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authorLinus Torvalds <torvalds@linux-foundation.org>2016-05-28 12:23:12 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2016-05-28 12:23:12 -0700
commit0723ab4a97a19bf9da135d68529977aeba17570d (patch)
tree453051ed67b556ddd13a5921742ba228c22d980a
parent9ba55cf7cfbfd12a7e914d0d55b7581e896b3f0d (diff)
parenteb4606e64a7d548f5d60a9583baa8104890b2c6e (diff)
Merge tag 'sound-4.7-rc1-2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull more sound updates from Takashi Iwai: "This is the second update round for 4.7-rc1. Most of changes are about the pending ASoC updates and fixes, including a few new drivers. Below are some highlights: ASoC: - New drivers for MAX98371 and TAS5720 - SPI support for TLV320AIC32x4, along with the module split - TDM support for STI Uniperf IPs - Remaining topology API fixes / updates HDA: - A couple of Dell quirks and new Realtek codec support" * tag 'sound-4.7-rc1-2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (63 commits) ALSA: hda - Fix headset mic detection problem for one Dell machine spi: spi-ep93xx: Fix the PTR_ERR() argument ALSA: hda/realtek - Add support for ALC295/ALC3254 ASoC: kirkwood: fix build failure ALSA: hda - Fix headphone noise on Dell XPS 13 9360 ASoC: ak4642: Enable cache usage to fix crashes on resume ASoC: twl6040: Disconnect AUX output pads on digital mute ASoC: tlv320aic32x4: Properly implement the positive and negative pins into the mixers rcar: src: skip disabled-SRC nodes ASoC: max98371 Remove duplicate entry in max98371_reg ASoC: twl6040: Select LPPLL during standby ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS ASoC: simple-card: Add pm callbacks to platform driver ASoC: pxa: Fix module autoload for platform drivers ASoC: topology: Fix memory leak in widget creation ASoC: Add max98371 codec driver ASoC: rsnd: count .probe/.remove for rsnd_mod_call() ASoC: topology: Check size mismatch of ABI objects before parsing ASoC: topology: Check failure to create a widget ASoC: add support for TAS5720 digital amplifier ...
-rw-r--r--Documentation/devicetree/bindings/sound/max98371.txt17
-rw-r--r--Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt5
-rw-r--r--Documentation/devicetree/bindings/sound/mt8173-rt5650.txt10
-rw-r--r--Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt48
-rw-r--r--Documentation/devicetree/bindings/sound/tas571x.txt10
-rw-r--r--Documentation/devicetree/bindings/sound/tas5720.txt25
-rw-r--r--drivers/spi/spi-ep93xx.c2
-rw-r--r--include/linux/mfd/twl6040.h1
-rw-r--r--include/uapi/sound/asoc.h44
-rw-r--r--sound/pci/hda/patch_realtek.c16
-rw-r--r--sound/soc/codecs/Kconfig38
-rw-r--r--sound/soc/codecs/Makefile11
-rw-r--r--sound/soc/codecs/ak4642.c3
-rw-r--r--sound/soc/codecs/max98371.c441
-rw-r--r--sound/soc/codecs/max98371.h67
-rw-r--r--sound/soc/codecs/rt298.c51
-rw-r--r--sound/soc/codecs/rt298.h2
-rw-r--r--sound/soc/codecs/rt5677.c24
-rw-r--r--sound/soc/codecs/tas571x.c141
-rw-r--r--sound/soc/codecs/tas571x.h22
-rw-r--r--sound/soc/codecs/tas5720.c620
-rw-r--r--sound/soc/codecs/tas5720.h90
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c10
-rw-r--r--sound/soc/codecs/tlv320aic32x4-i2c.c74
-rw-r--r--sound/soc/codecs/tlv320aic32x4-spi.c76
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c279
-rw-r--r--sound/soc/codecs/tlv320aic32x4.h7
-rw-r--r--sound/soc/codecs/twl6040.c16
-rw-r--r--sound/soc/codecs/wm8962.c9
-rw-r--r--sound/soc/codecs/wm8962.h6
-rw-r--r--sound/soc/generic/simple-card.c1
-rw-r--r--sound/soc/kirkwood/Kconfig1
-rw-r--r--sound/soc/mediatek/Kconfig1
-rw-r--r--sound/soc/mediatek/mt8173-rt5650-rt5676.c27
-rw-r--r--sound/soc/mediatek/mt8173-rt5650.c50
-rw-r--r--sound/soc/mediatek/mtk-afe-pcm.c2
-rw-r--r--sound/soc/omap/mcbsp.c8
-rw-r--r--sound/soc/omap/omap-pcm.c2
-rw-r--r--sound/soc/pxa/brownstone.c1
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c1
-rw-r--r--sound/soc/pxa/mmp-pcm.c1
-rw-r--r--sound/soc/pxa/mmp-sspa.c1
-rw-r--r--sound/soc/pxa/palm27x.c1
-rw-r--r--sound/soc/pxa/pxa-ssp.c1
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c1
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c1
-rw-r--r--sound/soc/qcom/lpass-platform.c6
-rw-r--r--sound/soc/sh/rcar/adg.c8
-rw-r--r--sound/soc/sh/rcar/dma.c12
-rw-r--r--sound/soc/sh/rcar/rsnd.h13
-rw-r--r--sound/soc/sh/rcar/src.c4
-rw-r--r--sound/soc/soc-topology.c48
-rw-r--r--sound/soc/sti/sti_uniperif.c144
-rw-r--r--sound/soc/sti/uniperif.h220
-rw-r--r--sound/soc/sti/uniperif_player.c182
-rw-r--r--sound/soc/sti/uniperif_reader.c229
56 files changed, 2800 insertions, 331 deletions
diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt
new file mode 100644
index 000000000000..6c285235e64b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max98371.txt
@@ -0,0 +1,17 @@
+max98371 codec
+
+This device supports I2C mode only.
+
+Required properties:
+
+- compatible : "maxim,max98371"
+- reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+ max98371: max98371@0x31 {
+ compatible = "maxim,max98371";
+ reg = <0x31>;
+ };
+};
diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt
index f205ce9e31dd..ac28cdb4910e 100644
--- a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt
+++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt
@@ -1,15 +1,16 @@
-MT8173 with RT5650 RT5676 CODECS
+MT8173 with RT5650 RT5676 CODECS and HDMI via I2S
Required properties:
- compatible : "mediatek,mt8173-rt5650-rt5676"
- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs
+ and of the hdmi encoder node
- mediatek,platform: the phandle of MT8173 ASoC platform
Example:
sound {
compatible = "mediatek,mt8173-rt5650-rt5676";
- mediatek,audio-codec = <&rt5650 &rt5676>;
+ mediatek,audio-codec = <&rt5650 &rt5676 &hdmi0>;
mediatek,platform = <&afe>;
};
diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
index fe5a5ef1714d..5bfa6b60530b 100644
--- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
+++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt
@@ -5,11 +5,21 @@ Required properties:
- mediatek,audio-codec: the phandles of rt5650 codecs
- mediatek,platform: the phandle of MT8173 ASoC platform
+Optional subnodes:
+- codec-capture : the subnode of rt5650 codec capture
+Required codec-capture subnode properties:
+- sound-dai: audio codec dai name on capture path
+ <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1)
+ <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2)
+
Example:
sound {
compatible = "mediatek,mt8173-rt5650";
mediatek,audio-codec = <&rt5650>;
mediatek,platform = <&afe>;
+ codec-capture {
+ sound-dai = <&rt5650 1>;
+ };
};
diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
index 028fa1c82f50..4d9a83d9a017 100644
--- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
@@ -37,17 +37,18 @@ Required properties:
- dai-name: DAI name that describes the IP.
+ - IP mode: IP working mode depending on associated codec.
+ "HDMI" connected to HDMI codec and support IEC HDMI formats (player only).
+ "SPDIF" connected to SPDIF codec and support SPDIF formats (player only).
+ "PCM" PCM standard mode for I2S or TDM bus.
+ "TDM" TDM mode for TDM bus.
+
Required properties ("st,sti-uni-player" compatibility only):
- clocks: CPU_DAI IP clock source, listed in the same order than the
CPU_DAI properties.
- uniperiph-id: internal SOC IP instance ID.
- - IP mode: IP working mode depending on associated codec.
- "HDMI" connected to HDMI codec IP and IEC HDMI formats.
- "SPDIF"connected to SPDIF codec and support SPDIF formats.
- "PCM" PCM standard mode for I2S or TDM bus.
-
Optional properties:
- pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for
external codecs connection.
@@ -56,6 +57,22 @@ Optional properties:
Example:
+ sti_uni_player1: sti-uni-player@1 {
+ compatible = "st,sti-uni-player";
+ status = "okay";
+ #sound-dai-cells = <0>;
+ st,syscfg = <&syscfg_core>;
+ clocks = <&clk_s_d0_flexgen CLK_PCM_1>;
+ reg = <0x8D81000 0x158>;
+ interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>;
+ dmas = <&fdma0 3 0 1>;
+ st,dai-name = "Uni Player #1 (I2S)";
+ dma-names = "tx";
+ st,uniperiph-id = <1>;
+ st,version = <5>;
+ st,mode = "TDM";
+ };
+
sti_uni_player2: sti-uni-player@2 {
compatible = "st,sti-uni-player";
status = "okay";
@@ -65,7 +82,7 @@ Example:
reg = <0x8D82000 0x158>;
interrupts = <GIC_SPI 86 IRQ_TYPE_NONE>;
dmas = <&fdma0 4 0 1>;
- dai-name = "Uni Player #1 (DAC)";
+ dai-name = "Uni Player #2 (DAC)";
dma-names = "tx";
uniperiph-id = <2>;
version = <5>;
@@ -82,7 +99,7 @@ Example:
interrupts = <GIC_SPI 89 IRQ_TYPE_NONE>;
dmas = <&fdma0 7 0 1>;
dma-names = "tx";
- dai-name = "Uni Player #1 (PIO)";
+ dai-name = "Uni Player #3 (SPDIF)";
uniperiph-id = <3>;
version = <5>;
mode = "SPDIF";
@@ -99,6 +116,7 @@ Example:
dma-names = "rx";
dai-name = "Uni Reader #1 (HDMI RX)";
version = <3>;
+ st,mode = "PCM";
};
2) sti-sas-codec: internal audio codec IPs driver
@@ -152,4 +170,20 @@ Example of audio card declaration:
sound-dai = <&sti_sasg_codec 0>;
};
};
+ simple-audio-card,dai-link@2 {
+ /* TDM playback */
+ format = "left_j";
+ frame-inversion = <1>;
+ cpu {
+ sound-dai = <&sti_uni_player1>;
+ dai-tdm-slot-num = <16>;
+ dai-tdm-slot-width = <16>;
+ dai-tdm-slot-tx-mask =
+ <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>;
+ };
+
+ codec {
+ sound-dai = <&sti_sasg_codec 3>;
+ };
+ };
};
diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt
index 0ac31d8d5ac4..b4959f10b74b 100644
--- a/Documentation/devicetree/bindings/sound/tas571x.txt
+++ b/Documentation/devicetree/bindings/sound/tas571x.txt
@@ -1,4 +1,4 @@
-Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers
+Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers
The codec is controlled through an I2C interface. It also has two other
signals that can be wired up to GPIOs: reset (strongly recommended), and
@@ -6,7 +6,11 @@ powerdown (optional).
Required properties:
-- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719"
+- compatible: should be one of the following:
+ - "ti,tas5711",
+ - "ti,tas5717",
+ - "ti,tas5719",
+ - "ti,tas5721"
- reg: The I2C address of the device
- #sound-dai-cells: must be equal to 0
@@ -25,6 +29,8 @@ Optional properties:
- PVDD_B-supply: regulator phandle for the PVDD_B supply (5711)
- PVDD_C-supply: regulator phandle for the PVDD_C supply (5711)
- PVDD_D-supply: regulator phandle for the PVDD_D supply (5711)
+- DRVDD-supply: regulator phandle for the DRVDD supply (5721)
+- PVDD-supply: regulator phandle for the PVDD supply (5721)
Example:
diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt
new file mode 100644
index 000000000000..806ea7381483
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tas5720.txt
@@ -0,0 +1,25 @@
+Texas Instruments TAS5720 Mono Audio amplifier
+
+The TAS5720 serial control bus communicates through the I2C protocol only. The
+serial bus is also used for periodic codec fault checking/reporting during
+audio playback. For more product information please see the links below:
+
+http://www.ti.com/product/TAS5720L
+http://www.ti.com/product/TAS5720M
+
+Required properties:
+
+- compatible : "ti,tas5720"
+- reg : I2C slave address
+- dvdd-supply : phandle to a 3.3-V supply for the digital circuitry
+- pvdd-supply : phandle to a supply used for the Class-D amp and the analog
+
+Example:
+
+tas5720: tas5720@6c {
+ status = "okay";
+ compatible = "ti,tas5720";
+ reg = <0x6c>;
+ dvdd-supply = <&vdd_3v3_reg>;
+ pvdd-supply = <&amp_supply_reg>;
+};
diff --git a/drivers/spi/spi-ep93xx.c b/drivers/spi/spi-ep93xx.c
index bb00be8d1851..17a6387e20b5 100644
--- a/drivers/spi/spi-ep93xx.c
+++ b/drivers/spi/spi-ep93xx.c
@@ -567,7 +567,7 @@ static void ep93xx_spi_dma_transfer(struct ep93xx_spi *espi)
txd = ep93xx_spi_dma_prepare(espi, DMA_MEM_TO_DEV);
if (IS_ERR(txd)) {
ep93xx_spi_dma_finish(espi, DMA_DEV_TO_MEM);
- dev_err(&espi->pdev->dev, "DMA TX failed: %ld\n", PTR_ERR(rxd));
+ dev_err(&espi->pdev->dev, "DMA TX failed: %ld\n", PTR_ERR(txd));
msg->status = PTR_ERR(txd);
return;
}
diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h
index 8f9fc3d26e6d..8e95cd87cd74 100644
--- a/include/linux/mfd/twl6040.h
+++ b/include/linux/mfd/twl6040.h
@@ -134,6 +134,7 @@
#define TWL6040_HFDACENA (1 << 0)
#define TWL6040_HFPGAENA (1 << 1)
#define TWL6040_HFDRVENA (1 << 4)
+#define TWL6040_HFSWENA (1 << 6)
/* VIBCTLL/R (0x18/0x1A) fields */
diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h
index c4cc1e40b35c..e4701a3c6331 100644
--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@@ -116,6 +116,14 @@
#define SND_SOC_TPLG_STREAM_PLAYBACK 0
#define SND_SOC_TPLG_STREAM_CAPTURE 1
+/* vendor tuple types */
+#define SND_SOC_TPLG_TUPLE_TYPE_UUID 0
+#define SND_SOC_TPLG_TUPLE_TYPE_STRING 1
+#define SND_SOC_TPLG_TUPLE_TYPE_BOOL 2
+#define SND_SOC_TPLG_TUPLE_TYPE_BYTE 3
+#define SND_SOC_TPLG_TUPLE_TYPE_WORD 4
+#define SND_SOC_TPLG_TUPLE_TYPE_SHORT 5
+
/*
* Block Header.
