mirrors
/
linux-stable
mirror of https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git
1
0
Fork 0
Code Issues Releases Wiki Activity

sound fixes for 5.14-rc3 

A collection of small fixes, mostly covering device-specific
 regressions and bugs over ASoC, HD-audio and USB-audio, while the
 ALSA PCM core received a few additional fixes for the possible
 (new and old) regressions.
 -----BEGIN PGP SIGNATURE-----
 
 iQJCBAABCAAsFiEEIXTw5fNLNI7mMiVaLtJE4w1nLE8FAmD6eIUOHHRpd2FpQHN1
 c2UuZGUACgkQLtJE4w1nLE+jdhAAwL64tV/0gw1IAq61YMRQmDhSO9D9ebdnqRdo
 kgjedrwDLulSczli9YNCSjy8yhPRcPBwt09W9vcvuvAnp7v7Z2rboReG9RCdbaTm
 Fwle30UQlOJmwE8wiAyxhrrb1R1vhb5omOlMUfnDjXHKaIE3PatZMFoHM4JzyDa8
 YeTdHGjRgu2NfJU3ki57iDol0YO6EfEa2cy/FTvdHBR+x8l4e4F70eeg7CEkcLnQ
 Ckz+hhAx4EIrff4I3MDKaEzn3iY2hTglqdv6qV0/S8eCKB2p8f45lu3wdtmtYnD1
 74wB6pY5InTpaYE5Tf4TdqGhlmuhOsYLv+f65FNriVoIuU9Jc0AgaU9bX9j6WkS4
 DC4ueI0TcfMDxDUWab4YrYhwuvNSFo4neonZ3nPWaSTAMsfmirJWDwEe1VZZrtfG
 t5tmCVpghic5os7p97V8VWOSaiRz3yyp1ap23/btmuBINemgjRPtftIKVEngbnkI
 NS8m1mZUpyCosIv7KJXScMUgA8nOlU0y6p9RdEQTJ2FECaIf2XOnrAbPiMVAlFtj
 TFGF0ubkh3FGCqjEBb0c5nDKnVJmk7yK3ddt78f/ocASxIBRs8ssdhrkkVlXLLEP
 iygBDhzePtXTpfnWUnseyYgctXDPSGD5M0pAfgWqFP7AilJIcThIpr0zhfeVs/KI
 ep0zZYY=
 =O7fC
 -----END PGP SIGNATURE-----

Merge tag 'sound-5.14-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes, mostly covering device-specific
  regressions and bugs over ASoC, HD-audio and USB-audio, while
  the ALSA PCM core received a few additional fixes for the
  possible (new and old) regressions"

* tag 'sound-5.14-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits)
  ALSA: usb-audio: Add registration quirk for JBL Quantum headsets
  ALSA: hda/hdmi: Add quirk to force pin connectivity on NUC10
  ALSA: pcm: Fix mmap without buffer preallocation
  ALSA: pcm: Fix mmap capability check
  ALSA: hda: intel-dsp-cfg: add missing ElkhartLake PCI ID
  ASoC: ti: j721e-evm: Check for not initialized parent_clk_id
  ASoC: ti: j721e-evm: Fix unbalanced domain activity tracking during startup
  ALSA: hda/realtek: Fix pop noise and 2 Front Mic issues on a machine
  ALSA: hdmi: Expose all pins on MSI MS-7C94 board
  ALSA: sb: Fix potential ABBA deadlock in CSP driver
  ASoC: rt5682: Fix the issue of garbled recording after powerd_dbus_suspend
  ASoC: amd: reverse stop sequence for stoneyridge platform
  ASoC: soc-pcm: add a flag to reverse the stop sequence
  ASoC: codecs: wcd938x: setup irq during component bind
  ASoC: dt-bindings: renesas: rsnd: Fix incorrect 'port' regex schema
  ALSA: usb-audio: Add missing proc text entry for BESPOKEN type
  ASoC: codecs: wcd938x: make sdw dependency explicit in Kconfig
  ASoC: SOF: Intel: Update ADL descriptor to use ACPI power states
  ASoC: rt5631: Fix regcache sync errors on resume
  ALSA: pcm: Call substream ack() method upon compat mmap commit
  ...
This commit is contained in:
Linus Torvalds 2021-07-23 09:58:23 -07:00
parent 8baef6386b b0084afde2
commit e7562a00c1
23 changed files with 193 additions and 94 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
View File

