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Diffstat (limited to 'sound')
-rw-r--r--sound/core/pcm_lib.c2
-rw-r--r--sound/hda/hdac_device.c16
-rw-r--r--sound/hda/hdac_regmap.c69
-rw-r--r--sound/pci/hda/patch_cirrus.c8
-rw-r--r--sound/pci/hda/patch_conexant.c7
-rw-r--r--sound/pci/hda/patch_hdmi.c43
-rw-r--r--sound/pci/hda/patch_realtek.c1
-rw-r--r--sound/pci/intel8x0.c1
-rw-r--r--sound/usb/clock.c2
-rw-r--r--sound/usb/endpoint.c3
-rw-r--r--sound/usb/mixer_quirks.c4
-rw-r--r--sound/usb/pcm.c2
-rw-r--r--sound/usb/quirks.c27
-rw-r--r--sound/usb/stream.c6
14 files changed, 145 insertions, 46 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 6b5a811e0..3a9b66c6e 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -322,7 +322,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
char name[16];
snd_pcm_debug_name(substream, name, sizeof(name));
pcm_err(substream->pcm,
- "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
+ "invalid position: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
name, pos, runtime->buffer_size,
runtime->period_size);
}
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index e361024ea..d1a4d6973 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -611,6 +611,22 @@ int snd_hdac_power_up_pm(struct hdac_device *codec)
}
EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm);
+/* like snd_hdac_power_up_pm(), but only increment the pm count when
+ * already powered up. Returns -1 if not powered up, 1 if incremented
+ * or 0 if unchanged. Only used in hdac_regmap.c
+ */
+int snd_hdac_keep_power_up(struct hdac_device *codec)
+{
+ if (!atomic_inc_not_zero(&codec->in_pm)) {
+ int ret = pm_runtime_get_if_in_use(&codec->dev);
+ if (!ret)
+ return -1;
+ if (ret < 0)
+ return 0;
+ }
+ return 1;
+}
+
/**
* snd_hdac_power_down_pm - power down the codec
* @codec: the codec object
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index eb8f7c30c..bdbcd6b75 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -21,13 +21,16 @@
#include <sound/hdaudio.h>
#include <sound/hda_regmap.h>
-#ifdef CONFIG_PM
-#define codec_is_running(codec) \
- (atomic_read(&(codec)->in_pm) || \
- !pm_runtime_suspended(&(codec)->dev))
-#else
-#define codec_is_running(codec) true
-#endif
+static int codec_pm_lock(struct hdac_device *codec)
+{
+ return snd_hdac_keep_power_up(codec);
+}
+
+static void codec_pm_unlock(struct hdac_device *codec, int lock)
+{
+ if (lock == 1)
+ snd_hdac_power_down_pm(codec);
+}
#define get_verb(reg) (((reg) >> 8) & 0xfff)
@@ -238,20 +241,28 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val)
struct hdac_device *codec = context;
int verb = get_verb(reg);
int err;
+ int pm_lock = 0;
- if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE)
- return -EAGAIN;
+ if (verb != AC_VERB_GET_POWER_STATE) {
+ pm_lock = codec_pm_lock(codec);
+ if (pm_lock < 0)
+ return -EAGAIN;
+ }
reg |= (codec->addr << 28);
- if (is_stereo_amp_verb(reg))
- return hda_reg_read_stereo_amp(codec, reg, val);
- if (verb == AC_VERB_GET_PROC_COEF)
- return hda_reg_read_coef(codec, reg, val);
+ if (is_stereo_amp_verb(reg)) {
+ err = hda_reg_read_stereo_amp(codec, reg, val);
+ goto out;
+ }
+ if (verb == AC_VERB_GET_PROC_COEF) {
+ err = hda_reg_read_coef(codec, reg, val);
+ goto out;
+ }
if ((verb & 0x700) == AC_VERB_SET_AMP_GAIN_MUTE)
reg &= ~AC_AMP_FAKE_MUTE;
err = snd_hdac_exec_verb(codec, reg, 0, val);
if (err < 0)
- return err;
+ goto out;
/* special handling for asymmetric reads */
if (verb == AC_VERB_GET_POWER_STATE) {
if (*val & AC_PWRST_ERROR)
@@ -259,7 +270,9 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val)
else /* take only the actual state */
*val = (*val >> 4) & 0x0f;
}
- return 0;
+ out:
