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-rw-r--r--sound/arm/Kconfig15
-rw-r--r--sound/hda/ext/hdac_ext_bus.c1
-rw-r--r--sound/pci/hda/hda_codec.c4
-rw-r--r--sound/pci/hda/hda_tegra.c30
-rw-r--r--sound/pci/hda/patch_cirrus.c1
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c32
-rw-r--r--sound/pci/hda/patch_sigmatel.c6
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile1
-rw-r--r--sound/soc/atmel/atmel_wm8904.c1
-rw-r--r--sound/soc/au1x/db1000.c10
-rw-r--r--sound/soc/au1x/db1200.c14
-rw-r--r--sound/soc/au1x/psc-i2s.c1
-rw-r--r--sound/soc/blackfin/bf5xx-ad1836.c11
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1373.c12
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1701.c12
-rw-r--r--sound/soc/blackfin/bfin-eval-adav80x.c12
-rw-r--r--sound/soc/codecs/Kconfig9
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/ak4613.c497
-rw-r--r--sound/soc/codecs/ak4642.c153
-rw-r--r--sound/soc/codecs/arizona.c16
-rw-r--r--sound/soc/codecs/arizona.h2
-rw-r--r--sound/soc/codecs/hdmi.c109
-rw-r--r--sound/soc/codecs/rt298.c26
-rw-r--r--sound/soc/codecs/rt5645.c63
-rw-r--r--sound/soc/codecs/rt5645.h22
-rw-r--r--sound/soc/codecs/sgtl5000.c6
-rw-r--r--sound/soc/codecs/tas2552.c2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c49
-rw-r--r--sound/soc/codecs/wm0010.c23
-rw-r--r--sound/soc/codecs/wm5110.c187
-rw-r--r--sound/soc/codecs/wm8960.c26
-rw-r--r--sound/soc/codecs/wm8962.c10
-rw-r--r--sound/soc/davinci/davinci-mcasp.c319
-rw-r--r--sound/soc/dwc/designware_i2s.c19
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c4
-rw-r--r--sound/soc/fsl/fsl_sai.c1
-rw-r--r--sound/soc/fsl/fsl_ssi.c5
-rw-r--r--sound/soc/fsl/imx-ssi.c19
-rw-r--r--sound/soc/generic/simple-card.c8
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c17
-rw-r--r--sound/soc/intel/boards/broadwell.c9
-rw-r--r--sound/soc/intel/haswell/sst-haswell-ipc.c20
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c33
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c1
-rw-r--r--sound/soc/kirkwood/armada-370-db.c1
-rw-r--r--sound/soc/mediatek/mt8173-max98090.c11
-rw-r--r--sound/soc/mediatek/mt8173-rt5650-rt5676.c11
-rw-r--r--sound/soc/mediatek/mtk-afe-pcm.c17
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c6
-rw-r--r--sound/soc/pxa/Kconfig2
-rw-r--r--sound/soc/pxa/brownstone.c9
-rw-r--r--sound/soc/pxa/corgi.c11
-rw-r--r--sound/soc/pxa/e740_wm9705.c5
-rw-r--r--sound/soc/pxa/e750_wm9705.c5
-rw-r--r--sound/soc/pxa/e800_wm9712.c5
-rw-r--r--sound/soc/pxa/hx4700.c4
-rw-r--r--sound/soc/pxa/imote2.c11
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c11
-rw-r--r--sound/soc/pxa/palm27x.c9
-rw-r--r--sound/soc/pxa/poodle.c11
-rw-r--r--sound/soc/pxa/pxa-ssp.c1
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c4
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c1
-rw-r--r--sound/soc/pxa/spitz.c5
-rw-r--r--sound/soc/pxa/tosa.c5
-rw-r--r--sound/soc/pxa/ttc-dkb.c12
-rw-r--r--sound/soc/qcom/lpass-cpu.c3
-rw-r--r--sound/soc/rockchip/Kconfig4
-rw-r--r--sound/soc/sh/Kconfig1
-rw-r--r--sound/soc/sh/rcar/adg.c303
-rw-r--r--sound/soc/sh/rcar/core.c12
-rw-r--r--sound/soc/sh/rcar/ctu.c6
-rw-r--r--sound/soc/sh/rcar/dvc.c6
-rw-r--r--sound/soc/sh/rcar/mix.c6
-rw-r--r--sound/soc/sh/rcar/rsnd.h15
-rw-r--r--sound/soc/sh/rcar/src.c17
-rw-r--r--sound/soc/sh/rcar/ssi.c98
-rw-r--r--sound/soc/sh/siu_dai.c85
-rw-r--r--sound/soc/soc-core.c25
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/soc/soc-ops.c28
-rw-r--r--sound/soc/soc-pcm.c49
-rw-r--r--sound/soc/soc-utils.c9
-rw-r--r--sound/soc/spear/Kconfig2
-rw-r--r--sound/soc/sti/uniperif_player.c14
-rw-r--r--sound/soc/sti/uniperif_reader.c6
-rw-r--r--sound/soc/sunxi/Kconfig11
-rw-r--r--sound/soc/sunxi/Makefile2
-rw-r--r--sound/soc/sunxi/sun4i-codec.c719
-rw-r--r--sound/soc/ux500/mop500.c1
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c1
-rw-r--r--sound/synth/emux/emux_oss.c3
95 files changed, 2526 insertions, 883 deletions
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 885683a3b0bd..e0406211716b 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -9,6 +9,14 @@ menuconfig SND_ARM
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 @@ config SND_PXA2XX_PCM
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/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 4449d1a99089..2433f7c81472 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -19,6 +19,7 @@
#include <linux/module.h>
#include <linux/slab.h>
+#include <linux/io.h>
#include <sound/hdaudio_ext.h>
MODULE_DESCRIPTION("HDA extended core");
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 37f43a1b34ef..a249d5486889 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3367,10 +3367,8 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
int dev, err;
err = snd_hda_codec_parse_pcms(codec);
- if (err < 0) {
- snd_hda_codec_reset(codec);
+ if (err < 0)
return err;
- }
/* attach a new PCM streams */
list_for_each_entry(cpcm, &codec->pcm_list_head, list) {
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 477742cb70a2..58c0aad37284 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -73,6 +73,7 @@ struct hda_tegra {
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_disconnect(struct snd_device *device)
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 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev)
/*
* 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 @@ static int hda_tegra_create(struct snd_card *card,
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 @@ static int hda_tegra_probe(struct platform_device *pdev)
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 @@ static int hda_tegra_probe(struct platform_device *pdev)
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_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 584a0343ab0c..85813de26da8 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -633,6 +633,7 @@ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11),
SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6),
SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6),
+ SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11),
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ca03c40609fc..2f0ec7c45fc7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -819,6 +819,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD),
+ SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a75b5611d1e4..16b8dcba5c12 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4188,6 +4188,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
}
}
+/* 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 @@ enum {
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 @@ static const struct hda_fixup alc269_fixups[] = {
},
[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
},
@@ -5299,6 +5306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9d947aef2c8b..def5cc8dff02 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4520,7 +4520,11 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
return err;
spec = codec->spec;
- codec->power_save_node = 1;
+ /* enable power_save_node only for new 92HD89xx chips, as it causes
+ * click noises on old 92HD73xx chips.
+ */
+ if ((codec->core.vendor_id & 0xfffffff0) != 0x111d7670)
+ codec->power_save_node = 1;
spec->linear_tone_beep = 0;
spec->gen.mixer_nid = 0x1d;
spec->have_spdif_mux = 1;
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 3e4f44434600..7ff7d88e46dd 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -61,6 +61,7 @@ source "sound/soc/sh/Kconfig"
source "sound/soc/sirf/Kconfig"
source "sound/soc/spear/Kconfig"
source "sound/soc/sti/Kconfig"
+source "sound/soc/sunxi/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
source "sound/soc/ux500/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 280ac8257d3e..8eb06db32fa0 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -41,6 +41,7 @@ obj-$(CONFIG_SND_SOC) += sh/
obj-$(CONFIG_SND_SOC) += sirf/
obj-$(CONFIG_SND_SOC) += spear/
obj-$(CONFIG_SND_SOC) += sti/
+obj-$(CONFIG_SND_SOC) += sunxi/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
obj-$(CONFIG_SND_SOC) += ux500/
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index aa354e1c6ff7..1933bcd46cca 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -176,6 +176,7 @@ static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = {
{ .compatible = "atmel,asoc-wm8904", },
{ }
};
+MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids);
#endif
static struct platform_driver atmel_asoc_wm8904_driver = {
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
index 452f404abfd2..e97c32798e98 100644
--- a/sound/soc/au1x/db1000.c
+++ b/sound/soc/au1x/db1000.c
@@ -38,14 +38,7 @@ static int db1000_audio_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &db1000_ac97;
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1000_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1000_audio_driver = {
@@ -54,7 +47,6 @@ static struct platform_driver db1000_audio_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = db1000_audio_probe,
- .remove = db1000_audio_remove,
};
module_platform_driver(db1000_audio_driver);
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 58c3164802b8..5c73061d912a 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -129,6 +129,8 @@ static struct snd_soc_dai_link db1300_i2s_dai = {
.cpu_dai_name = "au1xpsc_i2s.2",
.platform_name = "au1xpsc-pcm.2",
.codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &db1200_i2s_wm8731_ops,
};
@@ -146,6 +148,8 @@ static struct snd_soc_dai_link db1550_i2s_dai = {
.cpu_dai_name = "au1xpsc_i2s.3",
.platform_name = "au1xpsc-pcm.3",
.codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &db1200_i2s_wm8731_ops,
};
@@ -174,14 +178,7 @@ static int db1200_audio_probe(struct platform_device *pdev)
card = db1200_cards[pid->driver_data];
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1200_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1200_audio_driver = {
@@ -191,7 +188,6 @@ static struct platform_driver db1200_audio_driver = {
},
.id_table = db1200_pids,
.probe = db1200_audio_probe,
- .remove = db1200_audio_remove,
};
module_platform_driver(db1200_audio_driver);
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 38e853add96e..0bf9d62b91a0 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -296,7 +296,6 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
{
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/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index 5bf1501e5e3c..864df2616e10 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -87,27 +87,18 @@ static int bf5xx_ad1836_driver_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "Failed to register card\n");
return ret;
}
-static int bf5xx_ad1836_driver_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver bf5xx_ad1836_driver = {
.driver = {
.name = "bfin-snd-ad1836",
.pm = &snd_soc_pm_ops,
},
.probe = bf5xx_ad1836_driver_probe,
- .remove = bf5xx_ad1836_driver_remove,
};
module_platform_driver(bf5xx_ad1836_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
index 523baf5820d7..72ac78988426 100644
--- a/sound/soc/blackfin/bfin-eval-adau1373.c
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -154,16 +154,7 @@ static int bfin_eval_adau1373_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1373);
-}
-
-static int bfin_eval_adau1373_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373);
}
static struct platform_driver bfin_eval_adau1373_driver = {
@@ -172,7 +163,6 @@ static struct platform_driver bfin_eval_adau1373_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1373_probe,
- .remove = bfin_eval_adau1373_remove,
};
module_platform_driver(bfin_eval_adau1373_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
index f9e926dfd4ef..5c67f72cf9a9 100644
--- a/sound/soc/blackfin/bfin-eval-adau1701.c
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -94,16 +94,7 @@ static int bfin_eval_adau1701_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1701);
-}
-
-static int bfin_eval_adau1701_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701);
}
static struct platform_driver bfin_eval_adau1701_driver = {
@@ -112,7 +103,6 @@ static struct platform_driver bfin_eval_adau1701_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1701_probe,
- .remove = bfin_eval_adau1701_remove,
};
module_platform_driver(bfin_eval_adau1701_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
index 27eee66afdb2..1037477d10b2 100644
--- a/sound/soc/blackfin/bfin-eval-adav80x.c
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -119,16 +119,7 @@ static int bfin_eval_adav80x_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adav80x);
-}
-
-static int bfin_eval_adav80x_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x);
}
static const struct platform_device_id bfin_eval_adav80x_ids[] = {
@@ -144,7 +135,6 @@ static struct platform_driver bfin_eval_adav80x_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adav80x_probe,
- .remove = bfin_eval_adav80x_remove,
.id_table = bfin_eval_adav80x_ids,
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0c9733ecd17f..70e5a75901aa 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
select SND_SOC_AK4554
+ select SND_SOC_AK4613 if I2C
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
@@ -79,7 +80,6 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX9877 if I2C
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
- select SND_SOC_HDMI_CODEC
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
@@ -319,6 +319,10 @@ config SND_SOC_AK4535
config SND_SOC_AK4554
tristate "AKM AK4554 CODEC"
+config SND_SOC_AK4613
+ tristate "AKM AK4613 CODEC"
+ depends on I2C
+
config SND_SOC_AK4641
tristate
@@ -442,9 +446,6 @@ config SND_SOC_BT_SCO
config SND_SOC_DMIC
tristate
-config SND_SOC_HDMI_CODEC
- tristate "HDMI stub CODEC"
-
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4a32077954ae..be1491acb6ae 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4554-objs := ak4554.o
+snd-soc-ak4613-objs := ak4613.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
@@ -72,7 +73,6 @@ snd-soc-max98925-objs := max98925.o
snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
-snd-soc-hdmi-codec-objs := hdmi.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
@@ -216,6 +216,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o
+obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o
obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
@@ -264,7 +265,6 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
-obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
new file mode 100644
index 000000000000..07a266460ec3
--- /dev/null
+++ b/sound/soc/codecs/ak4613.c
@@ -0,0 +1,497 @@
+/*
+ * ak4613.c -- Asahi Kasei ALSA Soc Audio driver
+ *
+ * Copyright (C) 2015 Renesas Electronics Corporation
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on ak4642.c by Kuninori Morimoto
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#define PW_MGMT1 0x00 /* Power Management 1 */
+#define PW_MGMT2 0x01 /* Power Management 2 */
+#define PW_MGMT3 0x02 /* Power Management 3 */
+#define CTRL1 0x03 /* Control 1 */
+#define CTRL2 0x04 /* Control 2 */
+#define DEMP1 0x05 /* De-emphasis1 */
+#define DEMP2 0x06 /* De-emphasis2 */
+#define OFD 0x07 /* Overflow Detect */
+#define ZRD 0x08 /* Zero Detect */
+#define ICTRL 0x09 /* Input Control */
+#define OCTRL 0x0a /* Output Control */
+#define LOUT1 0x0b /* LOUT1 Volume Control */
+#define ROUT1 0x0c /* ROUT1 Volume Control */
+#define LOUT2 0x0d /* LOUT2 Volume Control */
+#define ROUT2 0x0e /* ROUT2 Volume Control */