@ -114,7 +114,7 @@ properties:
ports:
$ref: /schemas/graph.yaml#/properties/ports
properties:
patternProperties:
port(@[0-9a-f]+)?:
$ref: audio-graph-port.yaml#
unevaluatedProperties: false

6
include/sound/soc.h
View File

@ -712,6 +712,12 @@ struct snd_soc_dai_link {
/* Do not create a PCM for this DAI link (Backend link) */
unsigned int ignore:1;
/* This flag will reorder stop sequence. By enabling this flag
* DMA controller stop sequence will be invoked first followed by
* CPU DAI driver stop sequence
*/
unsigned int stop_dma_first:1;
#ifdef CONFIG_SND_SOC_TOPOLOGY
struct snd_soc_dobj dobj; /* For topology */
#endif

27
sound/core/pcm_native.c
View File

@ -246,12 +246,18 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream)
if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP))
return false;
if (substream->ops->mmap ||
(substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV &&
substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC))
if (substream->ops->mmap)
return true;
return dma_can_mmap(substream->dma_buffer.dev.dev);
switch (substream->dma_buffer.dev.type) {
case SNDRV_DMA_TYPE_UNKNOWN:
return false;
case SNDRV_DMA_TYPE_CONTINUOUS:
case SNDRV_DMA_TYPE_VMALLOC:
return true;
default:
return dma_can_mmap(substream->dma_buffer.dev.dev);
}
}
static int constrain_mask_params(struct snd_pcm_substream *substream,
@ -3063,9 +3069,14 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream,
boundary = 0x7fffffff;
snd_pcm_stream_lock_irq(substream);
/* FIXME: we should consider the boundary for the sync from app */
if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL))
control->appl_ptr = scontrol.appl_ptr;
else
if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) {
err = pcm_lib_apply_appl_ptr(substream,
scontrol.appl_ptr);
if (err < 0) {
snd_pcm_stream_unlock_irq(substream);
return err;
}
} else
scontrol.appl_ptr = control->appl_ptr % boundary;
if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN))
control->avail_min = scontrol.avail_min;
@ -3664,6 +3675,8 @@ static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf)
return VM_FAULT_SIGBUS;
if (substream->ops->page)
page = substream->ops->page(substream, offset);
else if (!snd_pcm_get_dma_buf(substream))
page = virt_to_page(runtime->dma_area + offset);
else
page = snd_sgbuf_get_page(snd_pcm_get_dma_buf(substream), offset);
if (!page)

4
sound/hda/intel-dsp-config.c
View File

@ -319,6 +319,10 @@ static const struct config_entry config_table[] = {
.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
.device = 0x4b55,
},
{
.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
.device = 0x4b58,
},
#endif
/* Alder Lake */

4
sound/isa/sb/sb16_csp.c
View File

@ -816,6 +816,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
set_mode_register(p->chip, 0xc0); /* c0 = STOP */
@ -855,6 +856,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
@ -880,6 +882,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
@ -894,6 +897,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);

2
sound/pci/hda/patch_hdmi.c
View File

@ -1940,6 +1940,8 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
static const struct snd_pci_quirk force_connect_list[] = {
SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1),
SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1),
{}
};

1
sound/pci/hda/patch_realtek.c
View File

@ -8626,6 +8626,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),

5
sound/soc/amd/acp-da7219-max98357a.c
View File

@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
| SND_SOC_DAIFMT_CBM_CFM,
.init = cz_rt5682_init,
.dpcm_playback = 1,
.stop_dma_first = 1,
.ops = &cz_rt5682_play_ops,
SND_SOC_DAILINK_REG(designware1, rt5682, platform),
},
@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
.stop_dma_first = 1,
.ops = &cz_rt5682_cap_ops,
SND_SOC_DAILINK_REG(designware2, rt5682, platform),
},
@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_playback = 1,
.stop_dma_first = 1,
.ops = &cz_rt5682_max_play_ops,
SND_SOC_DAILINK_REG(designware3, mx, platform),
},
@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
.stop_dma_first = 1,
.ops = &cz_rt5682_dmic0_cap_ops,
SND_SOC_DAILINK_REG(designware3, adau, platform),
},
@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
.stop_dma_first = 1,
.ops = &cz_rt5682_dmic1_cap_ops,
SND_SOC_DAILINK_REG(designware2, adau, platform),
},

