+ /* only register after the bus probe finished; otherwise it's racy */
+ if (!codec->bus->bus_probing && codec->card->registered) {
err = snd_card_register(codec->card);
if (err < 0)
goto error_module;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c6b046ddefdd..1b5e217d1bb2 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3004,6 +3004,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
hda_jackpoll_work(&codec->jackpoll_work.work);
else
snd_hda_jack_report_sync(codec);
+ codec->core.dev.power.power_state = PMSG_ON;
atomic_dec(&codec->core.in_pm);
}
@@ -3036,10 +3037,62 @@ static int hda_codec_runtime_resume(struct device *dev)
}
#endif /* CONFIG_PM */
+#ifdef CONFIG_PM_SLEEP
+static int hda_codec_force_resume(struct device *dev)
+{
+ int ret;
+
+ /* The get/put pair below enforces the runtime resume even if the
+ * device hasn't been used at suspend time. This trick is needed to
+ * update the jack state change during the sleep.
+ */
+ pm_runtime_get_noresume(dev);
+ ret = pm_runtime_force_resume(dev);
+ pm_runtime_put(dev);
+ return ret;
+}
+
+static int hda_codec_pm_suspend(struct device *dev)
+{
+ dev->power.power_state = PMSG_SUSPEND;
+ return pm_runtime_force_suspend(dev);
+}
+
+static int hda_codec_pm_resume(struct device *dev)
+{
+ dev->power.power_state = PMSG_RESUME;
+ return hda_codec_force_resume(dev);
+}
+
+static int hda_codec_pm_freeze(struct device *dev)
+{
+ dev->power.power_state = PMSG_FREEZE;
+ return pm_runtime_force_suspend(dev);
+}
+
+static int hda_codec_pm_thaw(struct device *dev)
+{
+ dev->power.power_state = PMSG_THAW;
+ return hda_codec_force_resume(dev);
+}
+
+static int hda_codec_pm_restore(struct device *dev)
+{
+ dev->power.power_state = PMSG_RESTORE;
+ return hda_codec_force_resume(dev);
+}
+#endif /* CONFIG_PM_SLEEP */
+
/* referred in hda_bind.c */
const struct dev_pm_ops hda_codec_driver_pm = {
- SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
- pm_runtime_force_resume)
+#ifdef CONFIG_PM_SLEEP
+ .suspend = hda_codec_pm_suspend,
+ .resume = hda_codec_pm_resume,
+ .freeze = hda_codec_pm_freeze,
+ .thaw = hda_codec_pm_thaw,
+ .poweroff = hda_codec_pm_suspend,
+ .restore = hda_codec_pm_restore,
+#endif /* CONFIG_PM_SLEEP */
SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume,
NULL)
};
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 776dffa88aee..171e11be938d 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -68,6 +68,7 @@ struct hda_bus {
unsigned int response_reset:1; /* controller was reset */
unsigned int in_reset:1; /* during reset operation */
unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
+ unsigned int bus_probing :1; /* during probing process */
int primary_dig_out_type; /* primary digital out PCM type */
unsigned int mixer_assigned; /* codec addr for mixer name */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3557e3943ad5..789eca17fc60 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2089,6 +2089,7 @@ static int azx_probe_continue(struct azx *chip)
int val;
int err;
+ to_hda_bus(bus)->bus_probing = 1;
hda->probe_continued = 1;
/* Request display power well for the HDA controller or codec. For
@@ -2189,6 +2190,7 @@ i915_power_fail:
if (err < 0)
hda->init_failed = 1;
complete_all(&hda->probe_wait);
+ to_hda_bus(bus)->bus_probing = 0;
return err;
}
@@ -2352,6 +2354,10 @@ static const struct pci_device_id azx_ids[] = {
/* AMD Hudson */
{ PCI_DEVICE(0x1022, 0x780d),
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+ /* AMD Stoney */
+ { PCI_DEVICE(0x1022, 0x157a),
+ .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
+ AZX_DCAPS_PM_RUNTIME },
/* AMD Raven */
{ PCI_DEVICE(0x1022, 0x15e3),
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 0621920f7617..e85fb04ec7be 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -249,10 +249,12 @@ static int hda_tegra_suspend(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
+ struct hdac_bus *bus = azx_bus(chip);
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
azx_stop_chip(chip);
+ synchronize_irq(bus->irq);
azx_enter_link_reset(chip);
hda_tegra_disable_clocks(hda);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d392e867e9ab..447b3a8a83c3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -853,6 +853,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK),
+ SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
+ SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6c2668b4e3bc..0fc05ebdf81a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4489,9 +4489,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec,
{ 0x19, 0x21a11010 }, /* dock mic */
{ }
};
+ /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise
+ * the speaker output becomes too low by some reason on Thinkpads with
+ * ALC298 codec
+ */
+ static hda_nid_t preferred_pairs[] = {
+ 0x14, 0x03, 0x17, 0x02, 0x21, 0x02,
+ 0
+ };
struct alc_spec *spec = codec->spec;
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.preferred_dacs = preferred_pairs;
spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
snd_hda_apply_pincfgs(codec, pincfgs);
} else if (action == HDA_FIXUP_ACT_INIT) {
@@ -4832,6 +4841,13 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec,
}
}
+static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action == HDA_FIXUP_ACT_PRE_PROBE)
+ snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
+}
+
/* for hda_fixup_thinkpad_acpi() */
#include "thinkpad_helper.c"
@@ -4938,6 +4954,7 @@ enum {
ALC293_FIXUP_LENOVO_SPK_NOISE,
ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
ALC255_FIXUP_DELL_SPK_NOISE,
+ ALC225_FIXUP_DISABLE_MIC_VREF,
ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC295_FIXUP_DISABLE_DAC3,
ALC280_FIXUP_HP_HEADSET_MIC,
@@ -5596,6 +5613,12 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
},
+ [ALC225_FIXUP_DISABLE_MIC_VREF] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_disable_mic_vref,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+ },
[ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
.type = HDA_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -5605,7 +5628,7 @@ static const struct hda_fixup alc269_fixups[] = {
{}
},
.chained = true,
- .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+ .chain_id = ALC225_FIXUP_DISABLE_MIC_VREF
},
[ALC280_FIXUP_HP_HEADSET_MIC] = {
.type = HDA_FIXUP_FUNC,
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index b94fc6357139..b044dea3c815 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -30,6 +30,7 @@
#include <linux/math64.h>
#include <linux/vmalloc.h>
#include <linux/io.h>
+#include <linux/nospec.h>
#include <sound/core.h>
#include <sound/control.h>
@@ -4065,15 +4066,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream,
struct snd_pcm_channel_info *info)
{
struct hdsp *hdsp = snd_pcm_substream_chip(substream);
- int mapped_channel;
+ unsigned int channel = info->channel;
- if (snd_BUG_ON(info->channel >= hdsp->max_channels))
+ if (snd_BUG_ON(channel >= hdsp->max_channels))
return -EINVAL;
+ channel = array_index_nospec(channel, hdsp->max_channels);
- if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
+ if (hdsp->channel_map[channel] < 0)
return -EINVAL;
- info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES;
+ info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES;
info->first = 0;
info->step = 32;
return 0;
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 09103aab0cb2..7d410e39d1a0 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -253,6 +253,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform)
rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp),
GFP_KERNEL);
+ if (!rt5514_dsp)
+ return -ENOMEM;
rt5514_dsp->dev = &rt5514_spi->dev;
mutex_init(&rt5514_dsp->dma_lock);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 0ba0bf13c3a9..887af25f7367 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -281,7 +281,7 @@ config SND_SOC_PHYCORE_AC97
config SND_SOC_EUKREA_TLV320
tristate "Eukrea TLV320"
- depends on ARCH_MXC && I2C
+ depends on ARCH_MXC && !ARM64 && I2C
select SND_SOC_TLV320AIC23_I2C
select SND_SOC_IMX_AUDMUX
select SND_SOC_IMX_SSI
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 17766f8b09a8..f01f7be66b37 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -88,49 +88,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
if (!buf)
return -ENOMEM;
- ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
+ ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
pdcr, ptcr);
if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxFS output from %s, ",
audmux_port_string((ptcr >> 27) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxFS input, ");
if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxClk output from %s",
audmux_port_string((ptcr >> 22) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxClk input");
- ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) {
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"Port is symmetric");
} else {
if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxFS output from %s, ",
audmux_port_string((ptcr >> 17) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxFS input, ");
if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxClk output from %s",
audmux_port_string((ptcr >> 12) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxClk input");
}
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"\nData received from %s\n",
audmux_port_string((pdcr >> 13) & 0x7));
