Merge tag 'sound-5.11-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are some piled fixes, hopefully the last big one for 5.11.
All changes are device-specific small fixes, and majority of commits
are for ASoC while USB-audio got a bit large changes for addressing
the regression for devices with quirks"
* tag 'sound-5.11-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (31 commits)
ALSA: hda/hdmi - enable runtime pm for CI AMD display audio
ALSA: firewire-tascam: Fix integer overflow in midi_port_work()
ALSA: fireface: Fix integer overflow in transmit_midi_msg()
ALSA: hda/tegra: fix tegra-hda on tegra30 soc
clk: tegra30: Add hda clock default rates to clock driver
ALSA: doc: Fix reference to mixart.rst
ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices
ALSA: usb-audio: Annotate the endpoint index in audioformat
ALSA: usb-audio: Avoid unnecessary interface re-setup
ALSA: usb-audio: Choose audioformat of a counter-part substream
ALSA: usb-audio: Fix the missing endpoints creations for quirks
ALSA: hda/realtek: fix right sounds and mute/micmute LEDs for HP machines
ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395
ASoC: amd: Replacing MSI with Legacy IRQ model
ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad E14 Gen 2
ASoC: meson: axg-tdm-interface: fix loopback
ASoC: meson: axg-tdmin: fix axg skew offset
ASoC: max98373: don't access volatile registers in bias level off
ASoC: rt711: mutex between calibration and power state changes
ASoC: Intel: haswell: Add missing pm_ops
...
diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml
index 805da4d..ec06789 100644
--- a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml
+++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml
@@ -1,4 +1,6 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2020 Texas Instruments Incorporated
+# Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
%YAML 1.2
---
$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-audio.yaml#
@@ -7,7 +9,7 @@
title: Texas Instruments J721e Common Processor Board Audio Support
maintainers:
- - Peter Ujfalusi <peter.ujfalusi@ti.com>
+ - Peter Ujfalusi <peter.ujfalusi@gmail.com>
description: |
The audio support on the board is using pcm3168a codec connected to McASP10
diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml
index bb780f6..ee9f960 100644
--- a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml
+++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml
@@ -1,4 +1,6 @@
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2020 Texas Instruments Incorporated
+# Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
%YAML 1.2
---
$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml#
@@ -7,7 +9,7 @@
title: Texas Instruments J721e Common Processor Board Audio Support
maintainers:
- - Peter Ujfalusi <peter.ujfalusi@ti.com>
+ - Peter Ujfalusi <peter.ujfalusi@gmail.com>
description: |
The Infotainment board plugs into the Common Processor Board, the support of the
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst
index fe52c31..b36af65 100644
--- a/Documentation/sound/alsa-configuration.rst
+++ b/Documentation/sound/alsa-configuration.rst
@@ -1501,7 +1501,7 @@
This module supports multiple cards.
Note: One miXart8 board will be represented as 4 alsa cards.
-See MIXART.txt for details.
+See Documentation/sound/cards/mixart.rst for details.
When the driver is compiled as a module and the hotplug firmware
is supported, the firmware data is loaded via hotplug automatically.
diff --git a/MAINTAINERS b/MAINTAINERS
index cc1e6a5..f4f67ec 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -12848,7 +12848,7 @@
F: include/uapi/misc/ocxl.h
OMAP AUDIO SUPPORT
-M: Peter Ujfalusi <peter.ujfalusi@ti.com>
+M: Peter Ujfalusi <peter.ujfalusi@gmail.com>
M: Jarkko Nikula <jarkko.nikula@bitmer.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
L: linux-omap@vger.kernel.org
@@ -17541,7 +17541,7 @@
F: drivers/irqchip/irq-xtensa-*
TEXAS INSTRUMENTS ASoC DRIVERS
-M: Peter Ujfalusi <peter.ujfalusi@ti.com>
+M: Peter Ujfalusi <peter.ujfalusi@gmail.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Maintained
F: sound/soc/ti/
@@ -17851,7 +17851,7 @@
F: drivers/nfc/trf7970a.c
TI TWL4030 SERIES SOC CODEC DRIVER
-M: Peter Ujfalusi <peter.ujfalusi@ti.com>
+M: Peter Ujfalusi <peter.ujfalusi@gmail.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Maintained
F: sound/soc/codecs/twl4030*
diff --git a/drivers/clk/tegra/clk-tegra30.c b/drivers/clk/tegra/clk-tegra30.c
index 37244a7..9cf249c 100644
--- a/drivers/clk/tegra/clk-tegra30.c
+++ b/drivers/clk/tegra/clk-tegra30.c
@@ -1256,6 +1256,8 @@
{ TEGRA30_CLK_I2S3_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 },
{ TEGRA30_CLK_I2S4_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 },
{ TEGRA30_CLK_VIMCLK_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 },
+ { TEGRA30_CLK_HDA, TEGRA30_CLK_PLL_P, 102000000, 0 },
+ { TEGRA30_CLK_HDA2CODEC_2X, TEGRA30_CLK_PLL_P, 48000000, 0 },
/* must be the last entry */
{ TEGRA30_CLK_CLK_MAX, TEGRA30_CLK_CLK_MAX, 0, 0 },
};
diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c
index 7f82762..ee7122c 100644
--- a/sound/firewire/fireface/ff-transaction.c
+++ b/sound/firewire/fireface/ff-transaction.c
@@ -88,7 +88,7 @@
/* Set interval to next transaction. */
ff->next_ktime[port] = ktime_add_ns(ktime_get(),
- ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250);
+ ff->rx_bytes[port] * 8 * (NSEC_PER_SEC / 31250));
if (quad_count == 1)
tcode = TCODE_WRITE_QUADLET_REQUEST;
diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c
index 90288b4..a073cec 100644
--- a/sound/firewire/tascam/tascam-transaction.c
+++ b/sound/firewire/tascam/tascam-transaction.c
@@ -209,7 +209,7 @@
/* Set interval to next transaction. */
port->next_ktime = ktime_add_ns(ktime_get(),
- port->consume_bytes * 8 * NSEC_PER_SEC / 31250);
+ port->consume_bytes * 8 * (NSEC_PER_SEC / 31250));
/* Start this transaction. */
port->idling = false;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 770ad25..e4dd2ff 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2598,7 +2598,8 @@
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
/* ATI HDMI */
{ PCI_DEVICE(0x1002, 0x0002),
- .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+ .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
+ AZX_DCAPS_PM_RUNTIME },
{ PCI_DEVICE(0x1002, 0x1308),
.driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
{ PCI_DEVICE(0x1002, 0x157a),
@@ -2660,9 +2661,11 @@
{ PCI_DEVICE(0x1002, 0xaab0),
.driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
{ PCI_DEVICE(0x1002, 0xaac0),
- .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+ .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
+ AZX_DCAPS_PM_RUNTIME },
{ PCI_DEVICE(0x1002, 0xaac8),
- .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+ .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
+ AZX_DCAPS_PM_RUNTIME },
{ PCI_DEVICE(0x1002, 0xaad8),
.driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
AZX_DCAPS_PM_RUNTIME },
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 70164d142..361cf20 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -388,7 +388,7 @@
* in powers of 2, next available ratio is 16 which can be
* used as a limiting factor here.
