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Date: Fri, 24 Apr 2015 18:14:27 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...uxfoundation.org>
Cc: linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 4.1-rc1
Linus,
please pull sound fixes for v4.1-rc1 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-fix-4.1-rc1
The topmost commit is e8191a8e475551b277d85cd76c3f0f52fdf42e86
----------------------------------------------------------------
sound fixes for 4.1-rc1
Here are a few fixes that have been pending since the previous pull
request: a regression fix for HD-audio multiple SPDIF / HDMI devices,
several ALC256 codec fixes, a couple of i915 HDMI audio fixes, and
various small fixes.
Nothing exciting, just boring, but things good to have.
----------------------------------------------------------------
Alexey Khoroshilov (1):
sound/oss: fix deadlock in sequencer_ioctl(SNDCTL_SEQ_OUTOFBAND)
Dan Carpenter (1):
ALSA: hda - potential (but unlikely) uninitialized variable
David Henningsson (2):
ALSA: hda - simplify azx_has_pm_runtime
ALSA: hda - fix "num_steps = 0" error on ALC256
Gabriele Mazzotta (1):
ALSA: hda - Mute headphone pin on suspend on XPS13 9333
Hui Wang (1):
ALSA: hda - fix headset mic detection problem for one more machine
Kailang Yang (1):
ALSA: hda/realtek - Fix Headphone Mic doesn't recording for ALC256
Mengdong Lin (2):
ALSA: hda - only sync BCLK to the display clock for Haswell & Broadwell
ALSA: hda - add AZX_DCAPS_I915_POWERWELL to Baytrail
Scott Wood (1):
ALSA: intel8x0: Check pci_iomap() success for DEVICE_ALI
Takamichi Horikawa (1):
ALSA: usb-audio: Fix audio output on Roland SC-D70 sound module
Takashi Iwai (1):
ALSA: hda - Fix regression for slave SPDIF setups
Vinod Koul (1):
ALSA: asound.h - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN
---
include/uapi/sound/asound.h | 2 +-
sound/oss/sequencer.c | 12 ++---------
sound/pci/hda/hda_codec.c | 2 +-
sound/pci/hda/hda_controller.h | 2 +-
sound/pci/hda/hda_i915.c | 6 ++++++
sound/pci/hda/hda_intel.c | 5 ++++-
sound/pci/hda/hda_proc.c | 4 ++--
sound/pci/hda/patch_realtek.c | 47 +++++++++++++++++++++++-------------------
sound/pci/intel8x0.c | 4 ++--
sound/usb/format.c | 5 ++++-
sound/usb/quirks-table.h | 30 ++-------------------------
11 files changed, 51 insertions(+), 68 deletions(-)
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 46145a5277fe..a45be6bdcf5b 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -864,7 +864,7 @@ struct snd_ctl_elem_id {
snd_ctl_elem_iface_t iface; /* interface identifier */
unsigned int device; /* device/client number */
unsigned int subdevice; /* subdevice (substream) number */
- unsigned char name[44]; /* ASCII name of item */
+ unsigned char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* ASCII name of item */
unsigned int index; /* index of item */
};
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index c0eea1dfe90f..f19da4b47c1d 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -681,13 +681,8 @@ static int seq_timing_event(unsigned char *event_rec)
break;
case TMR_ECHO:
- if (seq_mode == SEQ_2)
- seq_copy_to_input(event_rec, 8);
- else
- {
- parm = (parm << 8 | SEQ_ECHO);
- seq_copy_to_input((unsigned char *) &parm, 4);
- }
+ parm = (parm << 8 | SEQ_ECHO);
+ seq_copy_to_input((unsigned char *) &parm, 4);
break;
default:;
@@ -1324,7 +1319,6 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
int mode = translate_mode(file);
struct synth_info inf;
struct seq_event_rec event_rec;
- unsigned long flags;
int __user *p = arg;
orig_dev = dev = dev >> 4;
@@ -1479,9 +1473,7 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
case SNDCTL_SEQ_OUTOFBAND:
if (copy_from_user(&event_rec, arg, sizeof(event_rec)))
return -EFAULT;
- spin_lock_irqsave(&lock,flags);
play_event(event_rec.arr);
- spin_unlock_irqrestore(&lock,flags);
return 0;
case SNDCTL_MIDI_INFO:
