Merge tag 'sound-4.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This ended up with a larger set of fixes than wished, unfortunately.
As diffstat shows, the majority of changes are for various ASoC
drivers (Realtek, Wolfson codec drivers, etc), in addition to a couple
of HD-audio regression fixes. All these are reasonably small and
nothing to scare much"
* tag 'sound-4.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits)
ALSA: hda - Disable power_save_node for Thinkpads
ALSA: hda/tegra - async probe for avoiding module loading deadlock
ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging
ASoC: rt5645: Increase the delay time to remove the pop sound
ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up
ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
MAINTAINERS: Update website and git repo for Wolfson Microelectronics
ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
ASoC: wm0010: fix error path
ASoC: wm0010: fix memory leak
ASoC: wm8960: correct the max register value of mic boost pga
ASoC: wm8962: remove 64k sample rate support
ASoC: davinci-mcasp: Fix devm_kasprintf format string
ASoC: fix broken pxa SoC support
ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver
ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning
ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
ASoC: mediatek: Increase periods_min in capture
ASoC: davinci-mcasp: Revise the FIFO threshold calculation
ASoC: wm8960: correct gain value for input PGA and add microphone PGA
...
diff --git a/MAINTAINERS b/MAINTAINERS
index 4a59cdc..45b06ab 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -11249,7 +11249,6 @@
M: Liam Girdwood <lgirdwood@gmail.com>
M: Mark Brown <broonie@kernel.org>
L: linux-kernel@vger.kernel.org
-W: http://opensource.wolfsonmicro.com/node/15
W: http://www.slimlogic.co.uk/?p=48
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator.git
S: Supported
@@ -11378,17 +11377,15 @@
M: Mark Brown <broonie@kernel.org>
M: Liam Girdwood <lrg@slimlogic.co.uk>
L: linux-input@vger.kernel.org
-T: git git://opensource.wolfsonmicro.com/linux-2.6-touch
-W: http://opensource.wolfsonmicro.com/node/7
+W: https://github.com/CirrusLogic/linux-drivers/wiki
S: Supported
F: drivers/input/touchscreen/*wm97*
F: include/linux/wm97xx.h
WOLFSON MICROELECTRONICS DRIVERS
L: patches@opensource.wolfsonmicro.com
-T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
-T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
-W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
+T: git https://github.com/CirrusLogic/linux-drivers.git
+W: https://github.com/CirrusLogic/linux-drivers/wiki
S: Supported
F: Documentation/hwmon/wm83??
F: arch/arm/mach-s3c64xx/mach-crag6410*
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 885683a..e040621 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -9,6 +9,14 @@
Drivers that are implemented on ASoC can be found in
"ALSA for SoC audio support" section.
+config SND_PXA2XX_LIB
+ tristate
+ select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
+ select SND_DMAENGINE_PCM
+
+config SND_PXA2XX_LIB_AC97
+ bool
+
if SND_ARM
config SND_ARMAACI
@@ -21,13 +29,6 @@
tristate
select SND_PCM
-config SND_PXA2XX_LIB
- tristate
- select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
-
-config SND_PXA2XX_LIB_AC97
- bool
-
config SND_PXA2XX_AC97
tristate "AC97 driver for the Intel PXA2xx chip"
depends on ARCH_PXA
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 477742c..58c0aad 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -73,6 +73,7 @@
struct clk *hda2codec_2x_clk;
struct clk *hda2hdmi_clk;
void __iomem *regs;
+ struct work_struct probe_work;
};
#ifdef CONFIG_PM
@@ -294,7 +295,9 @@
static int hda_tegra_dev_free(struct snd_device *device)
{
struct azx *chip = device->device_data;
+ struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
+ cancel_work_sync(&hda->probe_work);
if (azx_bus(chip)->chip_init) {
azx_stop_all_streams(chip);
azx_stop_chip(chip);
@@ -426,6 +429,9 @@
/*
* constructor
*/
+
+static void hda_tegra_probe_work(struct work_struct *work);
+
static int hda_tegra_create(struct snd_card *card,
unsigned int driver_caps,
struct hda_tegra *hda)
@@ -452,6 +458,8 @@
chip->single_cmd = false;
chip->snoop = true;