*/
- if (of_device_is_compatible(np, "nvidia,tegra194-hda"))
+ if (of_device_is_compatible(np, "nvidia,tegra30-hda"))
chip->bus.core.sdo_limit = 16;
/* codec detection */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3c1d2a3..dd82ff2b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7970,6 +7970,10 @@
SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
+ SND_PCI_QUIRK(0x103c, 0x8780, "HP ZBook Fury 17 G7 Mobile Workstation",
+ ALC285_FIXUP_HP_GPIO_AMP_INIT),
+ SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation",
+ ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c
index 8c138e4..d3536fd 100644
--- a/sound/soc/amd/raven/pci-acp3x.c
+++ b/sound/soc/amd/raven/pci-acp3x.c
@@ -140,21 +140,14 @@
goto release_regions;
}
- /* check for msi interrupt support */
- ret = pci_enable_msi(pci);
- if (ret)
- /* msi is not enabled */
- irqflags = IRQF_SHARED;
- else
- /* msi is enabled */
- irqflags = 0;
+ irqflags = IRQF_SHARED;
addr = pci_resource_start(pci, 0);
adata->acp3x_base = devm_ioremap(&pci->dev, addr,
pci_resource_len(pci, 0));
if (!adata->acp3x_base) {
ret = -ENOMEM;
- goto disable_msi;
+ goto release_regions;
}
pci_set_master(pci);
pci_set_drvdata(pci, adata);
@@ -162,7 +155,7 @@
adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN);
ret = acp3x_init(adata);
if (ret)
- goto disable_msi;
+ goto release_regions;
val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG);
switch (val) {
@@ -251,8 +244,6 @@
de_init:
if (acp3x_deinit(adata->acp3x_base))
dev_err(&pci->dev, "ACP de-init failed\n");
-disable_msi:
- pci_disable_msi(pci);
release_regions:
pci_release_regions(pci);
disable_pci:
@@ -311,7 +302,6 @@
dev_err(&pci->dev, "ACP de-init failed\n");
pm_runtime_forbid(&pci->dev);
pm_runtime_get_noresume(&pci->dev);
- pci_disable_msi(pci);
pci_release_regions(pci);
pci_disable_device(pci);
}
diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c
index fa169bf..deca8c7 100644
--- a/sound/soc/amd/renoir/rn-pci-acp3x.c
+++ b/sound/soc/amd/renoir/rn-pci-acp3x.c
@@ -171,6 +171,20 @@
DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"),
}
},
+ {
+ /* Lenovo ThinkPad E14 Gen 2 */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+ DMI_EXACT_MATCH(DMI_BOARD_NAME, "20T6CTO1WW"),
+ }
+ },
+ {
+ /* Lenovo ThinkPad X395 */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+ DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"),
+ }
+ },
{}
};
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 142373e..9fe9471 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -143,7 +143,7 @@
- sama7g5
This S/PDIF TX driver is compliant with IEC-60958 standard and
- includes programable User Data and Channel Status fields.
+ includes programmable User Data and Channel Status fields.
config SND_MCHP_SOC_SPDIFRX
tristate "Microchip ASoC driver for boards using S/PDIF RX"
@@ -157,5 +157,5 @@
- sama7g5
This S/PDIF RX driver is compliant with IEC-60958 standard and
- includes programable User Data and Channel Status fields.
+ includes programmable User Data and Channel Status fields.
endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index ba4eb54..9bf6bfd 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -457,7 +457,7 @@
help
Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM
Converter. In this mode, the device works in standalone mode which
- means that there is no bus to comunicate with it. Stereo mode is not
+ means that there is no bus to communicate with it. Stereo mode is not
supported in this mode.