* This header precedes all object and object arrays below.
@@ -132,6 +140,35 @@ struct snd_soc_tplg_hdr {
__le32 count; /* number of elements in block */
} __attribute__((packed));
+/* vendor tuple for uuid */
+struct snd_soc_tplg_vendor_uuid_elem {
+ __le32 token;
+ char uuid[16];
+} __attribute__((packed));
+
+/* vendor tuple for a bool/byte/short/word value */
+struct snd_soc_tplg_vendor_value_elem {
+ __le32 token;
+ __le32 value;
+} __attribute__((packed));
+
+/* vendor tuple for string */
+struct snd_soc_tplg_vendor_string_elem {
+ __le32 token;
+ char string[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+} __attribute__((packed));
+
+struct snd_soc_tplg_vendor_array {
+ __le32 size; /* size in bytes of the array, including all elements */
+ __le32 type; /* SND_SOC_TPLG_TUPLE_TYPE_ */
+ __le32 num_elems; /* number of elements in array */
+ union {
+ struct snd_soc_tplg_vendor_uuid_elem uuid[0];
+ struct snd_soc_tplg_vendor_value_elem value[0];
+ struct snd_soc_tplg_vendor_string_elem string[0];
+ };
+} __attribute__((packed));
+
/*
* Private data.
* All topology objects may have private data that can be used by the driver or
@@ -139,7 +176,10 @@ struct snd_soc_tplg_hdr {
*/
struct snd_soc_tplg_private {
__le32 size; /* in bytes of private data */
- char data[0];
+ union {
+ char data[0];
+ struct snd_soc_tplg_vendor_array array[0];
+ };
} __attribute__((packed));
/*
@@ -383,7 +423,7 @@ struct snd_soc_tplg_pcm {
__le32 size; /* in bytes of this structure */
char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- __le32 pcm_id; /* unique ID - used to match */
+ __le32 pcm_id; /* unique ID - used to match with DAI link */
__le32 dai_id; /* unique ID - used to match */
__le32 playback; /* supports playback mode */
__le32 capture; /* supports capture mode */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 002f153bc659..d53c25e7a1c1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -335,6 +335,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0283:
case 0x10ec0286:
case 0x10ec0288:
+ case 0x10ec0295:
case 0x10ec0298:
alc_update_coef_idx(codec, 0x10, 1<<9, 0);
break;
@@ -907,6 +908,7 @@ static struct alc_codec_rename_pci_table rename_pci_tbl[] = {
{ 0x10ec0298, 0x1028, 0, "ALC3266" },
{ 0x10ec0256, 0x1028, 0, "ALC3246" },
{ 0x10ec0225, 0x1028, 0, "ALC3253" },
+ { 0x10ec0295, 0x1028, 0, "ALC3254" },
{ 0x10ec0670, 0x1025, 0, "ALC669X" },
{ 0x10ec0676, 0x1025, 0, "ALC679X" },
{ 0x10ec0282, 0x1043, 0, "ALC3229" },
@@ -3697,6 +3699,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
alc_process_coef_fw(codec, coef0668);
break;
case 0x10ec0225:
+ case 0x10ec0295:
alc_process_coef_fw(codec, coef0225);
break;
}
@@ -3797,6 +3800,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
break;
case 0x10ec0225:
+ case 0x10ec0295:
alc_update_coef_idx(codec, 0x45, 0x3f<<10, 0x31<<10);
snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
alc_process_coef_fw(codec, coef0225);
@@ -3854,6 +3858,7 @@ static void alc_headset_mode_default(struct hda_codec *codec)
switch (codec->core.vendor_id) {
case 0x10ec0225:
+ case 0x10ec0295:
alc_process_coef_fw(codec, coef0225);
break;
case 0x10ec0255:
@@ -3957,6 +3962,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
alc_process_coef_fw(codec, coef0688);
break;
case 0x10ec0225:
+ case 0x10ec0295:
alc_process_coef_fw(codec, coef0225);
break;
}
@@ -4038,6 +4044,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
alc_process_coef_fw(codec, coef0688);
break;
case 0x10ec0225:
+ case 0x10ec0295:
alc_process_coef_fw(codec, coef0225);
break;
}
@@ -4121,6 +4128,7 @@ static void alc_determine_headset_type(struct hda_codec *codec)
is_ctia = (val & 0x1c02) == 0x1c02;
break;
case 0x10ec0225:
+ case 0x10ec0295:
alc_process_coef_fw(codec, coef0225);
msleep(800);
val = alc_read_coef_idx(codec, 0x46);
@@ -5466,8 +5474,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
- SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
+ SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE),
+ SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5711,6 +5720,9 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x14, 0x90170110},
{0x21, 0x02211020}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ {0x14, 0x90170130},
+ {0x21, 0x02211040}),
+ SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x12, 0x90a60140},
{0x14, 0x90170110},
{0x21, 0x02211020}),
@@ -6033,6 +6045,7 @@ static int patch_alc269(struct hda_codec *codec)
alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
break;
case 0x10ec0225:
+ case 0x10ec0295:
spec->codec_variant = ALC269_TYPE_ALC225;
break;
case 0x10ec0234:
@@ -6979,6 +6992,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0294, "ALC294", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b3afae990e39..4d82a58ff6b0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK5386
select SND_SOC_ALC5623 if I2C
select SND_SOC_ALC5632 if I2C
+ select SND_SOC_BT_SCO
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS35L32 if I2C
select SND_SOC_CS42L51_I2C if I2C
@@ -64,7 +65,6 @@ config SND_SOC_ALL_CODECS
select SND_SOC_DA732X if I2C
select SND_SOC_DA9055 if I2C
select SND_SOC_DMIC
- select SND_SOC_BT_SCO
select SND_SOC_ES8328_SPI if SPI_MASTER
select SND_SOC_ES8328_I2C if I2C
select SND_SOC_GTM601
@@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98090 if I2C
select SND_SOC_MAX98095 if I2C
select SND_SOC_MAX98357A if GPIOLIB
+ select SND_SOC_MAX98371 if I2C
select SND_SOC_MAX9867 if I2C
select SND_SOC_MAX98925 if I2C
select SND_SOC_MAX98926 if I2C
@@ -126,12 +127,14 @@ config SND_SOC_ALL_CODECS
select SND_SOC_TAS2552 if I2C
select SND_SOC_TAS5086 if I2C
select SND_SOC_TAS571X if I2C
+ select SND_SOC_TAS5720 if I2C
select SND_SOC_TFA9879 if I2C
select SND_SOC_TLV320AIC23_I2C if I2C
select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC31XX if I2C
- select SND_SOC_TLV320AIC32X4 if I2C
+ select SND_SOC_TLV320AIC32X4_I2C if I2C
+ select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER
select SND_SOC_TLV320AIC3X if I2C
select SND_SOC_TPA6130A2 if I2C
select SND_SOC_TLV320DAC33 if I2C
@@ -367,6 +370,9 @@ config SND_SOC_ALC5623
config SND_SOC_ALC5632
tristate
+config SND_SOC_BT_SCO
+ tristate
+
config SND_SOC_CQ0093VC
tristate
@@ -473,9 +479,6 @@ config SND_SOC_DA732X
config SND_SOC_DA9055
tristate
-config SND_SOC_BT_SCO
- tristate
-
config SND_SOC_DMIC
tristate
@@ -529,6 +532,9 @@ config SND_SOC_MAX98095
config SND_SOC_MAX98357A
tristate
+config SND_SOC_MAX98371
+ tristate
+
config SND_SOC_MAX9867
tristate
@@ -748,9 +754,16 @@ config SND_SOC_TAS5086
depends on I2C
config SND_SOC_TAS571X
- tristate "Texas Instruments TAS5711/TAS5717/TAS5719 power amplifiers"
+ tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers"
depends on I2C
+config SND_SOC_TAS5720
+ tristate "Texas Instruments TAS5720 Mono Audio amplifier"
+ depends on I2C
+ help
+ Enable support for Texas Instruments TAS5720L/M high-efficiency mono
+ Class-D audio power amplifiers.
+
config SND_SOC_TFA9879
tristate "NXP Semiconductors TFA9879 amplifier"
depends on I2C
@@ -780,6 +793,16 @@ config SND_SOC_TLV320AIC31XX
config SND_SOC_TLV320AIC32X4
tristate
+config SND_SOC_TLV320AIC32X4_I2C
+ tristate
+ depends on I2C
+ select SND_SOC_TLV320AIC32X4
+
+config SND_SOC_TLV320AIC32X4_SPI
+ tristate
+ depends on SPI_MASTER
+ select SND_SOC_TLV320AIC32X4
+
config SND_SOC_TLV320AIC3X
tristate "Texas Instruments TLV320AIC3x CODECs"
depends on I2C
@@ -920,7 +943,8 @@ config SND_SOC_WM8955
tristate
config SND_SOC_WM8960
- tristate
+ tristate "Wolfson Microelectronics WM8960 CODEC"
+ depends on I2C
config SND_SOC_WM8961
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index b7b99416537f..0f548fd34ca3 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -32,6 +32,7 @@ snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
snd-soc-ak5386-objs := ak5386.o
snd-soc-arizona-objs := arizona.o
+snd-soc-bt-sco-objs := bt-sco.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs35l32-objs := cs35l32.o
snd-soc-cs42l51-objs := cs42l51.o
@@ -55,7 +56,6 @@ snd-soc-da7218-objs := da7218.o
snd-soc-da7219-objs := da7219.o da7219-aad.o
snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o
-snd-soc-bt-sco-objs := bt-sco.o
snd-soc-dmic-objs := dmic.o
snd-soc-es8328-objs := es8328.o
snd-soc-es8328-i2c-objs := es8328-i2c.o
@@ -74,6 +74,7 @@ snd-soc-max98088-objs := max98088.o
snd-soc-max98090-objs := max98090.o
snd-soc-max98095-objs := max98095.o
snd-soc-max98357a-objs := max98357a.o
+snd-soc-max98371-objs := max98371.o
snd-soc-max9867-objs := max9867.o
snd-soc-max98925-objs := max98925.o
snd-soc-max98926-objs := max98926.o
@@ -131,6 +132,7 @@ snd-soc-stac9766-objs := stac9766.o
snd-soc-sti-sas-objs := sti-sas.o
snd-soc-tas5086-objs := tas5086.o
snd-soc-tas571x-objs := tas571x.o
+snd-soc-tas5720-objs := tas5720.o
snd-soc-tfa9879-objs := tfa9879.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
@@ -138,6 +140,8 @@ snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
+snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o
+snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-tlv320dac33-objs := tlv320dac33.o
snd-soc-ts3a227e-objs := ts3a227e.o
@@ -243,6 +247,7 @@ obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o
obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o
obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o
+obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o
obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
@@ -266,7 +271,6 @@ obj-$(CONFIG_SND_SOC_DA7218) += snd-soc-da7218.o
obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o
obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
-obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o
obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
@@ -339,6 +343,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o
obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o
obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o
+obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o
obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o
@@ -346,6 +351,8 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o
obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o
+obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o
+obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 1ee8506c06c7..4d8b9e49e8d6 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = {
.max_register = FIL1_3,
.reg_defaults = ak4642_reg,
.num_reg_defaults = NUM_AK4642_REG_DEFAULTS,
+ .cache_type = REGCACHE_RBTREE,
};
static const struct regmap_config ak4643_regmap = {
@@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = {
.max_register = SPK_MS,
.reg_defaults = ak4643_reg,
.num_reg_defaults = ARRAY_SIZE(ak4643_reg),
+ .cache_type = REGCACHE_RBTREE,
};
static const struct regmap_config ak4648_regmap = {
@@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = {
.max_register = EQ_FBEQE,
.reg_defaults = ak4648_reg,
.num_reg_defaults = ARRAY_SIZE(ak4648_reg),
+ .cache_type = REGCACHE_RBTREE,
};
static const struct ak4642_drvdata ak4642_drvdata = {
diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
new file mode 100644
index 000000000000..cf0a39bb631a
--- /dev/null
+++ b/sound/soc/codecs/max98371.c
@@ -0,0 +1,441 @@
+/*
+ * max98371.c -- ALSA SoC Stereo MAX98371 driver
+ *
+ * Copyright 2015-16 Maxim Integrated Products
+ *
+ * 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/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "max98371.h"
+
+static const char *const monomix_text[] = {
+ "Left", "Right", "LeftRightDiv2",
+};
+
+static const char *const hpf_cutoff_txt[] = {
+ "Disable", "DC Block", "50Hz",
+ "100Hz", "200Hz", "400Hz", "800Hz",
+};
+
+static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0,
+ monomix_text);
+
+static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0,
+ hpf_cutoff_txt);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain,
+ 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0),
+ 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0),
+ 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0),
+ 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0),
+ 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0),
+ 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0),
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain,
+ 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0),
+ 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0),
+ 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0),
+ 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0),
+ 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0),
+ 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0),
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain,
+ 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
+ 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0),
+ 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0),
+ 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0),
+ 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
+ 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+);
+
+static const struct reg_default max98371_reg[] = {
+ { 0x01, 0x00 },
+ { 0x02, 0x00 },
+ { 0x03, 0x00 },
+ { 0x04, 0x00 },
+ { 0x05, 0x00 },
+ { 0x06, 0x00 },
+ { 0x07, 0x00 },
+ { 0x08, 0x00 },
+ { 0x09, 0x00 },
+ { 0x0A, 0x00 },
+ { 0x10, 0x06 },
+ { 0x11, 0x08 },
+ { 0x14, 0x80 },
+ { 0x15, 0x00 },
+ { 0x16, 0x00 },
+ { 0x18, 0x00 },
+ { 0x19, 0x00 },
+ { 0x1C, 0x00 },
+ { 0x1D, 0x00 },
+ { 0x1E, 0x00 },
+ { 0x1F, 0x00 },
+ { 0x20, 0x00 },
+ { 0x21, 0x00 },
+ { 0x22, 0x00 },
+ { 0x23, 0x00 },
+ { 0x24, 0x00 },
+ { 0x25, 0x00 },
+ { 0x26, 0x00 },
+ { 0x27, 0x00 },
+ { 0x28, 0x00 },
+ { 0x29, 0x00 },
+ { 0x2A, 0x00 },
+ { 0x2B, 0x00 },
+ { 0x2C, 0x00 },
+ { 0x2D, 0x00 },
+ { 0x2E, 0x0B },
+ { 0x31, 0x00 },
+ { 0x32, 0x18 },
+ { 0x33, 0x00 },
+ { 0x34, 0x00 },
+ { 0x36, 0x00 },
+ { 0x37, 0x00 },
+ { 0x38, 0x00 },
+ { 0x39, 0x00 },
+ { 0x3A, 0x00 },
+ { 0x3B, 0x00 },
+ { 0x3C, 0x00 },
+ { 0x3D, 0x00 },
+ { 0x3E, 0x00 },
+ { 0x3F, 0x00 },
+ { 0x40, 0x00 },
+ { 0x41, 0x00 },
+ { 0x42, 0x00 },
+ { 0x43, 0x00 },
+ { 0x4A, 0x00 },
+ { 0x4B, 0x00 },
+ { 0x4C, 0x00 },
+ { 0x4D, 0x00 },