+#define LOUT3 0x0f /* LOUT3 Volume Control */
+#define ROUT3 0x10 /* ROUT3 Volume Control */
+#define LOUT4 0x11 /* LOUT4 Volume Control */
+#define ROUT4 0x12 /* ROUT4 Volume Control */
+#define LOUT5 0x13 /* LOUT5 Volume Control */
+#define ROUT5 0x14 /* ROUT5 Volume Control */
+#define LOUT6 0x15 /* LOUT6 Volume Control */
+#define ROUT6 0x16 /* ROUT6 Volume Control */
+
+/* PW_MGMT1 */
+#define RSTN BIT(0)
+#define PMDAC BIT(1)
+#define PMADC BIT(2)
+#define PMVR BIT(3)
+
+/* PW_MGMT2 */
+#define PMAD_ALL 0x7
+
+/* PW_MGMT3 */
+#define PMDA_ALL 0x3f
+
+/* CTRL1 */
+#define DIF0 BIT(3)
+#define DIF1 BIT(4)
+#define DIF2 BIT(5)
+#define TDM0 BIT(6)
+#define TDM1 BIT(7)
+#define NO_FMT (0xff)
+#define FMT_MASK (0xf8)
+
+/* CTRL2 */
+#define DFS_NORMAL_SPEED (0 << 2)
+#define DFS_DOUBLE_SPEED (1 << 2)
+#define DFS_QUAD_SPEED (2 << 2)
+
+struct ak4613_priv {
+ struct mutex lock;
+
+ unsigned int fmt;
+ u8 fmt_ctrl;
+ int cnt;
+};
+
+struct ak4613_formats {
+ unsigned int width;
+ unsigned int fmt;
+};
+
+struct ak4613_interface {
+ struct ak4613_formats capture;
+ struct ak4613_formats playback;
+};
+
+/*
+ * Playback Volume
+ *
+ * max : 0x00 : 0 dB
+ * ( 0.5 dB step )
+ * min : 0xFE : -127.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new ak4613_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6,
+ 0, 0xFF, 1, out_tlv),
+};
+
+static const struct reg_default ak4613_reg[] = {
+ { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 },
+ { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 },
+ { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 },
+ { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 },
+ { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 },
+ { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
+};
+
+#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3)
+#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt }
+static const struct ak4613_interface ak4613_iface[] = {
+ /* capture */ /* playback */
+ [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) },
+ [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) },
+ [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) },
+ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) },
+ [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) },
+};
+
+static const struct regmap_config ak4613_regmap_cfg = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = 0x16,
+ .reg_defaults = ak4613_reg,
+ .num_reg_defaults = ARRAY_SIZE(ak4613_reg),
+};
+
+static const struct of_device_id ak4613_of_match[] = {
+ { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg },
+ {},
+};
+MODULE_DEVICE_TABLE(of, ak4613_of_match);
+
+static const struct i2c_device_id ak4613_i2c_id[] = {
+ { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id);
+
+static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = {
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("LOUT3"),
+ SND_SOC_DAPM_OUTPUT("LOUT4"),
+ SND_SOC_DAPM_OUTPUT("LOUT5"),
+ SND_SOC_DAPM_OUTPUT("LOUT6"),
+
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT3"),
+ SND_SOC_DAPM_OUTPUT("ROUT4"),
+ SND_SOC_DAPM_OUTPUT("ROUT5"),
+ SND_SOC_DAPM_OUTPUT("ROUT6"),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("LIN1"),
+ SND_SOC_DAPM_INPUT("LIN2"),
+
+ SND_SOC_DAPM_INPUT("RIN1"),
+ SND_SOC_DAPM_INPUT("RIN2"),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0),
+ SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0),
+ SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0),
+ SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0),
+ SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0),
+ SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0),
+ SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0),
+};
+
+static const struct snd_soc_dapm_route ak4613_intercon[] = {
+ {"LOUT1", NULL, "DAC1"},
+ {"LOUT2", NULL, "DAC2"},
+ {"LOUT3", NULL, "DAC3"},
+ {"LOUT4", NULL, "DAC4"},
+ {"LOUT5", NULL, "DAC5"},
+ {"LOUT6", NULL, "DAC6"},
+
+ {"ROUT1", NULL, "DAC1"},
+ {"ROUT2", NULL, "DAC2"},
+ {"ROUT3", NULL, "DAC3"},
+ {"ROUT4", NULL, "DAC4"},
+ {"ROUT5", NULL, "DAC5"},
+ {"ROUT6", NULL, "DAC6"},
+
+ {"DAC1", NULL, "Playback"},
+ {"DAC2", NULL, "Playback"},
+ {"DAC3", NULL, "Playback"},
+ {"DAC4", NULL, "Playback"},
+ {"DAC5", NULL, "Playback"},
+ {"DAC6", NULL, "Playback"},
+
+ {"Capture", NULL, "ADC1"},
+ {"Capture", NULL, "ADC2"},
+
+ {"ADC1", NULL, "LIN1"},
+ {"ADC2", NULL, "LIN2"},
+
+ {"ADC1", NULL, "RIN1"},
+ {"ADC2", NULL, "RIN2"},
+};
+
+static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+
+ mutex_lock(&priv->lock);
+ priv->cnt--;
+ if (priv->cnt < 0) {
+ dev_err(dev, "unexpected counter error\n");
+ priv->cnt = 0;
+ }
+ if (!priv->cnt)
+ priv->fmt_ctrl = NO_FMT;
+ mutex_unlock(&priv->lock);
+}
+
+static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ fmt &= SND_SOC_DAIFMT_FORMAT_MASK;
+
+ switch (fmt) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_I2S:
+ priv->fmt = fmt;
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ const struct ak4613_formats *fmts;
+ struct device *dev = codec->dev;
+ unsigned int width = params_width(params);
+ unsigned int fmt = priv->fmt;
+ unsigned int rate;
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ int i, ret;
+ u8 fmt_ctrl, ctrl2;
+
+ rate = params_rate(params);
+ switch (rate) {
+ case 32000:
+ case 44100:
+ case 48000:
+ ctrl2 = DFS_NORMAL_SPEED;
+ break;
+ case 88200:
+ case 96000:
+ ctrl2 = DFS_DOUBLE_SPEED;
+ break;
+ case 176400:
+ case 192000:
+ ctrl2 = DFS_QUAD_SPEED;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /*
+ * FIXME
+ *
+ * It doesn't support TDM at this point
+ */
+ fmt_ctrl = NO_FMT;
+ for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) {
+ fmts = (is_play) ? &ak4613_iface[i].playback :
+ &ak4613_iface[i].capture;
+
+ if (fmts->fmt != fmt)
+ continue;
+
+ if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
+ if (fmts->width != width)
+ continue;
+ } else {
+ if (fmts->width < width)
+ continue;
+ }
+
+ fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i);
+ break;
+ }
+
+ ret = -EINVAL;
+ if (fmt_ctrl == NO_FMT)
+ goto hw_params_end;
+
+ mutex_lock(&priv->lock);
+ if ((priv->fmt_ctrl == NO_FMT) ||
+ (priv->fmt_ctrl == fmt_ctrl)) {
+ priv->fmt_ctrl = fmt_ctrl;
+ priv->cnt++;
+ ret = 0;
+ }
+ mutex_unlock(&priv->lock);
+
+ if (ret < 0)
+ goto hw_params_end;
+
+ snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl);
+ snd_soc_write(codec, CTRL2, ctrl2);
+
+hw_params_end:
+ if (ret < 0)
+ dev_warn(dev, "unsupported data width/format combination\n");
+
+ return ret;
+}
+
+static int ak4613_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 mgmt1 = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ mgmt1 |= RSTN;
+ /* fall through */
+ case SND_SOC_BIAS_PREPARE:
+ mgmt1 |= PMADC | PMDAC;
+ /* fall through */
+ case SND_SOC_BIAS_STANDBY:
+ mgmt1 |= PMVR;
+ /* fall through */
+ case SND_SOC_BIAS_OFF:
+ default:
+ break;
+ }
+
+ snd_soc_write(codec, PW_MGMT1, mgmt1);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops ak4613_dai_ops = {
+ .shutdown = ak4613_dai_shutdown,
+ .set_fmt = ak4613_dai_set_fmt,
+ .hw_params = ak4613_dai_hw_params,
+};
+
+#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_64000 |\
+ SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000 |\
+ SNDRV_PCM_RATE_176400 |\
+ SNDRV_PCM_RATE_192000)
+#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver ak4613_dai = {
+ .name = "ak4613-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .ops = &ak4613_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+ struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+ regcache_mark_dirty(regmap);
+ return regcache_sync(regmap);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+ .resume = ak4613_resume,
+ .set_bias_level = ak4613_set_bias_level,
+ .controls = ak4613_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4613_snd_controls),
+ .dapm_widgets = ak4613_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets),
+ .dapm_routes = ak4613_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ak4613_intercon),
+};
+
+static int ak4613_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
+ const struct regmap_config *regmap_cfg;
+ struct regmap *regmap;
+ struct ak4613_priv *priv;
+
+ regmap_cfg = NULL;
+ if (np) {
+ const struct of_device_id *of_id;
+
+ of_id = of_match_device(ak4613_of_match, dev);
+ if (of_id)
+ regmap_cfg = of_id->data;
+ } else {
+ regmap_cfg = (const struct regmap_config *)id->driver_data;
+ }
+
+ if (!regmap_cfg)
+ return -EINVAL;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->fmt_ctrl = NO_FMT;
+ priv->cnt = 0;
+
+ mutex_init(&priv->lock);
+
+ i2c_set_clientdata(i2c, priv);
+
+ regmap = devm_regmap_init_i2c(i2c, regmap_cfg);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ return snd_soc_register_codec(dev, &soc_codec_dev_ak4613,
+ &ak4613_dai, 1);
+}
+
+static int ak4613_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver ak4613_i2c_driver = {
+ .driver = {
+ .name = "ak4613-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = ak4613_of_match,
+ },
+ .probe = ak4613_i2c_probe,
+ .remove = ak4613_i2c_remove,
+ .id_table = ak4613_i2c_id,
+};
+
+module_i2c_driver(ak4613_i2c_driver);
+
+MODULE_DESCRIPTION("Soc AK4613 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4a90143d0e90..cda27c22812a 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -23,6 +23,8 @@
* AK4648 is tested.
*/
+#include <linux/clk.h>
+#include <linux/clk-provider.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
@@ -128,11 +130,8 @@
#define I2S (3 << 0)
/* MD_CTL2 */
-#define FS0 (1 << 0)
-#define FS1 (1 << 1)
-#define FS2 (1 << 2)
-#define FS3 (1 << 5)
-#define FS_MASK (FS0 | FS1 | FS2 | FS3)
+#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2))
+#define PSs(val) ((val & 0x3) << 6)
/* MD_CTL3 */
#define BST1 (1 << 3)
@@ -147,6 +146,7 @@ struct ak4642_drvdata {
struct ak4642_priv {
const struct ak4642_drvdata *drvdata;
+ struct clk *mcko;
};
/*
@@ -430,56 +430,56 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
+static int ak4642_set_mcko(struct snd_soc_codec *codec,
+ u32 frequency)
+{
+ u32 fs_list[] = {
+ [0] = 8000,
+ [1] = 12000,
+ [2] = 16000,
+ [3] = 24000,
+ [4] = 7350,
+ [5] = 11025,
+ [6] = 14700,
+ [7] = 22050,
+ [10] = 32000,
+ [11] = 48000,
+ [14] = 29400,
+ [15] = 44100,
+ };
+ u32 ps_list[] = {
+ [0] = 256,
+ [1] = 128,
+ [2] = 64,
+ [3] = 32
+ };
+ int ps, fs;
+
+ for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
+ for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
+ if (frequency == ps_list[ps] * fs_list[fs]) {
+ snd_soc_write(codec, MD_CTL2,
+ PSs(ps) | FSs(fs));
+ return 0;
+ }
+ }
+ }
+
+ return 0;
+}
+
static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
- u8 rate;
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+ u32 rate = clk_get_rate(priv->mcko);
- switch (params_rate(params)) {
- case 7350:
- rate = FS2;
- break;
- case 8000:
- rate = 0;
- break;
- case 11025:
- rate = FS2 | FS0;
- break;
- case 12000:
- rate = FS0;
- break;
- case 14700:
- rate = FS2 | FS1;
- break;
- case 16000:
- rate = FS1;
- break;
- case 22050:
- rate = FS2 | FS1 | FS0;
- break;
- case 24000:
- rate = FS1 | FS0;
- break;
- case 29400:
- rate = FS3 | FS2 | FS1;
- break;
- case 32000:
- rate = FS3 | FS1;
- break;
- case 44100:
- rate = FS3 | FS2 | FS1 | FS0;
- break;
- case 48000:
- rate = FS3 | FS1 | FS0;
- break;
- default:
- return -EINVAL;
- }
- snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
+ if (!rate)
+ rate = params_rate(params) * 256;
- return 0;
+ return ak4642_set_mcko(codec, rate);
}
static int ak4642_set_bias_level(struct snd_soc_codec *codec,
@@ -532,7 +532,18 @@ static int ak4642_resume(struct snd_soc_codec *codec)
return 0;
}
+static int ak4642_probe(struct snd_soc_codec *codec)
+{
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (priv->mcko)
+ ak4642_set_mcko(codec, clk_get_rate(priv->mcko));
+
+ return 0;
+}
+
static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+ .probe = ak4642_probe,
.resume = ak4642_resume,
.set_bias_level = ak4642_set_bias_level,
.controls = ak4642_snd_controls,
@@ -580,19 +591,54 @@ static const struct ak4642_drvdata ak4648_drvdata = {
.extended_frequencies = 1,
};
+#ifdef CONFIG_COMMON_CLK
+static struct clk *ak4642_of_parse_mcko(struct device *dev)
+{
+ struct device_node *np = dev->of_node;
+ struct clk *clk;
+ const char *clk_name = np->name;
+ const char *parent_clk_name = NULL;
+ u32 rate;
+
+ if (of_property_read_u32(np, "clock-frequency", &rate))
+ return NULL;
+
+ if (of_property_read_bool(np, "clocks"))
+ parent_clk_name = of_clk_get_parent_name(np, 0);
+
+ of_property_read_string(np, "clock-output-names", &clk_name);
+
+ clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
+ (parent_clk_name) ? 0 : CLK_IS_ROOT,
+ rate);
+ if (!IS_ERR(clk))
+ of_clk_add_provider(np, of_clk_src_simple_get, clk);
+
+ return clk;
+}
+#else
+#define ak4642_of_parse_mcko(d) 0
+#endif
+
static const struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
- struct device_node *np = i2c->dev.of_node;
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
const struct ak4642_drvdata *drvdata = NULL;
struct regmap *regmap;
struct ak4642_priv *priv;
+ struct clk *mcko = NULL;
if (np) {
const struct of_device_id *of_id;
- of_id = of_match_device(ak4642_of_match, &i2c->dev);
+ mcko = ak4642_of_parse_mcko(dev);
+ if (IS_ERR(mcko))
+ mcko = NULL;
+
+ of_id = of_match_device(ak4642_of_match, dev);
if (of_id)
drvdata = of_id->data;
} else {
@@ -600,15 +646,16 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
}
if (!drvdata) {
- dev_err(&i2c->dev, "Unknown device type\n");
+ dev_err(dev, "Unknown device type\n");
return -EINVAL;
}
- priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
priv->drvdata = drvdata;
+ priv->mcko = mcko;
i2c_set_clientdata(i2c, priv);
@@ -616,7 +663,7 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
if (IS_ERR(regmap))
return PTR_ERR(regmap);
- return snd_soc_register_codec(&i2c->dev,
+ return snd_soc_register_codec(dev,
&soc_codec_dev_ak4642, &ak4642_dai, 1);
}
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8a2221ab3d10..ac21b85ff75f 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
0x4f5, 0x0da);
}
break;
+ default:
+ break;
}
return 0;
@@ -689,6 +691,15 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
ARIZONA_IN_VU, val);
}
+bool arizona_input_analog(struct snd_soc_codec *codec, int shift)
+{
+ unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8);
+ unsigned int val = snd_soc_read(codec, reg);
+
+ return !(val & ARIZONA_IN1_MODE_MASK);
+}
+EXPORT_SYMBOL_GPL(arizona_input_analog);
+
int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
int event)
{
@@ -725,6 +736,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES);
if (reg == 0)
arizona_in_set_vu(codec, 0);
+ break;
+ default:
+ break;
}
return 0;
@@ -806,6 +820,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
break;
}
break;
+ default:
+ break;
}
return 0;
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ada0a418ff4b..7b68d05a0939 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -294,4 +294,6 @@ extern int arizona_init_dai(struct arizona_priv *priv, int dai);
int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
bool diff);
+extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
+
#endif
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
deleted file mode 100644
index bd42ad34e004..000000000000
--- a/sound/soc/codecs/hdmi.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * ALSA SoC codec driver for HDMI audio codecs.