To compile this driver as a module, choose M here: the module
diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c
index 92921e3..85f6865 100644
--- a/sound/soc/codecs/max98373-i2c.c
+++ b/sound/soc/codecs/max98373-i2c.c
@@ -19,6 +19,12 @@
#include <sound/tlv.h>
#include "max98373.h"
+static const u32 max98373_i2c_cache_reg[] = {
+ MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK,
+ MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK,
+ MAX98373_R20B6_BDE_CUR_STATE_READBACK,
+};
+
static struct reg_default max98373_reg[] = {
{MAX98373_R2000_SW_RESET, 0x00},
{MAX98373_R2001_INT_RAW1, 0x00},
@@ -472,6 +478,11 @@
static int max98373_suspend(struct device *dev)
{
struct max98373_priv *max98373 = dev_get_drvdata(dev);
+ int i;
+
+ /* cache feedback register values before suspend */
+ for (i = 0; i < max98373->cache_num; i++)
+ regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val);
regcache_cache_only(max98373->regmap, true);
regcache_mark_dirty(max98373->regmap);
@@ -509,6 +520,7 @@
{
int ret = 0;
int reg = 0;
+ int i;
struct max98373_priv *max98373 = NULL;
max98373 = devm_kzalloc(&i2c->dev, sizeof(*max98373), GFP_KERNEL);
@@ -534,6 +546,14 @@
return ret;
}
+ max98373->cache_num = ARRAY_SIZE(max98373_i2c_cache_reg);
+ max98373->cache = devm_kcalloc(&i2c->dev, max98373->cache_num,
+ sizeof(*max98373->cache),
+ GFP_KERNEL);
+
+ for (i = 0; i < max98373->cache_num; i++)
+ max98373->cache[i].reg = max98373_i2c_cache_reg[i];
+
/* voltage/current slot & gpio configuration */
max98373_slot_config(&i2c->dev, max98373);
diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c
index ec2e79c..b8d471d 100644
--- a/sound/soc/codecs/max98373-sdw.c
+++ b/sound/soc/codecs/max98373-sdw.c
@@ -23,6 +23,12 @@
struct sdw_stream_runtime *sdw_stream;
};
+static const u32 max98373_sdw_cache_reg[] = {
+ MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK,
+ MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK,
+ MAX98373_R20B6_BDE_CUR_STATE_READBACK,
+};
+
static struct reg_default max98373_reg[] = {
{MAX98373_R0040_SCP_INIT_STAT_1, 0x00},
{MAX98373_R0041_SCP_INIT_MASK_1, 0x00},
@@ -245,6 +251,11 @@
static __maybe_unused int max98373_suspend(struct device *dev)
{
struct max98373_priv *max98373 = dev_get_drvdata(dev);
+ int i;
+
+ /* cache feedback register values before suspend */
+ for (i = 0; i < max98373->cache_num; i++)
+ regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val);
regcache_cache_only(max98373->regmap, true);
@@ -757,6 +768,7 @@
{
struct max98373_priv *max98373;
int ret;
+ int i;
struct device *dev = &slave->dev;
/* Allocate and assign private driver data structure */
@@ -768,6 +780,14 @@
max98373->regmap = regmap;
max98373->slave = slave;
+ max98373->cache_num = ARRAY_SIZE(max98373_sdw_cache_reg);
+ max98373->cache = devm_kcalloc(dev, max98373->cache_num,
+ sizeof(*max98373->cache),
+ GFP_KERNEL);
+
+ for (i = 0; i < max98373->cache_num; i++)
+ max98373->cache[i].reg = max98373_sdw_cache_reg[i];
+
/* Read voltage and slot configuration */
max98373_slot_config(dev, max98373);
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 929bb17..31d571d 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -168,6 +168,31 @@
MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0,
max98373_ADC_samplerate_text);
+static int max98373_feedback_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component);
+ int i;
+
+ if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
+ /*
+ * Register values will be cached before suspend. The cached value
+ * will be a valid value and userspace will happy with that.
+ */
+ for (i = 0; i < max98373->cache_num; i++) {
+ if (mc->reg == max98373->cache[i].reg) {
+ ucontrol->value.integer.value[0] = max98373->cache[i].val;
+ return 0;
+ }
+ }
+ }
+
+ return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
static const struct snd_kcontrol_new max98373_snd_controls[] = {
SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG,
MAX98373_AMP_VOL_SEL_SHIFT, 1, 0),
@@ -209,8 +234,10 @@
MAX98373_FLT_EN_SHIFT, 1, 0),
SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG,
MAX98373_FLT_EN_SHIFT, 1, 0),
-SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0),
-SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0),
+SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0,
+ max98373_feedback_get, NULL),
+SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0,
+ max98373_feedback_get, NULL),
SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG,
0, 0x3, 0),
SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG,
@@ -226,7 +253,8 @@
SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0),
SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0),
SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0),
-SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0),
+SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0,
+ max98373_feedback_get, NULL),
SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0),
SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0),
SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0),
diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h
index 4ab29b9..71f5a52 100644
--- a/sound/soc/codecs/max98373.h
+++ b/sound/soc/codecs/max98373.h
@@ -203,6 +203,11 @@
/* MAX98373_R2000_SW_RESET */
#define MAX98373_SOFT_RESET (0x1 << 0)
+struct max98373_cache {
+ u32 reg;
+ u32 val;
+};
+
struct max98373_priv {
struct regmap *regmap;
int reset_gpio;
@@ -212,6 +217,9 @@
bool interleave_mode;
unsigned int ch_size;
bool tdm_mode;
+ /* cache for reading a valid fake feedback value */
+ struct max98373_cache *cache;
+ int cache_num;
/* variables to support soundwire */
struct sdw_slave *slave;