+ { 0x4E, 0x00 },
+ { 0x50, 0x00 },
+ { 0x51, 0x00 },
+ { 0x55, 0x00 },
+ { 0x58, 0x00 },
+ { 0x59, 0x00 },
+ { 0x5C, 0x00 },
+ { 0xFF, 0x43 },
+};
+
+static bool max98371_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX98371_IRQ_CLEAR1:
+ case MAX98371_IRQ_CLEAR2:
+ case MAX98371_IRQ_CLEAR3:
+ case MAX98371_VERSION:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool max98371_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX98371_SOFT_RESET:
+ return false;
+ default:
+ return true;
+ }
+};
+
+static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv,
+ 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0),
+ 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0)
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv,
+ 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0),
+);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1);
+
+static const struct snd_kcontrol_new max98371_snd_controls[] = {
+ SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN,
+ MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0,
+ max98371_gain_tlv),
+ SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0,
+ (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv),
+ SOC_SINGLE_TLV("Speaker DHT Max Volume", MAX98371_GAIN,
+ 0, (1<<MAX98371_DHT_MAX_WIDTH)-1, 0,
+ max98371_dht_max_gain),
+ SOC_SINGLE_TLV("Speaker DHT Min Volume", MAX98371_DHT_GAIN,
+ 0, (1<<MAX98371_DHT_GAIN_WIDTH)-1, 0,
+ max98371_dht_min_gain),
+ SOC_SINGLE_TLV("Speaker DHT Rotation Volume", MAX98371_DHT_GAIN,
+ 0, (1<<MAX98371_DHT_ROT_WIDTH)-1, 0,
+ max98371_dht_rot_gain),
+ SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0),
+ SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0),
+ SOC_ENUM("Monomix Select", max98371_monomix),
+ SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff),
+};
+
+static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ dev_err(codec->dev, "DAI clock mode unsupported");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ val |= 0;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val |= MAX98371_DAI_RIGHT;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val |= MAX98371_DAI_LEFT;
+ break;
+ default:
+ dev_err(codec->dev, "DAI wrong mode unsupported");
+ return -EINVAL;
+ }
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MODE_MASK, val);
+ return 0;
+}
+
+static int max98371_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 max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec);
+ int blr_clk_ratio, ch_size, channels = params_channels(params);
+ int rate = params_rate(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16);
+ ch_size = 8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16);
+ ch_size = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32);
+ ch_size = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32);
+ ch_size = 32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* BCLK/LRCLK ratio calculation */
+ blr_clk_ratio = channels * ch_size;
+ switch (blr_clk_ratio) {
+ case 32:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_DAI_CLK,
+ MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32);
+ break;
+ case 48:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_DAI_CLK,
+ MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48);
+ break;
+ case 64:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_DAI_CLK,
+ MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (rate) {
+ case 32000:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32);
+ break;
+ case 44100:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44);
+ break;
+ case 48000:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48);
+ break;
+ case 88200:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88);
+ break;
+ case 96000:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* enabling both the RX channels*/
+ regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC,
+ MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1);
+ regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL,
+ MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK);
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0),
+ SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("SPK_OUT"),
+};
+
+static const struct snd_soc_dapm_route max98371_audio_map[] = {
+ {"DAC", NULL, "HiFi Playback"},
+ {"SPK_OUT", NULL, "DAC"},
+ {"SPK_OUT", NULL, "Global Enable"},
+};
+
+#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000
+#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static const struct snd_soc_dai_ops max98371_dai_ops = {
+ .set_fmt = max98371_dai_set_fmt,
+ .hw_params = max98371_dai_hw_params,
+};
+
+static struct snd_soc_dai_driver max98371_dai[] = {
+ {
+ .name = "max98371-aif1",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = MAX98371_FORMATS,
+ },
+ .ops = &max98371_dai_ops,
+ }
+};
+
+static const struct snd_soc_codec_driver max98371_codec = {
+ .controls = max98371_snd_controls,
+ .num_controls = ARRAY_SIZE(max98371_snd_controls),
+ .dapm_routes = max98371_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(max98371_audio_map),
+ .dapm_widgets = max98371_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets),
+};
+
+static const struct regmap_config max98371_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = MAX98371_VERSION,
+ .reg_defaults = max98371_reg,
+ .num_reg_defaults = ARRAY_SIZE(max98371_reg),
+ .volatile_reg = max98371_volatile_register,
+ .readable_reg = max98371_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int max98371_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct max98371_priv *max98371;
+ int ret, reg;
+
+ max98371 = devm_kzalloc(&i2c->dev,
+ sizeof(*max98371), GFP_KERNEL);
+ if (!max98371)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, max98371);
+ max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap);
+ if (IS_ERR(max98371->regmap)) {
+ ret = PTR_ERR(max98371->regmap);
+ dev_err(&i2c->dev,
+ "Failed to allocate regmap: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg);
+ if (ret < 0) {
+ dev_info(&i2c->dev, "device error %d\n", ret);
+ return ret;
+ }
+ dev_info(&i2c->dev, "device version %x\n", reg);
+
+ ret = snd_soc_register_codec(&i2c->dev, &max98371_codec,
+ max98371_dai, ARRAY_SIZE(max98371_dai));
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+ return ret;
+}
+
+static int max98371_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id max98371_i2c_id[] = {
+ { "max98371", 0 },
+};
+
+MODULE_DEVICE_TABLE(i2c, max98371_i2c_id);
+
+static const struct of_device_id max98371_of_match[] = {
+ { .compatible = "maxim,max98371", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, max98371_of_match);
+
+static struct i2c_driver max98371_i2c_driver = {
+ .driver = {
+ .name = "max98371",
+ .owner = THIS_MODULE,
+ .pm = NULL,
+ .of_match_table = of_match_ptr(max98371_of_match),
+ },
+ .probe = max98371_i2c_probe,
+ .remove = max98371_i2c_remove,
+ .id_table = max98371_i2c_id,
+};
+
+module_i2c_driver(max98371_i2c_driver);
+
+MODULE_AUTHOR("anish kumar <yesanishhere@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC MAX98371 driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h
new file mode 100644
index 000000000000..9f6330964d98
--- /dev/null
+++ b/sound/soc/codecs/max98371.h
@@ -0,0 +1,67 @@
+/*
+ * max98371.h -- MAX98371 ALSA SoC Audio driver
+ *
+ * Copyright 2011-2012 Maxim Integrated Products
+ *
+ * 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.
+ */
+
+#ifndef _MAX98371_H
+#define _MAX98371_H
+
+#define MAX98371_IRQ_CLEAR1 0x01
+#define MAX98371_IRQ_CLEAR2 0x02
+#define MAX98371_IRQ_CLEAR3 0x03
+#define MAX98371_DAI_CLK 0x10
+#define MAX98371_DAI_BSEL_MASK 0xF
+#define MAX98371_DAI_BSEL_32 2
+#define MAX98371_DAI_BSEL_48 3
+#define MAX98371_DAI_BSEL_64 4
+#define MAX98371_SPK_SR 0x11
+#define MAX98371_SPK_SR_MASK 0xF
+#define MAX98371_SPK_SR_32 6
+#define MAX98371_SPK_SR_44 7
+#define MAX98371_SPK_SR_48 8
+#define MAX98371_SPK_SR_88 10
+#define MAX98371_SPK_SR_96 11
+#define MAX98371_DAI_CHANNEL 0x15
+#define MAX98371_CHANNEL_MASK 0x3
+#define MAX98371_MONOMIX_SRC 0x18
+#define MAX98371_MONOMIX_CFG 0x19
+#define MAX98371_HPF 0x1C
+#define MAX98371_MONOMIX_SRC_MASK 0xFF
+#define MONOMIX_RX_0_1 ((0x1)<<(4))
+#define M98371_DAI_CHANNEL_I2S 0x3
+#define MAX98371_DIGITAL_GAIN 0x2D
+#define MAX98371_DIGITAL_GAIN_WIDTH 0x7
+#define MAX98371_GAIN 0x2E
+#define MAX98371_GAIN_SHIFT 0x4
+#define MAX98371_GAIN_WIDTH 0x4
+#define MAX98371_DHT_MAX_WIDTH 4
+#define MAX98371_FMT 0x14
+#define MAX98371_CHANSZ_WIDTH 6
+#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_FMT_MODE_MASK ((0x7)<<(3))
+#define MAX98371_DAI_LEFT ((0x1)<<(3))
+#define MAX98371_DAI_RIGHT ((0x2)<<(3))
+#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DHT 0x32
+#define MAX98371_DHT_STEP 0x3
+#define MAX98371_DHT_GAIN 0x31
+#define MAX98371_DHT_GAIN_WIDTH 0x4
+#define MAX98371_DHT_ROT_WIDTH 0x4
+#define MAX98371_SPK_ENABLE 0x4A
+#define MAX98371_GLOBAL_ENABLE 0x50
+#define MAX98371_SOFT_RESET 0x51
+#define MAX98371_VERSION 0xFF
+
+
+struct max98371_priv {
+ struct regmap *regmap;
+ struct snd_soc_codec *codec;
+};
+#endif
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index a1aaffc20862..f80cfe4d2ef2 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -276,6 +276,8 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic)
} else {
*mic = false;
regmap_write(rt298->regmap, RT298_SET_MIC1, 0x20);
+ regmap_update_bits(rt298->regmap,
+ RT298_CBJ_CTRL1, 0x0400, 0x0000);
}
} else {
regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf);
@@ -482,6 +484,26 @@ static int rt298_adc_event(struct snd_soc_dapm_widget *w,
snd_soc_update_bits(codec,
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
0x7080, 0x7000);
+ /* If MCLK doesn't exist, reset AD filter */
+ if (!(snd_soc_read(codec, RT298_VAD_CTRL) & 0x200)) {
+ pr_info("NO MCLK\n");
+ switch (nid) {
+ case RT298_ADC_IN1:
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x2, 0x2);
+ mdelay(10);
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x2, 0x0);
+ break;
+ case RT298_ADC_IN2:
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x4, 0x4);
+ mdelay(10);
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x4, 0x0);
+ break;
+ }
+ }
break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec,
@@ -520,30 +542,12 @@ static int rt298_mic1_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static int rt298_vref_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-
- switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
- snd_soc_update_bits(codec,
- RT298_CBJ_CTRL1, 0x0400, 0x0000);
- mdelay(50);
- break;
- default:
- return 0;
- }
-
- return 0;
-}
-
static const struct snd_soc_dapm_widget rt298_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY_S("HV", 1, RT298_POWER_CTRL1,
12, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("VREF", RT298_POWER_CTRL1,
- 0, 1, rt298_vref_event, SND_SOC_DAPM_PRE_PMU),
+ 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("BG_MBIAS", 1, RT298_POWER_CTRL2,
1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT298_POWER_CTRL2,
@@ -934,18 +938,9 @@ static int rt298_set_bias_level(struct snd_soc_codec *codec,
}
break;
- case SND_SOC_BIAS_ON:
- mdelay(30);
- snd_soc_update_bits(codec,
- RT298_CBJ_CTRL1, 0x0400, 0x0400);
-
- break;
-
case SND_SOC_BIAS_STANDBY:
snd_soc_write(codec,
RT298_SET_AUDIO_POWER, AC_PWRST_D3);
- snd_soc_update_bits(codec,
- RT298_CBJ_CTRL1, 0x0400, 0x0000);
break;
default:
diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h
index d66f8847b676..3638f3d61209 100644
--- a/sound/soc/codecs/rt298.h
+++ b/sound/soc/codecs/rt298.h
@@ -137,6 +137,7 @@
#define RT298_A_BIAS_CTRL2 0x02
#define RT298_POWER_CTRL1 0x03
#define RT298_A_BIAS_CTRL3 0x04
+#define RT298_D_FILTER_CTRL 0x05
#define RT298_POWER_CTRL2 0x08
#define RT298_I2S_CTRL1 0x09
#define RT298_I2S_CTRL2 0x0a
@@ -148,6 +149,7 @@
#define RT298_IRQ_CTRL 0x33
#define RT298_WIND_FILTER_CTRL 0x46
#define RT298_PLL_CTRL1 0x49
+#define RT298_VAD_CTRL 0x4e
#define RT298_CBJ_CTRL1 0x4f
#define RT298_CBJ_CTRL2 0x50
#define RT298_PLL_CTRL 0x63
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 60212266d5d1..da9483c1c6fb 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -1241,60 +1241,46 @@ static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source,
regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >>
RT5677_AD_STO1_CLK_SEL_SFT;
- if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
- asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
- return 1;
break;
case 10:
regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >>
RT5677_AD_STO2_CLK_SEL_SFT;
- if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
- asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
- return 1;
break;
case 9:
regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >>
RT5677_AD_STO3_CLK_SEL_SFT;
- if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
- asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
- return 1;
break;
case 8:
regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >>
RT5677_AD_STO4_CLK_SEL_SFT;
- if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
- asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
- return 1;
break;
case 7:
regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting);
asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >>
RT5677_AD_MONOL_CLK_SEL_SFT;
- if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
- asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
- return 1;
break;
case 6:
regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting);
asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >>
RT5677_AD_MONOR_CLK_SEL_SFT;
- if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
- asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
- return 1;
break;
default:
- break;
+ return 0;
}
+ if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
+ asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
+ return 1;
+
return 0;
}
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 39307ad41a34..b8d19b77bde9 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -4,6 +4,9 @@
* Copyright (C) 2015 Google, Inc.