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <linux/of.h>
-#include <linux/of_device.h>
-
-#define DRV_NAME "hdmi-audio-codec"
-
-static const struct snd_soc_dapm_widget hdmi_widgets[] = {
- SND_SOC_DAPM_INPUT("RX"),
- SND_SOC_DAPM_OUTPUT("TX"),
-};
-
-static const struct snd_soc_dapm_route hdmi_routes[] = {
- { "Capture", NULL, "RX" },
- { "TX", NULL, "Playback" },
-};
-
-static struct snd_soc_dai_driver hdmi_codec_dai = {
- .name = "hdmi-hifi",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 2,
- .channels_max = 8,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
- .sig_bits = 24,
- },
- .capture = {
- .stream_name = "Capture",
- .channels_min = 2,
- .channels_max = 2,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE,
- },
-
-};
-
-#ifdef CONFIG_OF
-static const struct of_device_id hdmi_audio_codec_ids[] = {
- { .compatible = "linux,hdmi-audio", },
- { }
-};
-MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids);
-#endif
-
-static struct snd_soc_codec_driver hdmi_codec = {
- .dapm_widgets = hdmi_widgets,
- .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
- .dapm_routes = hdmi_routes,
- .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
- .ignore_pmdown_time = true,
-};
-
-static int hdmi_codec_probe(struct platform_device *pdev)
-{
- return snd_soc_register_codec(&pdev->dev, &hdmi_codec,
- &hdmi_codec_dai, 1);
-}
-
-static int hdmi_codec_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_codec(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver hdmi_codec_driver = {
- .driver = {
- .name = DRV_NAME,
- .of_match_table = of_match_ptr(hdmi_audio_codec_ids),
- },
-
- .probe = hdmi_codec_probe,
- .remove = hdmi_codec_remove,
-};
-
-module_platform_driver(hdmi_codec_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("ASoC generic HDMI codec driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index 3c2f0f8d6266..f823eb502367 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -50,24 +50,24 @@ struct rt298_priv {
};
static struct reg_default rt298_index_def[] = {
- { 0x01, 0xaaaa },
- { 0x02, 0x8aaa },
+ { 0x01, 0xa5a8 },
+ { 0x02, 0x8e95 },
{ 0x03, 0x0002 },
- { 0x04, 0xaf01 },
- { 0x08, 0x000d },
- { 0x09, 0xd810 },
- { 0x0a, 0x0120 },
+ { 0x04, 0xaf67 },
+ { 0x08, 0x200f },
+ { 0x09, 0xd010 },
+ { 0x0a, 0x0100 },
{ 0x0b, 0x0000 },
{ 0x0d, 0x2800 },
- { 0x0f, 0x0000 },
- { 0x19, 0x0a17 },
+ { 0x0f, 0x0022 },
+ { 0x19, 0x0217 },
{ 0x20, 0x0020 },
{ 0x33, 0x0208 },
{ 0x46, 0x0300 },
- { 0x49, 0x0004 },
- { 0x4f, 0x50e9 },
- { 0x50, 0x2000 },
- { 0x63, 0x2902 },
+ { 0x49, 0x4004 },
+ { 0x4f, 0x50c9 },
+ { 0x50, 0x3000 },
+ { 0x63, 0x1b02 },
{ 0x67, 0x1111 },
{ 0x68, 0x1016 },
{ 0x69, 0x273f },
@@ -1214,7 +1214,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c,
mdelay(10);
if (!rt298->pdata.gpio2_en)
- regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x4000);
+ regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x40);
else
regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 4972bf3efa91..080cc1ce3963 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv),
/* ADC Boost Volume Control */
- SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1,
+ SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1,
RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
- SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1,
- RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0,
+ SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2,
+ RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
/* I2S2 function select */
@@ -732,14 +732,14 @@ static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = {
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 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
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 +2829,15 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
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);
+ if (rt5645->pdata.jd_invert)
+ regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+ RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
} 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,
@@ -2848,6 +2850,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
snd_soc_dapm_disable_pin(dapm, "LDO2");
snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
snd_soc_dapm_sync(dapm);
+ if (rt5645->pdata.jd_invert)
+ regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+ RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR);
}
return rt5645->jack_type;
@@ -2880,8 +2885,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec,
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);
}
@@ -3205,9 +3208,42 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Celes"),
},
},
+ {
+ .ident = "Google Ultima",
+ .callback = strago_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
+ },
+ },
+ { }
+};
+
+static struct rt5645_platform_data buddy_platform_data = {
+ .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
+ .dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
+ .jd_mode = 3,
+ .jd_invert = true,
+};
+
+static int buddy_quirk_cb(const struct dmi_system_id *id)
+{
+ rt5645_pdata = &buddy_platform_data;
+
+ return 1;
+}
+
+static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = {
+ {
+ .ident = "Chrome Buddy",
+ .callback = buddy_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"),
+ },
+ },
{ }
};
+
static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev)
{
rt5645->pdata.in2_diff = device_property_read_bool(dev,
@@ -3240,7 +3276,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
if (pdata)
rt5645->pdata = *pdata;
- else if (dmi_check_system(dmi_platform_intel_braswell))
+ else if (dmi_check_system(dmi_platform_intel_braswell) ||
+ dmi_check_system(dmi_platform_intel_broadwell))
rt5645->pdata = *rt5645_pdata;
else
rt5645_parse_dt(rt5645, &i2c->dev);
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 0e4cfc6ac649..61bc8ab77646 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -39,8 +39,8 @@
#define RT5645_STO1_ADC_DIG_VOL 0x1c
#define RT5645_MONO_ADC_DIG_VOL 0x1d
#define RT5645_ADC_BST_VOL1 0x1e
-/* Mixer - D-D */
#define RT5645_ADC_BST_VOL2 0x20
+/* Mixer - D-D */
#define RT5645_STO1_ADC_MIXER 0x27
#define RT5645_MONO_ADC_MIXER 0x28
#define RT5645_AD_DA_MIXER 0x29
@@ -315,12 +315,14 @@
#define RT5645_STO1_ADC_R_BST_SFT 12
#define RT5645_STO1_ADC_COMP_MASK (0x3 << 10)
#define RT5645_STO1_ADC_COMP_SFT 10
-#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8)
-#define RT5645_STO2_ADC_L_BST_SFT 8
-#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6)
-#define RT5645_STO2_ADC_R_BST_SFT 6
-#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4)
-#define RT5645_STO2_ADC_COMP_SFT 4
+
+/* ADC Boost Volume Control (0x20) */
+#define RT5645_MONO_ADC_L_BST_MASK (0x3 << 14)
+#define RT5645_MONO_ADC_L_BST_SFT 14
+#define RT5645_MONO_ADC_R_BST_MASK (0x3 << 12)
+#define RT5645_MONO_ADC_R_BST_SFT 12
+#define RT5645_MONO_ADC_COMP_MASK (0x3 << 10)
+#define RT5645_MONO_ADC_COMP_SFT 10
/* Stereo2 ADC Mixer Control (0x26) */
#define RT5645_STO2_ADC_SRC_MASK (0x1 << 15)
@@ -777,8 +779,6 @@
#define RT5645_PWR_CLS_D_R_BIT 9
#define RT5645_PWR_CLS_D_L (0x1 << 8)
#define RT5645_PWR_CLS_D_L_BIT 8
-#define RT5645_PWR_ADC_R (0x1 << 1)
-#define RT5645_PWR_ADC_R_BIT 1
#define RT5645_PWR_DAC_L2 (0x1 << 7)
#define RT5645_PWR_DAC_L2_BIT 7
#define RT5645_PWR_DAC_R2 (0x1 << 6)
@@ -1626,6 +1626,10 @@
#define RT5645_OT_P_NOR (0x0 << 10)
#define RT5645_OT_P_INV (0x1 << 10)
#define RT5645_IRQ_JD_1_1_EN (0x1 << 9)
+#define RT5645_JD_1_1_MASK (0x1 << 7)
+#define RT5645_JD_1_1_SFT 7
+#define RT5645_JD_1_1_NOR (0x0 << 7)
+#define RT5645_JD_1_1_INV (0x1 << 7)
/* IRQ Control 2 (0xbe) */
#define RT5645_IRQ_MB1_OC_MASK (0x1 << 15)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index bfda25ef0dd4..f540f82b1f27 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
- SGTL5000_BIAS_R_MASK,
- sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
+ SGTL5000_BIAS_VOLT_MASK,
+ sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT);
/*
* disable DAP
* TODO:
@@ -1549,7 +1549,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
else {
sgtl5000->micbias_voltage = 0;
dev_err(&client->dev,
- "Unsuitable MicBias resistor\n");
+ "Unsuitable MicBias voltage\n");
}
} else {
sgtl5000->micbias_voltage = 0;
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index e3a0bca28bcf..cc1d3981fa4b 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
/*
* DAC digital volumes. From -7 to 24 dB in 1 dB steps
*/
-static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0);
+static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0);
static const char * const tas2552_din_source_select[] = {
"Muted",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 1a82b19b2644..a564759845f9 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -80,6 +80,7 @@ struct aic3x_priv {
unsigned int sysclk;
unsigned int dai_fmt;
unsigned int tdm_delay;
+ unsigned int slot_width;
struct list_head list;
int master;
int gpio_reset;
@@ -1025,10 +1026,14 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
u16 d, pll_d = 1;
int clk;
+ int width = aic3x->slot_width;
+
+ if (!width)
+ width = params_width(params);
/* select data word length */
data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
- switch (params_width(params)) {
+ switch (width) {
case 16:
break;
case 20:
@@ -1170,12 +1175,16 @@ static int aic3x_prepare(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int delay = 0;
+ int width = aic3x->slot_width;
+
+ if (!width)
+ width = substream->runtime->sample_bits;
/* TDM slot selection only valid in DSP_A/_B mode */
if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
- delay += (aic3x->tdm_delay + 1);
+ delay += (aic3x->tdm_delay*width + 1);
else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
- delay += aic3x->tdm_delay;
+ delay += aic3x->tdm_delay*width;
/* Configure data delay */
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
@@ -1296,7 +1305,20 @@ static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- aic3x->tdm_delay = lsb * slot_width;
+ switch (slot_width) {
+ case 16:
+ case 20:
+ case 24:
+ case 32:
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported slot width %d\n", slot_width);
+ return -EINVAL;
+ }
+
+
+ aic3x->tdm_delay = lsb;
+ aic3x->slot_width = slot_width;
/* DOUT in high-impedance on inactive bit clocks */
snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA,
@@ -1509,14 +1531,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
- /* Line2 to HP Bypass default volume, disconnect from Output Mixer */
- snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
- /* Line2 Line Out default volume, disconnect from Output Mixer */
- snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+ /* On tlv320aic3104, these registers are reserved and must not be written */
+ if (aic3x->model != AIC3X_MODEL_3104) {
+ /* Line2 to HP Bypass default volume, disconnect from Output Mixer */
+ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
+ /* Line2 Line Out default volume, disconnect from Output Mixer */
+ snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+ }
switch (aic3x->model) {
case AIC3X_MODEL_3X:
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index f2c6ad4b8fde..581ec1502228 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -577,7 +577,6 @@ static int wm0010_boot(struct snd_soc_codec *codec)
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 @@ static int wm0010_boot(struct snd_soc_codec *codec)
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 @@ static int wm0010_boot(struct snd_soc_codec *codec)
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 @@ static int wm0010_boot(struct snd_soc_codec *codec)
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 @@ static int wm0010_boot(struct snd_soc_codec *codec)
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/wm5110.c b/sound/soc/codecs/wm5110.c
index 9756578fc752..c04c0bc6f58a 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -38,6 +38,12 @@
struct wm5110_priv {
struct arizona_priv core;
struct arizona_fll fll[2];
+
+ unsigned int in_value;
+ int in_pre_pending;
+ int in_post_pending;
+
+ unsigned int in_pga_cache[6];
};
static const struct wm_adsp_region wm5110_dsp1_regions[] = {
@@ -428,6 +434,127 @@ err:
return ret;
}
+static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct snd_soc_card *card = dapm->card;
+ int ret;
+
+ /*
+ * PGA Volume is also used as part of the enable sequence, so
+ * usage of it should be avoided whilst that is running.
+ */
+ mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_get_volsw_range(kcontrol, ucontrol);
+
+ mutex_unlock(&card->dapm_mutex);
+
+ return ret;
+}
+
+static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct snd_soc_card *card = dapm->card;
+ int ret;
+
+ /*
+ * PGA Volume is also used as part of the enable sequence, so
+ * usage of it should be avoided whilst that is running.
+ */
+ mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_put_volsw_range(kcontrol, ucontrol);
+
+ mutex_unlock(&card->dapm_mutex);
+
+ return ret;
+}
+
+static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ unsigned int reg, mask;
+ struct reg_sequence analog_seq[] = {
+ { 0x80, 0x3 },
+ { 0x35d, 0 },
+ { 0x80, 0x0 },
+ };
+
+ reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4);
+ mask = ARIZONA_IN1L_PGA_VOL_MASK;
+
+ switch (event) {
+ case SND_SOC_DAPM_WILL_PMU:
+ wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2);
+ wm5110->in_pre_pending++;
+ wm5110->in_post_pending++;
+ return 0;
+ case SND_SOC_DAPM_PRE_PMU:
+ wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg);
+
+ snd_soc_update_bits(codec, reg, mask,
+ 0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT);
+
+ wm5110->in_pre_pending--;
+ if (wm5110->in_pre_pending == 0) {
+ analog_seq[1].def = wm5110->in_value;
+ regmap_multi_reg_write_bypassed(arizona->regmap,
+ analog_seq,
+ ARRAY_SIZE(analog_seq));
+
+ msleep(55);
+
+ wm5110->in_value = 0;
+ }
+
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, reg, mask,
+ wm5110->in_pga_cache[w->shift]);
+
+ wm5110->in_post_pending--;
+ if (wm5110->in_post_pending == 0)
+ regmap_multi_reg_write_bypassed(arizona->regmap,
+ analog_seq,
+ ARRAY_SIZE(analog_seq));
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int wm5110_in_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+
+ switch (arizona->rev) {
+ case 0 ... 4:
+ if (arizona_input_analog(codec, w->shift))
+ wm5110_in_analog_ev(w, kcontrol, event);
+
+ break;
+ default:
+ break;
+ }
+
+ return arizona_in_ev(w, kcontrol, event);
+}
+
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
@@ -454,18 +581,24 @@ SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]),
SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]),
SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]),
-SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
- ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
- ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
- ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
- ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
- ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
- ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
+ ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
+ ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
+ ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
@@ -896,29 +1029,35 @@ SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index e3b7d0c57411..dbd88408861a 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -211,28 +211,38 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
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 b4eb975da981..39ebd7bf4f53 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2944,7 +2944,8 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute)
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)
@@ -3759,7 +3760,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8962, &wm8962_dai, 1);
if (ret < 0)
- goto err_enable;
+ goto err_pm_runtime;
regcache_cache_only(wm8962->regmap, true);
@@ -3768,6 +3769,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
return 0;
+err_pm_runtime:
+ pm_runtime_disable(&i2c->dev);
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
err:
@@ -3777,6 +3780,7 @@ err:
static int wm8962_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
+ pm_runtime_disable(&client->dev);
return 0;
}
@@ -3804,6 +3808,8 @@ static int wm8962_runtime_resume(struct device *dev)
wm8962_reset(wm8962);
+ regcache_mark_dirty(wm8962->regmap);
+
/* SYSCLK defaults to on; make sure it is off so we can safely
* write to registers if the device is declocked.
*/
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index add6bb99661d..4495a40a9468 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -80,12 +80,13 @@ struct davinci_mcasp {
/* McASP specific data */
int tdm_slots;
+ u32 tdm_mask[2];
+ int slot_width;
u8 op_mode;
u8 num_serializer;
u8 *serial_dir;
u8 version;
u8 bclk_div;
- u16 bclk_lrclk_ratio;
int streams;
u32 irq_request[2];
int dma_request[2];
@@ -556,8 +557,21 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
mcasp->bclk_div = div;
break;
- case 2: /* BCLK/LRCLK ratio */
- mcasp->bclk_lrclk_ratio = div;
+ case 2: /*
+ * BCLK/LRCLK ratio descries how many bit-clock cycles
+ * fit into one frame. The clock ratio is given for a
+ * full period of data (for I2S format both left and
+ * right channels), so it has to be divided by number
+ * of tdm-slots (for I2S - divided by 2).