+ codec_pm_unlock(codec, pm_lock);
+ return err;
}
static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
@@ -267,6 +280,7 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
struct hdac_device *codec = context;
unsigned int verb;
int i, bytes, err;
+ int pm_lock = 0;
if (codec->caps_overwriting)
return 0;
@@ -275,14 +289,21 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
reg |= (codec->addr << 28);
verb = get_verb(reg);
- if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE)
- return codec->lazy_cache ? 0 : -EAGAIN;
+ if (verb != AC_VERB_SET_POWER_STATE) {
+ pm_lock = codec_pm_lock(codec);
+ if (pm_lock < 0)
+ return codec->lazy_cache ? 0 : -EAGAIN;
+ }
- if (is_stereo_amp_verb(reg))
- return hda_reg_write_stereo_amp(codec, reg, val);
+ if (is_stereo_amp_verb(reg)) {
+ err = hda_reg_write_stereo_amp(codec, reg, val);
+ goto out;
+ }
- if (verb == AC_VERB_SET_PROC_COEF)
- return hda_reg_write_coef(codec, reg, val);
+ if (verb == AC_VERB_SET_PROC_COEF) {
+ err = hda_reg_write_coef(codec, reg, val);
+ goto out;
+ }
switch (verb & 0xf00) {
case AC_VERB_SET_AMP_GAIN_MUTE:
@@ -319,10 +340,12 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
reg |= (verb + i) << 8 | ((val >> (8 * i)) & 0xff);
err = snd_hdac_exec_verb(codec, reg, 0, NULL);
if (err < 0)
- return err;
+ goto out;
}
- return 0;
+ out:
+ codec_pm_unlock(codec, pm_lock);
+ return err;
}
static const struct regmap_config hda_regmap_cfg = {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index c1c855a6c..a47e8ae0e 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -174,8 +174,12 @@ static void cs_automute(struct hda_codec *codec)
snd_hda_gen_update_outputs(codec);
if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) {
- spec->gpio_data = spec->gen.hp_jack_present ?
- spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
+ if (spec->gen.automute_speaker)
+ spec->gpio_data = spec->gen.hp_jack_present ?
+ spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
+ else
+ spec->gpio_data =
+ spec->gpio_eapd_hp | spec->gpio_eapd_speaker;
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, spec->gpio_data);
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6122b8ca8..56fefbd85 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -204,8 +204,13 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
- if (codec->core.vendor_id != 0x14f150f2)
+ switch (codec->core.vendor_id) {
+ case 0x14f150f2: /* CX20722 */
+ case 0x14f150f4: /* CX20724 */
+ break;
+ default:
return;
+ }
/* Turn the CX20722 codec into D3 to avoid spurious noises
from the internal speaker during (and after) reboot */
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bcbc4ee10..e68fa449e 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -152,13 +152,17 @@ struct hdmi_spec {
struct hda_pcm_stream pcm_playback;
/* i915/powerwell (Haswell+/Valleyview+) specific */
+ bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */
struct i915_audio_component_audio_ops i915_audio_ops;
bool i915_bound; /* was i915 bound in this driver? */
};
#ifdef CONFIG_SND_HDA_I915
-#define codec_has_acomp(codec) \
- ((codec)->bus->core.audio_component != NULL)
+static inline bool codec_has_acomp(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+ return spec->use_acomp_notifier;
+}
#else
#define codec_has_acomp(codec) false
#endif
@@ -1562,6 +1566,7 @@ static void update_eld(struct hda_codec *codec,
eld->eld_size) != 0)
eld_changed = true;
+ pin_eld->monitor_present = eld->monitor_present;
pin_eld->eld_valid = eld->eld_valid;
pin_eld->eld_size = eld->eld_size;
if (eld->eld_valid)
@@ -1665,11 +1670,10 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
int size;
mutex_lock(&per_pin->lock);