8
sound/soc/codecs/Kconfig
View File

@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305
high-efficiency mono Class-D audio power amplifiers.
config SND_SOC_SSM2518
tristate
tristate "Analog Devices SSM2518 Class-D Amplifier"
depends on I2C
config SND_SOC_SSM2602
@ -1557,6 +1557,7 @@ config SND_SOC_WCD934X
Qualcomm SoCs like SDM845.
config SND_SOC_WCD938X
depends on SND_SOC_WCD938X_SDW
tristate
config SND_SOC_WCD938X_SDW
@ -1813,11 +1814,6 @@ config SND_SOC_ZL38060
which consists of a Digital Signal Processor (DSP), several Digital
Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs.
config SND_SOC_ZX_AUD96P22
tristate "ZTE ZX AUD96P22 CODEC"
depends on I2C
select REGMAP_I2C
# Amp
config SND_SOC_LM4857
tristate

2
sound/soc/codecs/rt5631.c
View File

@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = {
.reg_defaults = rt5631_reg,
.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
.cache_type = REGCACHE_RBTREE,
.use_single_read = true,
.use_single_write = true,
};
static int rt5631_i2c_probe(struct i2c_client *i2c,

8
sound/soc/codecs/rt5682.c
View File

@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
rt5682_enable_push_button_irq(component, false);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") &&
!snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
!snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") &&
!snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
!snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,

2
sound/soc/codecs/tlv320aic31xx.c
View File

@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
ret);
return ret;
}
regcache_cache_only(aic31xx->regmap, true);
aic31xx->dev = &i2c->dev;
aic31xx->irq = i2c->irq;

4
sound/soc/codecs/tlv320aic31xx.h
View File

@ -151,8 +151,8 @@ struct aic31xx_pdata {
#define AIC31XX_WORD_LEN_24BITS 0x02
#define AIC31XX_WORD_LEN_32BITS 0x03
#define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2)
#define AIC31XX_BCLK_MASTER BIT(2)
#define AIC31XX_WCLK_MASTER BIT(3)
#define AIC31XX_BCLK_MASTER BIT(3)
#define AIC31XX_WCLK_MASTER BIT(2)
/* AIC31XX_DATA_OFFSET */
#define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0)

27
sound/soc/codecs/tlv320aic32x4.c
View File

@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
/* -12dB min, 0.5dB steps */
static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
/* -6dB min, 1dB steps */
static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1);
static const char * const lo_cm_text[] = {
@ -1063,21 +1063,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = {
};
static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = {
SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
SOC_SINGLE_S8_TLV("PCM Playback Volume",
AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm),
SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum),
SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN,
AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0,
tlv_driver_gain),
SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
AIC32X4_HPLGAIN, 6, 0x01, 1),
SOC_SINGLE_TLV("HP Driver Gain Volume",
AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain),
SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1),
SOC_SINGLE_TLV("Speaker Driver Playback Volume",
TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain),
SOC_SINGLE_TLV("Speaker Amplifier Playback Volume",
TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol),
SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1,
0, 0, 117, 1, tlv_spk_vol),
SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2,
4, 5, 0, tlv_amp_vol),
};
static const struct snd_kcontrol_new hp_output_mixer_controls[] = {