* Copyright (c) 2013 Daniel Mack <zonque@gmail.com>
*
+ * TAS5721 support:
+ * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
+ *
* 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
@@ -57,6 +60,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg)
case TAS571X_CH1_VOL_REG:
case TAS571X_CH2_VOL_REG:
return priv->chip->vol_reg_size;
+ case TAS571X_INPUT_MUX_REG:
+ case TAS571X_CH4_SRC_SELECT_REG:
+ case TAS571X_PWM_MUX_REG:
+ return 4;
default:
return 1;
}
@@ -167,6 +174,23 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream,
TAS571X_SDI_FMT_MASK, val);
}
+static int tas571x_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 sysctl2;
+ int ret;
+
+ sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0;
+
+ ret = snd_soc_update_bits(codec,
+ TAS571X_SYS_CTRL_2_REG,
+ TAS571X_SYS_CTRL_2_SDN_MASK,
+ sysctl2);
+ usleep_range(1000, 2000);
+
+ return ret;
+}
+
static int tas571x_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -214,6 +238,7 @@ static int tas571x_set_bias_level(struct snd_soc_codec *codec,
static const struct snd_soc_dai_ops tas571x_dai_ops = {
.set_fmt = tas571x_set_dai_fmt,
.hw_params = tas571x_hw_params,
+ .digital_mute = tas571x_mute,
};
static const char *const tas5711_supply_names[] = {
@@ -241,6 +266,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = {
1, 1),
};
+static const struct regmap_range tas571x_readonly_regs_range[] = {
+ regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG),
+};
+
+static const struct regmap_range tas571x_volatile_regs_range[] = {
+ regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG),
+ regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG),
+};
+
+static const struct regmap_access_table tas571x_write_regs = {
+ .no_ranges = tas571x_readonly_regs_range,
+ .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range),
+};
+
+static const struct regmap_access_table tas571x_volatile_regs = {
+ .yes_ranges = tas571x_volatile_regs_range,
+ .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range),
+
+};
+
static const struct reg_default tas5711_reg_defaults[] = {
{ 0x04, 0x05 },
{ 0x05, 0x40 },
@@ -260,6 +305,8 @@ static const struct regmap_config tas5711_regmap_config = {
.reg_defaults = tas5711_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults),
.cache_type = REGCACHE_RBTREE,
+ .wr_table = &tas571x_write_regs,
+ .volatile_table = &tas571x_volatile_regs,
};
static const struct tas571x_chip tas5711_chip = {
@@ -314,6 +361,8 @@ static const struct regmap_config tas5717_regmap_config = {
.reg_defaults = tas5717_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults),
.cache_type = REGCACHE_RBTREE,
+ .wr_table = &tas571x_write_regs,
+ .volatile_table = &tas571x_volatile_regs,
};
/* This entry is reused for tas5719 as the software interface is identical. */
@@ -326,6 +375,77 @@ static const struct tas571x_chip tas5717_chip = {
.vol_reg_size = 2,
};
+static const char *const tas5721_supply_names[] = {
+ "AVDD",
+ "DVDD",
+ "DRVDD",
+ "PVDD",
+};
+
+static const struct snd_kcontrol_new tas5721_controls[] = {
+ SOC_SINGLE_TLV("Master Volume",
+ TAS571X_MVOL_REG,
+ 0, 0xff, 1, tas5711_volume_tlv),
+ SOC_DOUBLE_R_TLV("Speaker Volume",
+ TAS571X_CH1_VOL_REG,
+ TAS571X_CH2_VOL_REG,
+ 0, 0xff, 1, tas5711_volume_tlv),
+ SOC_DOUBLE("Speaker Switch",
+ TAS571X_SOFT_MUTE_REG,
+ TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT,
+ 1, 1),
+};
+
+static const struct reg_default tas5721_reg_defaults[] = {
+ {TAS571X_CLK_CTRL_REG, 0x6c},
+ {TAS571X_DEV_ID_REG, 0x00},
+ {TAS571X_ERR_STATUS_REG, 0x00},
+ {TAS571X_SYS_CTRL_1_REG, 0xa0},
+ {TAS571X_SDI_REG, 0x05},
+ {TAS571X_SYS_CTRL_2_REG, 0x40},
+ {TAS571X_SOFT_MUTE_REG, 0x00},
+ {TAS571X_MVOL_REG, 0xff},
+ {TAS571X_CH1_VOL_REG, 0x30},
+ {TAS571X_CH2_VOL_REG, 0x30},
+ {TAS571X_CH3_VOL_REG, 0x30},
+ {TAS571X_VOL_CFG_REG, 0x91},
+ {TAS571X_MODULATION_LIMIT_REG, 0x02},
+ {TAS571X_IC_DELAY_CH1_REG, 0xac},
+ {TAS571X_IC_DELAY_CH2_REG, 0x54},
+ {TAS571X_IC_DELAY_CH3_REG, 0xac},
+ {TAS571X_IC_DELAY_CH4_REG, 0x54},
+ {TAS571X_PWM_CH_SDN_GROUP_REG, 0x30},
+ {TAS571X_START_STOP_PERIOD_REG, 0x0f},
+ {TAS571X_OSC_TRIM_REG, 0x82},
+ {TAS571X_BKND_ERR_REG, 0x02},
+ {TAS571X_INPUT_MUX_REG, 0x17772},
+ {TAS571X_CH4_SRC_SELECT_REG, 0x4303},
+ {TAS571X_PWM_MUX_REG, 0x1021345},
+};
+
+static const struct regmap_config tas5721_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 32,
+ .max_register = 0xff,
+ .reg_read = tas571x_reg_read,
+ .reg_write = tas571x_reg_write,
+ .reg_defaults = tas5721_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(tas5721_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+ .wr_table = &tas571x_write_regs,
+ .volatile_table = &tas571x_volatile_regs,
+};
+
+
+static const struct tas571x_chip tas5721_chip = {
+ .supply_names = tas5721_supply_names,
+ .num_supply_names = ARRAY_SIZE(tas5721_supply_names),
+ .controls = tas5711_controls,
+ .num_controls = ARRAY_SIZE(tas5711_controls),
+ .regmap_config = &tas5721_regmap_config,
+ .vol_reg_size = 1,
+};
+
static const struct snd_soc_dapm_widget tas571x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0),
@@ -386,11 +506,10 @@ static int tas571x_i2c_probe(struct i2c_client *client,
i2c_set_clientdata(client, priv);
of_id = of_match_device(tas571x_of_match, dev);
- if (!of_id) {
- dev_err(dev, "Unknown device type\n");
- return -EINVAL;
- }
- priv->chip = of_id->data;
+ if (of_id)
+ priv->chip = of_id->data;
+ else
+ priv->chip = (void *) id->driver_data;
priv->mclk = devm_clk_get(dev, "mclk");
if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) {
@@ -445,10 +564,6 @@ static int tas571x_i2c_probe(struct i2c_client *client,
if (ret)
return ret;
- ret = regmap_update_bits(priv->regmap, TAS571X_SYS_CTRL_2_REG,
- TAS571X_SYS_CTRL_2_SDN_MASK, 0);
- if (ret)
- return ret;
memcpy(&priv->codec_driver, &tas571x_codec, sizeof(priv->codec_driver));
priv->codec_driver.controls = priv->chip->controls;
@@ -486,14 +601,16 @@ static const struct of_device_id tas571x_of_match[] = {
{ .compatible = "ti,tas5711", .data = &tas5711_chip, },
{ .compatible = "ti,tas5717", .data = &tas5717_chip, },
{ .compatible = "ti,tas5719", .data = &tas5717_chip, },
+ { .compatible = "ti,tas5721", .data = &tas5721_chip, },
{ }
};
MODULE_DEVICE_TABLE(of, tas571x_of_match);
static const struct i2c_device_id tas571x_i2c_id[] = {
- { "tas5711", 0 },
- { "tas5717", 0 },
- { "tas5719", 0 },
+ { "tas5711", (kernel_ulong_t) &tas5711_chip },
+ { "tas5717", (kernel_ulong_t) &tas5717_chip },
+ { "tas5719", (kernel_ulong_t) &tas5717_chip },
+ { "tas5721", (kernel_ulong_t) &tas5721_chip },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas571x_i2c_id);
diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h
index 0aee471232cd..cf800c364f0f 100644
--- a/sound/soc/codecs/tas571x.h
+++ b/sound/soc/codecs/tas571x.h
@@ -13,6 +13,10 @@
#define _TAS571X_H
/* device registers */
+#define TAS571X_CLK_CTRL_REG 0x00
+#define TAS571X_DEV_ID_REG 0x01
+#define TAS571X_ERR_STATUS_REG 0x02
+#define TAS571X_SYS_CTRL_1_REG 0x03
#define TAS571X_SDI_REG 0x04
#define TAS571X_SDI_FMT_MASK 0x0f
@@ -27,7 +31,25 @@
#define TAS571X_MVOL_REG 0x07
#define TAS571X_CH1_VOL_REG 0x08
#define TAS571X_CH2_VOL_REG 0x09
+#define TAS571X_CH3_VOL_REG 0x0a
+#define TAS571X_VOL_CFG_REG 0x0e
+#define TAS571X_MODULATION_LIMIT_REG 0x10
+#define TAS571X_IC_DELAY_CH1_REG 0x11
+#define TAS571X_IC_DELAY_CH2_REG 0x12
+#define TAS571X_IC_DELAY_CH3_REG 0x13
+#define TAS571X_IC_DELAY_CH4_REG 0x14
+#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */
+#define TAS571X_PWM_CH1_SDN_MASK (1<<0)
+#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1)
+#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2)
+#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3)
+
+#define TAS571X_START_STOP_PERIOD_REG 0x1a
#define TAS571X_OSC_TRIM_REG 0x1b
+#define TAS571X_BKND_ERR_REG 0x1c
+#define TAS571X_INPUT_MUX_REG 0x20
+#define TAS571X_CH4_SRC_SELECT_REG 0x21
+#define TAS571X_PWM_MUX_REG 0x25
#endif /* _TAS571X_H */
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
new file mode 100644
index 000000000000..f54fb46b77c2
--- /dev/null
+++ b/sound/soc/codecs/tas5720.c
@@ -0,0 +1,620 @@
+/*
+ * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
+ *
+ * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Andreas Dannenberg <dannenberg@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.
+ */
+
+#include <linux/module.h>
+#include <linux/errno.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/regulator/consumer.h>
+#include <linux/delay.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "tas5720.h"
+
+/* Define how often to check (and clear) the fault status register (in ms) */
+#define TAS5720_FAULT_CHECK_INTERVAL 200
+
+static const char * const tas5720_supply_names[] = {
+ "dvdd", /* Digital power supply. Connect to 3.3-V supply. */
+ "pvdd", /* Class-D amp and analog power supply (connected). */
+};
+
+#define TAS5720_NUM_SUPPLIES ARRAY_SIZE(tas5720_supply_names)
+
+struct tas5720_data {
+ struct snd_soc_codec *codec;
+ struct regmap *regmap;
+ struct i2c_client *tas5720_client;
+ struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES];
+ struct delayed_work fault_check_work;
+ unsigned int last_fault;
+};
+
+static int tas5720_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;
+ unsigned int rate = params_rate(params);
+ bool ssz_ds;
+ int ret;
+
+ switch (rate) {
+ case 44100:
+ case 48000:
+ ssz_ds = false;
+ break;
+ case 88200:
+ case 96000:
+ ssz_ds = true;
+ break;
+ default:
+ dev_err(codec->dev, "unsupported sample rate: %u\n", rate);
+ return -EINVAL;
+ }
+
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+ TAS5720_SSZ_DS, ssz_ds);
+ if (ret < 0) {
+ dev_err(codec->dev, "error setting sample rate: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int tas5720_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 serial_format;
+ int ret;
+
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+ dev_vdbg(codec->dev, "DAI Format master is not found\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+ SND_SOC_DAIFMT_INV_MASK)) {
+ case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
+ /* 1st data bit occur one BCLK cycle after the frame sync */
+ serial_format = TAS5720_SAIF_I2S;
+ break;
+ case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF):
+ /*
+ * Note that although the TAS5720 does not have a dedicated DSP
+ * mode it doesn't care about the LRCLK duty cycle during TDM
+ * operation. Therefore we can use the device's I2S mode with
+ * its delaying of the 1st data bit to receive DSP_A formatted
+ * data. See device datasheet for additional details.