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e70a7fb393dd..873ed1bce12b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2529,7 +2529,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
if (!d)
return;
for (; *d; d++)
- snd_hdac_regmap_update(&codec->core, nid,
+ snd_hdac_regmap_update(&codec->core, *d,
AC_VERB_SET_DIGI_CONVERT_1, mask, val);
}
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index be1b7ded8d82..0efdb094d21c 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -404,7 +404,7 @@ struct azx {
((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg))
#define azx_has_pm_runtime(chip) \
- (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME))
+ ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)
/* PCM setup */
static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream)
diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c
index 52a85d87c23c..3052a2b095f7 100644
--- a/sound/pci/hda/hda_i915.c
+++ b/sound/pci/hda/hda_i915.c
@@ -55,6 +55,12 @@ void haswell_set_bclk(struct hda_intel *hda)
int cdclk_freq;
unsigned int bclk_m, bclk_n;
struct i915_audio_component *acomp = &hda->audio_component;
+ struct pci_dev *pci = hda->chip.pci;
+
+ /* Only Haswell/Broadwell need set BCLK */
+ if (pci->device != 0x0a0c && pci->device != 0x0c0c
+ && pci->device != 0x0d0c && pci->device != 0x160c)
+ return;
if (!acomp->ops)
return;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e1c210515581..34040d26c94f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -297,6 +297,9 @@ enum {
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
AZX_DCAPS_SNOOP_TYPE(SCH))
+#define AZX_DCAPS_INTEL_BAYTRAIL \
+ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL)
+
#define AZX_DCAPS_INTEL_BRASWELL \
(AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL)
@@ -1992,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM },
/* BayTrail */
{ PCI_DEVICE(0x8086, 0x0f04),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL },
/* Braswell */
{ PCI_DEVICE(0x8086, 0x2284),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL },
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ee6230767c64..baaf7ed06875 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -582,8 +582,8 @@ static void print_conn_list(struct snd_info_buffer *buffer,
/* Get Cache connections info */
cache_len = snd_hda_get_conn_list(codec, nid, &list);
- if (cache_len != conn_len
- || memcmp(list, conn, conn_len)) {
+ if (cache_len >= 0 && (cache_len != conn_len ||
+ memcmp(list, conn, conn_len) != 0)) {
snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len);
if (cache_len > 0) {
snd_iprintf(buffer, " ");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b18b9c67b262..06199e4e930f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4176,17 +4176,15 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
}
}
-static unsigned int alc_power_filter_xps13(struct hda_codec *codec,
- hda_nid_t nid,
- unsigned int power_state)
+static void alc_shutup_dell_xps13(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
+ int hp_pin = spec->gen.autocfg.hp_pins[0];
- /* Avoid pop noises when headphones are plugged in */
- if (spec->gen.hp_jack_present)
- if (nid == codec->core.afg || nid == 0x02 || nid == 0x15)
- return AC_PWRST_D0;
- return snd_hda_gen_path_power_filter(codec, nid, power_state);
+ /* Prevent pop noises when headphones are plugged in */
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ msleep(20);
}
static void alc_fixup_dell_xps13(struct hda_codec *codec,
@@ -4197,8 +4195,7 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec,
struct hda_input_mux *imux = &spec->gen.input_mux;
int i;
- spec->shutup = alc_no_shutup;
- codec->power_filter = alc_power_filter_xps13;
+ spec->shutup = alc_shutup_dell_xps13;
/* Make the internal mic the default input source. */
for (i = 0; i < imux->num_items; i++) {
