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authorLinus Torvalds <torvalds@linux-foundation.org>2020-09-04 12:05:25 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2020-09-04 12:05:25 -0700
commit86edf52e7c7201fabfba39ae694a5206d48e77af (patch)
tree1cbcb6838b63fc74db715147855c4b4b01e4df08
parentcf85f5de83b19361c3d575fa0ea05d8194bb0d05 (diff)
parent6a6660d049f88b89fd9a4b9db3581b245f7782fa (diff)
Merge tag 'sound-5.9-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "A collection of small changes, nothing intrusive: - remaining tasklet API conversions, now all sound stuff have been converted - a few HD-audio and USB-audio quirks and minor fixes - FireWire Tascam and Digi00xx fixes - drop a kernel WARNING from PCM OSS for syzkaller" * tag 'sound-5.9-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits) ALSA: hda/realtek - Improved routing for Thinkpad X1 7th/8th Gen ALSA: hda: use consistent HDAudio spelling in comments/docs ALSA: hda: add dev_dbg log when driver is not selected ALSA: hda: fix a runtime pm issue in SOF when integrated GPU is disabled ALSA: hda: hdmi - add Rocketlake support ALSA: ua101: convert tasklets to use new tasklet_setup() API ALSA: usb-audio: convert tasklets to use new tasklet_setup() API ASoC: txx9: convert tasklets to use new tasklet_setup() API ASoC: siu: convert tasklets to use new tasklet_setup() API ASoC: fsl_esai: convert tasklets to use new tasklet_setup() API ALSA: hdsp: convert tasklets to use new tasklet_setup() API ALSA: riptide: convert tasklets to use new tasklet_setup() API ALSA: pci/asihpi: convert tasklets to use new tasklet_setup() API ALSA: firewire: convert tasklets to use new tasklet_setup() API ALSA: core: convert tasklets to use new tasklet_setup() API ALSA: pcm: oss: Remove superfluous WARN_ON() for mulaw sanity check ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO ALSA: hda/hdmi: always check pin power status in i915 pin fixup ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion NT950XCJ-X716A ALSA: usb-audio: Add basic capture support for Pioneer DJ DJM-250MK2 ...
-rw-r--r--Documentation/sound/cards/audigy-mixer.rst2
-rw-r--r--Documentation/sound/cards/sb-live-mixer.rst2
-rw-r--r--Documentation/sound/designs/timestamping.rst2
-rw-r--r--sound/core/oss/mulaw.c4
-rw-r--r--sound/core/timer.c7
-rw-r--r--sound/firewire/amdtp-stream.c8
-rw-r--r--sound/firewire/digi00x/digi00x.c5
-rw-r--r--sound/firewire/tascam/tascam.c33
-rw-r--r--sound/hda/hdac_device.c2
-rw-r--r--sound/hda/intel-dsp-config.c10
-rw-r--r--sound/pci/asihpi/asihpi.c9
-rw-r--r--sound/pci/ca0106/ca0106_main.c3
-rw-r--r--sound/pci/hda/hda_intel.c7
-rw-r--r--sound/pci/hda/hda_tegra.c7
-rw-r--r--sound/pci/hda/patch_hdmi.c7
-rw-r--r--sound/pci/hda/patch_realtek.c46
-rw-r--r--sound/pci/riptide/riptide.c6
-rw-r--r--sound/pci/rme9652/hdsp.c6
-rw-r--r--sound/pci/rme9652/hdspm.c7
-rw-r--r--sound/soc/fsl/fsl_esai.c7
-rw-r--r--sound/soc/sh/siu_pcm.c10
-rw-r--r--sound/soc/txx9/txx9aclc.c7
-rw-r--r--sound/usb/midi.c7
-rw-r--r--sound/usb/misc/ua101.c7
-rw-r--r--sound/usb/pcm.c2
-rw-r--r--sound/usb/quirks-table.h78
-rw-r--r--sound/usb/quirks.c11
-rw-r--r--sound/usb/usbaudio.h1
-rw-r--r--sound/x86/Kconfig2
29 files changed, 227 insertions, 78 deletions
diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst
index 998f76e19cdd..f3f4640ee2af 100644
--- a/Documentation/sound/cards/audigy-mixer.rst
+++ b/Documentation/sound/cards/audigy-mixer.rst
@@ -332,7 +332,7 @@ WO 9901953 (A1)
US Patents (https://www.uspto.gov/)
-----------------------------------
+-----------------------------------
US 5925841
Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst
index eccb0f0ffd0f..2ce41d3822d8 100644
--- a/Documentation/sound/cards/sb-live-mixer.rst
+++ b/Documentation/sound/cards/sb-live-mixer.rst
@@ -337,7 +337,7 @@ WO 9901953 (A1)
US Patents (https://www.uspto.gov/)
-----------------------------------
+-----------------------------------
US 5925841
Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
diff --git a/Documentation/sound/designs/timestamping.rst b/Documentation/sound/designs/timestamping.rst
index 2b0fff503415..7c7ecf5dbc4b 100644
--- a/Documentation/sound/designs/timestamping.rst
+++ b/Documentation/sound/designs/timestamping.rst
@@ -143,7 +143,7 @@ timestamp shows when the information is put together by the driver
before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases
this driver_timestamp will be identical to the regular system tstamp.