bool hw_init;
diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c
index 5771c02..85f7441 100644
--- a/sound/soc/codecs/rt711.c
+++ b/sound/soc/codecs/rt711.c
@@ -462,6 +462,8 @@
unsigned int read_ll, read_rl;
int i;
+ mutex_lock(&rt711->calibrate_mutex);
+
/* Can't use update bit function, so read the original value first */
addr_h = mc->reg;
addr_l = mc->rreg;
@@ -547,6 +549,8 @@
if (dapm->bias_level <= SND_SOC_BIAS_STANDBY)
regmap_write(rt711->regmap,
RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3);
+
+ mutex_unlock(&rt711->calibrate_mutex);
return 0;
}
@@ -859,9 +863,11 @@
break;
case SND_SOC_BIAS_STANDBY:
+ mutex_lock(&rt711->calibrate_mutex);
regmap_write(rt711->regmap,
RT711_SET_AUDIO_POWER_STATE,
AC_PWRST_D3);
+ mutex_unlock(&rt711->calibrate_mutex);
break;
default:
diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c
index 2c2a76a..ede4a9a 100644
--- a/sound/soc/fsl/imx-hdmi.c
+++ b/sound/soc/fsl/imx-hdmi.c
@@ -164,6 +164,7 @@
if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) {
dev_err(&pdev->dev, "Invalid HDMI DAI link\n");
+ ret = -EINVAL;
goto fail;
}
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index c55d123..c763bfe 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -189,6 +189,7 @@
.probe = haswell_audio_probe,
.driver = {
.name = "haswell-audio",
+ .pm = &snd_soc_pm_ops,
},
};
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
index fcd8dff..1275c14 100644
--- a/sound/soc/intel/skylake/cnl-sst.c
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -224,6 +224,7 @@
"dsp boot timeout, status=%#x error=%#x\n",
sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS),
sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE));
+ ret = -ETIMEDOUT;
goto err;
}
} else {
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index c8664ab..87cac44 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -467,8 +467,20 @@
return ret;
}
+static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = {
+ SND_SOC_DAPM_SIGGEN("Playback Signal"),
+};
+
+static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = {
+ { "Loopback", NULL, "Playback Signal" },
+};
+
static const struct snd_soc_component_driver axg_tdm_iface_component_drv = {
- .set_bias_level = axg_tdm_iface_set_bias_level,
+ .dapm_widgets = axg_tdm_iface_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets),
+ .dapm_routes = axg_tdm_iface_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes),
+ .set_bias_level = axg_tdm_iface_set_bias_level,
};
static const struct of_device_id axg_tdm_iface_of_match[] = {
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index 88ed95a..b4faf9d 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -228,15 +228,6 @@
.regmap_cfg = &axg_tdmin_regmap_cfg,
.ops = &axg_tdmin_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .skew_offset = 2,
- },
-};
-
-static const struct axg_tdm_formatter_driver g12a_tdmin_drv = {
- .component_drv = &axg_tdmin_component_drv,
- .regmap_cfg = &axg_tdmin_regmap_cfg,
- .ops = &axg_tdmin_ops,
- .quirks = &(const struct axg_tdm_formatter_hw) {
.skew_offset = 3,
},
};
@@ -247,10 +238,10 @@
.data = &axg_tdmin_drv,
}, {
.compatible = "amlogic,g12a-tdmin",
- .data = &g12a_tdmin_drv,
+ .data = &axg_tdmin_drv,
}, {
.compatible = "amlogic,sm1-tdmin",
- .data = &g12a_tdmin_drv,
+ .data = &axg_tdmin_drv,
}, {}
};
MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index af684fd..c5e99c2 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -270,18 +270,6 @@
struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
unsigned int id = dai->driver->id;
int ret = -EINVAL;
- unsigned int val = 0;
-
- ret = regmap_read(drvdata->lpaif_map,
- LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), &val);
- if (ret) {
- dev_err(dai->dev, "error reading from i2sctl reg: %d\n", ret);
- return ret;
- }
- if (val == LPAIF_I2SCTL_RESET_STATE) {
- dev_err(dai->dev, "error in i2sctl register state\n");
- return -ENOTRECOVERABLE;
- }
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -454,20 +442,16 @@
struct lpass_variant *v = drvdata->variant;
int i;
- for (i = 0; i < v->i2s_ports; ++i)
- if (reg == LPAIF_I2SCTL_REG(v, i))
- return true;
for (i = 0; i < v->irq_ports; ++i)
if (reg == LPAIF_IRQSTAT_REG(v, i))
return true;
for (i = 0; i < v->rdma_channels; ++i)
- if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i))
+ if (reg == LPAIF_RDMACURR_REG(v, i))
return true;
for (i = 0; i < v->wrdma_channels; ++i)
- if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) ||
- reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start))
+ if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start))
return true;
return false;
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 80b09de..d1c2485 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -452,7 +452,6 @@
unsigned int reg_irqclr = 0, val_irqclr = 0;
unsigned int reg_irqen = 0, val_irqen = 0, val_mask = 0;
unsigned int dai_id = cpu_dai->driver->id;
- unsigned int dma_ctrl_reg = 0;
ch = pcm_data->dma_ch;
if (dir == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -469,17 +468,7 @@
id = pcm_data->dma_ch - v->wrdma_channel_start;
map = drvdata->lpaif_map;
}
- ret = regmap_read(map, LPAIF_DMACTL_REG(v, ch, dir, dai_id), &dma_ctrl_reg);
- if (ret) {
- dev_err(soc_runtime->dev, "error reading from rdmactl reg: %d\n", ret);
- return ret;
- }
- if (dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE ||
- dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE + 1) {
- dev_err(soc_runtime->dev, "error in rdmactl register state\n");
- return -ENOTRECOVERABLE;
- }
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
@@ -500,7 +489,6 @@
"error writing to rdmactl reg: %d\n", ret);