+ * Instead of storing this ratio, we calculate a new
+ * tdm_slot width by dividing the the ratio by the
+ * number of configured tdm slots.
+ */
+ mcasp->slot_width = div / mcasp->tdm_slots;
+ if (div % mcasp->tdm_slots)
+ dev_warn(mcasp->dev,
+ "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots",
+ __func__, div, mcasp->tdm_slots);
break;
default:
@@ -596,12 +610,92 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
return 0;
}
+/* All serializers must have equal number of channels */
+static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
+ int serializers)
+{
+ struct snd_pcm_hw_constraint_list *cl = &mcasp->chconstr[stream];
+ unsigned int *list = (unsigned int *) cl->list;
+ int slots = mcasp->tdm_slots;
+ int i, count = 0;
+
+ if (mcasp->tdm_mask[stream])
+ slots = hweight32(mcasp->tdm_mask[stream]);
+
+ for (i = 2; i <= slots; i++)
+ list[count++] = i;
+
+ for (i = 2; i <= serializers; i++)
+ list[count++] = i*slots;
+
+ cl->count = count;
+
+ return 0;
+}
+
+static int davinci_mcasp_set_ch_constraints(struct davinci_mcasp *mcasp)
+{
+ int rx_serializers = 0, tx_serializers = 0, ret, i;
+
+ for (i = 0; i < mcasp->num_serializer; i++)
+ if (mcasp->serial_dir[i] == TX_MODE)
+ tx_serializers++;
+ else if (mcasp->serial_dir[i] == RX_MODE)
+ rx_serializers++;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_PLAYBACK,
+ tx_serializers);
+ if (ret)
+ return ret;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_CAPTURE,
+ rx_serializers);
+
+ return ret;
+}
+
+
+static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask,
+ unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(mcasp->dev,
+ "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n",
+ __func__, tx_mask, rx_mask, slots, slot_width);
+
+ if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
+ dev_err(mcasp->dev,
+ "Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n",
+ tx_mask, rx_mask, slots);
+ return -EINVAL;
+ }
+
+ if (slot_width &&
+ (slot_width < 8 || slot_width > 32 || slot_width % 4 != 0)) {
+ dev_err(mcasp->dev, "%s: Unsupported slot_width %d\n",
+ __func__, slot_width);
+ return -EINVAL;
+ }
+
+ mcasp->tdm_slots = slots;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+ mcasp->slot_width = slot_width;
+
+ return davinci_mcasp_set_ch_constraints(mcasp);
+}
+
static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
- int word_length)
+ int sample_width)
{
u32 fmt;
- u32 tx_rotate = (word_length / 4) & 0x7;
- u32 mask = (1ULL << word_length) - 1;
+ u32 tx_rotate = (sample_width / 4) & 0x7;
+ u32 mask = (1ULL << sample_width) - 1;
+ u32 slot_width = sample_width;
+
/*
* For captured data we should not rotate, inversion and masking is
* enoguh to get the data to the right position:
@@ -614,28 +708,23 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
u32 rx_rotate = 0;
/*
- * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
- * callback, take it into account here. That allows us to for example
- * send 32 bits per channel to the codec, while only 16 of them carry
- * audio payload.
- * The clock ratio is given for a full period of data (for I2S format
- * both left and right channels), so it has to be divided by number of
- * tdm-slots (for I2S - divided by 2).
+ * Setting the tdm slot width either with set_clkdiv() or
+ * set_tdm_slot() allows us to for example send 32 bits per
+ * channel to the codec, while only 16 of them carry audio
+ * payload.
*/
- if (mcasp->bclk_lrclk_ratio) {
- u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
-
+ if (mcasp->slot_width) {
/*
- * When we have more bclk then it is needed for the data, we
- * need to use the rotation to move the received samples to have
- * correct alignment.
+ * When we have more bclk then it is needed for the
+ * data, we need to use the rotation to move the
+ * received samples to have correct alignment.
*/
- rx_rotate = (slot_length - word_length) / 4;
- word_length = slot_length;
+ slot_width = mcasp->slot_width;
+ rx_rotate = (slot_width - sample_width) / 4;
}
/* mapping of the XSSZ bit-field as described in the datasheet */
- fmt = (word_length >> 1) - 1;
+ fmt = (slot_width >> 1) - 1;
if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt),
@@ -663,7 +752,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
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 +834,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
* 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)
@@ -777,33 +865,50 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
/*
* If more than one serializer is needed, then use them with
- * their specified tdm_slots count. Otherwise, one serializer
- * can cope with the transaction using as many slots as channels
- * in the stream, requires channels symmetry
+ * all the specified tdm_slots. Otherwise, one serializer can
+ * cope with the transaction using just as many slots as there
+ * are channels in the stream.
*/
- active_serializers = (channels + total_slots - 1) / total_slots;
- if (active_serializers == 1)
- active_slots = channels;
- else
- active_slots = total_slots;
-
- for (i = 0; i < active_slots; i++)
- mask |= (1 << i);
+ if (mcasp->tdm_mask[stream]) {
+ active_slots = hweight32(mcasp->tdm_mask[stream]);
+ active_serializers = (channels + active_slots - 1) /
+ active_slots;
+ if (active_serializers == 1) {
+ active_slots = channels;
+ for (i = 0; i < total_slots; i++) {
+ if ((1 << i) & mcasp->tdm_mask[stream]) {
+ mask |= (1 << i);
+ if (--active_slots <= 0)
+ break;
+ }
+ }
+ }
+ } else {
+ active_serializers = (channels + total_slots - 1) / total_slots;
+ if (active_serializers == 1)
+ active_slots = channels;
+ else
+ active_slots = total_slots;
+ for (i = 0; i < active_slots; i++)
+ mask |= (1 << i);
+ }
mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
if (!mcasp->dat_port)
busel = TXSEL;
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
- FSXMOD(total_slots), FSXMOD(0x1FF));
-
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
- FSRMOD(total_slots), FSRMOD(0x1FF));
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(total_slots), FSXMOD(0x1FF));
+ } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+ FSRMOD(total_slots), FSRMOD(0x1FF));
+ }
return 0;
}
@@ -923,6 +1028,9 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
int sbits = params_width(params);
int ppm, div;
+ if (mcasp->slot_width)
+ sbits = mcasp->slot_width;
+
div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots,
&ppm);
if (ppm)
@@ -1028,6 +1136,9 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
struct snd_interval range;
int i;
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
snd_interval_any(&range);
range.empty = 1;
@@ -1070,10 +1181,14 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
if (snd_mask_test(fmt, i)) {
- uint bclk_freq = snd_pcm_format_width(i)*slots*rate;
+ uint sbits = snd_pcm_format_width(i);
int ppm;
- davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
+ davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
+ &ppm);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
snd_mask_set(&nfmt, i);
count++;
@@ -1095,6 +1210,10 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
&mcasp->ruledata[substream->stream];
u32 max_channels = 0;
int i, dir;
+ int tdm_slots = mcasp->tdm_slots;
+
+ if (mcasp->tdm_mask[substream->stream])
+ tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
mcasp->substreams[substream->stream] = substream;
@@ -1115,7 +1234,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
max_channels++;
}
ruledata->serializers = max_channels;
- max_channels *= mcasp->tdm_slots;
+ max_channels *= tdm_slots;
/*
* If the already active stream has less channels than the calculated
* limnit based on the seirializers * tdm_slots, we need to use that as
@@ -1125,15 +1244,25 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
*/
if (mcasp->channels && mcasp->channels < max_channels)
max_channels = mcasp->channels;
+ /*
+ * But we can always allow channels upto the amount of
+ * the available tdm_slots.
+ */
+ if (max_channels < tdm_slots)
+ max_channels = tdm_slots;
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
2, max_channels);
- if (mcasp->chconstr[substream->stream].count)
- snd_pcm_hw_constraint_list(substream->runtime,
- 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- &mcasp->chconstr[substream->stream]);
+ snd_pcm_hw_constraint_list(substream->runtime,
+ 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &mcasp->chconstr[substream->stream]);
+
+ if (mcasp->slot_width)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ 8, mcasp->slot_width);
/*
* If we rely on implicit BCLK divider setting we should
@@ -1185,6 +1314,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
.set_fmt = davinci_mcasp_set_dai_fmt,
.set_clkdiv = davinci_mcasp_set_clkdiv,
.set_sysclk = davinci_mcasp_set_sysclk,
+ .set_tdm_slot = davinci_mcasp_set_tdm_slot,
};
static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
@@ -1299,6 +1429,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.ops = &davinci_mcasp_dai_ops,
.symmetric_samplebits = 1,
+ .symmetric_rates = 1,
},
{
.name = "davinci-mcasp.1",
@@ -1514,59 +1645,6 @@ nodata:
return pdata;
}
-/* All serializers must have equal number of channels */
-static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp,
- struct snd_pcm_hw_constraint_list *cl,
- int serializers)
-{
- unsigned int *list;
- int i, count = 0;
-
- if (serializers <= 1)
- return 0;
-
- list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
- (mcasp->tdm_slots + serializers - 2),
- GFP_KERNEL);
- if (!list)
- return -ENOMEM;
-
- for (i = 2; i <= mcasp->tdm_slots; i++)
- list[count++] = i;
-
- for (i = 2; i <= serializers; i++)
- list[count++] = i*mcasp->tdm_slots;
-
- cl->count = count;
- cl->list = list;
-
- return 0;
-}
-
-
-static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp)
-{
- int rx_serializers = 0, tx_serializers = 0, ret, i;
-
- for (i = 0; i < mcasp->num_serializer; i++)
- if (mcasp->serial_dir[i] == TX_MODE)
- tx_serializers++;
- else if (mcasp->serial_dir[i] == RX_MODE)
- rx_serializers++;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_PLAYBACK],
- tx_serializers);
- if (ret)
- return ret;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_CAPTURE],
- rx_serializers);
-
- return ret;
-}
-
enum {
PCM_EDMA,
PCM_SDMA,
@@ -1685,7 +1763,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
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 +1780,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
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 +1795,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
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,
@@ -1783,7 +1861,28 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
}
- ret = davinci_mcasp_init_ch_constraints(mcasp);
+ /* Allocate memory for long enough list for all possible
+ * scenarios. Maximum number tdm slots is 32 and there cannot
+ * be more serializers than given in the configuration. The
+ * serializer directions could be taken into account, but it
+ * would make code much more complex and save only couple of
+ * bytes.
+ */
+ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
+ !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list)
+ return -ENOMEM;
+
+ ret = davinci_mcasp_set_ch_constraints(mcasp);
if (ret)
goto err;
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index a3e97b46b64e..ba34252b7bba 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -131,23 +131,32 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
for (i = 0; i < 4; i++)
- i2s_write_reg(dev->i2s_base, TOR(i), 0);
+ i2s_read_reg(dev->i2s_base, TOR(i));
} else {
for (i = 0; i < 4; i++)
- i2s_write_reg(dev->i2s_base, ROR(i), 0);
+ i2s_read_reg(dev->i2s_base, ROR(i));
}
}
static void i2s_start(struct dw_i2s_dev *dev,
struct snd_pcm_substream *substream)
{
-
+ u32 i, irq;
i2s_write_reg(dev->i2s_base, IER, 1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
+ }
i2s_write_reg(dev->i2s_base, ITER, 1);
- else
+ } else {
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
+ }
i2s_write_reg(dev->i2s_base, IRER, 1);
+ }
i2s_write_reg(dev->i2s_base, CER, 1);
}
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5aeb6ed4827e..0901d5e20df2 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
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 */
@@ -592,6 +593,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-wm8960", },
{}
};
+MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
static struct platform_driver fsl_asoc_card_driver = {
.probe = fsl_asoc_card_probe,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a18fd92c4a85..9366b5a42e1d 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -801,6 +801,7 @@ static const struct of_device_id fsl_sai_ids[] = {
{ .compatible = "fsl,imx6sx-sai", },
{ /* sentinel */ }
};
+MODULE_DEVICE_TABLE(of, fsl_sai_ids);
static struct platform_driver fsl_sai_driver = {
.probe = fsl_sai_probe,
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 8ec6fb208ea0..37c5cd4d0e59 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
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 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
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/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 48b2d24dd1f0..b95132e2f9dc 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -95,7 +95,8 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
/* data on rising edge of bclk, frame low 1clk before data */
- strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSI |
+ SSI_STCR_TEFS;
scr |= SSI_SCR_NET;
if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) {
scr &= ~SSI_I2S_MODE_MASK;
@@ -104,33 +105,31 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_LEFT_J:
/* data on rising edge of bclk, frame high with data */
- strcr |= SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP;
break;
case SND_SOC_DAIFMT_DSP_B:
/* data on rising edge of bclk, frame high with data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL;
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL |
+ SSI_STCR_TEFS;
break;
}
/* DAI clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_IF:
- strcr |= SSI_STCR_TFSI;
- strcr &= ~SSI_STCR_TSCKP;
+ strcr ^= SSI_STCR_TSCKP | SSI_STCR_TFSI;
break;
case SND_SOC_DAIFMT_IB_NF:
- strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI);
+ strcr ^= SSI_STCR_TSCKP;
break;
case SND_SOC_DAIFMT_NB_IF:
- strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
+ strcr ^= SSI_STCR_TFSI;
break;
case SND_SOC_DAIFMT_NB_NF:
- strcr &= ~SSI_STCR_TFSI;
- strcr |= SSI_STCR_TSCKP;
break;
}
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 3ff76d419436..54c33204541f 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -151,7 +151,9 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
}
if (set->slots) {
- ret = snd_soc_dai_set_tdm_slot(dai, 0, 0,
+ ret = snd_soc_dai_set_tdm_slot(dai,
+ set->tx_slot_mask,
+ set->rx_slot_mask,
set->slots,
set->slot_width);
if (ret && ret != -ENOTSUPP) {
@@ -243,7 +245,9 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
return ret;
/* Parse TDM slot */
- ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
+ ret = snd_soc_of_parse_tdm_slot(np, &dai->tx_slot_mask,
+ &dai->rx_slot_mask,
+ &dai->slots, &dai->slot_width);
if (ret)
return ret;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 4c734f5aeb8b..0487cfaac538 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -368,23 +368,6 @@ static void sst_media_close(struct snd_pcm_substream *substream,
kfree(stream);
}
-static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai,
- struct snd_pcm_substream *substream)
-{
- struct sst_data *sst = snd_soc_dai_get_drvdata(dai);
- struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
- struct sst_runtime_stream *stream =
- substream->runtime->private_data;
- u32 str_id = stream->stream_info.str_id;
- unsigned int pipe_id;
-
- pipe_id = map[str_id].device_id;
-
- dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n",
- pipe_id, str_id);
- return pipe_id;
-}
-
static int sst_media_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 8bafaf6ceab1..3f8a1e10bed0 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -266,18 +266,11 @@ static int broadwell_audio_probe(struct platform_device *pdev)
{
broadwell_rt286.dev = &pdev->dev;
- return snd_soc_register_card(&broadwell_rt286);
-}
-
-static int broadwell_audio_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&broadwell_rt286);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286);
}
static struct platform_driver broadwell_audio = {
.probe = broadwell_audio_probe,
- .remove = broadwell_audio_remove,
.driver = {
.name = "broadwell-audio",
},
diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index f6efa9d4acad..b27f25f70730 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 {
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 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
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 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw)
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/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 7d617bf493bc..bea26730873c 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -510,17 +510,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
},
},
{
- .name = "DMIC23 Pin",
- .ops = &skl_dmic_dai_ops,
- .capture = {
- .stream_name = "DMIC23 Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
-},
-{
.name = "HD-Codec Pin",
.ops = &skl_link_dai_ops,
.playback = {
@@ -538,28 +527,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
-{
- .name = "HD-Codec-SPK Pin",
- .ops = &skl_link_dai_ops,
- .playback = {
- .stream_name = "HD-Codec-SPK Tx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
-{
- .name = "HD-Codec-AMIC Pin",
- .ops = &skl_link_dai_ops,
- .capture = {
- .stream_name = "HD-Codec-AMIC Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
};
static int skl_platform_open(struct snd_pcm_substream *substream)
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index b05fb1c1a848..794a3499e567 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -485,6 +485,7 @@ static const struct of_device_id jz4740_of_matches[] = {
{ .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 },
{ /* sentinel */ }
};
+MODULE_DEVICE_TABLE(of, jz4740_of_matches);
#endif
static int jz4740_i2s_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
index de7563bdc5c2..e0304d544f26 100644
--- a/sound/soc/kirkwood/armada-370-db.c