-Examples of typestamping with HDaudio:
+Examples of timestamping with HDAudio:
1. DMA timestamp, no compensation for DMA+analog delay
::
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 3788906421a7..fe27034f2846 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
snd_BUG();
return -EINVAL;
}
- if (snd_BUG_ON(!snd_pcm_format_linear(format->format)))
- return -ENXIO;
+ if (!snd_pcm_format_linear(format->format))
+ return -EINVAL;
err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion",
src_format, dst_format,
diff --git a/sound/core/timer.c b/sound/core/timer.c
index d9f85f2d66a3..6e27d87b18ed 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -816,9 +816,9 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer,
* timer tasklet
*
*/
-static void snd_timer_tasklet(unsigned long arg)
+static void snd_timer_tasklet(struct tasklet_struct *t)
{
- struct snd_timer *timer = (struct snd_timer *) arg;
+ struct snd_timer *timer = from_tasklet(timer, t, task_queue);
unsigned long flags;
if (timer->card && timer->card->shutdown) {
@@ -967,8 +967,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid,
INIT_LIST_HEAD(&timer->ack_list_head);
INIT_LIST_HEAD(&timer->sack_list_head);
spin_lock_init(&timer->lock);
- tasklet_init(&timer->task_queue, snd_timer_tasklet,
- (unsigned long)timer);
+ tasklet_setup(&timer->task_queue, snd_timer_tasklet);
timer->max_instances = 1000; /* default limit per timer */
if (card != NULL) {
timer->module = card->module;
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index f8586f75441d..ee1c428b1fd3 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -64,7 +64,7 @@
#define IT_PKT_HEADER_SIZE_CIP 8 // For 2 CIP header.
#define IT_PKT_HEADER_SIZE_NO_CIP 0 // Nothing.
-static void pcm_period_tasklet(unsigned long data);
+static void pcm_period_tasklet(struct tasklet_struct *t);
/**
* amdtp_stream_init - initialize an AMDTP stream structure
@@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
s->flags = flags;
s->context = ERR_PTR(-1);
mutex_init(&s->mutex);
- tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s);
+ tasklet_setup(&s->period_tasklet, pcm_period_tasklet);
s->packet_index = 0;
init_waitqueue_head(&s->callback_wait);
@@ -441,9 +441,9 @@ static void update_pcm_pointers(struct amdtp_stream *s,
}
}
-static void pcm_period_tasklet(unsigned long data)
+static void pcm_period_tasklet(struct tasklet_struct *t)
{
- struct amdtp_stream *s = (void *)data;
+ struct amdtp_stream *s = from_tasklet(s, t, period_tasklet);
struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
if (pcm)
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index c84b913a9fe0..ab8408966ec3 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2");
#define VENDOR_DIGIDESIGN 0x00a07e
#define MODEL_CONSOLE 0x000001
#define MODEL_RACK 0x000002
+#define SPEC_VERSION 0x000001
static int name_card(struct snd_dg00x *dg00x)
{
@@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = {
/* Both of 002/003 use the same ID. */
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_VERSION |
IEEE1394_MATCH_MODEL_ID,
.vendor_id = VENDOR_DIGIDESIGN,
+ .version = SPEC_VERSION,
.model_id = MODEL_CONSOLE,
},
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_VERSION |
IEEE1394_MATCH_MODEL_ID,
.vendor_id = VENDOR_DIGIDESIGN,
+ .version = SPEC_VERSION,
.model_id = MODEL_RACK,
},
{}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index 5dac0d9fc58e..75f2edd8e78f 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = {
.midi_capture_ports = 2,
.midi_playback_ports = 4,
},
- // This kernel module doesn't support FE-8 because the most of features
- // can be implemented in userspace without any specific support of this
- // module.