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index f5a8050351b5..e83e314a76a5 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ int ret;
+
+ ret =
+ snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(params));
+ if (ret)
+ return ret;
memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
return 0;
}
diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c
index 33917146d9c4..054b1d514e8a 100644
--- a/sound/soc/intel/atom/sst/sst_loader.c
+++ b/sound/soc/intel/atom/sst/sst_loader.c
@@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst)
const struct firmware *fw;
retval = request_firmware(&fw, sst->firmware_name, sst->dev);
- if (fw == NULL) {
- dev_err(sst->dev, "fw is returning as null\n");
- return -EINVAL;
- }
if (retval) {
dev_err(sst->dev, "request fw failed %d\n", retval);
return retval;
}
+ if (fw == NULL) {
+ dev_err(sst->dev, "fw is returning as null\n");
+ return -EINVAL;
+ }
mutex_lock(&sst->sst_lock);
retval = sst_cache_and_parse_fw(sst, fw);
mutex_unlock(&sst->sst_lock);
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 7486a0022fde..993d2c105ae1 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -191,7 +191,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.stream_name = "Loopback",
.cpu_dai_name = "Loopback Pin",
.platform_name = "haswell-pcm-audio",
- .dynamic = 0,
+ .dynamic = 1,
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 863f1d5e2a2c..11d0cc2b0e39 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
.stream_name = "Loopback",
.cpu_dai_name = "Loopback Pin",
.platform_name = "haswell-pcm-audio",
- .dynamic = 0,
+ .dynamic = 1,
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 89fe95e877db..07af30017b48 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -36,6 +36,8 @@
#include "../codecs/twl6040.h"
struct abe_twl6040 {
+ struct snd_soc_card card;
+ struct snd_soc_dai_link dai_links[2];
int jack_detection; /* board can detect jack events */
int mclk_freq; /* MCLK frequency speed for twl6040 */
};
@@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
ARRAY_SIZE(dmic_audio_map));
}
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
- {
- .name = "TWL6040",
- .stream_name = "TWL6040",
- .codec_dai_name = "twl6040-legacy",
- .codec_name = "twl6040-codec",
- .init = omap_abe_twl6040_init,
- .ops = &omap_abe_ops,
- },
- {
- .name = "DMIC",
- .stream_name = "DMIC Capture",
- .codec_dai_name = "dmic-hifi",
- .codec_name = "dmic-codec",
- .init = omap_abe_dmic_init,
- .ops = &omap_abe_dmic_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card omap_abe_card = {
- .owner = THIS_MODULE,
-
- .dapm_widgets = twl6040_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
static int omap_abe_probe(struct platform_device *pdev)
{
struct device_node *node = pdev->dev.of_node;
- struct snd_soc_card *card = &omap_abe_card;
+ struct snd_soc_card *card;
struct device_node *dai_node;
struct abe_twl6040 *priv;
int num_links = 0;
@@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev)
return -ENODEV;
}
- card->dev = &pdev->dev;
-
priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
if (priv == NULL)
return -ENOMEM;
+ card = &priv->card;
+ card->dev = &pdev->dev;
+ card->owner = THIS_MODULE;
+ card->dapm_widgets = twl6040_dapm_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
+ card->dapm_routes = audio_map;
+ card->num_dapm_routes = ARRAY_SIZE(audio_map);
+
if (snd_soc_of_parse_card_name(card, "ti,model")) {
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
@@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "McPDM node is not provided\n");
return -EINVAL;
}
- abe_twl6040_dai_links[0].cpu_of_node = dai_node;
- abe_twl6040_dai_links[0].platform_of_node = dai_node;
+
+ priv->dai_links[0].name = "DMIC";
+ priv->dai_links[0].stream_name = "TWL6040";
+ priv->dai_links[0].cpu_of_node = dai_node;
+ priv->dai_links[0].platform_of_node = dai_node;
+ priv->dai_links[0].codec_dai_name = "twl6040-legacy";
+ priv->dai_links[0].codec_name = "twl6040-codec";
+ priv->dai_links[0].init = omap_abe_twl6040_init;
+ priv->dai_links[0].ops = &omap_abe_ops;
dai_node = of_parse_phandle(node, "ti,dmic", 0);
if (dai_node) {
num_links = 2;
- abe_twl6040_dai_links[1].cpu_of_node = dai_node;
- abe_twl6040_dai_links[1].platform_of_node = dai_node;
+ priv->dai_links[1].name = "TWL6040";
+ priv->dai_links[1].stream_name = "DMIC Capture";
+ priv->dai_links[1].cpu_of_node = dai_node;
+ priv->dai_links[1].platform_of_node = dai_node;
+ priv->dai_links[1].codec_dai_name = "dmic-hifi";
+ priv->dai_links[1].codec_name = "dmic-codec";
+ priv->dai_links[1].init = omap_abe_dmic_init;
+ priv->dai_links[1].ops = &omap_abe_dmic_ops;
} else {
num_links = 1;
}