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -130,6 +130,7 @@ static const struct of_device_id a370db_dt_ids[] = {
{ .compatible = "marvell,a370db-audio" },
{ },
};
+MODULE_DEVICE_TABLE(of, a370db_dt_ids);
static struct platform_driver a370db_driver = {
.driver = {
diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c
index 684e8a78bed0..71a1a35047ba 100644
--- a/sound/soc/mediatek/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173-max98090.c
@@ -179,21 +179,13 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
}
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
-static int mt8173_max98090_dev_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static const struct of_device_id mt8173_max98090_dt_match[] = {
{ .compatible = "mediatek,mt8173-max98090", },
{ }
@@ -209,7 +201,6 @@ static struct platform_driver mt8173_max98090_driver = {
#endif
},
.probe = mt8173_max98090_dev_probe,
- .remove = mt8173_max98090_dev_remove,
};
module_platform_driver(mt8173_max98090_driver);
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 86cf9752f18a..50ba538eccb3 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -246,21 +246,13 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
-static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = {
{ .compatible = "mediatek,mt8173-rt5650-rt5676", },
{ }
@@ -276,7 +268,6 @@ static struct platform_driver mt8173_rt5650_rt5676_driver = {
#endif
},
.probe = mt8173_rt5650_rt5676_dev_probe,
- .remove = mt8173_rt5650_rt5676_dev_remove,
};
module_platform_driver(mt8173_rt5650_rt5676_driver);
diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index d190fe017559..f5baf3c38863 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -549,6 +549,23 @@ static int mtk_afe_dais_startup(struct snd_pcm_substream *substream,
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/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 6e6fce6a14ba..2b23ffbac6b1 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
ret);
@@ -154,12 +154,8 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
static int mxs_sgtl5000_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
mxs_saif_put_mclk(0);
- snd_soc_unregister_card(card);
-
return 0;
}
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 39cea80846c3..f2bf8661dd21 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_AC97
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/brownstone.c b/sound/soc/pxa/brownstone.c
index 2b26318bc200..6147e86e9b0f 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -116,26 +116,19 @@ static int brownstone_probe(struct platform_device *pdev)
int ret;
brownstone.dev = &pdev->dev;
- ret = snd_soc_register_card(&brownstone);
+ ret = devm_snd_soc_register_card(&pdev->dev, &brownstone);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int brownstone_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&brownstone);
- return 0;
-}
-
static struct platform_driver mmp_driver = {
.driver = {
.name = "brownstone-audio",
.pm = &snd_soc_pm_ops,
},
.probe = brownstone_probe,
- .remove = brownstone_remove,
};
module_platform_driver(mmp_driver);
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 3580d10c9f28..c97dc13d3608 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -295,28 +295,19 @@ static int corgi_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int corgi_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver corgi_driver = {
.driver = {
.name = "corgi-audio",
.pm = &snd_soc_pm_ops,
},
.probe = corgi_probe,
- .remove = corgi_remove,
};
module_platform_driver(corgi_driver);
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index d72e124a3676..1de876529aa1 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -138,7 +138,7 @@ static int e740_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -149,10 +149,7 @@ static int e740_probe(struct platform_device *pdev)
static int e740_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 48f2d7c2e68c..b7eb7cd5df7d 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -120,7 +120,7 @@ static int e750_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -131,10 +131,7 @@ static int e750_probe(struct platform_device *pdev)
static int e750_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 45d4bd46fff6..41bf71466a7b 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -119,7 +119,7 @@ static int e800_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -130,10 +130,7 @@ static int e800_probe(struct platform_device *pdev)
static int e800_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 9f8be7cd567e..ecbf2873b7ff 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -193,7 +193,7 @@ static int hx4700_audio_probe(struct platform_device *pdev)
return ret;
snd_soc_card_hx4700.dev = &pdev->dev;
- ret = snd_soc_register_card(&snd_soc_card_hx4700);
+ ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700);
if (ret)
gpio_free_array(hx4700_audio_gpios,
ARRAY_SIZE(hx4700_audio_gpios));
@@ -203,8 +203,6 @@ static int hx4700_audio_probe(struct platform_device *pdev)
static int hx4700_audio_remove(struct platform_device *pdev)
{
- snd_soc_unregister_card(&snd_soc_card_hx4700);
-
gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index 29fabbfd21f1..9d0e40771ef5 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -72,28 +72,19 @@ static int imote2_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int imote2_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver imote2_driver = {
.driver = {
.name = "imote2-audio",
.pm = &snd_soc_pm_ops,
},
.probe = imote2_probe,
- .remove = imote2_remove,
};
module_platform_driver(imote2_driver);
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index a9615a574546..29bc60e85e92 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -181,7 +181,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev)
return -ENODEV;
mioa701.dev = &pdev->dev;
- rc = snd_soc_register_card(&mioa701);
+ rc = devm_snd_soc_register_card(&pdev->dev, &mioa701);
if (!rc)
dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
"lead to overheating and possible destruction of your device."
@@ -189,17 +189,8 @@ static int mioa701_wm9713_probe(struct platform_device *pdev)
return rc;
}
-static int mioa701_wm9713_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver mioa701_wm9713_driver = {
.probe = mioa701_wm9713_probe,
- .remove = mioa701_wm9713_remove,
.driver = {
.name = "mioa701-wm9713",
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index c20bbc042425..4e74d9573f03 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -140,22 +140,15 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
palm27x_asoc.dev = &pdev->dev;
- ret = snd_soc_register_card(&palm27x_asoc);
+ ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int palm27x_asoc_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&palm27x_asoc);
- return 0;
-}
-
static struct platform_driver palm27x_wm9712_driver = {
.probe = palm27x_asoc_probe,
- .remove = palm27x_asoc_remove,
.driver = {
.name = "palm27x-asoc",
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 80b457ac522a..84d0e2e50808 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -267,28 +267,19 @@ static int poodle_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int poodle_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver poodle_driver = {
.driver = {
.name = "poodle-audio",
.pm = &snd_soc_pm_ops,
},
.probe = poodle_probe,
- .remove = poodle_remove,
};
module_platform_driver(poodle_driver);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 3da485ec1de7..da03fad1b9cd 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -809,6 +809,7 @@ static const struct of_device_id pxa_ssp_of_ids[] = {
{ .compatible = "mrvl,pxa-ssp-dai" },
{}
};
+MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids);
#endif
static int asoc_ssp_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1f6054650991..9e4b04e0fbd1 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
.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 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
.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/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 831ee37d2e3e..29a3fdbb7b59 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -132,6 +132,7 @@ static const struct of_device_id snd_soc_pxa_audio_match[] = {
{ .compatible = "mrvl,pxa-pcm-audio" },
{ }
};
+MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match);
#endif
static struct platform_driver pxa_pcm_driver = {
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 461123ad5ff2..b00222620fd0 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -305,7 +305,7 @@ static int spitz_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -322,9 +322,6 @@ err1:
static int spitz_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
gpio_free(spitz_mic_gpio);
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f59f566551ef..49518dd642aa 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -233,7 +233,7 @@ static int tosa_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -244,10 +244,7 @@ static int tosa_probe(struct platform_device *pdev)
static int tosa_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free(TOSA_GPIO_L_MUTE);
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index 1753c7d9e760..65c20f779177 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -128,7 +128,7 @@ static int ttc_dkb_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -136,22 +136,12 @@ static int ttc_dkb_probe(struct platform_device *pdev)
return ret;
}
-static int ttc_dkb_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
-}
-
static struct platform_driver ttc_dkb_driver = {
.driver = {
.name = "ttc-dkb-audio",
.pm = &snd_soc_pm_ops,
},
.probe = ttc_dkb_probe,
- .remove = ttc_dkb_remove,
};
module_platform_driver(ttc_dkb_driver);
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 97bc2023f08a..e5101e0d2d37 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -438,7 +438,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev)
if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) {
dev_err(&pdev->dev,
"%s() error getting mi2s-bit-clk: %ld\n",
- __func__, PTR_ERR(drvdata->mi2s_bit_clk[i]));
+ __func__,
+ PTR_ERR(drvdata->mi2s_bit_clk[dai_id]));
return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]);
}
}
diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
index 58bae8e2cf5f..570905709d3a 100644
--- a/sound/soc/rockchip/Kconfig
+++ b/sound/soc/rockchip/Kconfig
@@ -17,7 +17,7 @@ config SND_SOC_ROCKCHIP_I2S
config SND_SOC_ROCKCHIP_MAX98090
tristate "ASoC support for Rockchip boards using a MAX98090 codec"
- depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+ depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
select SND_SOC_ROCKCHIP_I2S
select SND_SOC_MAX98090
select SND_SOC_TS3A227E
@@ -27,7 +27,7 @@ config SND_SOC_ROCKCHIP_MAX98090
config SND_SOC_ROCKCHIP_RT5645
tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec"
- depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+ depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
select SND_SOC_ROCKCHIP_I2S
select SND_SOC_RT5645
help
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 07114b0b0dc1..6ca90aaf141f 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU
config SND_SOC_RCAR
tristate "R-Car series SRU/SCU/SSIU/SSI support"
depends on DMA_OF
+ depends on COMMON_CLK
select SND_SIMPLE_CARD
select REGMAP_MMIO
help
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index fefc881dbac2..c4ebbb7a7b6f 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -7,7 +7,7 @@
* License. See the file "COPYING" in the main directory of this archive
* for more details.
*/
-#include <linux/sh_clk.h>
+#include <linux/clk-provider.h>
#include "rsnd.h"
#define CLKA 0
@@ -16,12 +16,26 @@
#define CLKI 3
#define CLKMAX 4
+#define CLKOUT 0
+#define CLKOUT1 1
+#define CLKOUT2 2
+#define CLKOUT3 3
+#define CLKOUTMAX 4
+
+#define BRRx_MASK(x) (0x3FF & x)
+
+static struct rsnd_mod_ops adg_ops = {
+ .name = "adg",
+};
+
struct rsnd_adg {
struct clk *clk[CLKMAX];
+ struct clk *clkout[CLKOUTMAX];
+ struct clk_onecell_data onecell;
+ struct rsnd_mod mod;
- int rbga_rate_for_441khz_div_6; /* RBGA */
- int rbgb_rate_for_48khz_div_6; /* RBGB */
- u32 ckr;
+ int rbga_rate_for_441khz; /* RBGA */
+ int rbgb_rate_for_48khz; /* RBGB */
};
#define for_each_rsnd_clk(pos, adg, i) \
@@ -29,8 +43,28 @@ struct rsnd_adg {
(i < CLKMAX) && \
((pos) = adg->clk[i]); \
i++)
+#define for_each_rsnd_clkout(pos, adg, i) \
+ for (i = 0; \
+ (i < CLKOUTMAX) && \
+ ((pos) = adg->clkout[i]); \
+ i++)
#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
+static u32 rsnd_adg_calculate_rbgx(unsigned long div)
+{
+ int i, ratio;
+
+ if (!div)
+ return 0;
+
+ for (i = 3; i >= 0; i--) {
+ ratio = 2 << (i * 2);
+ if (0 == (div % ratio))
+ return (u32)((i << 8) | ((div / ratio) - 1));
+ }
+
+ return ~0;
+}
static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
{
@@ -60,6 +94,9 @@ static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io)
{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
int id = rsnd_mod_id(mod);
int shift = (id % 2) ? 16 : 0;
u32 mask, val;
@@ -69,21 +106,26 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod,
val = val << shift;
mask = 0xffff << shift;
- rsnd_mod_bset(mod, CMDOUT_TIMSEL, mask, val);
+ rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val);
return 0;
}
-static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod,
+static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io,
u32 timsel)
{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
int is_play = rsnd_io_is_play(io);
- int id = rsnd_mod_id(mod);
+ int id = rsnd_mod_id(src_mod);
int shift = (id % 2) ? 16 : 0;
u32 mask, ws;
u32 in, out;
+ rsnd_mod_confirm_src(src_mod);
+
ws = rsnd_adg_ssi_ws_timing_gen2(io);
in = (is_play) ? timsel : ws;
@@ -95,37 +137,38 @@ static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod,
switch (id / 2) {
case 0:
- rsnd_mod_bset(mod, SRCIN_TIMSEL0, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL0, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out);
break;
case 1:
- rsnd_mod_bset(mod, SRCIN_TIMSEL1, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL1, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out);
break;
case 2:
- rsnd_mod_bset(mod, SRCIN_TIMSEL2, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL2, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out);
break;
case 3:
- rsnd_mod_bset(mod, SRCIN_TIMSEL3, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL3, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out);
break;
case 4:
- rsnd_mod_bset(mod, SRCIN_TIMSEL4, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL4, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out);
break;
}
return 0;
}
-int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io,
unsigned int src_rate,
unsigned int dst_rate)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
struct device *dev = rsnd_priv_to_dev(priv);
int idx, sel, div, step, ret;
u32 val, en;
@@ -134,10 +177,12 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */
clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */
clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */
- adg->rbga_rate_for_441khz_div_6,/* 0011: RBGA */
- adg->rbgb_rate_for_48khz_div_6, /* 0100: RBGB */
+ adg->rbga_rate_for_441khz, /* 0011: RBGA */
+ adg->rbgb_rate_for_48khz, /* 0100: RBGB */
};
+ rsnd_mod_confirm_src(src_mod);
+
min = ~0;
val = 0;
en = 0;
@@ -175,25 +220,27 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
return -EIO;
}
- ret = rsnd_adg_set_src_timsel_gen2(mod, io, val);
+ ret = rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
if (ret < 0) {
dev_err(dev, "timsel error\n");
return ret;
}
- rsnd_mod_bset(mod, DIV_EN, en, en);
+ rsnd_mod_bset(adg_mod, DIV_EN, en, en);
dev_dbg(dev, "convert rate %d <-> %d\n", src_rate, dst_rate);
return 0;
}
-int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io)
{
u32 val = rsnd_adg_ssi_ws_timing_gen2(io);
- return rsnd_adg_set_src_timsel_gen2(mod, io, val);
+ rsnd_mod_confirm_src(src_mod);
+
+ return rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
}
int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
@@ -202,6 +249,7 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
unsigned int dst_rate)
{
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
struct device *dev = rsnd_priv_to_dev(priv);
int idx, sel, div, shift;
u32 mask, val;
@@ -211,8 +259,8 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
clk_get_rate(adg->clk[CLKB]), /* 001: CLKB */
clk_get_rate(adg->clk[CLKC]), /* 010: CLKC */
0, /* 011: MLBCLK (not used) */
- adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */
- adg->rbgb_rate_for_48khz_div_6, /* 101: RBGB */
+ adg->rbga_rate_for_441khz, /* 100: RBGA */
+ adg->rbgb_rate_for_48khz, /* 101: RBGB */
};
/* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */
@@ -238,13 +286,13 @@ find_rate:
switch (id / 4) {
case 0:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL3, mask, val);
break;
case 1:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL4, mask, val);
break;
case 2:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL5, mask, val);
break;
}
@@ -257,12 +305,17 @@ find_rate:
return 0;
}
-static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
+static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val)
{
- int id = rsnd_mod_id(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+ int id = rsnd_mod_id(ssi_mod);
int shift = (id % 4) * 8;
u32 mask = 0xFF << shift;
+ rsnd_mod_confirm_ssi(ssi_mod);
+
val = val << shift;
/*
@@ -274,13 +327,13 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
switch (id / 4) {
case 0:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val);
break;
case 1:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val);
break;
case 2:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val);
break;
}
}
@@ -326,14 +379,14 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
}
/*
- * find 1/6 clock from BRGA/BRGB
+ * find divided clock from BRGA/BRGB
*/
- if (rate == adg->rbga_rate_for_441khz_div_6) {
+ if (rate == adg->rbga_rate_for_441khz) {
data = 0x10;
goto found_clock;
}
- if (rate == adg->rbgb_rate_for_48khz_div_6) {
+ if (rate == adg->rbgb_rate_for_48khz) {
data = 0x20;
goto found_clock;
}
@@ -342,29 +395,60 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
found_clock:
- /* see rsnd_adg_ssi_clk_init() */
- rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr);
- rsnd_mod_write(mod, BRRA, 0x00000002); /* 1/6 */
- rsnd_mod_write(mod, BRRB, 0x00000002); /* 1/6 */
-
/*
* This "mod" = "ssi" here.