+ INIT_WORK(&hda->probe_work, hda_tegra_probe_work);
+
err = azx_bus_init(chip, NULL, &hda_tegra_io_ops);
if (err < 0)
return err;
@@ -499,6 +507,21 @@
card->private_data = chip;
dev_set_drvdata(&pdev->dev, card);
+ schedule_work(&hda->probe_work);
+
+ return 0;
+
+out_free:
+ snd_card_free(card);
+ return err;
+}
+
+static void hda_tegra_probe_work(struct work_struct *work)
+{
+ struct hda_tegra *hda = container_of(work, struct hda_tegra, probe_work);
+ struct azx *chip = &hda->chip;
+ struct platform_device *pdev = to_platform_device(hda->dev);
+ int err;
err = hda_tegra_first_init(chip, pdev);
if (err < 0)
@@ -520,11 +543,8 @@
chip->running = 1;
snd_hda_set_power_save(&chip->bus, power_save * 1000);
- return 0;
-
-out_free:
- snd_card_free(card);
- return err;
+ out_free:
+ return; /* no error return from async probe */
}
static int hda_tegra_remove(struct platform_device *pdev)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a75b561..afec6dc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4188,6 +4188,24 @@
}
}
+/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */
+static void alc_fixup_tpt440_dock(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const struct hda_pintbl pincfgs[] = {
+ { 0x16, 0x21211010 }, /* dock headphone */
+ { 0x19, 0x21a11010 }, /* dock mic */
+ { }
+ };
+ struct alc_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
+ codec->power_save_node = 0; /* avoid click noises */
+ snd_hda_apply_pincfgs(codec, pincfgs);
+ }
+}
+
static void alc_shutup_dell_xps13(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -4562,7 +4580,6 @@
ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC292_FIXUP_TPT440_DOCK,
- ALC292_FIXUP_TPT440_DOCK2,
ALC283_FIXUP_BXBT2807_MIC,
ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
ALC282_FIXUP_ASPIRE_V5_PINS,
@@ -5029,17 +5046,7 @@
},
[ALC292_FIXUP_TPT440_DOCK] = {
.type = HDA_FIXUP_FUNC,
- .v.func = alc269_fixup_pincfg_no_hp_to_lineout,
- .chained = true,
- .chain_id = ALC292_FIXUP_TPT440_DOCK2
- },
- [ALC292_FIXUP_TPT440_DOCK2] = {
- .type = HDA_FIXUP_PINS,
- .v.pins = (const struct hda_pintbl[]) {
- { 0x16, 0x21211010 }, /* dock headphone */
- { 0x19, 0x21a11010 }, /* dock mic */
- { }
- },
+ .v.func = alc_fixup_tpt440_dock,
.chained = true,
.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
},
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 38e853a..0bf9d62 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -296,7 +296,6 @@
{
struct resource *iores, *dmares;
unsigned long sel;
- int ret;
struct au1xpsc_audio_data *wd;
wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data),
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 4972bf3..268a28b 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -732,14 +732,14 @@
static const struct snd_kcontrol_new rt5645_dac_l_mix[] = {
SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
RT5645_M_ADCMIX_L_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
RT5645_M_DAC1_L_SFT, 1, 1),
};
static const struct snd_kcontrol_new rt5645_dac_r_mix[] = {
SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
RT5645_M_ADCMIX_R_SFT, 1, 1),
- SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+ SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
RT5645_M_DAC1_R_SFT, 1, 1),
};
@@ -1381,7 +1381,7 @@
regmap_write(rt5645->regmap, RT5645_PR_BASE +
RT5645_MAMP_INT_REG2, 0xfc00);
snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
- mdelay(5);
+ msleep(40);
rt5645->hp_on = true;
} else {
/* depop parameters */
@@ -2829,13 +2829,12 @@
snd_soc_dapm_sync(dapm);
rt5645->jack_type = SND_JACK_HEADPHONE;
}
-
- snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
- snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d);
- snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001);
} else { /* jack out */
rt5645->jack_type = 0;
+ regmap_update_bits(rt5645->regmap, RT5645_HP_VOL,
+ RT5645_L_MUTE | RT5645_R_MUTE,
+ RT5645_L_MUTE | RT5645_R_MUTE);
regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2,
RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD);
regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1,
@@ -2880,8 +2879,6 @@
rt5645->en_button_func = true;
regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1,
RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
- regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
- RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
}
@@ -3205,6 +3202,13 @@
DMI_MATCH(DMI_PRODUCT_NAME, "Celes"),
},
},
+ {
+ .ident = "Google Ultima",
+ .callback = strago_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
+ },
+ },
{ }
};
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index f2c6ad4..581ec15 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -577,7 +577,6 @@
struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
unsigned long flags;
int ret;
- const struct firmware *fw;
struct spi_message m;
struct spi_transfer t;
struct dfw_pllrec pll_rec;
@@ -623,14 +622,6 @@
wm0010->state = WM0010_OUT_OF_RESET;
spin_unlock_irqrestore(&wm0010->irq_lock, flags);
- /* First the bootloader */
- ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev);
- if (ret != 0) {
- dev_err(codec->dev, "Failed to request stage2 loader: %d\n",
- ret);
- goto abort;
- }
-
if (!wait_for_completion_timeout(&wm0010->boot_completion,
msecs_to_jiffies(20)))
dev_err(codec->dev, "Failed to get interrupt from DSP\n");
@@ -673,7 +664,7 @@
img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
if (!img_swap)
- goto abort;
+ goto abort_out;
/* We need to re-order for 0010 */
byte_swap_64((u64 *)&pll_rec, img_swap, len);
@@ -688,16 +679,16 @@
spi_message_add_tail(&t, &m);
ret = spi_sync(spi, &m);
- if (ret != 0) {
+ if (ret) {
dev_err(codec->dev, "First PLL write failed: %d\n", ret);
- goto abort;
+ goto abort_swap;
}
/* Use a second send of the message to get the return status */
ret = spi_sync(spi, &m);
- if (ret != 0) {
+ if (ret) {
dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
- goto abort;
+ goto abort_swap;
}
p = (u32 *)out;
@@ -730,6 +721,10 @@
return 0;
+abort_swap:
+ kfree(img_swap);
+abort_out:
+ kfree(out);
abort:
/* Put the chip back into reset */
wm0010_halt(codec);
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index e3b7d0c..dbd8840 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -211,28 +211,38 @@
return wm8960_set_deemph(codec);
}
-static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
-static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
-static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
+static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1);
+static const unsigned int micboost_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0),
+ 2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0),
+};
static const struct snd_kcontrol_new wm8960_snd_controls[] = {
SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
- 0, 63, 0, adc_tlv),
+ 0, 63, 0, inpga_tlv),
SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
6, 1, 0),
SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
7, 1, 0),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
- WM8960_INBMIX1, 4, 7, 0, boost_tlv),
+ WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
- WM8960_INBMIX1, 1, 7, 0, boost_tlv),
+ WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
- WM8960_INBMIX2, 4, 7, 0, boost_tlv),
+ WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
- WM8960_INBMIX2, 1, 7, 0, boost_tlv),
+ WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
+ WM8960_RINPATH, 4, 3, 0, micboost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
+ WM8960_LINPATH, 4, 3, 0, micboost_tlv),
SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
0, 255, 0, dac_tlv),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index b4eb975..293e47a 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2944,7 +2944,8 @@
WM8962_DAC_MUTE, val);
}
-#define WM8962_RATES SNDRV_PCM_RATE_8000_96000
+#define WM8962_RATES (SNDRV_PCM_RATE_8000_48000 |\
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index add6bb9..7d45d98 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -663,7 +663,7 @@
u8 rx_ser = 0;
u8 slots = mcasp->tdm_slots;
u8 max_active_serializers = (channels + slots - 1) / slots;
- int active_serializers, numevt, n;
+ int active_serializers, numevt;
u32 reg;
/* Default configuration */
if (mcasp->version < MCASP_VERSION_3)
@@ -745,9 +745,8 @@
* The number of words for numevt need to be in steps of active
* serializers.