+ */
+ serial_format = TAS5720_SAIF_I2S;
+ break;
+ case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF):
+ /*
+ * Similar to DSP_A, we can use the fact that the TAS5720 does
+ * not care about the LRCLK duty cycle during TDM to receive
+ * DSP_B formatted data in LEFTJ mode (no delaying of the 1st
+ * data bit).
+ */
+ serial_format = TAS5720_SAIF_LEFTJ;
+ break;
+ case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
+ /* No delay after the frame sync */
+ serial_format = TAS5720_SAIF_LEFTJ;
+ break;
+ default:
+ dev_vdbg(codec->dev, "DAI Format is not found\n");
+ return -EINVAL;
+ }
+
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+ TAS5720_SAIF_FORMAT_MASK,
+ serial_format);
+ if (ret < 0) {
+ dev_err(codec->dev, "error setting SAIF format: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int first_slot;
+ int ret;
+
+ if (!tx_mask) {
+ dev_err(codec->dev, "tx masks must not be 0\n");
+ return -EINVAL;
+ }
+
+ /*
+ * Determine the first slot that is being requested. We will only
+ * use the first slot that is found since the TAS5720 is a mono
+ * amplifier.
+ */
+ first_slot = __ffs(tx_mask);
+
+ if (first_slot > 7) {
+ dev_err(codec->dev, "slot selection out of bounds (%u)\n",
+ first_slot);
+ return -EINVAL;
+ }
+
+ /* Enable manual TDM slot selection (instead of I2C ID based) */
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+ TAS5720_TDM_CFG_SRC, TAS5720_TDM_CFG_SRC);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ /* Configure the TDM slot to process audio from */
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+ TAS5720_TDM_SLOT_SEL_MASK, first_slot);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ return 0;
+
+error_snd_soc_update_bits:
+ dev_err(codec->dev, "error configuring TDM mode: %d\n", ret);
+ return ret;
+}
+
+static int tas5720_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int ret;
+
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+ TAS5720_MUTE, mute ? TAS5720_MUTE : 0);
+ if (ret < 0) {
+ dev_err(codec->dev, "error (un-)muting device: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void tas5720_fault_check_work(struct work_struct *work)
+{
+ struct tas5720_data *tas5720 = container_of(work, struct tas5720_data,
+ fault_check_work.work);
+ struct device *dev = tas5720->codec->dev;
+ unsigned int curr_fault;
+ int ret;
+
+ ret = regmap_read(tas5720->regmap, TAS5720_FAULT_REG, &curr_fault);
+ if (ret < 0) {
+ dev_err(dev, "failed to read FAULT register: %d\n", ret);
+ goto out;
+ }
+
+ /* Check/handle all errors except SAIF clock errors */
+ curr_fault &= TAS5720_OCE | TAS5720_DCE | TAS5720_OTE;
+
+ /*
+ * Only flag errors once for a given occurrence. This is needed as
+ * the TAS5720 will take time clearing the fault condition internally
+ * during which we don't want to bombard the system with the same
+ * error message over and over.
+ */
+ if ((curr_fault & TAS5720_OCE) && !(tas5720->last_fault & TAS5720_OCE))
+ dev_crit(dev, "experienced an over current hardware fault\n");
+
+ if ((curr_fault & TAS5720_DCE) && !(tas5720->last_fault & TAS5720_DCE))
+ dev_crit(dev, "experienced a DC detection fault\n");
+
+ if ((curr_fault & TAS5720_OTE) && !(tas5720->last_fault & TAS5720_OTE))
+ dev_crit(dev, "experienced an over temperature fault\n");
+
+ /* Store current fault value so we can detect any changes next time */
+ tas5720->last_fault = curr_fault;
+
+ if (!curr_fault)
+ goto out;
+
+ /*
+ * Periodically toggle SDZ (shutdown bit) H->L->H to clear any latching
+ * faults as long as a fault condition persists. Always going through
+ * the full sequence no matter the first return value to minimizes
+ * chances for the device to end up in shutdown mode.
+ */
+ ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, 0);
+ if (ret < 0)
+ dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret);
+
+ ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, TAS5720_SDZ);
+ if (ret < 0)
+ dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret);
+
+out:
+ /* Schedule the next fault check at the specified interval */
+ schedule_delayed_work(&tas5720->fault_check_work,
+ msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL));
+}
+
+static int tas5720_codec_probe(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ unsigned int device_id;
+ int ret;
+
+ tas5720->codec = codec;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to read device ID register: %d\n",
+ ret);
+ goto probe_fail;
+ }
+
+ if (device_id != TAS5720_DEVICE_ID) {
+ dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n",
+ TAS5720_DEVICE_ID, device_id);
+ ret = -ENODEV;
+ goto probe_fail;
+ }
+
+ /* Set device to mute */
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+ TAS5720_MUTE, TAS5720_MUTE);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ /*
+ * Enter shutdown mode - our default when not playing audio - to
+ * minimize current consumption. On the TAS5720 there is no real down
+ * side doing so as all device registers are preserved and the wakeup
+ * of the codec is rather quick which we do using a dapm widget.
+ */
+ ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, 0);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ INIT_DELAYED_WORK(&tas5720->fault_check_work, tas5720_fault_check_work);
+
+ return 0;
+
+error_snd_soc_update_bits:
+ dev_err(codec->dev, "error configuring device registers: %d\n", ret);
+
+probe_fail:
+ regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ return ret;
+}
+
+static int tas5720_codec_remove(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ cancel_delayed_work_sync(&tas5720->fault_check_work);
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret < 0)
+ dev_err(codec->dev, "failed to disable supplies: %d\n", ret);
+
+ return ret;
+};
+
+static int tas5720_dac_event(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 tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ if (event & SND_SOC_DAPM_POST_PMU) {
+ /* Take TAS5720 out of shutdown mode */
+ ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, TAS5720_SDZ);
+ if (ret < 0) {
+ dev_err(codec->dev, "error waking codec: %d\n", ret);
+ return ret;
+ }
+
+ /*
+ * Observe codec shutdown-to-active time. The datasheet only
+ * lists a nominal value however just use-it as-is without
+ * additional padding to minimize the delay introduced in
+ * starting to play audio (actually there is other setup done
+ * by the ASoC framework that will provide additional delays,
+ * so we should always be safe).
+ */
+ msleep(25);
+
+ /* Turn on TAS5720 periodic fault checking/handling */
+ tas5720->last_fault = 0;
+ schedule_delayed_work(&tas5720->fault_check_work,
+ msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL));
+ } else if (event & SND_SOC_DAPM_PRE_PMD) {
+ /* Disable TAS5720 periodic fault checking/handling */
+ cancel_delayed_work_sync(&tas5720->fault_check_work);
+
+ /* Place TAS5720 in shutdown mode to minimize current draw */
+ ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, 0);
+ if (ret < 0) {
+ dev_err(codec->dev, "error shutting down codec: %d\n",
+ ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int tas5720_suspend(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ regcache_cache_only(tas5720->regmap, true);
+ regcache_mark_dirty(tas5720->regmap);
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret < 0)
+ dev_err(codec->dev, "failed to disable supplies: %d\n", ret);
+
+ return ret;
+}
+
+static int tas5720_resume(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(tas5720->regmap, false);
+
+ ret = regcache_sync(tas5720->regmap);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to sync regcache: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+#else
+#define tas5720_suspend NULL
+#define tas5720_resume NULL
+#endif
+
+static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case TAS5720_DEVICE_ID_REG:
+ case TAS5720_FAULT_REG:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config tas5720_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = TAS5720_MAX_REG,
+ .cache_type = REGCACHE_RBTREE,
+ .volatile_reg = tas5720_is_volatile_reg,
+};
+
+/*
+ * DAC analog gain. There are four discrete values to select from, ranging
+ * from 19.2 dB to 26.3dB.
+ */
+static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
+ 0x0, 0x0, TLV_DB_SCALE_ITEM(1920, 0, 0),
+ 0x1, 0x1, TLV_DB_SCALE_ITEM(2070, 0, 0),
+ 0x2, 0x2, TLV_DB_SCALE_ITEM(2350, 0, 0),
+ 0x3, 0x3, TLV_DB_SCALE_ITEM(2630, 0, 0),
+);
+
+/*
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
+ * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
+ * as per device datasheet.
+ */
+static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+
+static const struct snd_kcontrol_new tas5720_snd_controls[] = {
+ SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+ TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv),
+ SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
+ TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
+};
+
+static const struct snd_soc_dapm_widget tas5720_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas5720_dac_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_OUTPUT("OUT")
+};
+
+static const struct snd_soc_dapm_route tas5720_audio_map[] = {
+ { "DAC", NULL, "DAC IN" },
+ { "OUT", NULL, "DAC" },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
+ .probe = tas5720_codec_probe,
+ .remove = tas5720_codec_remove,
+ .suspend = tas5720_suspend,
+ .resume = tas5720_resume,
+
+ .controls = tas5720_snd_controls,
+ .num_controls = ARRAY_SIZE(tas5720_snd_controls),
+ .dapm_widgets = tas5720_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets),
+ .dapm_routes = tas5720_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map),
+};
+
+/* PCM rates supported by the TAS5720 driver */
+#define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+/* Formats supported by TAS5720 driver */
+#define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
+ .hw_params = tas5720_hw_params,
+ .set_fmt = tas5720_set_dai_fmt,
+ .set_tdm_slot = tas5720_set_dai_tdm_slot,
+ .digital_mute = tas5720_mute,
+};
+
+/*
+ * TAS5720 DAI structure
+ *
+ * Note that were are advertising .playback.channels_max = 2 despite this being
+ * a mono amplifier. The reason for that is that some serial ports such as TI's
+ * McASP module have a minimum number of channels (2) that they can output.
+ * Advertising more channels than we have will allow us to interface with such
+ * a serial port without really any negative side effects as the TAS5720 will
+ * simply ignore any extra channel(s) asides from the one channel that is
+ * configured to be played back.
+ */
+static struct snd_soc_dai_driver tas5720_dai[] = {
+ {
+ .name = "tas5720-amplifier",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = TAS5720_RATES,
+ .formats = TAS5720_FORMATS,
+ },
+ .ops = &tas5720_speaker_dai_ops,
+ },
+};
+
+static int tas5720_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &client->dev;
+ struct tas5720_data *data;
+ int ret;
+ int i;
+
+ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ data->tas5720_client = client;
+ data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config);
+ if (IS_ERR(data->regmap)) {
+ ret = PTR_ERR(data->regmap);
+ dev_err(dev, "failed to allocate register map: %d\n", ret);
+ return ret;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
+ data->supplies[i].supply = tas5720_supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
+ data->supplies);
+ if (ret != 0) {
+ dev_err(dev, "failed to request supplies: %d\n", ret);
+ return ret;
+ }
+
+ dev_set_drvdata(dev, data);
+
+ ret = snd_soc_register_codec(&client->dev,
+ &soc_codec_dev_tas5720,
+ tas5720_dai, ARRAY_SIZE(tas5720_dai));
+ if (ret < 0) {
+ dev_err(dev, "failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int tas5720_remove(struct i2c_client *client)
+{
+ struct device *dev = &client->dev;
+
+ snd_soc_unregister_codec(dev);
+
+ return 0;
+}
+
+static const struct i2c_device_id tas5720_id[] = {
+ { "tas5720", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, tas5720_id);
+
+#if IS_ENABLED(CONFIG_OF)
+static const struct of_device_id tas5720_of_match[] = {
+ { .compatible = "ti,tas5720", },
+ { },
+};
+MODULE_DEVICE_TABLE(of, tas5720_of_match);
+#endif
+
+static struct i2c_driver tas5720_i2c_driver = {
+ .driver = {
+ .name = "tas5720",
+ .of_match_table = of_match_ptr(tas5720_of_match),
+ },
+ .probe = tas5720_probe,
+ .remove = tas5720_remove,
+ .id_table = tas5720_id,
+};
+
+module_i2c_driver(tas5720_i2c_driver);
+
+MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>");
+MODULE_DESCRIPTION("TAS5720 Audio amplifier driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h
new file mode 100644
index 000000000000..3d077c779b12
--- /dev/null
+++ b/sound/soc/codecs/tas5720.h
@@ -0,0 +1,90 @@
+/*
+ * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
+ *
+ * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Andreas Dannenberg <dannenberg@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.