@@ -5231,6 +5228,16 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{0x1b, 0x411111f0}, \
{0x1e, 0x411111f0}
+#define ALC256_STANDARD_PINS \
+ {0x12, 0x90a60140}, \
+ {0x14, 0x90170110}, \
+ {0x19, 0x411111f0}, \
+ {0x1a, 0x411111f0}, \
+ {0x1b, 0x411111f0}, \
+ {0x1d, 0x40700001}, \
+ {0x1e, 0x411111f0}, \
+ {0x21, 0x02211020}
+
#define ALC282_STANDARD_PINS \
{0x14, 0x90170110}, \
{0x18, 0x411111f0}, \
@@ -5331,15 +5338,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x1d, 0x40700001},
{0x21, 0x02211050}),
SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
- {0x12, 0x90a60140},
- {0x13, 0x40000000},
- {0x14, 0x90170110},
- {0x19, 0x411111f0},
- {0x1a, 0x411111f0},
- {0x1b, 0x411111f0},
- {0x1d, 0x40700001},
- {0x1e, 0x411111f0},
- {0x21, 0x02211020}),
+ ALC256_STANDARD_PINS,
+ {0x13, 0x40000000}),
+ SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC256_STANDARD_PINS,
+ {0x13, 0x411111f0}),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
{0x13, 0x40000000},
@@ -5667,6 +5670,8 @@ static int patch_alc269(struct hda_codec *codec)
break;
case 0x10ec0256:
spec->codec_variant = ALC269_TYPE_ALC256;
+ spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */
+ alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
break;
}
@@ -5680,8 +5685,8 @@ static int patch_alc269(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog && spec->gen.beep_nid)
- set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+ if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid)
+ set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT);
codec->patch_ops = alc_patch_ops;
codec->patch_ops.stream_pm = snd_hda_gen_stream_pm;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 749069aa6997..b120925223ae 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -3101,13 +3101,13 @@ static int snd_intel8x0_create(struct snd_card *card,
chip->bmaddr = pci_iomap(pci, 3, 0);
else
chip->bmaddr = pci_iomap(pci, 1, 0);
+
+ port_inited:
if (!chip->bmaddr) {
dev_err(card->dev, "Controller space ioremap problem\n");
snd_intel8x0_free(chip);
return -EIO;
}
-
- port_inited:
chip->bdbars_count = bdbars[device_type];
/* initialize offsets */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 8bcc87cf5667..789d19ec035d 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -79,7 +79,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
format = 1 << UAC_FORMAT_TYPE_I_PCM;
}
if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) {
- if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ &&
+ if (((chip->usb_id == USB_ID(0x0582, 0x0016)) ||
+ /* Edirol SD-90 */
+ (chip->usb_id == USB_ID(0x0582, 0x000c))) &&
+ /* Roland SC-D70 */
sample_width == 24 && sample_bytes == 2)
sample_bytes = 3;
else if (sample_width > sample_bytes * 8) {
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 07f984d5f516..2f6d3e9a1bcd 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -816,37 +816,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.data = (const struct snd_usb_audio_quirk[]) {
{
.ifnum = 0,
- .type = QUIRK_AUDIO_FIXED_ENDPOINT,
- .data = & (const struct audioformat) {
- .formats = SNDRV_PCM_FMTBIT_S24_3LE,
- .channels = 2,
- .iface = 0,
- .altsetting = 1,
- .altset_idx = 1,
- .attributes = 0,
- .endpoint = 0x01,
- .ep_attr = 0x01,
- .rates = SNDRV_PCM_RATE_CONTINUOUS,
- .rate_min = 44100,
- .rate_max = 44100,
- }
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
},
{
.ifnum = 1,
- .type = QUIRK_AUDIO_FIXED_ENDPOINT,
- .data = & (const struct audioformat) {
- .formats = SNDRV_PCM_FMTBIT_S24_3LE,
- .channels = 2,
- .iface = 1,
- .altsetting = 1,
- .altset_idx = 1,
- .attributes = 0,
- .endpoint = 0x81,
- .ep_attr = 0x01,
- .rates = SNDRV_PCM_RATE_CONTINUOUS,
- .rate_min = 44100,
- .rate_max = 44100,
- }
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
},
{
.ifnum = 2,
--
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