*/
- n = numevt % active_serializers;
- if (n)
- numevt += (active_serializers - n);
+ numevt = (numevt / active_serializers) * active_serializers;
+
while (period_words % numevt && numevt > 0)
numevt -= active_serializers;
if (numevt <= 0)
@@ -1299,6 +1298,7 @@
.ops = &davinci_mcasp_dai_ops,
.symmetric_samplebits = 1,
+ .symmetric_rates = 1,
},
{
.name = "davinci-mcasp.1",
@@ -1685,7 +1685,7 @@
irq = platform_get_irq_byname(pdev, "common");
if (irq >= 0) {
- irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n",
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
dev_name(&pdev->dev));
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_common_irq_handler,
@@ -1702,7 +1702,7 @@
irq = platform_get_irq_byname(pdev, "rx");
if (irq >= 0) {
- irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
dev_name(&pdev->dev));
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_rx_irq_handler,
@@ -1717,7 +1717,7 @@
irq = platform_get_irq_byname(pdev, "tx");
if (irq >= 0) {
- irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n",
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
dev_name(&pdev->dev));
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_tx_irq_handler,
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5aeb6ed..96f55ae 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -488,7 +488,8 @@
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto asrc_fail;
}
/* Common settings for corresponding Freescale CPU DAI driver */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 8ec6fb2..37c5cd4 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -249,7 +249,8 @@
static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private)
{
- return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97);
+ return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+ SND_SOC_DAIFMT_AC97;
}
static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
@@ -947,7 +948,7 @@
CCSR_SSI_SCR_TCH_EN);
}
- if (fmt & SND_SOC_DAIFMT_AC97)
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97)
fsl_ssi_setup_ac97(ssi_private);
return 0;
diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index f6efa9d..b27f25f 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -302,6 +302,10 @@
struct sst_hsw_ipc_dx_reply dx;
void *dx_context;
dma_addr_t dx_context_paddr;
+ enum sst_hsw_device_id dx_dev;
+ enum sst_hsw_device_mclk dx_mclk;
+ enum sst_hsw_device_mode dx_mode;
+ u32 dx_clock_divider;
/* boot */
wait_queue_head_t boot_wait;
@@ -1400,10 +1404,10 @@
trace_ipc_request("set device config", dev);
- config.ssp_interface = dev;
- config.clock_frequency = mclk;
- config.mode = mode;
- config.clock_divider = clock_divider;
+ hsw->dx_dev = config.ssp_interface = dev;
+ hsw->dx_mclk = config.clock_frequency = mclk;
+ hsw->dx_mode = config.mode = mode;
+ hsw->dx_clock_divider = config.clock_divider = clock_divider;
if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER)
config.channels = 4;
else
@@ -1704,10 +1708,10 @@
return -EIO;
}
- /* Set ADSP SSP port settings */
- ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0,
- SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
- SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ /* Set ADSP SSP port settings - sadly the FW does not store SSP port
+ settings as part of the PM context. */
+ ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk,
+ hsw->dx_mode, hsw->dx_clock_divider);
if (ret < 0)
dev_err(dev, "error: SSP re-initialization failed\n");
diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index d190fe0..f5baf3c 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -549,6 +549,23 @@
memif->substream = substream;
snd_soc_set_runtime_hwparams(substream, &mtk_afe_hardware);
+
+ /*
+ * Capture cannot use ping-pong buffer since hw_ptr at IRQ may be
+ * smaller than period_size due to AFE's internal buffer.
