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sound fixes for 5.4-rc1 

A few small remaining wrap-up for this merge window.
 
 Most of patches are device-specific (HD-audio and USB-audio quirks,
 FireWire, pcm316a, fsl, rsnd, Atmel, and TI fixes), while there is
 a simple fix (actually two commits) for ASoC core.
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Merge tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A few small remaining wrap-up for this merge window.

  Most of patches are device-specific (HD-audio and USB-audio quirks,
  FireWire, pcm316a, fsl, rsnd, Atmel, and TI fixes), while there is a
  simple fix (actually two commits) for ASoC core"

* tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Add DSD support for EVGA NU Audio
  ALSA: hda - Add laptop imic fixup for ASUS M9V laptop
  ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies
  ASoC: pcm3168a: The codec does not support S32_LE
  ASoC: core: use list_del_init and move it back to soc_cleanup_component
  ALSA: hda/realtek - PCI quirk for Medion E4254
  ALSA: hda - Apply AMD controller workaround for Raven platform
  ASoC: rsnd: do error check after rsnd_channel_normalization()
  ASoC: atmel_ssc_dai: Remove wrong spinlock usage
  ASoC: core: delete component->card_list in soc_remove_component only
  ASoC: fsl_sai: Fix noise when using EDMA
  ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
  ALSA: hda/realtek - Fix alienware headset mic
  ALSA: dice: fix wrong packet parameter for Alesis iO26
This commit is contained in:
Linus Torvalds 2019-09-24 16:46:16 -07:00
parent b6cb84b4fc f41f900568
commit 3cf7487c5d
13 changed files with 67 additions and 24 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
sound/firewire/dice/dice-alesis.c
View File

@ -15,7 +15,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
static const unsigned int
alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
{10, 10, 8}, /* Tx0 = Analog + S/PDIF. */
{10, 10, 4}, /* Tx0 = Analog + S/PDIF. */
{16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */
};

3
sound/pci/hda/hda_intel.c
View File

@ -2485,8 +2485,7 @@ static const struct pci_device_id azx_ids[] = {
AZX_DCAPS_PM_RUNTIME },
/* AMD Raven */
{ PCI_DEVICE(0x1022, 0x15e3),
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
AZX_DCAPS_PM_RUNTIME },
.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 },

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

@ -357,6 +357,7 @@ static const struct hda_fixup ad1986a_fixups[] = {
static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9V", AD1986A_FIXUP_LAPTOP_IMIC),
SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD),
SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD),
SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),

28
sound/pci/hda/patch_realtek.c
View File

@ -5817,6 +5817,7 @@ enum {
ALC292_FIXUP_DELL_E7X,
ALC292_FIXUP_DISABLE_AAMIX,
ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK,
ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE,
ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
ALC275_FIXUP_DELL_XPS,
@ -5871,6 +5872,7 @@ enum {
ALC256_FIXUP_ASUS_MIC_NO_PRESENCE,
ALC299_FIXUP_PREDATOR_SPK,
ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC,
ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE,
};
static const struct hda_fixup alc269_fixups[] = {
@ -6506,6 +6508,15 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC292_FIXUP_DISABLE_AAMIX
},
[ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
{ 0x18, 0x01a1913c }, /* headset mic w/o jack detect */
{ }
},
.chained_before = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE,
},
[ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@ -6927,6 +6938,16 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
},
[ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
{ 0x19, 0x04a11040 },
{ 0x21, 0x04211020 },
{ }
},
.chained = true,
.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
},
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@ -7190,6 +7211,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
#if 0
/* Below is a quirk table taken from the old code.
@ -7358,6 +7380,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"},
{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
{}
};
#define ALC225_STANDARD_PINS \
@ -7770,6 +7793,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x17, 0x90170110},
{0x1a, 0x03011020},
{0x21, 0x03211030}),
SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE,
{0x12, 0xb7a60140},
{0x17, 0x90170110},
{0x1a, 0x03a11030},
{0x21, 0x03211020}),
SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
ALC225_STANDARD_PINS,
{0x12, 0xb7a60130},

12
sound/soc/atmel/atmel_ssc_dai.c
View File

@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = {
{
.name = "ssc0",
.lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
.dir_mask = SSC_DIR_MASK_UNUSED,
.initialized = 0,
},
{
.name = "ssc1",
.lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
.dir_mask = SSC_DIR_MASK_UNUSED,
.initialized = 0,
},
{
.name = "ssc2",
.lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
.dir_mask = SSC_DIR_MASK_UNUSED,
.initialized = 0,
},
@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
snd_soc_dai_set_dma_data(dai, substream, dma_params);
spin_lock_irq(&ssc_p->lock);
if (ssc_p->dir_mask & dir_mask) {
spin_unlock_irq(&ssc_p->lock);
if (ssc_p->dir_mask & dir_mask)
return -EBUSY;
}
ssc_p->dir_mask |= dir_mask;
spin_unlock_irq(&ssc_p->lock);
return 0;
}
@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
dir_mask = 1 << dir;
spin_lock_irq(&ssc_p->lock);
ssc_p->dir_mask &= ~dir_mask;
if (!ssc_p->dir_mask) {
if (ssc_p->initialized) {
@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
ssc_p->forced_divider = 0;
}
spin_unlock_irq(&ssc_p->lock);
/* Shutdown the SSC clock. */
pr_debug("atmel_ssc_dai: Stopping clock\n");