@@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev)
return -ENODEV;
}
- card->dai_link = abe_twl6040_dai_links;
+ card->dai_link = priv->dai_links;
card->num_links = num_links;
snd_soc_card_set_drvdata(card, priv);
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 09db2aec12a3..776e809a8aab 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -48,6 +48,8 @@ struct omap_dmic {
struct device *dev;
void __iomem *io_base;
struct clk *fclk;
+ struct pm_qos_request pm_qos_req;
+ int latency;
int fclk_freq;
int out_freq;
int clk_div;
@@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
mutex_lock(&dmic->mutex);
+ pm_qos_remove_request(&dmic->pm_qos_req);
+
if (!dai->active)
dmic->active = 0;
@@ -226,6 +230,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
/* packet size is threshold * channels */
dma_data = snd_soc_dai_get_dma_data(dai, substream);
dma_data->maxburst = dmic->threshold * channels;
+ dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC /
+ params_rate(params);
return 0;
}
@@ -236,6 +242,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
u32 ctrl;
+ if (pm_qos_request_active(&dmic->pm_qos_req))
+ pm_qos_update_request(&dmic->pm_qos_req, dmic->latency);
+
/* Configure uplink threshold */
omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 64609c77a79d..44ffeb71cd1d 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -54,6 +54,8 @@ struct omap_mcpdm {
unsigned long phys_base;
void __iomem *io_base;
int irq;
+ struct pm_qos_request pm_qos_req;
+ int latency[2];
struct mutex mutex;
@@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+ int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+ int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock(&mcpdm->mutex);
@@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
}
}
+ if (mcpdm->latency[stream2])
+ pm_qos_update_request(&mcpdm->pm_qos_req,
+ mcpdm->latency[stream2]);
+ else if (mcpdm->latency[stream1])
+ pm_qos_remove_request(&mcpdm->pm_qos_req);
+
+ mcpdm->latency[stream1] = 0;
+
mutex_unlock(&mcpdm->mutex);
}
@@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
int stream = substream->stream;
struct snd_dmaengine_dai_dma_data *dma_data;
u32 threshold;
- int channels;
+ int channels, latency;
int link_mask = 0;
channels = params_channels(params);
@@ -340,14 +353,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
dma_data->maxburst =
(MCPDM_DN_THRES_MAX - threshold) * channels;
+ latency = threshold;
} else {
/* If playback is not running assume a stereo stream to come */
if (!mcpdm->config[!stream].link_mask)
mcpdm->config[!stream].link_mask = (0x3 << 3);
dma_data->maxburst = threshold * channels;
+ latency = (MCPDM_DN_THRES_MAX - threshold);
}
+ /*
+ * The DMA must act to a DMA request within latency time (usec) to avoid
+ * under/overflow
+ */
+ mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params);
+
+ if (!mcpdm->latency[stream])
+ mcpdm->latency[stream] = 10;
+
/* Check if we need to restart McPDM with this stream */
if (mcpdm->config[stream].link_mask &&
mcpdm->config[stream].link_mask != link_mask)
@@ -362,6 +386,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+ struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req;
+ int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+ int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ int latency = mcpdm->latency[stream2];
+
+ /* Prevent omap hardware from hitting off between FIFO fills */
+ if (!latency || mcpdm->latency[stream1] < latency)
+ latency = mcpdm->latency[stream1];
+
+ if (pm_qos_request_active(pm_qos_req))
+ pm_qos_update_request(pm_qos_req, latency);
+ else if (latency)
+ pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency);
if (!omap_mcpdm_active(mcpdm)) {
omap_mcpdm_start(mcpdm);
@@ -423,6 +461,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
free_irq(mcpdm->irq, (void *)mcpdm);
pm_runtime_disable(mcpdm->dev);
+ if (pm_qos_request_active(&mcpdm->pm_qos_req))
+ pm_qos_remove_request(&mcpdm->pm_qos_req);
+
return 0;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e3de566809c..168559b5e9f3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2018,6 +2018,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
card->instantiated = 1;
+ dapm_mark_endpoints_dirty(card);
snd_soc_dapm_sync(&card->dapm);
mutex_unlock(&card->mutex);
mutex_unlock(&client_mutex);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8bfc534e3b34..ab647f1fe11b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1976,19 +1976,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
out = is_connected_output_ep(w, NULL, NULL);
}
- ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d",
+ ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d",