};
static int identify_model(struct snd_tscm *tscm)
@@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit)
}
static const struct ieee1394_device_id snd_tscm_id_table[] = {
+ // Tascam, FW-1884.
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ .version = 0x800000,
+ },
+ // Tascam, FE-8 (.version = 0x800001)
+ // This kernel module doesn't support FE-8 because the most of features
+ // can be implemented in userspace without any specific support of this
+ // module.
+ //
+ // .version = 0x800002 is unknown.
+ //
+ // Tascam, FW-1082.
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ .version = 0x800003,
+ },
+ // Tascam, FW-1804.
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
- IEEE1394_MATCH_SPECIFIER_ID,
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
.vendor_id = 0x00022e,
.specifier_id = 0x00022e,
+ .version = 0x800004,
},
{}
};
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 333220f0f8af..3e9e9ac804f6 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init);
void snd_hdac_device_exit(struct hdac_device *codec)
{
pm_runtime_put_noidle(&codec->dev);
+ /* keep balance of runtime PM child_count in parent device */
+ pm_runtime_set_suspended(&codec->dev);
snd_hdac_bus_remove_device(codec->bus, codec);
kfree(codec->vendor_name);
kfree(codec->chip_name);
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index 99aec7349167..1c5114dedda9 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -54,7 +54,7 @@ static const struct config_entry config_table[] = {
#endif
/*
* Apollolake (Broxton-P)
- * the legacy HDaudio driver is used except on Up Squared (SOF) and
+ * the legacy HDAudio driver is used except on Up Squared (SOF) and
* Chromebooks (SST)
*/
#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE)
@@ -89,7 +89,7 @@ static const struct config_entry config_table[] = {
},
#endif
/*
- * Skylake and Kabylake use legacy HDaudio driver except for Google
+ * Skylake and Kabylake use legacy HDAudio driver except for Google
* Chromebooks (SST)
*/
@@ -135,7 +135,7 @@ static const struct config_entry config_table[] = {
#endif
/*
- * Geminilake uses legacy HDaudio driver except for Google
+ * Geminilake uses legacy HDAudio driver except for Google
* Chromebooks
*/
/* Geminilake */
@@ -157,7 +157,7 @@ static const struct config_entry config_table[] = {
/*
* CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy
- * HDaudio driver except for Google Chromebooks and when DMICs are
+ * HDAudio driver except for Google Chromebooks and when DMICs are
* present. Two cases are required since Coreboot does not expose NHLT
* tables.