* we can get "ssi id" from mod
*/
rsnd_adg_set_ssi_clk(mod, data);
- dev_dbg(dev, "ADG: ssi%d selects clk%d = %d",
- rsnd_mod_id(mod), i, rate);
+ dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod),
+ data, rate);
return 0;
}
-static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
+static void rsnd_adg_get_clkin(struct rsnd_priv *priv,
+ struct rsnd_adg *adg)
{
+ struct device *dev = rsnd_priv_to_dev(priv);
struct clk *clk;
- unsigned long rate;
- u32 ckr;
+ static const char * const clk_name[] = {
+ [CLKA] = "clk_a",
+ [CLKB] = "clk_b",
+ [CLKC] = "clk_c",
+ [CLKI] = "clk_i",
+ };
int i;
+
+ for (i = 0; i < CLKMAX; i++) {
+ clk = devm_clk_get(dev, clk_name[i]);
+ adg->clk[i] = IS_ERR(clk) ? NULL : clk;
+ }
+
+ for_each_rsnd_clk(clk, adg, i)
+ dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+}
+
+static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
+ struct rsnd_adg *adg)
+{
+ struct clk *clk;
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct device_node *np = dev->of_node;
+ u32 ckr, rbgx, rbga, rbgb;
+ u32 rate, req_rate, div;
+ uint32_t count = 0;
+ unsigned long req_48kHz_rate, req_441kHz_rate;
+ int i;
+ const char *parent_clk_name = NULL;
+ static const char * const clkout_name[] = {
+ [CLKOUT] = "audio_clkout",
+ [CLKOUT1] = "audio_clkout1",
+ [CLKOUT2] = "audio_clkout2",
+ [CLKOUT3] = "audio_clkout3",
+ };
int brg_table[] = {
[CLKA] = 0x0,
[CLKB] = 0x1,
@@ -372,19 +456,34 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
[CLKI] = 0x2,
};
+ of_property_read_u32(np, "#clock-cells", &count);
+
+ /*
+ * ADG supports BRRA/BRRB output only
+ * this means all clkout0/1/2/3 will be same rate
+ */
+ of_property_read_u32(np, "clock-frequency", &req_rate);
+ req_48kHz_rate = 0;
+ req_441kHz_rate = 0;
+ if (0 == (req_rate % 44100))
+ req_441kHz_rate = req_rate;
+ if (0 == (req_rate % 48000))
+ req_48kHz_rate = req_rate;
+
/*
* This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC
* have 44.1kHz or 48kHz base clocks for now.
*
* SSI itself can divide parent clock by 1/1 - 1/16
- * So, BRGA outputs 44.1kHz base parent clock 1/32,
- * and, BRGB outputs 48.0kHz base parent clock 1/32 here.
* see
* rsnd_adg_ssi_clk_try_start()
+ * rsnd_ssi_master_clk_start()
*/
ckr = 0;
- adg->rbga_rate_for_441khz_div_6 = 0;
- adg->rbgb_rate_for_48khz_div_6 = 0;
+ rbga = 2; /* default 1/6 */
+ rbgb = 2; /* default 1/6 */
+ adg->rbga_rate_for_441khz = 0;
+ adg->rbgb_rate_for_48khz = 0;
for_each_rsnd_clk(clk, adg, i) {
rate = clk_get_rate(clk);
@@ -392,19 +491,86 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
continue;
/* RBGA */
- if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) {
- adg->rbga_rate_for_441khz_div_6 = rate / 6;
- ckr |= brg_table[i] << 20;
+ if (!adg->rbga_rate_for_441khz && (0 == rate % 44100)) {
+ div = 6;
+ if (req_441kHz_rate)
+ div = rate / req_441kHz_rate;
+ rbgx = rsnd_adg_calculate_rbgx(div);
+ if (BRRx_MASK(rbgx) == rbgx) {
+ rbga = rbgx;
+ adg->rbga_rate_for_441khz = rate / div;
+ ckr |= brg_table[i] << 20;
+ if (req_441kHz_rate)
+ parent_clk_name = __clk_get_name(clk);
+ }
}
/* RBGB */
- if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) {
- adg->rbgb_rate_for_48khz_div_6 = rate / 6;
- ckr |= brg_table[i] << 16;
+ if (!adg->rbgb_rate_for_48khz && (0 == rate % 48000)) {
+ div = 6;
+ if (req_48kHz_rate)
+ div = rate / req_48kHz_rate;
+ rbgx = rsnd_adg_calculate_rbgx(div);
+ if (BRRx_MASK(rbgx) == rbgx) {
+ rbgb = rbgx;
+ adg->rbgb_rate_for_48khz = rate / div;
+ ckr |= brg_table[i] << 16;
+ if (req_48kHz_rate) {
+ parent_clk_name = __clk_get_name(clk);
+ ckr |= 0x80000000;
+ }
+ }
+ }
+ }
+
+ /*
+ * ADG supports BRRA/BRRB output only.
+ * this means all clkout0/1/2/3 will be * same rate
+ */
+
+ /*
+ * for clkout
+ */
+ if (!count) {
+ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT],
+ parent_clk_name,
+ (parent_clk_name) ?
+ 0 : CLK_IS_ROOT, req_rate);
+ if (!IS_ERR(clk)) {
+ adg->clkout[CLKOUT] = clk;
+ of_clk_add_provider(np, of_clk_src_simple_get, clk);
+ }
+ }
+ /*
+ * for clkout0/1/2/3
+ */
+ else {
+ for (i = 0; i < CLKOUTMAX; i++) {
+ clk = clk_register_fixed_rate(dev, clkout_name[i],
+ parent_clk_name,
+ (parent_clk_name) ?
+ 0 : CLK_IS_ROOT,
+ req_rate);
+ if (!IS_ERR(clk)) {
+ adg->onecell.clks = adg->clkout;
+ adg->onecell.clk_num = CLKOUTMAX;
+
+ adg->clkout[i] = clk;
+
+ of_clk_add_provider(np, of_clk_src_onecell_get,
+ &adg->onecell);
+ }
}
}
- adg->ckr = ckr;
+ rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr);
+ rsnd_mod_write(adg_mod, BRRA, rbga);
+ rsnd_mod_write(adg_mod, BRRB, rbgb);
+
+ for_each_rsnd_clkout(clk, adg, i)
+ dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+ dev_dbg(dev, "SSICKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
+ ckr, rbga, rbgb);
}
int rsnd_adg_probe(struct platform_device *pdev,
@@ -413,8 +579,6 @@ int rsnd_adg_probe(struct platform_device *pdev,
{
struct rsnd_adg *adg;
struct device *dev = rsnd_priv_to_dev(priv);
- struct clk *clk;
- int i;
adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
if (!adg) {
@@ -422,15 +586,16 @@ int rsnd_adg_probe(struct platform_device *pdev,
return -ENOMEM;
}
- adg->clk[CLKA] = devm_clk_get(dev, "clk_a");
- adg->clk[CLKB] = devm_clk_get(dev, "clk_b");
- adg->clk[CLKC] = devm_clk_get(dev, "clk_c");
- adg->clk[CLKI] = devm_clk_get(dev, "clk_i");
-
- for_each_rsnd_clk(clk, adg, i)
- dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+ /*
+ * ADG is special module.
+ * Use ADG mod without rsnd_mod_init() to make debug easy
+ * for rsnd_write/rsnd_read
+ */
+ adg->mod.ops = &adg_ops;
+ adg->mod.priv = priv;
- rsnd_adg_ssi_clk_init(priv, adg);
+ rsnd_adg_get_clkin(priv, adg);
+ rsnd_adg_get_clkout(priv, adg);
priv->adg = adg;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f3feed5ce9b6..eec294da81e3 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -110,6 +110,7 @@ static const struct rsnd_of_data rsnd_of_data_gen2 = {
static const struct of_device_id rsnd_of_match[] = {
{ .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 },
{ .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 },
+ { .compatible = "renesas,rcar_sound-gen3", .data = &rsnd_of_data_gen2 }, /* gen2 compatible */
{},
};
MODULE_DEVICE_TABLE(of, rsnd_of_match);
@@ -126,6 +127,17 @@ MODULE_DEVICE_TABLE(of, rsnd_of_match);
#define rsnd_info_id(priv, io, name) \
((io)->info->name - priv->info->name##_info)
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type)
+{
+ if (mod->type != type) {
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ dev_warn(dev, "%s[%d] is not your expected module\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
+ }
+}
+
/*
* rsnd_mod functions
*/
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 05498bba5874..a3e7c716e1f7 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -66,7 +66,7 @@ struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_ctu_nr(priv)))
id = 0;
- return &((struct rsnd_ctu *)(priv->ctu) + id)->mod;
+ return rsnd_mod_get((struct rsnd_ctu *)(priv->ctu) + id);
}
static void rsnd_of_parse_ctu(struct platform_device *pdev,
@@ -150,7 +150,7 @@ int rsnd_ctu_probe(struct platform_device *pdev,
ctu->info = &info->ctu_info[i];
- ret = rsnd_mod_init(priv, &ctu->mod, &rsnd_ctu_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops,
clk, RSND_MOD_CTU, i);
if (ret)
return ret;
@@ -166,6 +166,6 @@ void rsnd_ctu_remove(struct platform_device *pdev,
int i;
for_each_rsnd_ctu(ctu, priv, i) {
- rsnd_mod_quit(&ctu->mod);
+ rsnd_mod_quit(rsnd_mod_get(ctu));
}
}
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index 57796387d482..8d8eee6350c9 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -282,7 +282,7 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_dvc_nr(priv)))
id = 0;
- return &((struct rsnd_dvc *)(priv->dvc) + id)->mod;
+ return rsnd_mod_get((struct rsnd_dvc *)(priv->dvc) + id);
}
static void rsnd_of_parse_dvc(struct platform_device *pdev,
@@ -361,7 +361,7 @@ int rsnd_dvc_probe(struct platform_device *pdev,
dvc->info = &info->dvc_info[i];
- ret = rsnd_mod_init(priv, &dvc->mod, &rsnd_dvc_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops,
clk, RSND_MOD_DVC, i);
if (ret)
return ret;
@@ -377,6 +377,6 @@ void rsnd_dvc_remove(struct platform_device *pdev,
int i;
for_each_rsnd_dvc(dvc, priv, i) {
- rsnd_mod_quit(&dvc->mod);
+ rsnd_mod_quit(rsnd_mod_get(dvc));
}
}
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 0d5c102db6f5..8544403ffb26 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -99,7 +99,7 @@ struct rsnd_mod *rsnd_mix_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_mix_nr(priv)))
id = 0;
- return &((struct rsnd_mix *)(priv->mix) + id)->mod;
+ return rsnd_mod_get((struct rsnd_mix *)(priv->mix) + id);
}
static void rsnd_of_parse_mix(struct platform_device *pdev,
@@ -179,7 +179,7 @@ int rsnd_mix_probe(struct platform_device *pdev,
mix->info = &info->mix_info[i];
- ret = rsnd_mod_init(priv, &mix->mod, &rsnd_mix_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops,
clk, RSND_MOD_MIX, i);
if (ret)
return ret;
@@ -195,6 +195,6 @@ void rsnd_mix_remove(struct platform_device *pdev,
int i;
for_each_rsnd_mix(mix, priv, i) {
- rsnd_mod_quit(&mix->mod);
+ rsnd_mod_quit(rsnd_mod_get(mix));
}
}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 7a0e52b4640a..e4068d78616c 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -214,6 +214,7 @@ struct rsnd_dma {
};
#define rsnd_dma_to_dmaen(dma) (&(dma)->dma.en)
#define rsnd_dma_to_dmapp(dma) (&(dma)->dma.pp)
+#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
void rsnd_dma_start(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
void rsnd_dma_stop(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
@@ -225,8 +226,6 @@ int rsnd_dma_probe(struct platform_device *pdev,
struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node,
struct rsnd_mod *mod, char *name);
-#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
-
/*
* R-Car sound mod
*/
@@ -332,6 +331,7 @@ struct rsnd_mod {
#define rsnd_mod_id(mod) ((mod) ? (mod)->id : -1)
#define rsnd_mod_hw_start(mod) clk_enable((mod)->clk)
#define rsnd_mod_hw_stop(mod) clk_disable((mod)->clk)
+#define rsnd_mod_get(ip) (&(ip)->mod)
int rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
@@ -627,4 +627,15 @@ void rsnd_dvc_remove(struct platform_device *pdev,
struct rsnd_priv *priv);
struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id);
+#ifdef DEBUG
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type);
+#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI)
+#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC)
+#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC)
+#else
+#define rsnd_mod_confirm_ssi(mssi)
+#define rsnd_mod_confirm_src(msrc)
+#define rsnd_mod_confirm_dvc(mdvc)
+#endif
+
#endif
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 89a18e102feb..ca7a20f03c9b 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -918,11 +918,10 @@ static void rsnd_src_reconvert_update(struct rsnd_dai_stream *io,
rsnd_mod_write(mod, SRC_IFSVR, fsrate);
}
-static int rsnd_src_pcm_new(struct rsnd_mod *mod,
+static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct snd_soc_pcm_runtime *rtd)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct rsnd_src *src = rsnd_mod_to_src(mod);
int ret;
@@ -932,12 +931,6 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod,
*/
/*
- * Gen1 is not supported
- */
- if (rsnd_is_gen1(priv))
- return 0;
-
- /*
* SRC sync convert needs clock master
*/
if (!rsnd_rdai_is_clk_master(rdai))
@@ -975,7 +968,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = {
.start = rsnd_src_start_gen2,
.stop = rsnd_src_stop_gen2,
.hw_params = rsnd_src_hw_params,
- .pcm_new = rsnd_src_pcm_new,
+ .pcm_new = rsnd_src_pcm_new_gen2,
};
struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
@@ -983,7 +976,7 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_src_nr(priv)))
id = 0;
- return &((struct rsnd_src *)(priv->src) + id)->mod;
+ return rsnd_mod_get((struct rsnd_src *)(priv->src) + id);
}
static void rsnd_of_parse_src(struct platform_device *pdev,
@@ -1078,7 +1071,7 @@ int rsnd_src_probe(struct platform_device *pdev,
src->info = &info->src_info[i];
- ret = rsnd_mod_init(priv, &src->mod, ops, clk, RSND_MOD_SRC, i);
+ ret = rsnd_mod_init(priv, rsnd_mod_get(src), ops, clk, RSND_MOD_SRC, i);
if (ret)
return ret;
}
@@ -1093,6 +1086,6 @@ void rsnd_src_remove(struct platform_device *pdev,
int i;
for_each_rsnd_src(src, priv, i) {
- rsnd_mod_quit(&src->mod);
+ rsnd_mod_quit(rsnd_mod_get(src));
}
}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index d45b9a7e324e..5e05f9422073 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -128,10 +128,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
struct rsnd_priv *priv = rsnd_io_to_priv(io);
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
struct device *dev = rsnd_priv_to_dev(priv);
- int i, j, ret;
- int adg_clk_div_table[] = {
- 1, 6, /* see adg.c */
- };
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+ int j, ret;
int ssi_clk_mul_table[] = {
1, 2, 4, 8, 16, 6, 12,
};
@@ -141,28 +139,25 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
/*
* Find best clock, and try to start ADG
*/
- for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) {
- for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
-
- /*
- * this driver is assuming that
- * system word is 64fs (= 2 x 32bit)
- * see rsnd_ssi_init()
- */
- main_rate = rate / adg_clk_div_table[i]
- * 32 * 2 * ssi_clk_mul_table[j];
-
- ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate);
- if (0 == ret) {
- ssi->cr_clk = FORCE | SWL_32 |
- SCKD | SWSD | CKDV(j);
-
- dev_dbg(dev, "%s[%d] outputs %u Hz\n",
- rsnd_mod_name(&ssi->mod),
- rsnd_mod_id(&ssi->mod), rate);
-
- return 0;
- }
+ for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