w->name, w->power ? "On" : "Off",
w->force ? " (forced)" : "", in, out);
if (w->reg >= 0)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
" - R%d(0x%x) mask 0x%x",
w->reg, w->reg, w->mask << w->shift);
- ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
if (w->sname)
- ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
w->sname,
w->active ? "active" : "inactive");
@@ -2001,7 +2001,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
if (!p->connect)
continue;
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
" %s \"%s\" \"%s\"\n",
(rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out",
p->name ? p->name : "static",
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index d6b48c796bfc..086fe4d27f60 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1989,6 +1989,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id)
{
struct soc_tplg tplg;
+ int ret;
/* setup parsing context */
memset(&tplg, 0, sizeof(tplg));
@@ -2002,7 +2003,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
tplg.bytes_ext_ops = ops->bytes_ext_ops;
tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count;
- return soc_tplg_load(&tplg);
+ ret = soc_tplg_load(&tplg);
+ /* free the created components if fail to load topology */
+ if (ret)
+ snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL);
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load);
diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c
index e557946718a9..d9fcae071b47 100644
--- a/sound/synth/emux/emux_hwdep.c
+++ b/sound/synth/emux/emux_hwdep.c
@@ -22,9 +22,9 @@
#include <sound/core.h>
#include <sound/hwdep.h>
#include <linux/uaccess.h>
+#include <linux/nospec.h>
#include "emux_voice.h"
-
#define TMP_CLIENT_ID 0x1001
/*
@@ -66,13 +66,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg)
return -EFAULT;
if (info.mode < 0 || info.mode >= EMUX_MD_END)
return -EINVAL;
+ info.mode = array_index_nospec(info.mode, EMUX_MD_END);
if (info.port < 0) {
for (i = 0; i < emu->num_ports; i++)
emu->portptrs[i]->ctrls[info.mode] = info.value;
} else {
- if (info.port < emu->num_ports)
+ if (info.port < emu->num_ports) {
+ info.port = array_index_nospec(info.port, emu->num_ports);
emu->portptrs[info.port]->ctrls[info.mode] = info.value;
+ }
}
return 0;
}
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 8906199a83e6..549b9b061694 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -644,9 +644,12 @@ static int usb_audio_probe(struct usb_interface *intf,
__error:
if (chip) {
+ /* chip->active is inside the chip->card object,
+ * decrement before memory is possibly returned.
+ */
+ atomic_dec(&chip->active);
if (!chip->num_interfaces)
snd_card_free(chip->card);
- atomic_dec(&chip->active);
}
mutex_unlock(&register_mutex);
return err;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index db8404e31fae..64b90b8ec661 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1882,7 +1882,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
char *name)
{
struct uac_processing_unit_descriptor *desc = raw_desc;
- int num_ins = desc->bNrInPins;
+ int num_ins;
struct usb_mixer_elem_info *cval;
struct snd_kcontrol *kctl;
int i, err, nameid, type, len;
@@ -1897,7 +1897,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
0, NULL, default_value_info
};
- if (desc->bLength < 13 || desc->bLength < 13 + num_ins ||
+ if (desc->bLength < 13) {
+ usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
+ return -EINVAL;
+ }
+
+ num_ins = desc->bNrInPins;
+ if (desc->bLength < 13 + num_ins ||
desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) {
usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
return -EINVAL;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index e6ac7b9b4648..497bad9f2789 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -313,6 +313,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
return 0;
}
+/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk
+ * applies. Returns 1 if a quirk was found.
+ */
static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
struct usb_device *dev,
struct usb_interface_descriptor *altsd,
@@ -391,7 +394,7 @@ add_sync_ep:
subs->data_endpoint->sync_master = subs->sync_endpoint;
- return 0;
+ return 1;
}
static int set_sync_endpoint(struct snd_usb_substream *subs,
@@ -430,6 +433,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
if (err < 0)
return err;
+ /* endpoint set by quirk */
+ if (err > 0)
+ return 0;
+
if (altsd->bNumEndpoints < 2)
return 0;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 15cbe2565703..d32727c74a16 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3321,6 +3321,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
}
}
},
+ {
+ .ifnum = -1
+ },
}
}
},