+ */
+
+#ifndef __TAS5720_H__
+#define __TAS5720_H__
+
+/* Register Address Map */
+#define TAS5720_DEVICE_ID_REG 0x00
+#define TAS5720_POWER_CTRL_REG 0x01
+#define TAS5720_DIGITAL_CTRL1_REG 0x02
+#define TAS5720_DIGITAL_CTRL2_REG 0x03
+#define TAS5720_VOLUME_CTRL_REG 0x04
+#define TAS5720_ANALOG_CTRL_REG 0x06
+#define TAS5720_FAULT_REG 0x08
+#define TAS5720_DIGITAL_CLIP2_REG 0x10
+#define TAS5720_DIGITAL_CLIP1_REG 0x11
+#define TAS5720_MAX_REG TAS5720_DIGITAL_CLIP1_REG
+
+/* TAS5720_DEVICE_ID_REG */
+#define TAS5720_DEVICE_ID 0x01
+
+/* TAS5720_POWER_CTRL_REG */
+#define TAS5720_DIG_CLIP_MASK GENMASK(7, 2)
+#define TAS5720_SLEEP BIT(1)
+#define TAS5720_SDZ BIT(0)
+
+/* TAS5720_DIGITAL_CTRL1_REG */
+#define TAS5720_HPF_BYPASS BIT(7)
+#define TAS5720_TDM_CFG_SRC BIT(6)
+#define TAS5720_SSZ_DS BIT(3)
+#define TAS5720_SAIF_RIGHTJ_24BIT (0x0)
+#define TAS5720_SAIF_RIGHTJ_20BIT (0x1)
+#define TAS5720_SAIF_RIGHTJ_18BIT (0x2)
+#define TAS5720_SAIF_RIGHTJ_16BIT (0x3)
+#define TAS5720_SAIF_I2S (0x4)
+#define TAS5720_SAIF_LEFTJ (0x5)
+#define TAS5720_SAIF_FORMAT_MASK GENMASK(2, 0)
+
+/* TAS5720_DIGITAL_CTRL2_REG */
+#define TAS5720_MUTE BIT(4)
+#define TAS5720_TDM_SLOT_SEL_MASK GENMASK(2, 0)
+
+/* TAS5720_ANALOG_CTRL_REG */
+#define TAS5720_PWM_RATE_6_3_FSYNC (0x0 << 4)
+#define TAS5720_PWM_RATE_8_4_FSYNC (0x1 << 4)
+#define TAS5720_PWM_RATE_10_5_FSYNC (0x2 << 4)
+#define TAS5720_PWM_RATE_12_6_FSYNC (0x3 << 4)
+#define TAS5720_PWM_RATE_14_7_FSYNC (0x4 << 4)
+#define TAS5720_PWM_RATE_16_8_FSYNC (0x5 << 4)
+#define TAS5720_PWM_RATE_20_10_FSYNC (0x6 << 4)
+#define TAS5720_PWM_RATE_24_12_FSYNC (0x7 << 4)
+#define TAS5720_PWM_RATE_MASK GENMASK(6, 4)
+#define TAS5720_ANALOG_GAIN_19_2DBV (0x0 << 2)
+#define TAS5720_ANALOG_GAIN_20_7DBV (0x1 << 2)
+#define TAS5720_ANALOG_GAIN_23_5DBV (0x2 << 2)
+#define TAS5720_ANALOG_GAIN_26_3DBV (0x3 << 2)
+#define TAS5720_ANALOG_GAIN_MASK GENMASK(3, 2)
+#define TAS5720_ANALOG_GAIN_SHIFT (0x2)
+
+/* TAS5720_FAULT_REG */
+#define TAS5720_OC_THRESH_100PCT (0x0 << 4)
+#define TAS5720_OC_THRESH_75PCT (0x1 << 4)
+#define TAS5720_OC_THRESH_50PCT (0x2 << 4)
+#define TAS5720_OC_THRESH_25PCT (0x3 << 4)
+#define TAS5720_OC_THRESH_MASK GENMASK(5, 4)
+#define TAS5720_CLKE BIT(3)
+#define TAS5720_OCE BIT(2)
+#define TAS5720_DCE BIT(1)
+#define TAS5720_OTE BIT(0)
+#define TAS5720_FAULT_MASK GENMASK(3, 0)
+
+/* TAS5720_DIGITAL_CLIP1_REG */
+#define TAS5720_CLIP1_MASK GENMASK(7, 2)
+#define TAS5720_CLIP1_SHIFT (0x2)
+
+#endif /* __TAS5720_H__ */
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index ee4def4f819f..3c5e1df01c19 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -28,6 +28,7 @@
#include <linux/i2c.h>
#include <linux/gpio.h>
#include <linux/regulator/consumer.h>
+#include <linux/acpi.h>
#include <linux/of.h>
#include <linux/of_gpio.h>
#include <linux/slab.h>
@@ -1280,10 +1281,19 @@ static const struct i2c_device_id aic31xx_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
+#ifdef CONFIG_ACPI
+static const struct acpi_device_id aic31xx_acpi_match[] = {
+ { "10TI3100", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match);
+#endif
+
static struct i2c_driver aic31xx_i2c_driver = {
.driver = {
.name = "tlv320aic31xx-codec",
.of_match_table = of_match_ptr(tlv320aic31xx_of_match),
+ .acpi_match_table = ACPI_PTR(aic31xx_acpi_match),
},
.probe = aic31xx_i2c_probe,
.remove = aic31xx_i2c_remove,
diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c
new file mode 100644
index 000000000000..59606cf3008f
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4-i2c.c
@@ -0,0 +1,74 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4-i2c.c
+ *
+ * Copyright 2011 NW Digital Radio
+ *
+ * Author: Jeremy McDermond <nh6z@nh6z.net>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * 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/i2c.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic32x4.h"
+
+static int aic32x4_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct regmap *regmap;
+ struct regmap_config config;
+
+ config = aic32x4_regmap_config;
+ config.reg_bits = 8;
+ config.val_bits = 8;
+
+ regmap = devm_regmap_init_i2c(i2c, &config);
+ return aic32x4_probe(&i2c->dev, regmap);
+}
+
+static int aic32x4_i2c_remove(struct i2c_client *i2c)
+{
+ return aic32x4_remove(&i2c->dev);
+}
+
+static const struct i2c_device_id aic32x4_i2c_id[] = {
+ { "tlv320aic32x4", 0 },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+
+static const struct of_device_id aic32x4_of_id[] = {
+ { .compatible = "ti,tlv320aic32x4", },
+ { /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
+static struct i2c_driver aic32x4_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic32x4",
+ .of_match_table = aic32x4_of_id,
+ },
+ .probe = aic32x4_i2c_probe,
+ .remove = aic32x4_i2c_remove,
+ .id_table = aic32x4_i2c_id,
+};
+
+module_i2c_driver(aic32x4_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C");
+MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c
new file mode 100644
index 000000000000..724fcdd491b2
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4-spi.c
@@ -0,0 +1,76 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4-spi.c
+ *
+ * Copyright 2011 NW Digital Radio
+ *
+ * Author: Jeremy McDermond <nh6z@nh6z.net>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * 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/spi/spi.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic32x4.h"
+
+static int aic32x4_spi_probe(struct spi_device *spi)
+{
+ struct regmap *regmap;
+ struct regmap_config config;
+
+ config = aic32x4_regmap_config;
+ config.reg_bits = 7;
+ config.pad_bits = 1;
+ config.val_bits = 8;
+ config.read_flag_mask = 0x01;
+
+ regmap = devm_regmap_init_spi(spi, &config);
+ return aic32x4_probe(&spi->dev, regmap);
+}
+
+static int aic32x4_spi_remove(struct spi_device *spi)
+{
+ return aic32x4_remove(&spi->dev);
+}
+
+static const struct spi_device_id aic32x4_spi_id[] = {
+ { "tlv320aic32x4", 0 },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(spi, aic32x4_spi_id);
+
+static const struct of_device_id aic32x4_of_id[] = {
+ { .compatible = "ti,tlv320aic32x4", },
+ { /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
+static struct spi_driver aic32x4_spi_driver = {
+ .driver = {
+ .name = "tlv320aic32x4",
+ .owner = THIS_MODULE,
+ .of_match_table = aic32x4_of_id,
+ },
+ .probe = aic32x4_spi_probe,
+ .remove = aic32x4_spi_remove,
+ .id_table = aic32x4_spi_id,
+};
+
+module_spi_driver(aic32x4_spi_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI");
+MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f2d3191961e1..85d4978d0384 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -30,7 +30,6 @@
#include <linux/pm.h>
#include <linux/gpio.h>
#include <linux/of_gpio.h>
-#include <linux/i2c.h>
#include <linux/cdev.h>
#include <linux/slab.h>
#include <linux/clk.h>
@@ -160,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = {
/* 48k rate */
{AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4},
{AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4},
- {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}
+ {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4},
+
+ /* 96k rate */
+ {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1},
};
static const struct snd_kcontrol_new hpl_output_mixer_controls[] = {
@@ -181,16 +183,71 @@ static const struct snd_kcontrol_new lor_output_mixer_controls[] = {
SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0),
};
-static const struct snd_kcontrol_new left_input_mixer_controls[] = {
- SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0),
- SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0),
- SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0),
+static const char * const resistor_text[] = {
+ "Off", "10 kOhm", "20 kOhm", "40 kOhm",
};
-static const struct snd_kcontrol_new right_input_mixer_controls[] = {
- SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0),
- SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0),
- SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0),
+/* Left mixer pins */
+static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text);
+
+static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text);
+
+static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum),
+};
+static const struct snd_kcontrol_new cml_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum),
+};
+static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum),
+};
+static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum),
+};
+
+/* Right mixer pins */
+static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text);
+static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text);
+
+static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum),
+};
+static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum),
+};
+static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum),
+};
+static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum),
};
static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
@@ -214,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
&lor_output_mixer_controls[0],
ARRAY_SIZE(lor_output_mixer_controls)),
SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
- &left_input_mixer_controls[0],
- ARRAY_SIZE(left_input_mixer_controls)),
- SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
- &right_input_mixer_controls[0],
- ARRAY_SIZE(right_input_mixer_controls)),
- SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+
SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0),
+ SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in1r_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in2r_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in3r_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in2l_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ cmr_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in1l_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in3l_to_rmixer_controls),
+
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+ SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in1l_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in2l_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in3l_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in1r_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ cml_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in2r_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in3r_to_lmixer_controls),
+
SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0),
SND_SOC_DAPM_OUTPUT("HPL"),
@@ -261,19 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
{"LOR Power", NULL, "LOR Output Mixer"},
{"LOR", NULL, "LOR Power"},
- /* Left input */
- {"Left Input Mixer", "IN1_L P Switch", "IN1_L"},
- {"Left Input Mixer", "IN2_L P Switch", "IN2_L"},
- {"Left Input Mixer", "IN3_L P Switch", "IN3_L"},
-
- {"Left ADC", NULL, "Left Input Mixer"},
-
/* Right Input */
- {"Right Input Mixer", "IN1_R P Switch", "IN1_R"},
- {"Right Input Mixer", "IN2_R P Switch", "IN2_R"},
- {"Right Input Mixer", "IN3_R P Switch", "IN3_R"},
-
- {"Right ADC", NULL, "Right Input Mixer"},
+ {"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"},
+ {"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"},
+ {"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"},
+ {"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"},
+
+ {"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"},
+ {"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"},
+ {"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"},
+ {"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"},
+
+ {"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"},
+ {"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"},
+ {"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"},
+ {"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"},
+
+ {"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"},
+ {"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"},
+ {"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"},
+ {"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"},
+
+ {"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"},
+ {"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"},
+ {"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"},
+ {"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"},
+
+ {"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"},
+ {"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"},
+ {"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"},
+ {"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"},
+
+ {"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"},
+ {"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"},
+ {"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"},
+ {"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"},
+
+ /* Left Input */
+ {"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"},
+ {"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"},
+ {"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"},
+ {"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"},
+
+ {"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"},
+ {"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"},
+ {"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"},
+ {"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"},
+
+ {"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"},
+ {"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"},
+ {"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"},
+ {"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"},
+
+ {"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"},
+ {"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"},
+ {"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"},
+ {"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"},
+
+ {"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"},
+ {"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"},
+ {"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"},
+ {"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"},
+
+ {"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"},
+ {"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"},
+ {"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"},
+ {"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"},
+
+ {"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"},
+ {"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"},
+ {"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"},
+ {"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"},
};
static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
@@ -287,14 +427,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
},
};
-static const struct regmap_config aic32x4_regmap = {
- .reg_bits = 8,
- .val_bits = 8,
-
+const struct regmap_config aic32x4_regmap_config = {
.max_register = AIC32X4_RMICPGAVOL,
.ranges = aic32x4_regmap_pages,
.num_ranges = ARRAY_SIZE(aic32x4_regmap_pages),
};
+EXPORT_SYMBOL(aic32x4_regmap_config);
static inline int aic32x4_get_divs(int mclk, int rate)
{
@@ -567,7 +705,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000
+#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000
#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
| SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -596,7 +734,7 @@ static struct snd_soc_dai_driver aic32x4_dai = {
.symmetric_rates = 1,
};
-static int aic32x4_probe(struct snd_soc_codec *codec)
+static int aic32x4_codec_probe(struct snd_soc_codec *codec)
{
struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
u32 tmp_reg;
@@ -655,7 +793,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
}
static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
- .probe = aic32x4_probe,
+ .probe = aic32x4_codec_probe,
.set_bias_level = aic32x4_set_bias_level,
.suspend_bias_off = true,
@@ -777,24 +915,22 @@ error_ldo:
return ret;
}
-static int aic32x4_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