+ eld->monitor_present = false;
size = snd_hdac_acomp_get_eld(&codec->bus->core, per_pin->pin_nid,
&eld->monitor_present, eld->eld_buffer,
ELD_MAX_SIZE);
- if (size < 0)
- goto unlock;
if (size > 0) {
size = min(size, ELD_MAX_SIZE);
if (snd_hdmi_parse_eld(codec, &eld->info,
@@ -1873,7 +1877,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
/* Call sync_audio_rate to set the N/CTS/M manually if necessary */
/* Todo: add DP1.2 MST audio support later */
- snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate);
+ if (codec_has_acomp(codec))
+ snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate);
non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
mutex_lock(&per_pin->lock);
@@ -2432,6 +2437,10 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
struct hda_codec *codec = audio_ptr;
int pin_nid = port + 0x04;
+ /* we assume only from port-B to port-D */
+ if (port < 1 || port > 3)
+ return;
+
/* skip notification during system suspend (but not in runtime PM);
* the state will be updated at resume
*/
@@ -2456,11 +2465,24 @@ static int patch_generic_hdmi(struct hda_codec *codec)
codec->spec = spec;
hdmi_array_init(spec, 4);
- /* Try to bind with i915 for any Intel codecs (if not done yet) */
- if (!codec_has_acomp(codec) &&
- (codec->core.vendor_id >> 16) == 0x8086)
- if (!snd_hdac_i915_init(&codec->bus->core))
- spec->i915_bound = true;
+#ifdef CONFIG_SND_HDA_I915
+ /* Try to bind with i915 for Intel HSW+ codecs (if not done yet) */
+ if ((codec->core.vendor_id >> 16) == 0x8086 &&
+ is_haswell_plus(codec)) {
+#if 0
+ /* on-demand binding leads to an unbalanced refcount when
+ * both i915 and hda drivers are probed concurrently;
+ * disabled temporarily for now
+ */
+ if (!codec->bus->core.audio_component)
+ if (!snd_hdac_i915_init(&codec->bus->core))
+ spec->i915_bound = true;
+#endif
+ /* use i915 audio component notifier for hotplug */
+ if (codec->bus->core.audio_component)
+ spec->use_acomp_notifier = true;
+ }
+#endif
if (is_haswell_plus(codec)) {
intel_haswell_enable_all_pins(codec, true);
@@ -3659,6 +3681,7 @@ HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0082, "GPU 82 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch),
HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 93d2156b6..4f5ca0b9c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5556,6 +5556,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
+ SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 42bcbac80..ccdab29a8 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2879,6 +2879,7 @@ static void intel8x0_measure_ac97_clock(struct intel8x0 *chip)
static struct snd_pci_quirk intel8x0_clock_list[] = {
SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000),
+ SND_PCI_QUIRK(0x1014, 0x0581, "AD1981B", 48000),
SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100),
SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000),
SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 2ed260b10..7ccbcaf6a 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -285,6 +285,8 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
unsigned char data[3];
int err, crate;
+ if (get_iface_desc(alts)->bNumEndpoints < 1)
+ return -EINVAL;
ep = get_endpoint(alts, 0)->bEndpointAddress;
/* if endpoint doesn't have sampling rate control, bail out */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7b1cb365f..c07a7eda4 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -438,6 +438,9 @@ exit_clear:
*
* New endpoints will be added to chip->ep_list and must be freed by
* calling snd_usb_endpoint_free().
+ *
+ * For SND_USB_ENDPOINT_TYPE_SYNC, the caller needs to guarantee that
+ * bNumEndpoints > 1 beforehand.