*
@@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
if (pci->class == 0x040300)
return SND_INTEL_DSP_DRIVER_LEGACY;
if (pci->class != 0x040100 && pci->class != 0x040380) {
- dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class);
+ dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class);
return SND_INTEL_DSP_DRIVER_LEGACY;
}
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 023c35a2a951..35e76480306e 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -921,10 +921,10 @@ static void snd_card_asihpi_timer_function(struct timer_list *t)
add_timer(&dpcm->timer);
}
-static void snd_card_asihpi_int_task(unsigned long data)
+static void snd_card_asihpi_int_task(struct tasklet_struct *t)
{
- struct hpi_adapter *a = (struct hpi_adapter *)data;
- struct snd_card_asihpi *asihpi;
+ struct snd_card_asihpi *asihpi = from_tasklet(asihpi, t, t);
+ struct hpi_adapter *a = asihpi->hpi;
WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
@@ -2871,8 +2871,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
if (hpi->interrupt_mode) {
asihpi->pcm_start = snd_card_asihpi_pcm_int_start;
asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop;
- tasklet_init(&asihpi->t, snd_card_asihpi_int_task,
- (unsigned long)hpi);
+ tasklet_setup(&asihpi->t, snd_card_asihpi_int_task);
hpi->interrupt_callback = snd_card_asihpi_isr;
} else {
asihpi->pcm_start = snd_card_asihpi_pcm_timer_start;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 70d775ff967e..c189f70c82cb 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id,
else
/* Power down */
chip->spi_dac_reg[reg] |= bit;
- return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0)
+ return -ENXIO;
}
return 0;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e34a4d5d047c..36a9dbc33aa0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2127,9 +2127,10 @@ static int azx_probe(struct pci_dev *pci,
*/
if (dmic_detect) {
err = snd_intel_dsp_driver_probe(pci);
- if (err != SND_INTEL_DSP_DRIVER_ANY &&
- err != SND_INTEL_DSP_DRIVER_LEGACY)
+ if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) {
+ dev_dbg(&pci->dev, "HDAudio driver not selected, aborting probe\n");
return -ENODEV;
+ }
} else {
dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n");
}
@@ -2745,8 +2746,6 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
/* Zhaoxin */
{ PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN },
- /* Loongson */
- { PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC },
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index c94553bcca88..70164d1428d4 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -179,6 +179,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev)
struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
if (chip && chip->running) {
+ /* enable controller wake up event */
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+ STATESTS_INT_MASK);
+
azx_stop_chip(chip);
azx_enter_link_reset(chip);
}
@@ -200,6 +204,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev)
if (chip && chip->running) {
hda_tegra_init(hda);
azx_init_chip(chip, 1);
+ /* disable controller wake up event*/
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+ ~STATESTS_INT_MASK);
}
return 0;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index b8c8490e568b..402050088090 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
hda_nid_t cvt_nid)
{
if (per_pin) {
+ haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid);
snd_hda_set_dev_select(codec, per_pin->pin_nid,
per_pin->dev_id);
intel_verify_pin_cvt_connect(codec, per_pin);
@@ -3734,6 +3735,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec)
static int patch_tegra_hdmi(struct hda_codec *codec)
{
+ struct hdmi_spec *spec;
int err;
err = patch_generic_hdmi(codec);
@@ -3741,6 +3743,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec)
return err;
codec->patch_ops.build_pcms = tegra_hdmi_build_pcms;
+ spec = codec->spec;
+ spec->chmap.ops.chmap_cea_alloc_validate_get_type =
+ nvhdmi_chmap_cea_alloc_validate_get_type;
+ spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
return 0;
}
@@ -4263,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi),
HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi),
HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi),
HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a1fa983d2a94..c521a1f17096 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2475,6 +2475,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
+ SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
@@ -5867,6 +5868,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec,
}
}
+/* Quirk for Thinkpad X1 7th and 8th Gen
+ * The following fixed routing needed
+ * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly
+ * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC
+ * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp
+ */
+static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+ static const hda_nid_t preferred_pairs[] = {
+ 0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0
+ };
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+ spec->gen.preferred_dacs = preferred_pairs;
+ break;
+ case HDA_FIXUP_ACT_BUILD:
+ /* The generic parser creates somewhat unintuitive volume ctls
+ * with the fixed routing above, and the shared DAC2 may be
+ * confusing for PA.
+ * Rename those to unique names so that PA doesn't touch them
+ * and use only Master volume.