18
sound/soc/codecs/wcd938x.c
View File

@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component)
(WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0);
}
ret = wcd938x_irq_init(wcd938x, component->dev);
if (ret) {
dev_err(component->dev, "%s: IRQ init failed: %d\n",
__func__, ret);
return ret;
}
wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
WCD938X_IRQ_HPHR_PDM_WD_INT);
wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev);
wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x;
wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode);
if (!wcd938x->txdev) {
@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev);
wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x;
wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev);
if (!wcd938x->tx_sdw_dev) {
dev_err(dev, "could not get txslave with matching of dev\n");
@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev)
return PTR_ERR(wcd938x->regmap);
}
ret = wcd938x_irq_init(wcd938x, dev);
if (ret) {
dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret);
return ret;
}
wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
ret = wcd938x_set_micbias_data(wcd938x);
if (ret < 0) {
dev_err(dev, "%s: bad micbias pdata\n", __func__);

6
sound/soc/codecs/wm_adsp.c
View File

@ -282,6 +282,7 @@
/*
* HALO_CCM_CORE_CONTROL
*/
#define HALO_CORE_RESET 0x00000200
#define HALO_CORE_EN 0x00000001
/*
@ -1213,7 +1214,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
mutex_lock(&ctl->dsp->pwr_lock);
ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size);
ret = wm_coeff_read_ctrl(ctl, ctl->cache, size);
if (!ret && copy_to_user(bytes, ctl->cache, size))
ret = -EFAULT;
@ -3333,7 +3334,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp)
{
return regmap_update_bits(dsp->regmap,
dsp->base + HALO_CCM_CORE_CONTROL,
HALO_CORE_EN, HALO_CORE_EN);
HALO_CORE_RESET | HALO_CORE_EN,
HALO_CORE_RESET | HALO_CORE_EN);
}
static void wm_halo_stop_core(struct wm_adsp *dsp)

77
sound/soc/intel/boards/sof_sdw_max98373.c
View File

@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd)
static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable)
{
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
int ret;
int j;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
/* enable max98373 first */
ret = max_98373_trigger(substream, cmd);
if (ret < 0)
break;
/* set spk pin by playback only */
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
return 0;
ret = sdw_trigger(substream, cmd);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ret = sdw_trigger(substream, cmd);
if (ret < 0)
break;
cpu_dai = asoc_rtd_to_cpu(rtd, 0);
for_each_rtd_codec_dais(rtd, j, codec_dai) {
struct snd_soc_dapm_context *dapm =
snd_soc_component_get_dapm(cpu_dai->component);
char pin_name[16];
ret = max_98373_trigger(substream, cmd);
break;
default:
ret = -EINVAL;
break;
snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk",
codec_dai->component->name_prefix);
if (enable)
ret = snd_soc_dapm_enable_pin(dapm, pin_name);
else
ret = snd_soc_dapm_disable_pin(dapm, pin_name);
if (!ret)
snd_soc_dapm_sync(dapm);
}
return ret;
return 0;
}
static int mx8373_sdw_prepare(struct snd_pcm_substream *substream)
{
int ret = 0;
/* according to soc_pcm_prepare dai link prepare is called first */
ret = sdw_prepare(substream);
if (ret < 0)
return ret;
return mx8373_enable_spk_pin(substream, true);
}
static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream)
{
int ret = 0;
/* according to soc_pcm_hw_free dai link free is called first */
ret = sdw_hw_free(substream);
if (ret < 0)
return ret;
return mx8373_enable_spk_pin(substream, false);
}
static const struct snd_soc_ops max_98373_sdw_ops = {
.startup = sdw_startup,
.prepare = sdw_prepare,
.trigger = max98373_sdw_trigger,
.hw_free = sdw_hw_free,
.prepare = mx8373_sdw_prepare,
.trigger = sdw_trigger,
.hw_free = mx8373_sdw_hw_free,
.shutdown = sdw_shutdown,
};

22
sound/soc/soc-pcm.c
View File

@ -1015,6 +1015,7 @@ out:
static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
int ret = -EINVAL, _ret = 0;
int rollback = 0;
@ -1055,14 +1056,23 @@ start_err:
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
if (ret < 0)
break;
if (rtd->dai_link->stop_dma_first) {
ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
if (ret < 0)
break;
ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
if (ret < 0)
break;
ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
if (ret < 0)
break;
} else {
ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
if (ret < 0)
break;
ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
if (ret < 0)
break;
}
ret = snd_soc_link_trigger(substream, cmd, rollback);
break;
}