*/
struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
struct usb_host_interface *alts,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 279025650..f6c3bf79a 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1519,7 +1519,11 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol,
/* use known values for that card: interface#1 altsetting#1 */
iface = usb_ifnum_to_if(chip->dev, 1);
+ if (!iface || iface->num_altsetting < 2)
+ return -EINVAL;
alts = &iface->altsetting[1];
+ if (get_iface_desc(alts)->bNumEndpoints < 1)
+ return -EINVAL;
ep = get_endpoint(alts, 0)->bEndpointAddress;
err = snd_usb_ctl_msg(chip->dev,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 9245f52d4..44d178ee9 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -159,6 +159,8 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
unsigned char data[1];
int err;
+ if (get_iface_desc(alts)->bNumEndpoints < 1)
+ return -EINVAL;
ep = get_endpoint(alts, 0)->bEndpointAddress;
data[0] = 1;
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index c458d60d5..cd7eac28e 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -150,6 +150,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
usb_audio_err(chip, "cannot memdup\n");
return -ENOMEM;
}
+ INIT_LIST_HEAD(&fp->list);
if (fp->nr_rates > MAX_NR_RATES) {
kfree(fp);
return -EINVAL;
@@ -167,19 +168,20 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
stream = (fp->endpoint & USB_DIR_IN)
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
err = snd_usb_add_audio_stream(chip, stream, fp);
- if (err < 0) {
- kfree(fp);
- kfree(rate_table);
- return err;
- }
+ if (err < 0)
+ goto error;
if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber ||
fp->altset_idx >= iface->num_altsetting) {
- kfree(fp);
- kfree(rate_table);
- return -EINVAL;
+ err = -EINVAL;
+ goto error;
}
alts = &iface->altsetting[fp->altset_idx];
altsd = get_iface_desc(alts);
+ if (altsd->bNumEndpoints < 1) {
+ err = -EINVAL;
+ goto error;
+ }
+
fp->protocol = altsd->bInterfaceProtocol;
if (fp->datainterval == 0)
@@ -190,6 +192,12 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
snd_usb_init_pitch(chip, fp->iface, alts, fp);
snd_usb_init_sample_rate(chip, fp->iface, alts, fp, fp->rate_max);
return 0;
+
+ error:
+ list_del(&fp->list); /* unlink for avoiding double-free */
+ kfree(fp);
+ kfree(rate_table);
+ return err;
}
static int create_auto_pcm_quirk(struct snd_usb_audio *chip,
@@ -462,6 +470,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
fp->datainterval = 0;
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ INIT_LIST_HEAD(&fp->list);
switch (fp->maxpacksize) {
case 0x120:
@@ -485,6 +494,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
err = snd_usb_add_audio_stream(chip, stream, fp);
if (err < 0) {
+ list_del(&fp->list); /* unlink for avoiding double-free */
kfree(fp);
return err;
}
@@ -1121,6 +1131,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
switch (chip->usb_id) {
case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */
case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */
+ case USB_ID(0x045E, 0x076E): /* MS Lifecam HD-5001 */
case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */
case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */
case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index c4dc577ab..8e9548bc1 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -314,7 +314,9 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits,
/*
* add this endpoint to the chip instance.
* if a stream with the same endpoint already exists, append to it.
- * if not, create a new pcm stream.
+ * if not, create a new pcm stream. note, fp is added to the substream
+ * fmt_list and will be freed on the chip instance release. do not free
+ * fp or do remove it from the substream fmt_list to avoid double-free.
*/
int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
int stream,
@@ -675,6 +677,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
* (fp->maxpacksize & 0x7ff);
fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
fp->clock = clock;
+ INIT_LIST_HEAD(&fp->list);
/* some quirks for attributes here */
@@ -723,6 +726,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
err = snd_usb_add_audio_stream(chip, stream, fp);
if (err < 0) {
+ list_del(&fp->list); /* unlink for avoiding double-free */
kfree(fp->rate_table);
kfree(fp->chmap);
kfree(fp);
generated by cgit v1.2.3-54-g00ecf (git 2.45.2) at 2024-11-10 18:05:08 +0000