+
+ /*
+ * this driver is assuming that
+ * system word is 64fs (= 2 x 32bit)
+ * see rsnd_ssi_init()
+ */
+ main_rate = rate * 32 * 2 * ssi_clk_mul_table[j];
+
+ ret = rsnd_adg_ssi_clk_try_start(mod, main_rate);
+ if (0 == ret) {
+ ssi->cr_clk = FORCE | SWL_32 |
+ SCKD | SWSD | CKDV(j);
+
+ dev_dbg(dev, "%s[%d] outputs %u Hz\n",
+ rsnd_mod_name(mod),
+ rsnd_mod_id(mod), rate);
+
+ return 0;
}
}
@@ -172,8 +167,10 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi)
{
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
ssi->cr_clk = 0;
- rsnd_adg_ssi_clk_stop(&ssi->mod);
+ rsnd_adg_ssi_clk_stop(mod);
}
static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
@@ -182,11 +179,12 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
struct rsnd_priv *priv = rsnd_io_to_priv(io);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
u32 cr_mode;
u32 cr;
if (0 == ssi->usrcnt) {
- rsnd_mod_hw_start(&ssi->mod);
+ rsnd_mod_hw_start(mod);
if (rsnd_rdai_is_clk_master(rdai)) {
struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -198,7 +196,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
}
}
- if (rsnd_ssi_is_dma_mode(&ssi->mod)) {
+ if (rsnd_ssi_is_dma_mode(mod)) {
cr_mode = UIEN | OIEN | /* over/under run */
DMEN; /* DMA : enable DMA */
} else {
@@ -210,24 +208,25 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
cr_mode |
EN;
- rsnd_mod_write(&ssi->mod, SSICR, cr);
+ rsnd_mod_write(mod, SSICR, cr);
/* enable WS continue */
if (rsnd_rdai_is_clk_master(rdai))
- rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
+ rsnd_mod_write(mod, SSIWSR, CONT);
/* clear error status */
- rsnd_mod_write(&ssi->mod, SSISR, 0);
+ rsnd_mod_write(mod, SSISR, 0);
ssi->usrcnt++;
dev_dbg(dev, "%s[%d] hw started\n",
- rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
}
static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct device *dev = rsnd_priv_to_dev(priv);
u32 cr;
@@ -247,15 +246,15 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
cr = ssi->cr_own |
ssi->cr_clk;
- rsnd_mod_write(&ssi->mod, SSICR, cr | EN);
- rsnd_ssi_status_check(&ssi->mod, DIRQ);
+ rsnd_mod_write(mod, SSICR, cr | EN);
+ rsnd_ssi_status_check(mod, DIRQ);
/*
* disable SSI,
* and, wait idle state
*/
- rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */
- rsnd_ssi_status_check(&ssi->mod, IIRQ);
+ rsnd_mod_write(mod, SSICR, cr); /* disabled all */
+ rsnd_ssi_status_check(mod, IIRQ);
if (rsnd_rdai_is_clk_master(rdai)) {
struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -266,13 +265,13 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
rsnd_ssi_master_clk_stop(ssi);
}
- rsnd_mod_hw_stop(&ssi->mod);
+ rsnd_mod_hw_stop(mod);
ssi->chan = 0;
}
dev_dbg(dev, "%s[%d] hw stopped\n",
- rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
}
/*
@@ -371,7 +370,7 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
/* It will be removed on rsnd_ssi_hw_stop */
ssi->chan = chan;
if (ssi_parent)
- return rsnd_ssi_hw_params(&ssi_parent->mod, io,
+ return rsnd_ssi_hw_params(rsnd_mod_get(ssi_parent), io,
substream, params);
return 0;
@@ -379,12 +378,14 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status)
{
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
/* under/over flow error */
if (status & (UIRQ | OIRQ)) {
ssi->err++;
/* clear error status */
- rsnd_mod_write(&ssi->mod, SSISR, 0);
+ rsnd_mod_write(mod, SSISR, 0);
}
}
@@ -656,7 +657,7 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv)))
id = 0;
- return &((struct rsnd_ssi *)(priv->ssi) + id)->mod;
+ return rsnd_mod_get((struct rsnd_ssi *)(priv->ssi) + id);
}
int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
@@ -668,10 +669,12 @@ int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
static void rsnd_ssi_parent_setup(struct rsnd_priv *priv, struct rsnd_ssi *ssi)
{
- if (!rsnd_ssi_is_pin_sharing(&ssi->mod))
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
+ if (!rsnd_ssi_is_pin_sharing(mod))
return;
- switch (rsnd_mod_id(&ssi->mod)) {
+ switch (rsnd_mod_id(mod)) {
case 1:
case 2:
ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 0));
@@ -794,7 +797,8 @@ int rsnd_ssi_probe(struct platform_device *pdev,
else if (rsnd_ssi_pio_available(ssi))
ops = &rsnd_ssi_pio_ops;
- ret = rsnd_mod_init(priv, &ssi->mod, ops, clk, RSND_MOD_SSI, i);
+ ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk,
+ RSND_MOD_SSI, i);
if (ret)
return ret;
@@ -811,6 +815,6 @@ void rsnd_ssi_remove(struct platform_device *pdev,
int i;
for_each_rsnd_ssi(ssi, priv, i) {
- rsnd_mod_quit(&ssi->mod);
+ rsnd_mod_quit(rsnd_mod_get(ssi));
}
}
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index abb0d956231c..76b2ab8c2b4a 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -738,7 +738,7 @@ static int siu_probe(struct platform_device *pdev)
struct siu_info *info;
int ret;
- info = kmalloc(sizeof(*info), GFP_KERNEL);
+ info = devm_kmalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
if (!info)
return -ENOMEM;
siu_i2s_data = info;
@@ -746,7 +746,7 @@ static int siu_probe(struct platform_device *pdev)
ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev);
if (ret)
- goto ereqfw;
+ return ret;
/*
* Loaded firmware is "const" - read only, but we have to modify it in
@@ -757,89 +757,52 @@ static int siu_probe(struct platform_device *pdev)
release_firmware(fw_entry);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- ret = -ENODEV;
- goto egetres;
- }
+ if (!res)
+ return -ENODEV;
- region = request_mem_region(res->start, resource_size(res),
- pdev->name);
+ region = devm_request_mem_region(&pdev->dev, res->start,
+ resource_size(res), pdev->name);
if (!region) {
dev_err(&pdev->dev, "SIU region already claimed\n");
- ret = -EBUSY;
- goto ereqmemreg;
+ return -EBUSY;
}
- ret = -ENOMEM;
- info->pram = ioremap(res->start, PRAM_SIZE);
+ info->pram = devm_ioremap(&pdev->dev, res->start, PRAM_SIZE);
if (!info->pram)
- goto emappram;
- info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE);
+ return -ENOMEM;
+ info->xram = devm_ioremap(&pdev->dev, res->start + XRAM_OFFSET,
+ XRAM_SIZE);
if (!info->xram)
- goto emapxram;
- info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE);
+ return -ENOMEM;
+ info->yram = devm_ioremap(&pdev->dev, res->start + YRAM_OFFSET,
+ YRAM_SIZE);
if (!info->yram)
- goto emapyram;
- info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) -
- REG_OFFSET);
+ return -ENOMEM;
+ info->reg = devm_ioremap(&pdev->dev, res->start + REG_OFFSET,
+ resource_size(res) - REG_OFFSET);
if (!info->reg)
- goto emapreg;
+ return -ENOMEM;
dev_set_drvdata(&pdev->dev, info);
/* register using ARRAY version so we can keep dai name */
- ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component,
- &siu_i2s_dai, 1);
+ ret = devm_snd_soc_register_component(&pdev->dev, &siu_i2s_component,
+ &siu_i2s_dai, 1);
if (ret < 0)
- goto edaiinit;
+ return ret;
- ret = snd_soc_register_platform(&pdev->dev, &siu_platform);
+ ret = devm_snd_soc_register_platform(&pdev->dev, &siu_platform);
if (ret < 0)
- goto esocregp;
+ return ret;
pm_runtime_enable(&pdev->dev);
- return ret;
-
-esocregp:
- snd_soc_unregister_component(&pdev->dev);
-edaiinit:
- iounmap(info->reg);
-emapreg:
- iounmap(info->yram);
-emapyram:
- iounmap(info->xram);
-emapxram:
- iounmap(info->pram);
-emappram:
- release_mem_region(res->start, resource_size(res));
-ereqmemreg:
-egetres:
-ereqfw:
- kfree(info);
-
- return ret;
+ return 0;
}
static int siu_remove(struct platform_device *pdev)
{
- struct siu_info *info = dev_get_drvdata(&pdev->dev);
- struct resource *res;
-
pm_runtime_disable(&pdev->dev);
-
- snd_soc_unregister_platform(&pdev->dev);
- snd_soc_unregister_component(&pdev->dev);
-
- iounmap(info->reg);
- iounmap(info->yram);
- iounmap(info->xram);
- iounmap(info->pram);
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (res)
- release_mem_region(res->start, resource_size(res));
- kfree(info);
-
return 0;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6e543e18b4fa..24b096066a07 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3291,13 +3291,38 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
+static int snd_soc_of_get_slot_mask(struct device_node *np,
+ const char *prop_name,
+ unsigned int *mask)
+{
+ u32 val;
+ const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
+ int i;
+
+ if (!of_slot_mask)
+ return 0;
+ val /= sizeof(u32);
+ for (i = 0; i < val; i++)
+ if (be32_to_cpup(&of_slot_mask[i]))
+ *mask |= (1 << i);
+
+ return val;
+}
+
int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *tx_mask,
+ unsigned int *rx_mask,
unsigned int *slots,
unsigned int *slot_width)
{
u32 val;
int ret;
+ if (tx_mask)
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask);
+ if (rx_mask)
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask);
+
if (of_property_read_bool(np, "dai-tdm-slot-num")) {
ret = of_property_read_u32(np, "dai-tdm-slot-num", &val);
if (ret)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f4bf21a5539b..ff8bda471b25 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3501,7 +3501,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
default:
WARN(1, "Unknown event %d\n", event);
- return -EINVAL;
+ ret = -EINVAL;
}
out:
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 20f702add3f8..ecd38e52285a 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -207,6 +207,34 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
/**
+ * snd_soc_info_volsw_sx - Mixer info callback for SX TLV controls
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single mixer control, or a double
+ * mixer control that spans 2 registers of the SX TLV type. SX TLV controls
+ * have a range that represents both positive and negative values either side
+ * of zero but without a sign bit.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ snd_soc_info_volsw(kcontrol, uinfo);
+ /* Max represents the number of levels in an SX control not the
+ * maximum value, so add the minimum value back on
+ */
+ uinfo->value.integer.max += mc->min;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_sx);
+
+/**
* snd_soc_get_volsw - single mixer get callback
* @kcontrol: mixer control
* @ucontrol: control element information
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 70e4b9d8bdcd..317395824cd7 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -34,6 +34,24 @@
#define DPCM_MAX_BE_USERS 8
+/*
+ * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
+ *
+ * Returns true if the DAI supports the indicated stream type.
+ */
+static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_pcm_stream *codec_stream;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_stream = &dai->driver->playback;
+ else
+ codec_stream = &dai->driver->capture;
+
+ /* If the codec specifies any rate at all, it supports the stream. */
+ return codec_stream->rates;
+}
+
/**
* snd_soc_runtime_activate() - Increment active count for PCM runtime components
* @rtd: ASoC PCM runtime that is activated
@@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
/* first calculate min/max only for CODECs in the DAI link */
for (i = 0; i < rtd->num_codecs; i++) {
+
+ /*
+ * Skip CODECs which don't support the current stream type.
+ * Otherwise, since the rate, channel, and format values will
+ * zero in that case, we would have no usable settings left,
+ * causing the resulting setup to fail.
+ * At least one CODEC should match, otherwise we should have
+ * bailed out on a higher level, since there would be no
+ * CODEC to support the transfer direction in that case.
+ */
+ if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+ substream->stream))
+ continue;
+
codec_dai_drv = rtd->codec_dais[i]->driver;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_stream = &codec_dai_drv->playback;
@@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
struct snd_pcm_hw_params codec_params;
+ /*
+ * Skip CODECs which don't support the current stream type,
+ * the idea being that if a CODEC is not used for the currently
+ * set up transfer direction, it should not need to be
+ * configured, especially since the configuration used might
+ * not even be supported by that CODEC. There may be cases
+ * however where a CODEC needs to be set up although it is
+ * actually not being used for the transfer, e.g. if a
+ * capture-only CODEC is acting as an LRCLK and/or BCLK master
+ * for the DAI link including a playback-only CODEC.
+ * If this becomes necessary, we will have to augment the
+ * machine driver setup with information on how to act, so
+ * we can do the right thing here.
+ */
+ if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+ continue;
+
/* copy params for each codec */
codec_params = *params;
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 362c69ac1d6c..53dd085d3ee2 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec;
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 0a53053495f3..4fb91412ebec 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 f6eefe1b8f8f..843f037a317d 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev,
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 @@ static int uni_player_parse_dt(struct platform_device *pdev,
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 c502626f339b..f791239a3087 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
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;
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
new file mode 100644
index 000000000000..84c72ec6ad73
--- /dev/null
+++ b/sound/soc/sunxi/Kconfig
@@ -0,0 +1,11 @@
+menu "Allwinner SoC Audio support"
+
+config SND_SUN4I_CODEC
+ tristate "Allwinner A10 Codec Support"
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ select REGMAP_MMIO
+ help
+ Select Y or M to add support for the Codec embedded in the Allwinner
+ A10 and affiliated SoCs.