1
sound/soc/atmel/atmel_ssc_dai.h
View File

@ -93,7 +93,6 @@ struct atmel_ssc_state {
struct atmel_ssc_info {
char *name;
struct ssc_device *ssc;
spinlock_t lock; /* lock for dir_mask */
unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
unsigned short initialized; /* true if SSC has been initialized */
unsigned short daifmt;

3
sound/soc/codecs/pcm3168a.c
View File

@ -21,8 +21,7 @@
#define PCM3168A_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_3LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
SNDRV_PCM_FMTBIT_S24_LE)
#define PCM3168A_FMT_I2S 0x0
#define PCM3168A_FMT_LEFT_J 0x1

15
sound/soc/fsl/fsl_sai.c
View File

@ -628,6 +628,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
FSL_SAI_CR3_TRCE_MASK,
FSL_SAI_CR3_TRCE);
/*
* EDMA controller needs period size to be a multiple of
* tx/rx maxburst
*/
if (sai->soc_data->use_edma)
snd_pcm_hw_constraint_step(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
tx ? sai->dma_params_tx.maxburst :
sai->dma_params_rx.maxburst);
ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints);
@ -1026,30 +1036,35 @@ static int fsl_sai_remove(struct platform_device *pdev)
static const struct fsl_sai_soc_data fsl_sai_vf610_data = {
.use_imx_pcm = false,
.use_edma = false,
.fifo_depth = 32,
.reg_offset = 0,
};
static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
.use_imx_pcm = true,
.use_edma = false,
.fifo_depth = 32,
.reg_offset = 0,
};
static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = {
.use_imx_pcm = true,
.use_edma = false,
.fifo_depth = 16,
.reg_offset = 8,
};
static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = {
.use_imx_pcm = true,
.use_edma = false,
.fifo_depth = 128,
.reg_offset = 8,
};
static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = {
.use_imx_pcm = true,
.use_edma = true,
.fifo_depth = 64,
.reg_offset = 0,
};

1
sound/soc/fsl/fsl_sai.h
View File

@ -157,6 +157,7 @@
struct fsl_sai_soc_data {
bool use_imx_pcm;
bool use_edma;
unsigned int fifo_depth;
unsigned int reg_offset;
};

10
sound/soc/sh/rcar/ssi.c
View File

@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
if (rsnd_ssi_is_multi_slave(mod, io))
return 0;
if (rsnd_runtime_is_tdm_split(io))
chan = rsnd_io_converted_chan(io);
chan = rsnd_channel_normalization(chan);
if (ssi->usrcnt > 0) {
if (ssi->rate != rate) {
dev_err(dev, "SSI parent/child should use same rate\n");
@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
return 0;
}
if (rsnd_runtime_is_tdm_split(io))
chan = rsnd_io_converted_chan(io);
chan = rsnd_channel_normalization(chan);
main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
if (!main_rate) {
dev_err(dev, "unsupported clock rate\n");

2
sound/soc/soc-core.c
View File

@ -978,7 +978,7 @@ static void soc_cleanup_component(struct snd_soc_component *component)
/* For framework level robustness */
snd_soc_component_set_jack(component, NULL, NULL);
list_del(&component->card_list);
list_del_init(&component->card_list);
snd_soc_dapm_free(snd_soc_component_get_dapm(component));
soc_cleanup_component_debugfs(component);
component->card = NULL;

11
sound/soc/ti/Kconfig
View File

@ -200,11 +200,18 @@ config SND_SOC_DM365_AIC3X_CODEC
config SND_SOC_DM365_VOICE_CODEC
bool "Voice Codec - CQ93VC"
select MFD_DAVINCI_VOICECODEC
select SND_SOC_CQ0093VC
help
Say Y if you want to add support for SoC On-chip voice codec
endchoice
config SND_SOC_DM365_VOICE_CODEC_MODULE
def_tristate y
depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC
select MFD_DAVINCI_VOICECODEC
select SND_SOC_CQ0093VC
help
The is an internal symbol needed to ensure that the codec
and MFD driver can be built as loadable modules if necessary.
endmenu

2
sound/usb/quirks.c
View File

@ -1658,6 +1658,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
case 0x25ce: /* Mytek devices */
case 0x278b: /* Rotel? */
case 0x2ab6: /* T+A devices */
case 0x3842: /* EVGA */
case 0xc502: /* HiBy devices */
if (fp->dsd_raw)
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;