+ */
+ rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume");
+ rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume");
+ break;
+ }
+}
+
static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -6135,6 +6169,7 @@ enum {
ALC289_FIXUP_DUAL_SPK,
ALC294_FIXUP_SPK2_TO_DAC1,
ALC294_FIXUP_ASUS_DUAL_SPK,
+ ALC285_FIXUP_THINKPAD_X1_GEN7,
ALC285_FIXUP_THINKPAD_HEADSET_JACK,
ALC294_FIXUP_ASUS_HPE,
ALC294_FIXUP_ASUS_COEF_1B,
@@ -7280,11 +7315,17 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC294_FIXUP_SPK2_TO_DAC1
},
+ [ALC285_FIXUP_THINKPAD_X1_GEN7] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_thinkpad_x1_gen7,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_THINKPAD_ACPI
+ },
[ALC285_FIXUP_THINKPAD_HEADSET_JACK] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_headset_jack,
.chained = true,
- .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1
+ .chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7
},
[ALC294_FIXUP_ASUS_HPE] = {
.type = HDA_FIXUP_VERBS,
@@ -7695,7 +7736,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
- SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index b4f300281822..098c69b3b7aa 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1070,9 +1070,9 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval,
return 0;
}
-static void riptide_handleirq(unsigned long dev_id)
+static void riptide_handleirq(struct tasklet_struct *t)
{
- struct snd_riptide *chip = (void *)dev_id;
+ struct snd_riptide *chip = from_tasklet(chip, t, riptide_tq);
struct cmdif *cif = chip->cif;
struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1];
struct snd_pcm_runtime *runtime;
@@ -1843,7 +1843,7 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci,
chip->received_irqs = 0;
chip->handled_irqs = 0;
chip->cif = NULL;
- tasklet_init(&chip->riptide_tq, riptide_handleirq, (unsigned long)chip);
+ tasklet_setup(&chip->riptide_tq, riptide_handleirq);
if ((chip->res_port =
request_region(chip->port, 64, "RIPTIDE")) == NULL) {
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 227aece17e39..dda56ecfd33b 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -3791,9 +3791,9 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
return 0;
}
-static void hdsp_midi_tasklet(unsigned long arg)
+static void hdsp_midi_tasklet(struct tasklet_struct *t)
{
- struct hdsp *hdsp = (struct hdsp *)arg;
+ struct hdsp *hdsp = from_tasklet(hdsp, t, midi_tasklet);
if (hdsp->midi[0].pending)
snd_hdsp_midi_input_read (&hdsp->midi[0]);
@@ -5182,7 +5182,7 @@ static int snd_hdsp_create(struct snd_card *card,
spin_lock_init(&hdsp->lock);
- tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp);
+ tasklet_setup(&hdsp->midi_tasklet, hdsp_midi_tasklet);
pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev);
hdsp->firmware_rev &= 0xff;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0fa49f4d15cf..572350aaf18d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -2169,9 +2169,9 @@ static int snd_hdspm_create_midi(struct snd_card *card,
}
-static void hdspm_midi_tasklet(unsigned long arg)
+static void hdspm_midi_tasklet(struct tasklet_struct *t)
{
- struct hdspm *hdspm = (struct hdspm *)arg;
+ struct hdspm *hdspm = from_tasklet(hdspm, t, midi_tasklet);
int i = 0;
while (i < hdspm->midiPorts) {
@@ -6836,8 +6836,7 @@ static int snd_hdspm_create(struct snd_card *card,
}
- tasklet_init(&hdspm->midi_tasklet,
- hdspm_midi_tasklet, (unsigned long) hdspm);
+ tasklet_setup(&hdspm->midi_tasklet, hdspm_midi_tasklet);
if (hdspm->io_type != MADIface) {
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 4ae36099ae82..79b861afd986 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -708,9 +708,9 @@ static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx)
ESAI_xFCR_xFR, 0);
}
-static void fsl_esai_hw_reset(unsigned long arg)
+static void fsl_esai_hw_reset(struct tasklet_struct *t)
{
- struct fsl_esai *esai_priv = (struct fsl_esai *)arg;
+ struct fsl_esai *esai_priv = from_tasklet(esai_priv, t, task);
bool tx = true, rx = false, enabled[2];
unsigned long lock_flags;
u32 tfcr, rfcr;
@@ -1070,8 +1070,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
return ret;
}
- tasklet_init(&esai_priv->task, fsl_esai_hw_reset,
- (unsigned long)esai_priv);
+ tasklet_setup(&esai_priv->task, fsl_esai_hw_reset);
pm_runtime_enable(&pdev->dev);
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index bd9de77c35f3..50fc7810723e 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -198,9 +198,9 @@ static int siu_pcm_rd_set(struct siu_port *port_info,
return 0;
}
-static void siu_io_tasklet(unsigned long data)
+static void siu_io_tasklet(struct tasklet_struct *t)
{
- struct siu_stream *siu_stream = (struct siu_stream *)data;
+ struct siu_stream *siu_stream = from_tasklet(siu_stream, t, tasklet);
struct snd_pcm_substream *substream = siu_stream->substream;
struct device *dev = substream->pcm->card->dev;
struct snd_pcm_runtime *rt = substream->runtime;