+
+endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
new file mode 100644
index 000000000000..ea8a08c881d6
--- /dev/null
+++ b/sound/soc/sunxi/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
+
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
new file mode 100644
index 000000000000..8d59d83b5aa4
--- /dev/null
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -0,0 +1,719 @@
+/*
+ * Copyright 2014 Emilio López <emilio@elopez.com.ar>
+ * Copyright 2014 Jon Smirl <jonsmirl@gmail.com>
+ * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com>
+ *
+ * Based on the Allwinner SDK driver, released under the GPL.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_address.h>
+#include <linux/clk.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/dmaengine_pcm.h>
+
+/* Codec DAC register offsets and bit fields */
+#define SUN4I_CODEC_DAC_DPC (0x00)
+#define SUN4I_CODEC_DAC_DPC_EN_DA (31)
+#define SUN4I_CODEC_DAC_DPC_DVOL (12)
+#define SUN4I_CODEC_DAC_FIFOC (0x04)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_FS (29)
+#define SUN4I_CODEC_DAC_FIFOC_FIR_VERSION (28)
+#define SUN4I_CODEC_DAC_FIFOC_SEND_LASAT (26)
+#define SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE (24)
+#define SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT (21)
+#define SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL (8)
+#define SUN4I_CODEC_DAC_FIFOC_MONO_EN (6)
+#define SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS (5)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN (4)
+#define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH (0)
+#define SUN4I_CODEC_DAC_FIFOS (0x08)
+#define SUN4I_CODEC_DAC_TXDATA (0x0c)
+#define SUN4I_CODEC_DAC_ACTL (0x10)
+#define SUN4I_CODEC_DAC_ACTL_DACAENR (31)
+#define SUN4I_CODEC_DAC_ACTL_DACAENL (30)
+#define SUN4I_CODEC_DAC_ACTL_MIXEN (29)
+#define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15)
+#define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14)
+#define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13)
+#define SUN4I_CODEC_DAC_ACTL_DACPAS (8)
+#define SUN4I_CODEC_DAC_ACTL_MIXPAS (7)
+#define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6)
+#define SUN4I_CODEC_DAC_ACTL_PA_VOL (0)
+#define SUN4I_CODEC_DAC_TUNE (0x14)
+#define SUN4I_CODEC_DAC_DEBUG (0x18)
+
+/* Codec ADC register offsets and bit fields */
+#define SUN4I_CODEC_ADC_FIFOC (0x1c)
+#define SUN4I_CODEC_ADC_FIFOC_EN_AD (28)
+#define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24)
+#define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8)
+#define SUN4I_CODEC_ADC_FIFOC_MONO_EN (7)
+#define SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS (6)
+#define SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN (4)
+#define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH (0)
+#define SUN4I_CODEC_ADC_FIFOS (0x20)
+#define SUN4I_CODEC_ADC_RXDATA (0x24)
+#define SUN4I_CODEC_ADC_ACTL (0x28)
+#define SUN4I_CODEC_ADC_ACTL_ADC_R_EN (31)
+#define SUN4I_CODEC_ADC_ACTL_ADC_L_EN (30)
+#define SUN4I_CODEC_ADC_ACTL_PREG1EN (29)
+#define SUN4I_CODEC_ADC_ACTL_PREG2EN (28)
+#define SUN4I_CODEC_ADC_ACTL_VMICEN (27)
+#define SUN4I_CODEC_ADC_ACTL_VADCG (20)
+#define SUN4I_CODEC_ADC_ACTL_ADCIS (17)
+#define SUN4I_CODEC_ADC_ACTL_PA_EN (4)
+#define SUN4I_CODEC_ADC_ACTL_DDE (3)
+#define SUN4I_CODEC_ADC_DEBUG (0x2c)
+
+/* Other various ADC registers */
+#define SUN4I_CODEC_DAC_TXCNT (0x30)
+#define SUN4I_CODEC_ADC_RXCNT (0x34)
+#define SUN4I_CODEC_AC_SYS_VERI (0x38)
+#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c)
+
+struct sun4i_codec {
+ struct device *dev;
+ struct regmap *regmap;
+ struct clk *clk_apb;
+ struct clk *clk_module;
+
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+};
+
+static void sun4i_codec_start_playback(struct sun4i_codec *scodec)
+{
+ /*
+ * FIXME: according to the BSP, we might need to drive a PA
+ * GPIO high here on some boards
+ */
+
+ /* Flush TX FIFO */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+ /* Enable DAC DRQ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN));
+}
+
+static void sun4i_codec_stop_playback(struct sun4i_codec *scodec)
+{
+ /*
+ * FIXME: according to the BSP, we might need to drive a PA
+ * GPIO low here on some boards
+ */
+
+ /* Disable DAC DRQ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+ 0);
+}
+
+static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ sun4i_codec_start_playback(scodec);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sun4i_codec_stop_playback(scodec);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int sun4i_codec_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+ u32 val;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ /* Flush the TX FIFO */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+ /* Set TX FIFO Empty Trigger Level */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 0x3f << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL,
+ 0xf << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL);
+
+ if (substream->runtime->rate > 32000)
+ /* Use 64 bits FIR filter */
+ val = 0;
+ else
+ /* Use 32 bits FIR filter */
+ val = BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION);
+
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION),
+ val);
+
+ /* Send zeros when we have an underrun */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT),
+ 0);
+
+ return 0;
+}
+
+static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params)
+{
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 176400:
+ case 88200:
+ case 44100:
+ case 33075:
+ case 22050:
+ case 14700:
+ case 11025:
+ case 7350:
+ return 22579200;
+
+ case 192000:
+ case 96000:
+ case 48000:
+ case 32000:
+ case 24000:
+ case 16000:
+ case 12000:
+ case 8000:
+ return 24576000;
+
+ default:
+ return 0;
+ }
+}
+
+static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params)
+{
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 192000:
+ case 176400:
+ return 6;
+
+ case 96000:
+ case 88200:
+ return 7;
+
+ case 48000:
+ case 44100:
+ return 0;
+
+ case 32000:
+ case 33075:
+ return 1;
+
+ case 24000:
+ case 22050:
+ return 2;
+
+ case 16000:
+ case 14700:
+ return 3;
+
+ case 12000:
+ case 11025:
+ return 4;
+
+ case 8000:
+ case 7350:
+ return 5;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int sun4i_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+ unsigned long clk_freq;
+ int hwrate;
+ u32 val;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ clk_freq = sun4i_codec_get_mod_freq(params);
+ if (!clk_freq)
+ return -EINVAL;
+
+ if (clk_set_rate(scodec->clk_module, clk_freq))
+ return -EINVAL;
+
+ hwrate = sun4i_codec_get_hw_rate(params);
+ if (hwrate < 0)
+ return hwrate;
+
+ /* Set DAC sample rate */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 7 << SUN4I_CODEC_DAC_FIFOC_DAC_FS,
+ hwrate << SUN4I_CODEC_DAC_FIFOC_DAC_FS);
+
+ /* Set the number of channels we want to use */
+ if (params_channels(params) == 1)
+ val = BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN);
+ else
+ val = 0;
+
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN),
+ val);
+
+ /* Set the number of sample bits to either 16 or 24 bits */
+ if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) {
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS));
+
+ /* Set TX FIFO mode to padding the LSBs with 0 */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+ 0);
+
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ } else {
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+ 0);
+
+ /* Set TX FIFO mode to repeat the MSB */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE));
+
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ }
+
+ return 0;
+}
+
+static int sun4i_codec_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ /*
+ * Stop issuing DRQ when we have room for less than 16 samples
+ * in our TX FIFO
+ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT,
+ 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT);
+
+ return clk_prepare_enable(scodec->clk_module);
+}
+
+static void sun4i_codec_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ clk_disable_unprepare(scodec->clk_module);
+}
+
+static const struct snd_soc_dai_ops sun4i_codec_dai_ops = {
+ .startup = sun4i_codec_startup,
+ .shutdown = sun4i_codec_shutdown,
+ .trigger = sun4i_codec_trigger,
+ .hw_params = sun4i_codec_hw_params,
+ .prepare = sun4i_codec_prepare,
+};
+
+static struct snd_soc_dai_driver sun4i_codec_dai = {
+ .name = "Codec",
+ .ops = &sun4i_codec_dai_ops,
+ .playback = {
+ .stream_name = "Codec Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .rates = SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000 |
+ SNDRV_PCM_RATE_KNOT,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 24,
+ },
+};
+
+/*** Codec ***/
+static const struct snd_kcontrol_new sun4i_codec_pa_mute =
+ SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0);
+
+static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1);
+
+static const struct snd_kcontrol_new sun4i_codec_widgets[] = {
+ SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0,
+ sun4i_codec_pa_volume_scale),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACPAS, 1, 0),
+ SOC_DAPM_SINGLE("Mixer Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = {
+ /* Digital parts of the DACs */
+ SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC,
+ SUN4I_CODEC_DAC_DPC_EN_DA, 0,
+ NULL, 0),
+
+ /* Analog parts of the DACs */
+ SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACAENL, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACAENR, 0),
+
+ /* Mixers */
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ sun4i_codec_left_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ sun4i_codec_right_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_right_mixer_controls)),
+
+ /* Global Mixer Enable */
+ SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0),
+
+ /* Pre-Amplifier */
+ SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL,
+ SUN4I_CODEC_ADC_ACTL_PA_EN, 0,
+ sun4i_codec_pa_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_pa_mixer_controls)),
+ SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0,
+ &sun4i_codec_pa_mute),
+
+ SND_SOC_DAPM_OUTPUT("HP Right"),
+ SND_SOC_DAPM_OUTPUT("HP Left"),
+};
+
+static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = {
+ /* Left DAC Routes */
+ { "Left DAC", NULL, "DAC" },
+
+ /* Right DAC Routes */
+ { "Right DAC", NULL, "DAC" },
+
+ /* Right Mixer Routes */
+ { "Right Mixer", NULL, "Mixer Enable" },
+ { "Right Mixer", "Left DAC Playback Switch", "Left DAC" },
+ { "Right Mixer", "Right DAC Playback Switch", "Right DAC" },
+
+ /* Left Mixer Routes */
+ { "Left Mixer", NULL, "Mixer Enable" },
+ { "Left Mixer", "Left DAC Playback Switch", "Left DAC" },
+
+ /* Pre-Amplifier Mixer Routes */
+ { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" },
+ { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" },
+ { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" },
+ { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" },
+
+ /* PA -> HP path */
+ { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" },
+ { "HP Right", NULL, "Pre-Amplifier Mute" },
+ { "HP Left", NULL, "Pre-Amplifier Mute" },
+};
+
+static struct snd_soc_codec_driver sun4i_codec_codec = {
+ .controls = sun4i_codec_widgets,
+ .num_controls = ARRAY_SIZE(sun4i_codec_widgets),
+ .dapm_widgets = sun4i_codec_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets),
+ .dapm_routes = sun4i_codec_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes),
+};
+
+static const struct snd_soc_component_driver sun4i_codec_component = {
+ .name = "sun4i-codec",
+};
+
+#define SUN4I_CODEC_RATES SNDRV_PCM_RATE_8000_192000
+#define SUN4I_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static int sun4i_codec_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai);
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+ snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data,
+ NULL);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver dummy_cpu_dai = {
+ .name = "sun4i-codec-cpu-dai",
+ .probe = sun4i_codec_dai_probe,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SUN4I_CODEC_RATES,
+ .formats = SUN4I_CODEC_FORMATS,
+ .sig_bits = 24,
+ },
+};
+
+static const struct regmap_config sun4i_codec_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL,
+};
+
+static const struct of_device_id sun4i_codec_of_match[] = {
+ { .compatible = "allwinner,sun4i-a10-codec" },
+ { .compatible = "allwinner,sun7i-a20-codec" },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sun4i_codec_of_match);
+
+static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev,
+ int *num_links)
+{
+ struct snd_soc_dai_link *link = devm_kzalloc(dev, sizeof(*link),
+ GFP_KERNEL);
+ if (!link)
+ return NULL;
+
+ link->name = "cdc";
+ link->stream_name = "CDC PCM";
+ link->codec_dai_name = "Codec";
+ link->cpu_dai_name = dev_name(dev);
+ link->codec_name = dev_name(dev);
+ link->platform_name = dev_name(dev);
+ link->dai_fmt = SND_SOC_DAIFMT_I2S;
+
+ *num_links = 1;
+
+ return link;
+};
+
+static struct snd_soc_card *sun4i_codec_create_card(struct device *dev)
+{
+ struct snd_soc_card *card;
+ int ret;
+
+ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return NULL;
+
+ card->dai_link = sun4i_codec_create_link(dev, &card->num_links);
+ if (!card->dai_link)
+ return NULL;
+
+ card->dev = dev;
+ card->name = "sun4i-codec";
+
+ ret = snd_soc_of_parse_audio_routing(card, "routing");
+ if (ret) {
+ dev_err(dev, "Failed to create our audio routing\n");
+ return NULL;
+ }
+
+ return card;
+};
+
+static int sun4i_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card;
+ struct sun4i_codec *scodec;
+ struct resource *res;
+ void __iomem *base;
+ int ret;
+
+ scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL);
+ if (!scodec)
+ return -ENOMEM;
+
+ scodec->dev = &pdev->dev;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(base)) {
+ dev_err(&pdev->dev, "Failed to map the registers\n");
+ return PTR_ERR(base);
+ }
+
+ scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sun4i_codec_regmap_config);
+ if (IS_ERR(scodec->regmap)) {
+ dev_err(&pdev->dev, "Failed to create our regmap\n");
+ return PTR_ERR(scodec->regmap);
+ }
+
+ /* Get the clocks from the DT */
+ scodec->clk_apb = devm_clk_get(&pdev->dev, "apb");
+ if (IS_ERR(scodec->clk_apb)) {
+ dev_err(&pdev->dev, "Failed to get the APB clock\n");
+ return PTR_ERR(scodec->clk_apb);
+ }
+
+ scodec->clk_module = devm_clk_get(&pdev->dev, "codec");
+ if (IS_ERR(scodec->clk_module)) {
+ dev_err(&pdev->dev, "Failed to get the module clock\n");
+ return PTR_ERR(scodec->clk_module);
+ }
+
+ /* Enable the bus clock */
+ if (clk_prepare_enable(scodec->clk_apb)) {
+ dev_err(&pdev->dev, "Failed to enable the APB clock\n");
+ return -EINVAL;
+ }
+
+ /* DMA configuration for TX FIFO */
+ scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA;
+ scodec->playback_dma_data.maxburst = 4;
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+
+ ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec,
+ &sun4i_codec_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our codec\n");
+ goto err_clk_disable;
+ }
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &sun4i_codec_component,
+ &dummy_cpu_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our DAI\n");
+ goto err_unregister_codec;
+ }
+
+ ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register against DMAEngine\n");
+ goto err_unregister_codec;
+ }
+
+ card = sun4i_codec_create_card(&pdev->dev);
+ if (!card) {
+ dev_err(&pdev->dev, "Failed to create our card\n");
+ goto err_unregister_codec;
+ }
+
+ platform_set_drvdata(pdev, card);
+ snd_soc_card_set_drvdata(card, scodec);
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our card\n");
+ goto err_unregister_codec;
+ }
+
+ return 0;
+
+err_unregister_codec:
+ snd_soc_unregister_codec(&pdev->dev);
+err_clk_disable:
+ clk_disable_unprepare(scodec->clk_apb);
+ return ret;
+}
+
+static int sun4i_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+ snd_soc_unregister_card(card);
+ snd_soc_unregister_codec(&pdev->dev);
+ clk_disable_unprepare(scodec->clk_apb);
+
+ return 0;
+}
+
+static struct platform_driver sun4i_codec_driver = {
+ .driver = {
+ .name = "sun4i-codec",
+ .of_match_table = sun4i_codec_of_match,
+ },
+ .probe = sun4i_codec_probe,
+ .remove = sun4i_codec_remove,
+};
+module_platform_driver(sun4i_codec_driver);
+
+MODULE_DESCRIPTION("Allwinner A10 codec driver");
+MODULE_AUTHOR("Emilio López <emilio@elopez.com.ar>");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_AUTHOR("Maxime Ripard <maxime.ripard@free-electrons.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 4e0c0e502ade..ba9fc099cf67 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -152,6 +152,7 @@ static const struct of_device_id snd_soc_mop500_match[] = {
{ .compatible = "stericsson,snd-soc-mop500", },
{},
};
+MODULE_DEVICE_TABLE(of, snd_soc_mop500_match);
static struct platform_driver snd_soc_mop500_driver = {
.driver = {
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index f5df08ded770..6ba8ae9ecc7a 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -843,6 +843,7 @@ static const struct of_device_id ux500_msp_i2s_match[] = {
{ .compatible = "stericsson,ux500-msp-i2s", },
{},
};
+MODULE_DEVICE_TABLE(of, ux500_msp_i2s_match);
static struct platform_driver msp_i2s_driver = {
.driver = {
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
index 82e350e9501c..ac75816ada7c 100644
--- a/sound/synth/emux/emux_oss.c
+++ b/sound/synth/emux/emux_oss.c
@@ -69,7 +69,8 @@ snd_emux_init_seq_oss(struct snd_emux *emu)
struct snd_seq_oss_reg *arg;
struct snd_seq_device *dev;
- if (snd_seq_device_new(emu->card, 0, SNDRV_SEQ_DEV_ID_OSS,
+ /* using device#1 here for avoiding conflicts with OPL3 */
+ if (snd_seq_device_new(emu->card, 1, SNDRV_SEQ_DEV_ID_OSS,
sizeof(struct snd_seq_oss_reg), &dev) < 0)
return;