return ret;
}
- map = drvdata->hdmiif_map;
reg_irqclr = LPASS_HDMITX_APP_IRQCLEAR_REG(v);
val_irqclr = (LPAIF_IRQ_ALL(ch) |
LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(ch) |
@@ -519,7 +507,6 @@
break;
case MI2S_PRIMARY:
case MI2S_SECONDARY:
- map = drvdata->lpaif_map;
reg_irqclr = LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST);
val_irqclr = LPAIF_IRQ_ALL(ch);
@@ -563,7 +550,6 @@
"error writing to rdmactl reg: %d\n", ret);
return ret;
}
- map = drvdata->hdmiif_map;
reg_irqen = LPASS_HDMITX_APP_IRQEN_REG(v);
val_mask = (LPAIF_IRQ_ALL(ch) |
LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(ch) |
@@ -573,7 +559,6 @@
break;
case MI2S_PRIMARY:
case MI2S_SECONDARY:
- map = drvdata->lpaif_map;
reg_irqen = LPAIF_IRQEN_REG(v, LPAIF_IRQ_PORT_HOST);
val_mask = LPAIF_IRQ_ALL(ch);
val_irqen = 0;
@@ -838,6 +823,39 @@
}
}
+static int lpass_platform_pcmops_suspend(struct snd_soc_component *component)
+{
+ struct lpass_data *drvdata = snd_soc_component_get_drvdata(component);
+ struct regmap *map;
+ unsigned int dai_id = component->id;
+
+ if (dai_id == LPASS_DP_RX)
+ map = drvdata->hdmiif_map;
+ else
+ map = drvdata->lpaif_map;
+
+ regcache_cache_only(map, true);
+ regcache_mark_dirty(map);
+
+ return 0;
+}
+
+static int lpass_platform_pcmops_resume(struct snd_soc_component *component)
+{
+ struct lpass_data *drvdata = snd_soc_component_get_drvdata(component);
+ struct regmap *map;
+ unsigned int dai_id = component->id;
+
+ if (dai_id == LPASS_DP_RX)
+ map = drvdata->hdmiif_map;
+ else
+ map = drvdata->lpaif_map;
+
+ regcache_cache_only(map, false);
+ return regcache_sync(map);
+}
+
+
static const struct snd_soc_component_driver lpass_component_driver = {
.name = DRV_NAME,
.open = lpass_platform_pcmops_open,
@@ -850,6 +868,8 @@
.mmap = lpass_platform_pcmops_mmap,
.pcm_construct = lpass_platform_pcm_new,
.pcm_destruct = lpass_platform_pcm_free,
+ .suspend = lpass_platform_pcmops_suspend,
+ .resume = lpass_platform_pcmops_resume,
};
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index b9aacf3..abdfd9cf 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -366,25 +366,27 @@
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
struct device *dev = rsnd_priv_to_dev(priv);
struct clk *clk;
- int i, ret;
+ int i;
for_each_rsnd_clk(clk, adg, i) {
- ret = 0;
if (enable) {
- ret = clk_prepare_enable(clk);
+ int ret = clk_prepare_enable(clk);
/*
* We shouldn't use clk_get_rate() under
* atomic context. Let's keep it when
* rsnd_adg_clk_enable() was called
*/
- adg->clk_rate[i] = clk_get_rate(adg->clk[i]);
+ adg->clk_rate[i] = 0;
+ if (ret < 0)
+ dev_warn(dev, "can't use clk %d\n", i);
+ else
+ adg->clk_rate[i] = clk_get_rate(clk);
} else {
- clk_disable_unprepare(clk);
+ if (adg->clk_rate[i])
+ clk_disable_unprepare(clk);
+ adg->clk_rate[i] = 0;
}
-
- if (ret < 0)
- dev_warn(dev, "can't use clk %d\n", i);
}
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 9f0c86c..2b75d01 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2486,6 +2486,7 @@
enum snd_soc_dapm_direction dir;
list_del(&w->list);
+ list_del(&w->dirty);
/*
* remove source and sink paths associated to this widget.
* While removing the path, remove reference to it from both
diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig
index 031dad5..3e8b6c0 100644
--- a/sound/soc/sof/Kconfig
+++ b/sound/soc/sof/Kconfig
@@ -122,7 +122,7 @@
bool "SOF stop on XRUN"
help
This option forces PCMs to stop on any XRUN event. This is useful to
- preserve any trace data ond pipeline status prior to the XRUN.
+ preserve any trace data and pipeline status prior to the XRUN.
Say Y if you are debugging SOF FW pipeline XRUNs.
If unsure select "N".
diff --git a/sound/usb/card.c b/sound/usb/card.c
index d731ca6..e08fbf8 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -450,10 +450,8 @@
static void snd_usb_audio_free(struct snd_card *card)
{
struct snd_usb_audio *chip = card->private_data;
- struct snd_usb_endpoint *ep, *n;
- list_for_each_entry_safe(ep, n, &chip->ep_list, list)
- snd_usb_endpoint_free(ep);
+ snd_usb_endpoint_free_all(chip);
mutex_destroy(&chip->mutex);
if (!atomic_read(&chip->shutdown))
@@ -611,6 +609,7 @@
chip->usb_id = usb_id;
INIT_LIST_HEAD(&chip->pcm_list);
INIT_LIST_HEAD(&chip->ep_list);
+ INIT_LIST_HEAD(&chip->iface_ref_list);
INIT_LIST_HEAD(&chip->midi_list);
INIT_LIST_HEAD(&chip->mixer_list);
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 6a027c3..37091b1 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -18,6 +18,7 @@
unsigned int frame_size; /* samples per frame for non-audio */
unsigned char iface; /* interface number */
unsigned char altsetting; /* corresponding alternate setting */
+ unsigned char ep_idx; /* endpoint array index */
unsigned char altset_idx; /* array index of altenate setting */
unsigned char attributes; /* corresponding attributes of cs endpoint */
unsigned char endpoint; /* endpoint */
@@ -42,6 +43,7 @@
};
struct snd_usb_substream;
+struct snd_usb_iface_ref;
struct snd_usb_endpoint;
struct snd_usb_power_domain;
@@ -58,6 +60,7 @@
struct snd_usb_endpoint {
struct snd_usb_audio *chip;
+ struct snd_usb_iface_ref *iface_ref;
int opened; /* open refcount; protect with chip->mutex */
atomic_t running; /* running status */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 162da7a..fe73fe3 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -24,6 +24,14 @@
#define EP_FLAG_RUNNING 1
#define EP_FLAG_STOPPING 2
+/* interface refcounting */
+struct snd_usb_iface_ref {
+ unsigned char iface;
+ bool need_setup;
+ int opened;
+ struct list_head list;
+};
+
/*
* snd_usb_endpoint is a model that abstracts everything related to an
* USB endpoint and its streaming.