@@ -520,10 +520,8 @@ static int siu_pcm_new(struct snd_soc_component *component,
(*port_info)->pcm = pcm;
/* IO tasklets */
- tasklet_init(&(*port_info)->playback.tasklet, siu_io_tasklet,
- (unsigned long)&(*port_info)->playback);
- tasklet_init(&(*port_info)->capture.tasklet, siu_io_tasklet,
- (unsigned long)&(*port_info)->capture);
+ tasklet_setup(&(*port_info)->playback.tasklet, siu_io_tasklet);
+ tasklet_setup(&(*port_info)->capture.tasklet, siu_io_tasklet);
}
dev_info(card->dev, "SuperH SIU driver initialized.\n");
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 4b1cd4da3e36..939b33ec39f5 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -134,9 +134,9 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
#define NR_DMA_CHAIN 2
-static void txx9aclc_dma_tasklet(unsigned long data)
+static void txx9aclc_dma_tasklet(struct tasklet_struct *t)
{
- struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data;
+ struct txx9aclc_dmadata *dmadata = from_tasklet(dmadata, t, tasklet);
struct dma_chan *chan = dmadata->dma_chan;
struct dma_async_tx_descriptor *desc;
struct snd_pcm_substream *substream = dmadata->substream;
@@ -352,8 +352,7 @@ static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev,
"playback" : "capture");
return -EBUSY;
}
- tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet,
- (unsigned long)dmadata);
+ tasklet_setup(&dmadata->tasklet, txx9aclc_dma_tasklet);
return 0;
}
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index df639fe03118..e8287a05e36b 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -344,10 +344,9 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep)
spin_unlock_irqrestore(&ep->buffer_lock, flags);
}
-static void snd_usbmidi_out_tasklet(unsigned long data)
+static void snd_usbmidi_out_tasklet(struct tasklet_struct *t)
{
- struct snd_usb_midi_out_endpoint *ep =
- (struct snd_usb_midi_out_endpoint *) data;
+ struct snd_usb_midi_out_endpoint *ep = from_tasklet(ep, t, tasklet);
snd_usbmidi_do_output(ep);
}
@@ -1441,7 +1440,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
}
spin_lock_init(&ep->buffer_lock);
- tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep);
+ tasklet_setup(&ep->tasklet, snd_usbmidi_out_tasklet);
init_waitqueue_head(&ep->drain_wait);
for (i = 0; i < 0x10; ++i)
diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index 884e740a785c..3b2dce1043f5 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -247,9 +247,9 @@ static inline void add_with_wraparound(struct ua101 *ua,
*value -= ua->playback.queue_length;
}
-static void playback_tasklet(unsigned long data)
+static void playback_tasklet(struct tasklet_struct *t)
{
- struct ua101 *ua = (void *)data;
+ struct ua101 *ua = from_tasklet(ua, t, playback_tasklet);
unsigned long flags;
unsigned int frames;
struct ua101_urb *urb;
@@ -1218,8 +1218,7 @@ static int ua101_probe(struct usb_interface *interface,
spin_lock_init(&ua->lock);
mutex_init(&ua->mutex);
INIT_LIST_HEAD(&ua->ready_playback_urbs);
- tasklet_init(&ua->playback_tasklet,
- playback_tasklet, (unsigned long)ua);
+ tasklet_setup(&ua->playback_tasklet, playback_tasklet);
init_waitqueue_head(&ua->alsa_capture_wait);
init_waitqueue_head(&ua->rate_feedback_wait);
init_waitqueue_head(&ua->alsa_playback_wait);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 5600751803cf..b401ee894e1b 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -369,11 +369,13 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+ case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */
case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
ep = 0x81;
ifnum = 2;
goto add_sync_ep_from_ifnum;
case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+ case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
ep = 0x82;
ifnum = 0;
goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index f4fb002e3ef4..23eafd50126f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2827,14 +2827,24 @@ YAMAHA_DEVICE(0x7010, "UB99"),
/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
{
USB_DEVICE(0x17aa, 0x1046),
- QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear",
- "Lenovo-ThinkStation-P620-Rear"),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Lenovo",
+ .product_name = "ThinkStation P620 Rear",
+ .profile_name = "Lenovo-ThinkStation-P620-Rear",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+ }
},
/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
{
USB_DEVICE(0x17aa, 0x104d),
- QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main",
- "Lenovo-ThinkStation-P620-Main"),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Lenovo",
+ .product_name = "ThinkStation P620 Main",
+ .profile_name = "Lenovo-ThinkStation-P620-Main",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
+ }
},
/* Native Instruments MK2 series */
@@ -3549,14 +3559,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
{
/*
* Pioneer DJ DJM-250MK2
- * PCM is 8 channels out @ 48 fixed (endpoints 0x01).