+ * This easily leads to overrun when avail_min is period_size.
+ * One more period can hold the possible unread buffer.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS,
+ 3,
+ mtk_afe_hardware.periods_max);
+ if (ret < 0) {
+ dev_err(afe->dev, "hw_constraint_minmax failed\n");
+ return ret;
+ }
+ }
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 39cea80..f2bf866 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,7 +1,6 @@
config SND_PXA2XX_SOC
tristate "SoC Audio for the Intel PXA2xx chip"
depends on ARCH_PXA
- select SND_ARM
select SND_PXA2XX_LIB
help
Say Y or M if you want to add support for codecs attached to
@@ -25,7 +24,6 @@
config SND_PXA2XX_SOC_AC97
tristate
select AC97_BUS
- select SND_ARM
select SND_PXA2XX_LIB_AC97
select SND_SOC_AC97_BUS
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1f60546..9e4b04e 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@
.reset = pxa2xx_ac97_cold_reset,
};
-static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -57,7 +57,7 @@
.filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
};
-static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f4bf21a..ff8bda4 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3501,7 +3501,7 @@
default:
WARN(1, "Unknown event %d\n", event);
- return -EINVAL;
+ ret = -EINVAL;
}
out:
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 362c69a..53dd085 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -101,6 +101,15 @@
SNDRV_PCM_FMTBIT_S32_LE | \
SNDRV_PCM_FMTBIT_U32_LE | \
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+/*
+ * The dummy CODEC is only meant to be used in situations where there is no
+ * actual hardware.
+ *
+ * If there is actual hardware even if it does not have a control bus
+ * the hardware will still have constraints like supported samplerates, etc.
+ * which should be modelled. And the data flow graph also should be modelled
+ * using DAPM.
+ */
static struct snd_soc_dai_driver dummy_dai = {
.name = "snd-soc-dummy-dai",
.playback = {
diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig
index 0a53053..4fb9141 100644
--- a/sound/soc/spear/Kconfig
+++ b/sound/soc/spear/Kconfig
@@ -1,6 +1,6 @@
config SND_SPEAR_SOC
tristate
- select SND_DMAENGINE_PCM
+ select SND_SOC_GENERIC_DMAENGINE_PCM
config SND_SPEAR_SPDIF_OUT
tristate
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index f6eefe1..843f037 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -989,8 +989,8 @@
if (!info)
return -ENOMEM;
- of_property_read_u32(pnode, "version", &player->ver);
- if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
+ if (of_property_read_u32(pnode, "version", &player->ver) ||
+ player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
dev_err(dev, "Unknown uniperipheral version ");
return -EINVAL;
}
@@ -998,10 +998,16 @@
if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
info->underflow_enabled = 1;
- of_property_read_u32(pnode, "uniperiph-id", &info->id);
+ if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) {
+ dev_err(dev, "uniperipheral id not defined");
+ return -EINVAL;
+ }
/* Read the device mode property */
- of_property_read_string(pnode, "mode", &mode);
+ if (of_property_read_string(pnode, "mode", &mode)) {
+ dev_err(dev, "uniperipheral mode not defined");
+ return -EINVAL;
+ }
if (strcasecmp(mode, "hdmi") == 0)
info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI;
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index c502626..f791239 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -316,7 +316,11 @@
if (!info)
return -ENOMEM;
- of_property_read_u32(node, "version", &reader->ver);
+ if (of_property_read_u32(node, "version", &reader->ver) ||
+ reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
+ dev_err(&pdev->dev, "Unknown uniperipheral version ");
+ return -EINVAL;
+ }
/* Save the info structure */
reader->info = info;