@@ -489,6 +497,28 @@
}
/*
+ * Find or create a refcount object for the given interface
+ *
+ * The objects are released altogether in snd_usb_endpoint_free_all()
+ */
+static struct snd_usb_iface_ref *
+iface_ref_find(struct snd_usb_audio *chip, int iface)
+{
+ struct snd_usb_iface_ref *ip;
+
+ list_for_each_entry(ip, &chip->iface_ref_list, list)
+ if (ip->iface == iface)
+ return ip;
+
+ ip = kzalloc(sizeof(*ip), GFP_KERNEL);
+ if (!ip)
+ return NULL;
+ ip->iface = iface;
+ list_add_tail(&ip->list, &chip->iface_ref_list);
+ return ip;
+}
+
+/*
* Get the existing endpoint object corresponding EP
* Returns NULL if not present.
*/
@@ -520,8 +550,8 @@
*
* Returns zero on success or a negative error code.
*
- * New endpoints will be added to chip->ep_list and must be freed by
- * calling snd_usb_endpoint_free().
+ * New endpoints will be added to chip->ep_list and freed by
+ * calling snd_usb_endpoint_free_all().
*
* For SND_USB_ENDPOINT_TYPE_SYNC, the caller needs to guarantee that
* bNumEndpoints > 1 beforehand.
@@ -653,11 +683,17 @@
} else {
ep->iface = fp->iface;
ep->altsetting = fp->altsetting;
- ep->ep_idx = 0;
+ ep->ep_idx = fp->ep_idx;
}
usb_audio_dbg(chip, "Open EP 0x%x, iface=%d:%d, idx=%d\n",
ep_num, ep->iface, ep->altsetting, ep->ep_idx);
+ ep->iface_ref = iface_ref_find(chip, ep->iface);
+ if (!ep->iface_ref) {
+ ep = NULL;
+ goto unlock;
+ }
+
ep->cur_audiofmt = fp;
ep->cur_channels = fp->channels;
ep->cur_rate = params_rate(params);
@@ -681,6 +717,11 @@
ep->implicit_fb_sync);
} else {
+ if (WARN_ON(!ep->iface_ref)) {
+ ep = NULL;
+ goto unlock;
+ }
+
if (!endpoint_compatible(ep, fp, params)) {
usb_audio_err(chip, "Incompatible EP setup for 0x%x\n",
ep_num);
@@ -692,6 +733,9 @@
ep_num, ep->opened);
}
+ if (!ep->iface_ref->opened++)
+ ep->iface_ref->need_setup = true;
+
ep->opened++;
unlock:
@@ -760,12 +804,16 @@
mutex_lock(&chip->mutex);
usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n",
ep->ep_num, ep->opened);
- if (!--ep->opened) {
+
+ if (!--ep->iface_ref->opened)
endpoint_set_interface(chip, ep, false);
+
+ if (!--ep->opened) {
ep->iface = 0;
ep->altsetting = 0;
ep->cur_audiofmt = NULL;
ep->cur_rate = 0;
+ ep->iface_ref = NULL;
usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num);
}
mutex_unlock(&chip->mutex);
@@ -775,6 +823,8 @@
void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep)
{
ep->need_setup = true;
+ if (ep->iface_ref)
+ ep->iface_ref->need_setup = true;
}
/*
@@ -1195,11 +1245,13 @@
int err = 0;
mutex_lock(&chip->mutex);
+ if (WARN_ON(!ep->iface_ref))
+ goto unlock;
if (!ep->need_setup)
goto unlock;
- /* No need to (re-)configure the sync EP belonging to the same altset */
- if (ep->ep_idx) {
+ /* If the interface has been already set up, just set EP parameters */
+ if (!ep->iface_ref->need_setup) {
err = snd_usb_endpoint_set_params(chip, ep);
if (err < 0)
goto unlock;
@@ -1242,6 +1294,8 @@
goto unlock;
}
+ ep->iface_ref->need_setup = false;
+
done:
ep->need_setup = false;
err = 1;
@@ -1387,15 +1441,21 @@
}
/**
- * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint
+ * snd_usb_endpoint_free_all: Free the resources of an snd_usb_endpoint
+ * @card: The chip
*
- * @ep: the endpoint to free
- *
- * This free all resources of the given ep.