1
sound/soc/sof/intel/pci-tgl.c
View File

@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = {
static const struct sof_dev_desc adl_desc = {
.machines = snd_soc_acpi_intel_adl_machines,
.alt_machines = snd_soc_acpi_intel_adl_sdw_machines,
.use_acpi_target_states = true,
.resindex_lpe_base = 0,
.resindex_pcicfg_base = -1,
.resindex_imr_base = -1,

30
sound/soc/tegra/tegra_pcm.c
View File

@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component,
}
EXPORT_SYMBOL_GPL(tegra_pcm_pointer);
static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
static int tegra_pcm_preallocate_dma_buffer(struct device *dev, struct snd_pcm *pcm, int stream,
size_t size)
{
struct snd_pcm_substream *substream = pcm->streams[stream].substream;
struct snd_dma_buffer *buf = &substream->dma_buffer;
buf->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL);
buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL);
if (!buf->area)
return -ENOMEM;
buf->private_data = NULL;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
buf->dev.dev = pcm->card->dev;
buf->dev.dev = dev;
buf->bytes = size;
return 0;
@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
if (!buf->area)
return;
dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr);
dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr);
buf->area = NULL;
}
static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd,
static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd,
size_t size)
{
struct snd_card *card = rtd->card->snd_card;
struct snd_pcm *pcm = rtd->pcm;
int ret;
ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32));
if (ret < 0)
return ret;
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
ret = tegra_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK, size);
ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size);
if (ret)
goto err;
}
if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
ret = tegra_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE, size);
ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size);
if (ret)
goto err_free_play;
}
@ -284,7 +281,16 @@ err:
int tegra_pcm_construct(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max);
struct device *dev = component->dev;
/*
* Fallback for backwards-compatibility with older device trees that
* have the iommus property in the virtual, top-level "sound" node.
*/
if (!of_get_property(dev->of_node, "iommus", NULL))
dev = rtd->card->snd_card->dev;
return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max);
}
EXPORT_SYMBOL_GPL(tegra_pcm_construct);

18
sound/soc/ti/j721e-evm.c
View File

@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv,
return ret;
}
if (priv->hsdiv_rates[domain->parent_clk_id] != scki) {
if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) {
dev_dbg(priv->dev,
"%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI",
@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream)
j721e_rule_rate, &priv->rate_range,
SNDRV_PCM_HW_PARAM_RATE, -1);
mutex_unlock(&priv->mutex);
if (ret)
return ret;
goto out;
/* Reset TDM slots to 32 */
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
if (ret && ret != -ENOTSUPP)
return ret;
goto out;
for_each_rtd_codec_dais(rtd, i, codec_dai) {
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
if (ret && ret != -ENOTSUPP)
return ret;
goto out;
}
return 0;
if (ret == -ENOTSUPP)
ret = 0;
out:
if (ret)
domain->active--;
mutex_unlock(&priv->mutex);
return ret;
}
static int j721e_audio_hw_params(struct snd_pcm_substream *substream,

10
sound/usb/mixer.c
View File

@ -3295,7 +3295,15 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer,
{
struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
static const char * const val_types[] = {
"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16", "S32", "U32",
[USB_MIXER_BOOLEAN] = "BOOLEAN",
[USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN",
[USB_MIXER_S8] = "S8",
[USB_MIXER_U8] = "U8",
[USB_MIXER_S16] = "S16",
[USB_MIXER_U16] = "U16",
[USB_MIXER_S32] = "S32",
[USB_MIXER_U32] = "U32",
[USB_MIXER_BESPOKEN] = "BESPOKEN",
};
snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, "
"channels=%i, type=\"%s\"\n", cval->head.id,

3
sound/usb/quirks.c
View File

@ -1897,6 +1897,9 @@ static const struct registration_quirk registration_quirks[] = {
REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */
REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */
REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */
REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */
REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */
REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */
{ 0 } /* terminator */
};