+int aic32x4_probe(struct device *dev, struct regmap *regmap)
{
- struct aic32x4_pdata *pdata = i2c->dev.platform_data;
struct aic32x4_priv *aic32x4;
- struct device_node *np = i2c->dev.of_node;
+ struct aic32x4_pdata *pdata = dev->platform_data;
+ struct device_node *np = dev->of_node;
int ret;
- aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv),
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ aic32x4 = devm_kzalloc(dev, sizeof(struct aic32x4_priv),
GFP_KERNEL);
if (aic32x4 == NULL)
return -ENOMEM;
- aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap);
- if (IS_ERR(aic32x4->regmap))
- return PTR_ERR(aic32x4->regmap);
-
- i2c_set_clientdata(i2c, aic32x4);
+ dev_set_drvdata(dev, aic32x4);
if (pdata) {
aic32x4->power_cfg = pdata->power_cfg;
@@ -804,7 +940,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
} else if (np) {
ret = aic32x4_parse_dt(aic32x4, np);
if (ret) {
- dev_err(&i2c->dev, "Failed to parse DT node\n");
+ dev_err(dev, "Failed to parse DT node\n");
return ret;
}
} else {
@@ -814,71 +950,48 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
aic32x4->rstn_gpio = -1;
}
- aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk");
+ aic32x4->mclk = devm_clk_get(dev, "mclk");
if (IS_ERR(aic32x4->mclk)) {
- dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
+ dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
return PTR_ERR(aic32x4->mclk);
}
if (gpio_is_valid(aic32x4->rstn_gpio)) {
- ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio,
+ ret = devm_gpio_request_one(dev, aic32x4->rstn_gpio,
GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
if (ret != 0)
return ret;
}
- ret = aic32x4_setup_regulators(&i2c->dev, aic32x4);
+ ret = aic32x4_setup_regulators(dev, aic32x4);
if (ret) {
- dev_err(&i2c->dev, "Failed to setup regulators\n");
+ dev_err(dev, "Failed to setup regulators\n");
return ret;
}
- ret = snd_soc_register_codec(&i2c->dev,
+ ret = snd_soc_register_codec(dev,
&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
if (ret) {
- dev_err(&i2c->dev, "Failed to register codec\n");
+ dev_err(dev, "Failed to register codec\n");
aic32x4_disable_regulators(aic32x4);
return ret;
}
- i2c_set_clientdata(i2c, aic32x4);
-
return 0;
}
+EXPORT_SYMBOL(aic32x4_probe);
-static int aic32x4_i2c_remove(struct i2c_client *client)
+int aic32x4_remove(struct device *dev)
{
- struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client);
+ struct aic32x4_priv *aic32x4 = dev_get_drvdata(dev);
aic32x4_disable_regulators(aic32x4);
- snd_soc_unregister_codec(&client->dev);
+ snd_soc_unregister_codec(dev);
+
return 0;
}
-
-static const struct i2c_device_id aic32x4_i2c_id[] = {
- { "tlv320aic32x4", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
-
-static const struct of_device_id aic32x4_of_id[] = {
- { .compatible = "ti,tlv320aic32x4", },
- { /* senitel */ }
-};
-MODULE_DEVICE_TABLE(of, aic32x4_of_id);
-
-static struct i2c_driver aic32x4_i2c_driver = {
- .driver = {
- .name = "tlv320aic32x4",
- .of_match_table = aic32x4_of_id,
- },
- .probe = aic32x4_i2c_probe,
- .remove = aic32x4_i2c_remove,
- .id_table = aic32x4_i2c_id,
-};
-
-module_i2c_driver(aic32x4_i2c_driver);
+EXPORT_SYMBOL(aic32x4_remove);
MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver");
MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index 995f033a855d..a197dd51addc 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -10,6 +10,13 @@
#ifndef _TLV320AIC32X4_H
#define _TLV320AIC32X4_H
+struct device;
+struct regmap_config;
+
+extern const struct regmap_config aic32x4_regmap_config;
+int aic32x4_probe(struct device *dev, struct regmap *regmap);
+int aic32x4_remove(struct device *dev);
+
/* tlv320aic32x4 register space (in decimal to match datasheet) */
#define AIC32X4_PAGE1 128
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index bc3de2e844e6..1f7081043566 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
{
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- int ret;
+ int ret = 0;
switch (level) {
case SND_SOC_BIAS_ON:
@@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (priv->codec_powered)
+ if (priv->codec_powered) {
+ /* Select low power PLL in standby */
+ ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL,
+ 32768, 19200000);
break;
+ }
ret = twl6040_power(twl6040, 1);
if (ret)
- return ret;
+ break;
priv->codec_powered = 1;
@@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
break;
}
- return 0;
+ return ret;
}
static int twl6040_startup(struct snd_pcm_substream *substream,
@@ -983,9 +987,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i
if (mute) {
/* Power down drivers and DACs */
hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA |
- TWL6040_HFDRVENA);
+ TWL6040_HFDRVENA | TWL6040_HFSWENA);
hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA |
- TWL6040_HFDRVENA);
+ TWL6040_HFDRVENA | TWL6040_HFSWENA);
}
twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index fc164d69a557..f3109da24769 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3793,9 +3793,8 @@ static int wm8962_runtime_resume(struct device *dev)
ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies),
wm8962->supplies);
if (ret != 0) {
- dev_err(dev,
- "Failed to enable supplies: %d\n", ret);
- return ret;
+ dev_err(dev, "Failed to enable supplies: %d\n", ret);
+ goto disable_clock;
}
regcache_cache_only(wm8962->regmap, false);
@@ -3833,6 +3832,10 @@ static int wm8962_runtime_resume(struct device *dev)
msleep(5);
return 0;
+
+disable_clock:
+ clk_disable_unprepare(wm8962->pdata.mclk);
+ return ret;
}
static int wm8962_runtime_suspend(struct device *dev)
diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h
index 910aafd09d21..e63a318a3015 100644
--- a/sound/soc/codecs/wm8962.h
+++ b/sound/soc/codecs/wm8962.h
@@ -16,9 +16,9 @@
#include <asm/types.h>
#include <sound/soc.h>
-#define WM8962_SYSCLK_MCLK 1
-#define WM8962_SYSCLK_FLL 2
-#define WM8962_SYSCLK_PLL3 3
+#define WM8962_SYSCLK_MCLK 0
+#define WM8962_SYSCLK_FLL 1
+#define WM8962_SYSCLK_PLL3 2
#define WM8962_FLL 1
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 2389ab47e25f..466492b7d4f5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -643,6 +643,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match);
static struct platform_driver asoc_simple_card = {
.driver = {
.name = "asoc-simple-card",
+ .pm = &snd_soc_pm_ops,
.of_match_table = asoc_simple_of_match,
},
.probe = asoc_simple_card_probe,
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 132bb83f8e99..bc3c7b5ac752 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,7 @@
config SND_KIRKWOOD_SOC
tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
+ depends on HAS_DMA
help
Say Y or M if you want to add support for codecs attached to
the Kirkwood I2S interface. You will also need to select the
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index f7e789e97fbc..3abf51c07851 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -43,6 +43,7 @@ config SND_SOC_MT8173_RT5650_RT5676
depends on SND_SOC_MEDIATEK && I2C
select SND_SOC_RT5645
select SND_SOC_RT5677
+ select SND_SOC_HDMI_CODEC
help
This adds ASoC driver for Mediatek MT8173 boards
with the RT5650 and RT5676 codecs.
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 5c4c58c69c51..bb593926c62d 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -134,7 +134,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = {
enum {
DAI_LINK_PLAYBACK,
DAI_LINK_CAPTURE,
+ DAI_LINK_HDMI,
DAI_LINK_CODEC_I2S,
+ DAI_LINK_HDMI_I2S,
DAI_LINK_INTERCODEC
};
@@ -161,6 +163,16 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
.dynamic = 1,
.dpcm_capture = 1,
},
+ [DAI_LINK_HDMI] = {
+ .name = "HDMI",
+ .stream_name = "HDMI PCM",
+ .cpu_dai_name = "HDMI",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ },
/* Back End DAI links */
[DAI_LINK_CODEC_I2S] = {
@@ -177,6 +189,13 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
.dpcm_playback = 1,
.dpcm_capture = 1,
},
+ [DAI_LINK_HDMI_I2S] = {
+ .name = "HDMI BE",
+ .cpu_dai_name = "HDMIO",
+ .no_pcm = 1,
+ .codec_dai_name = "i2s-hifi",
+ .dpcm_playback = 1,
+ },
/* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */
[DAI_LINK_INTERCODEC] = {
.name = "rt5650_rt5676 intercodec",
@@ -251,6 +270,14 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
mt8173_rt5650_rt5676_dais[DAI_LINK_INTERCODEC].codec_of_node =
mt8173_rt5650_rt5676_codecs[1].of_node;
+ mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node =
+ of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 2);
+ if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node) {
+ dev_err(&pdev->dev,
+ "Property 'audio-codec' missing or invalid\n");
+ return -EINVAL;
+ }
+
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c
index bb09bb1b7f1c..a27a6673dbe3 100644
--- a/sound/soc/mediatek/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173-rt5650.c
@@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
struct snd_soc_codec *codec = runtime->codec_dais[0]->codec;
+ const char *codec_capture_dai = runtime->codec_dais[1]->name;
int ret;
rt5645_sel_asrc_clk_src(codec,
- RT5645_DA_STEREO_FILTER |
- RT5645_AD_STEREO_FILTER,
+ RT5645_DA_STEREO_FILTER,
RT5645_CLK_SEL_I2S1_ASRC);
+
+ if (!strcmp(codec_capture_dai, "rt5645-aif1")) {
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S1_ASRC);
+ } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) {
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S2_ASRC);
+ } else {
+ dev_warn(card->dev,
+ "Only one dai codec found in DTS, enabled rt5645 AD filter\n");
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S1_ASRC);
+ }
+
/* enable jack detection */
ret = snd_soc_card_jack_new(card, "Headset Jack",
SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
@@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = {
{
+ /* Playback */
+ .dai_name = "rt5645-aif1",
+ },
+ {
+ /* Capture */
.dai_name = "rt5645-aif1",
},
};
@@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = {
.cpu_dai_name = "I2S",
.no_pcm = 1,
.codecs = mt8173_rt5650_codecs,
- .num_codecs = 1,
+ .num_codecs = 2,
.init = mt8173_rt5650_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
@@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_rt5650_card;
struct device_node *platform_node;
+ struct device_node *np;
+ const char *codec_capture_dai;
int i, ret;
platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
+ mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node;
+
+ if (of_find_node_by_name(platform_node, "codec-capture")) {
+ np = of_get_child_by_name(pdev->dev.of_node, "codec-capture");
+ if (!np) {
+ dev_err(&pdev->dev,
+ "%s: Can't find codec-capture DT node\n",
+ __func__);
+ return -EINVAL;
+ }
+ ret = snd_soc_of_get_dai_name(np, &codec_capture_dai);
+ if (ret < 0) {
+ dev_err(&pdev->dev,
+ "%s codec_capture_dai name fail %d\n",
+ __func__, ret);
+ return ret;
+ }
+ mt8173_rt5650_codecs[1].dai_name = codec_capture_dai;
+ }
+
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index f1c58a2c12fb..2b5df2ef51a3 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -123,6 +123,7 @@
#define AFE_TDM_CON1_WLEN_32BIT (0x2 << 8)
#define AFE_TDM_CON1_MSB_ALIGNED (0x1 << 4)
#define AFE_TDM_CON1_1_BCK_DELAY (0x1 << 3)
+#define AFE_TDM_CON1_LRCK_INV (0x1 << 2)
#define AFE_TDM_CON1_BCK_INV (0x1 << 1)
#define AFE_TDM_CON1_EN (0x1 << 0)
@@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream,
runtime->rate * runtime->channels * 32);
val = AFE_TDM_CON1_BCK_INV |
+ AFE_TDM_CON1_LRCK_INV |
AFE_TDM_CON1_1_BCK_DELAY |
AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */
AFE_TDM_CON1_WLEN_32BIT |
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index c7563e230c7d..4a16e778966b 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -260,6 +260,10 @@ static void omap_st_on(struct omap_mcbsp *mcbsp)
if (mcbsp->pdata->enable_st_clock)
mcbsp->pdata->enable_st_clock(mcbsp->id, 1);
+ /* Disable Sidetone clock auto-gating for normal operation */
+ w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+ MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE));
+
/* Enable McBSP Sidetone */
w = MCBSP_READ(mcbsp, SSELCR);
MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN);
@@ -279,6 +283,10 @@ static void omap_st_off(struct omap_mcbsp *mcbsp)
w = MCBSP_READ(mcbsp, SSELCR);
MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN));
+ /* Enable Sidetone clock auto-gating to reduce power consumption */
+ w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+ MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE);
+
if (mcbsp->pdata->enable_st_clock)
mcbsp->pdata->enable_st_clock(mcbsp->id, 0);
}
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 99381a27295b..a84f677234f0 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
struct dma_chan *chan;
int err = 0;
+ memset(&config, 0x00, sizeof(config));
+
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
/* return if this is a bufferless transfer e.g.
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index ec522e94b0e2..b6cb9950f05d 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -133,3 +133,4 @@ module_platform_driver(mmp_driver);
MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
MODULE_DESCRIPTION("ALSA SoC Brownstone");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:brownstone-audio");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 5c8f9db50a47..d1661fa6ee08 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver);
MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mioa701-wm9713");
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 51e790d006f5..96df9b2d8fc4 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver);
MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
MODULE_DESCRIPTION("MMP Soc Audio DMA module");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mmp-pcm-audio");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index eca60c29791a..ca8b23f8c525 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver);
MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
MODULE_DESCRIPTION("MMP SSPA SoC Interface");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mmp-sspa-dai");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 4e74d9573f03..bcc81e920a67 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver);
MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:palm27x-asoc");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index da03fad1b9cd..3cad990dad2c 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa-ssp-dai");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index f3de615aacd7..9615e6de1306 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver);
MODULE_AUTHOR("Nicolas Pitre");
MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-ac97");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 9f390398d518..410d48b93031 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver);
MODULE_AUTHOR("Nicolas Pitre");
MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa-pcm-audio");
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index ddfe34434765..db000c6987a1 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -474,7 +474,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime)
struct lpass_data *drvdata =
snd_soc_platform_get_drvdata(soc_runtime->platform);
struct lpass_variant *v = drvdata->variant;
- int ret;
+ int ret = -EINVAL;
struct lpass_pcm_data *data;
size_t size = lpass_platform_pcm_hardware.buffer_bytes_max;
@@ -518,8 +518,10 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime)
data->wrdma_ch = v->alloc_dma_channel(drvdata,
SNDRV_PCM_STREAM_CAPTURE);
- if (data->wrdma_ch < 0)
+ if (data->wrdma_ch < 0) {
+ ret = data->wrdma_ch;
goto capture_alloc_err;
+ }
drvdata->substream[data->wrdma_ch] = csubstream;
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 606399de684d..49354d17ea55 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
*/
if (!count) {
clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT],
- parent_clk_name,
- (parent_clk_name) ?