+ * This free all endpoints and those resources
*/
-void snd_usb_endpoint_free(struct snd_usb_endpoint *ep)
+void snd_usb_endpoint_free_all(struct snd_usb_audio *chip)
{
- kfree(ep);
+ struct snd_usb_endpoint *ep, *en;
+ struct snd_usb_iface_ref *ip, *in;
+
+ list_for_each_entry_safe(ep, en, &chip->ep_list, list)
+ kfree(ep);
+
+ list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list)
+ kfree(ip);
}
/*
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 11e3bb8..eea4ca4 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -42,7 +42,7 @@
void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep);
int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
void snd_usb_endpoint_release(struct snd_usb_endpoint *ep);
-void snd_usb_endpoint_free(struct snd_usb_endpoint *ep);
+void snd_usb_endpoint_free_all(struct snd_usb_audio *chip);
int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep);
int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep,
diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c
index 931042a..1ac2cc6 100644
--- a/sound/usb/implicit.c
+++ b/sound/usb/implicit.c
@@ -58,8 +58,6 @@
IMPLICIT_FB_FIXED_DEV(0x0499, 0x172f, 0x81, 2), /* Steinberg UR22C */
IMPLICIT_FB_FIXED_DEV(0x0d9a, 0x00df, 0x81, 2), /* RTX6001 */
IMPLICIT_FB_FIXED_DEV(0x22f0, 0x0006, 0x81, 3), /* Allen&Heath Qu-16 */
- IMPLICIT_FB_FIXED_DEV(0x2b73, 0x000a, 0x82, 0), /* Pioneer DJ DJM-900NXS2 */
- IMPLICIT_FB_FIXED_DEV(0x2b73, 0x0017, 0x82, 0), /* Pioneer DJ DJM-250MK2 */
IMPLICIT_FB_FIXED_DEV(0x1686, 0xf029, 0x82, 2), /* Zoom UAC-2 */
IMPLICIT_FB_FIXED_DEV(0x2466, 0x8003, 0x86, 2), /* Fractal Audio Axe-Fx II */
IMPLICIT_FB_FIXED_DEV(0x0499, 0x172a, 0x86, 2), /* Yamaha MODX */
@@ -100,7 +98,7 @@
/* set up sync EP information on the audioformat */
static int add_implicit_fb_sync_ep(struct snd_usb_audio *chip,
struct audioformat *fmt,
- int ep, int ifnum,
+ int ep, int ep_idx, int ifnum,
const struct usb_host_interface *alts)
{
struct usb_interface *iface;
@@ -115,7 +113,7 @@
fmt->sync_ep = ep;
fmt->sync_iface = ifnum;
fmt->sync_altsetting = alts->desc.bAlternateSetting;
- fmt->sync_ep_idx = 0;
+ fmt->sync_ep_idx = ep_idx;
fmt->implicit_fb = 1;
usb_audio_dbg(chip,
"%d:%d: added %s implicit_fb sync_ep %x, iface %d:%d\n",
@@ -147,7 +145,7 @@
(epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
USB_ENDPOINT_USAGE_IMPLICIT_FB)
return 0;
- return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress,
+ return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0,
ifnum, alts);
}
@@ -173,10 +171,32 @@
(epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
USB_ENDPOINT_USAGE_IMPLICIT_FB)
return 0;
- return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress,
+ return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0,
ifnum, alts);
}
+/* Pioneer devices: playback and capture streams sharing the same iface/altset
+ */
+static int add_pioneer_implicit_fb(struct snd_usb_audio *chip,
+ struct audioformat *fmt,
+ struct usb_host_interface *alts)
+{
+ struct usb_endpoint_descriptor *epd;
+
+ if (alts->desc.bNumEndpoints != 2)
+ return 0;
+
+ epd = get_endpoint(alts, 1);
+ if (!usb_endpoint_is_isoc_in(epd) ||
+ (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC ||
+ ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
+ USB_ENDPOINT_USAGE_DATA &&
+ (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
+ USB_ENDPOINT_USAGE_IMPLICIT_FB))
+ return 0;
+ return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1,
+ alts->desc.bInterfaceNumber, alts);
+}
static int __add_generic_implicit_fb(struct snd_usb_audio *chip,
struct audioformat *fmt,
@@ -197,7 +217,7 @@
if (!usb_endpoint_is_isoc_in(epd) ||
(epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC)
return 0;
- return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress,
+ return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0,
iface, alts);
}
@@ -250,7 +270,7 @@
case IMPLICIT_FB_NONE:
return 0; /* No quirk */
case IMPLICIT_FB_FIXED:
- return add_implicit_fb_sync_ep(chip, fmt, p->ep_num,
+ return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0,
p->iface, NULL);
}
}
@@ -278,6 +298,14 @@
return 1;
}
+ /* Pioneer devices implicit feedback with vendor spec class */
+ if (attr == USB_ENDPOINT_SYNC_ASYNC &&
+ alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+ USB_ID_VENDOR(chip->usb_id) == 0x2b73 /* Pioneer */) {
+ if (add_pioneer_implicit_fb(chip, fmt, alts))
+ return 1;
+ }
+
/* Try the generic implicit fb if available */
if (chip->generic_implicit_fb)
return add_generic_implicit_fb(chip, fmt, alts);
@@ -295,8 +323,8 @@
p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts);
if (p && p->type == IMPLICIT_FB_FIXED)
- return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, p->iface,
- NULL);
+ return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0,
+ p->iface, NULL);
return 0;
}
@@ -378,20 +406,19 @@
int stream)
{
struct snd_usb_substream *subs;
- const struct audioformat *fp, *sync_fmt;
+ const struct audioformat *fp, *sync_fmt = NULL;