- * The output from computer to the mixer is usable.
+ * PCM is 8 channels out @ 48 fixed (endpoint 0x01)
+ * and 8 channels in @ 48 fixed (endpoint 0x82).
+ *
+ * Both playback and recording is working, even simultaneously.
*
- * The input (phono or line to computer) is not working.
- * It should be at endpoint 0x82 and probably also 8 channels,
- * but it seems that it works only with Pioneer proprietary software.
- * Even on officially supported OS, the Audacity was unable to record
- * and Mixxx to recognize the control vinyls.
+ * Playback channels could be mapped to:
+ * - CH1
+ * - CH2
+ * - AUX
+ *
+ * Recording channels could be mapped to:
+ * - Post CH1 Fader
+ * - Post CH2 Fader
+ * - Cross Fader A
+ * - Cross Fader B
+ * - MIC
+ * - AUX
+ * - REC OUT
+ *
+ * There is remaining problem with recording directly from PHONO/LINE.
+ * If we map a channel to:
+ * - CH1 Control Tone PHONO
+ * - CH1 Control Tone LINE
+ * - CH2 Control Tone PHONO
+ * - CH2 Control Tone LINE
+ * it is silent.
+ * There is no signal even on other operating systems with official drivers.
+ * The signal appears only when a supported application is started.
+ * This needs to be investigated yet...
+ * (there is quite a lot communication on the USB in both directions)
+ *
+ * In current version this mixer could be used for playback
+ * and for recording from vinyls (through Post CH* Fader)
+ * but not for DVS (Digital Vinyl Systems) like in Mixxx.
*/
USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017),
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
@@ -3580,6 +3616,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
.rate_max = 48000,
.nr_rates = 1,
.rate_table = (unsigned int[]) { 48000 }
+ }
+ },
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 8, // inputs
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC|
+ USB_ENDPOINT_SYNC_ASYNC|
+ USB_ENDPOINT_USAGE_IMPLICIT_FB,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .nr_rates = 1,
+ .rate_table = (unsigned int[]) { 48000 }
}
},
{
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index abf99b814a0f..75bbdc691243 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -518,6 +518,15 @@ static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip,
return 1; /* Continue with creating streams and mixer */
}
+static int setup_disable_autosuspend(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ struct usb_driver *driver,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ driver->supports_autosuspend = 0;
+ return 1; /* Continue with creating streams and mixer */
+}
+
/*
* audio-interface quirks
*
@@ -557,6 +566,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
[QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk,
+ [QUIRK_SETUP_DISABLE_AUTOSUSPEND] = setup_disable_autosuspend,
};
if (quirk->type < QUIRK_TYPE_COUNT) {
@@ -1493,6 +1503,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
set_format_emu_quirk(subs, fmt);
break;
case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
+ case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
pioneer_djm_set_format_quirk(subs);
break;
case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b91c4c0807ec..6839915a0128 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -102,6 +102,7 @@ enum quirk_type {
QUIRK_AUDIO_ALIGN_TRANSFER,
QUIRK_AUDIO_STANDARD_MIXER,
QUIRK_SETUP_FMT_AFTER_RESUME,
+ QUIRK_SETUP_DISABLE_AUTOSUSPEND,
QUIRK_TYPE_COUNT
};
diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig
index 77777192f650..4ffcc5e623c2 100644
--- a/sound/x86/Kconfig
+++ b/sound/x86/Kconfig
@@ -9,7 +9,7 @@ menuconfig SND_X86
if SND_X86
config HDMI_LPE_AUDIO
- tristate "HDMI audio without HDaudio on Intel Atom platforms"
+ tristate "HDMI audio without HDAudio on Intel Atom platforms"
depends on DRM_I915
select SND_PCM
help