int score, high_score;
- /* When sharing the same altset, use the original audioformat */
+ /* Use the original audioformat as fallback for the shared altset */
if (target->iface == target->sync_iface &&
target->altsetting == target->sync_altsetting)
- return target;
+ sync_fmt = target;
subs = find_matching_substream(chip, stream, target->sync_ep,
target->fmt_type);
if (!subs)
- return NULL;
+ return sync_fmt;
- sync_fmt = NULL;
high_score = 0;
list_for_each_entry(fp, &subs->fmt_list, list) {
score = match_endpoint_audioformats(subs, fp,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 0e11cb9..c8a4bdf 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3362,6 +3362,7 @@
.altsetting = 1,
.altset_idx = 1,
.endpoint = 0x86,
+ .ep_idx = 1,
.ep_attr = USB_ENDPOINT_XFER_ISOC|
USB_ENDPOINT_SYNC_ASYNC|
USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3450,6 +3451,7 @@
.altsetting = 1,
.altset_idx = 1,
.endpoint = 0x82,
+ .ep_idx = 1,
.ep_attr = USB_ENDPOINT_XFER_ISOC|
USB_ENDPOINT_SYNC_ASYNC|
USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3506,6 +3508,7 @@
.altsetting = 1,
.altset_idx = 1,
.endpoint = 0x82,
+ .ep_idx = 1,
.ep_attr = USB_ENDPOINT_XFER_ISOC|
USB_ENDPOINT_SYNC_ASYNC|
USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3562,6 +3565,7 @@
.altsetting = 1,
.altset_idx = 1,
.endpoint = 0x82,
+ .ep_idx = 1,
.ep_attr = USB_ENDPOINT_XFER_ISOC|
USB_ENDPOINT_SYNC_ASYNC|
USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3619,6 +3623,7 @@
.altsetting = 1,
.altset_idx = 1,
.endpoint = 0x82,
+ .ep_idx = 1,
.ep_attr = USB_ENDPOINT_XFER_ISOC|
USB_ENDPOINT_SYNC_ASYNC|
USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3679,6 +3684,7 @@
.altsetting = 1,
.altset_idx = 1,
.endpoint = 0x82,
+ .ep_idx = 1,
.ep_attr = USB_ENDPOINT_XFER_ISOC|
USB_ENDPOINT_SYNC_ASYNC|
USB_ENDPOINT_USAGE_IMPLICIT_FB,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index e4a690b..89e1726 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -120,6 +120,40 @@
return 0;
}
+/* create the audio stream and the corresponding endpoints from the fixed
+ * audioformat object; this is used for quirks with the fixed EPs
+ */
+static int add_audio_stream_from_fixed_fmt(struct snd_usb_audio *chip,
+ struct audioformat *fp)
+{
+ int stream, err;
+
+ stream = (fp->endpoint & USB_DIR_IN) ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+
+ snd_usb_audioformat_set_sync_ep(chip, fp);
+
+ err = snd_usb_add_audio_stream(chip, stream, fp);
+ if (err < 0)
+ return err;
+
+ err = snd_usb_add_endpoint(chip, fp->endpoint,
+ SND_USB_ENDPOINT_TYPE_DATA);
+ if (err < 0)
+ return err;
+
+ if (fp->sync_ep) {
+ err = snd_usb_add_endpoint(chip, fp->sync_ep,
+ fp->implicit_fb ?
+ SND_USB_ENDPOINT_TYPE_DATA :
+ SND_USB_ENDPOINT_TYPE_SYNC);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/*
* create a stream for an endpoint/altsetting without proper descriptors
*/
@@ -131,8 +165,8 @@
struct audioformat *fp;
struct usb_host_interface *alts;
struct usb_interface_descriptor *altsd;
- int stream, err;
unsigned *rate_table = NULL;
+ int err;
fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
if (!fp)
@@ -153,11 +187,6 @@
fp->rate_table = rate_table;
}
- 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)
- goto error;
if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber ||
fp->altset_idx >= iface->num_altsetting) {
err = -EINVAL;
@@ -165,7 +194,7 @@
}
alts = &iface->altsetting[fp->altset_idx];
altsd = get_iface_desc(alts);
- if (altsd->bNumEndpoints < 1) {
+ if (altsd->bNumEndpoints <= fp->ep_idx) {
err = -EINVAL;
goto error;
}
@@ -175,7 +204,14 @@
if (fp->datainterval == 0)
fp->datainterval = snd_usb_parse_datainterval(chip, alts);
if (fp->maxpacksize == 0)
- fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, fp->ep_idx)->wMaxPacketSize);
+ if (!fp->fmt_type)
+ fp->fmt_type = UAC_FORMAT_TYPE_I;
+
+ err = add_audio_stream_from_fixed_fmt(chip, fp);
+ if (err < 0)
+ goto error;
+
usb_set_interface(chip->dev, fp->iface, 0);
snd_usb_init_pitch(chip, fp);
snd_usb_init_sample_rate(chip, fp, fp->rate_max);
@@ -417,7 +453,7 @@
struct usb_host_interface *alts;
struct usb_interface_descriptor *altsd;
struct audioformat *fp;
- int stream, err;
+ int err;
/* both PCM and MIDI interfaces have 2 or more altsettings */
if (iface->num_altsetting < 2)
@@ -482,9 +518,7 @@
return -ENXIO;
}
- stream = (fp->endpoint & USB_DIR_IN)
- ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
- err = snd_usb_add_audio_stream(chip, stream, fp);
+ err = add_audio_stream_from_fixed_fmt(chip, fp);
if (err < 0) {
list_del(&fp->list); /* unlink for avoiding double-free */
kfree(fp);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 980287a..215c177 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -44,6 +44,7 @@
struct list_head pcm_list; /* list of pcm streams */
struct list_head ep_list; /* list of audio-related endpoints */
+ struct list_head iface_ref_list; /* list of interface refcounts */
int pcm_devs;
struct list_head midi_list; /* list of midi interfaces */