- 0 : CLK_IS_ROOT, req_rate);
+ parent_clk_name, 0, req_rate);
if (!IS_ERR(clk)) {
adg->clkout[CLKOUT] = clk;
of_clk_add_provider(np, of_clk_src_simple_get, clk);
@@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
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,
+ parent_clk_name, 0,
req_rate);
if (!IS_ERR(clk)) {
adg->onecell.clks = adg->clkout;
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index 7658e8fd7bdc..6bc93cbb3049 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io,
size = ARRAY_SIZE(gen2_id_table_cmd);
}
- if (!entry)
- return 0xFF;
+ if ((!entry) || (size <= id)) {
+ struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io));
- if (size <= id)
- return 0xFF;
+ dev_err(dev, "unknown connection (%s[%d])\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ /* use non-prohibited SRS number as error */
+ return 0x00; /* SSI00 */
+ }
return entry[id];
}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index fc89a67258ca..a8f61d79333b 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -276,8 +276,9 @@ struct rsnd_mod {
/*
* status
*
- * 0xH0000CB0
+ * 0xH0000CBA
*
+ * A 0: probe 1: remove
* B 0: init 1: quit
* C 0: start 1: stop
*
@@ -287,19 +288,19 @@ struct rsnd_mod {
* H 0: fallback
* H 0: hw_params
*/
+#define __rsnd_mod_shift_probe 0
+#define __rsnd_mod_shift_remove 0
#define __rsnd_mod_shift_init 4
#define __rsnd_mod_shift_quit 4
#define __rsnd_mod_shift_start 8
#define __rsnd_mod_shift_stop 8
-#define __rsnd_mod_shift_probe 28 /* always called */
-#define __rsnd_mod_shift_remove 28 /* always called */
#define __rsnd_mod_shift_irq 28 /* always called */
#define __rsnd_mod_shift_pcm_new 28 /* always called */
#define __rsnd_mod_shift_fallback 28 /* always called */
#define __rsnd_mod_shift_hw_params 28 /* always called */
-#define __rsnd_mod_add_probe 0
-#define __rsnd_mod_add_remove 0
+#define __rsnd_mod_add_probe 1
+#define __rsnd_mod_add_remove -1
#define __rsnd_mod_add_init 1
#define __rsnd_mod_add_quit -1
#define __rsnd_mod_add_start 1
@@ -310,7 +311,7 @@ struct rsnd_mod {
#define __rsnd_mod_add_hw_params 0
#define __rsnd_mod_call_probe 0
-#define __rsnd_mod_call_remove 0
+#define __rsnd_mod_call_remove 1
#define __rsnd_mod_call_init 0
#define __rsnd_mod_call_quit 1
#define __rsnd_mod_call_start 0
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 15d6ffe8be74..e39f916d0f2f 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *priv)
i = 0;
for_each_child_of_node(node, np) {
+ if (!of_device_is_available(np))
+ goto skip;
+
src = rsnd_src_get(priv, i);
snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d",
@@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *priv)
if (ret)
goto rsnd_src_probe_done;
+skip:
i++;
}
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 1cf94d7fb9f4..ee7f15aa46fc 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos;
+ if (control_hdr->size != sizeof(*control_hdr)) {
+ dev_err(tplg->dev, "ASoC: invalid control size\n");
+ return -EINVAL;
+ }
+
switch (control_hdr->ops.info) {
case SND_SOC_TPLG_CTL_VOLSW:
case SND_SOC_TPLG_CTL_STROBE:
@@ -1476,6 +1481,8 @@ widget:
widget->dobj.type = SND_SOC_DOBJ_WIDGET;
widget->dobj.ops = tplg->ops;
widget->dobj.index = tplg->index;
+ kfree(template.sname);
+ kfree(template.name);
list_add(&widget->dobj.list, &tplg->comp->dobj_list);
return 0;
@@ -1499,10 +1506,17 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg,
for (i = 0; i < count; i++) {
widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos;
+ if (widget->size != sizeof(*widget)) {
+ dev_err(tplg->dev, "ASoC: invalid widget size\n");
+ return -EINVAL;
+ }
+
ret = soc_tplg_dapm_widget_create(tplg, widget);
- if (ret < 0)
+ if (ret < 0) {
dev_err(tplg->dev, "ASoC: failed to load widget %s\n",
widget->name);
+ return ret;
+ }
}
return 0;
@@ -1586,6 +1600,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
return snd_soc_register_dai(tplg->comp, dai_drv);
}
+/* create the FE DAI link */
static int soc_tplg_link_create(struct soc_tplg *tplg,
struct snd_soc_tplg_pcm *pcm)
{
@@ -1598,6 +1613,16 @@ static int soc_tplg_link_create(struct soc_tplg *tplg,
link->name = pcm->pcm_name;
link->stream_name = pcm->pcm_name;
+ link->id = pcm->pcm_id;
+
+ link->cpu_dai_name = pcm->dai_name;
+ link->codec_name = "snd-soc-dummy";
+ link->codec_dai_name = "snd-soc-dummy-dai";
+
+ /* enable DPCM */
+ link->dynamic = 1;
+ link->dpcm_playback = pcm->playback;
+ link->dpcm_capture = pcm->capture;
/* pass control to component driver for optional further init */
ret = soc_tplg_dai_link_load(tplg, link);
@@ -1639,8 +1664,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
if (tplg->pass != SOC_TPLG_PASS_PCM_DAI)
return 0;
- pcm = (struct snd_soc_tplg_pcm *)tplg->pos;
-
if (soc_tplg_check_elem_count(tplg,
sizeof(struct snd_soc_tplg_pcm), count,
hdr->payload_size, "PCM DAI")) {
@@ -1650,7 +1673,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
}
/* create the FE DAIs and DAI links */
+ pcm = (struct snd_soc_tplg_pcm *)tplg->pos;
for (i = 0; i < count; i++) {
+ if (pcm->size != sizeof(*pcm)) {
+ dev_err(tplg->dev, "ASoC: invalid pcm size\n");
+ return -EINVAL;
+ }
+
soc_tplg_pcm_create(tplg, pcm);
pcm++;
}
@@ -1670,6 +1699,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
return 0;
manifest = (struct snd_soc_tplg_manifest *)tplg->pos;
+ if (manifest->size != sizeof(*manifest)) {
+ dev_err(tplg->dev, "ASoC: invalid manifest size\n");
+ return -EINVAL;
+ }
+
tplg->pos += sizeof(struct snd_soc_tplg_manifest);
if (tplg->comp && tplg->ops && tplg->ops->manifest)
@@ -1686,6 +1720,14 @@ static int soc_valid_header(struct soc_tplg *tplg,
if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size)
return 0;
+ if (hdr->size != sizeof(*hdr)) {
+ dev_err(tplg->dev,
+ "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n",
+ hdr->type, soc_tplg_get_hdr_offset(tplg),
+ tplg->fw->size);
+ return -EINVAL;
+ }
+
/* big endian firmware objects not supported atm */
if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) {
dev_err(tplg->dev,
diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c
index 39bcefe5eea0..488ef4ed8fba 100644
--- a/sound/soc/sti/sti_uniperif.c
+++ b/sound/soc/sti/sti_uniperif.c
@@ -11,6 +11,142 @@
#include "uniperif.h"
/*
+ * User frame size shall be 2, 4, 6 or 8 32-bits words length
+ * (i.e. 8, 16, 24 or 32 bytes)
+ * This constraint comes from allowed values for
+ * UNIPERIF_I2S_FMT_NUM_CH register
+ */
+#define UNIPERIF_MAX_FRAME_SZ 0x20
+#define UNIPERIF_ALLOWED_FRAME_SZ (0x08 | 0x10 | 0x18 | UNIPERIF_MAX_FRAME_SZ)
+
+int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots,
+ int slot_width)
+{
+ struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+ struct uniperif *uni = priv->dai_data.uni;
+ int i, frame_size, avail_slots;
+
+ if (!UNIPERIF_TYPE_IS_TDM(uni)) {
+ dev_err(uni->dev, "cpu dai not in tdm mode\n");
+ return -EINVAL;
+ }
+
+ /* store info in unip context */
+ uni->tdm_slot.slots = slots;
+ uni->tdm_slot.slot_width = slot_width;
+ /* unip is unidirectionnal */
+ uni->tdm_slot.mask = (tx_mask != 0) ? tx_mask : rx_mask;
+
+ /* number of available timeslots */
+ for (i = 0, avail_slots = 0; i < uni->tdm_slot.slots; i++) {
+ if ((uni->tdm_slot.mask >> i) & 0x01)
+ avail_slots++;
+ }
+ uni->tdm_slot.avail_slots = avail_slots;
+
+ /* frame size in bytes */
+ frame_size = uni->tdm_slot.avail_slots * uni->tdm_slot.slot_width / 8;
+
+ /* check frame size is allowed */
+ if ((frame_size > UNIPERIF_MAX_FRAME_SZ) ||
+ (frame_size & ~(int)UNIPERIF_ALLOWED_FRAME_SZ)) {
+ dev_err(uni->dev, "frame size not allowed: %d bytes\n",
+ frame_size);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct uniperif *uni = rule->private;
+ struct snd_interval t;
+
+ t.min = uni->tdm_slot.avail_slots;
+ t.max = uni->tdm_slot.avail_slots;
+ t.openmin = 0;
+ t.openmax = 0;
+ t.integer = 0;
+
+ return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct uniperif *uni = rule->private;
+ struct snd_mask *maskp = hw_param_mask(params, rule->var);
+ u64 format;
+
+ switch (uni->tdm_slot.slot_width) {
+ case 16:
+ format = SNDRV_PCM_FMTBIT_S16_LE;
+ break;
+ case 32:
+ format = SNDRV_PCM_FMTBIT_S32_LE;
+ break;
+ default:
+ dev_err(uni->dev, "format not supported: %d bits\n",
+ uni->tdm_slot.slot_width);
+ return -EINVAL;
+ }
+
+ maskp->bits[0] &= (u_int32_t)format;
+ maskp->bits[1] &= (u_int32_t)(format >> 32);
+ /* clear remaining indexes */
+ memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX - 64) / 8);
+
+ if (!maskp->bits[0] && !maskp->bits[1])
+ return -EINVAL;
+
+ return 0;
+}
+
+int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni,
+ unsigned int *word_pos)
+{
+ int slot_width = uni->tdm_slot.slot_width / 8;
+ int slots_num = uni->tdm_slot.slots;
+ unsigned int slots_mask = uni->tdm_slot.mask;
+ int i, j, k;
+ unsigned int word16_pos[4];
+
+ /* word16_pos:
+ * word16_pos[0] = WORDX_LSB
+ * word16_pos[1] = WORDX_MSB,
+ * word16_pos[2] = WORDX+1_LSB
+ * word16_pos[3] = WORDX+1_MSB
+ */
+
+ /* set unip word position */
+ for (i = 0, j = 0, k = 0; (i < slots_num) && (k < WORD_MAX); i++) {
+ if ((slots_mask >> i) & 0x01) {
+ word16_pos[j] = i * slot_width;
+
+ if (slot_width == 4) {
+ word16_pos[j + 1] = word16_pos[j] + 2;
+ j++;
+ }
+ j++;
+
+ if (j > 3) {
+ word_pos[k] = word16_pos[1] |
+ (word16_pos[0] << 8) |
+ (word16_pos[3] << 16) |
+ (word16_pos[2] << 24);
+ j = 0;
+ k++;
+ }
+ }
+ }
+
+ return 0;
+}
+
+/*
* sti_uniperiph_dai_create_ctrl
* This function is used to create Ctrl associated to DAI but also pcm device.
* Request is done by front end to associate ctrl with pcm device id
@@ -45,10 +181,16 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
+ struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+ struct uniperif *uni = priv->dai_data.uni;
struct snd_dmaengine_dai_dma_data *dma_data;
int transfer_size;
- transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES;
+ if (uni->info->type == SND_ST_UNIPERIF_TYPE_TDM)
+ /* transfer size = user frame size (in 32-bits FIFO cell) */
+ transfer_size = snd_soc_params_to_frame_size(params) / 32;
+ else
+ transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES;
dma_data = snd_soc_dai_get_dma_data(dai, substream);
dma_data->maxburst = transfer_size;
diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h
index f0fd5a9944e9..eb9933c62ad6 100644
--- a/sound/soc/sti/uniperif.h
+++ b/sound/soc/sti/uniperif.h
@@ -25,7 +25,7 @@
writel_relaxed((((value) & mask) << shift), ip->base + offset)
/*
- * AUD_UNIPERIF_SOFT_RST reg
+ * UNIPERIF_SOFT_RST reg
*/
#define UNIPERIF_SOFT_RST_OFFSET(ip) 0x0000
@@ -50,7 +50,7 @@
UNIPERIF_SOFT_RST_SOFT_RST_MASK(ip))
/*
- * AUD_UNIPERIF_FIFO_DATA reg
+ * UNIPERIF_FIFO_DATA reg
*/
#define UNIPERIF_FIFO_DATA_OFFSET(ip) 0x0004
@@ -58,7 +58,7 @@
writel_relaxed(value, ip->base + UNIPERIF_FIFO_DATA_OFFSET(ip))
/*
- * AUD_UNIPERIF_CHANNEL_STA_REGN reg
+ * UNIPERIF_CHANNEL_STA_REGN reg
*/
#define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n))
@@ -105,7 +105,7 @@
writel_relaxed(value, ip->base + UNIPERIF_CHANNEL_STA_REG5_OFFSET(ip))
/*
- * AUD_UNIPERIF_ITS reg
+ * UNIPERIF_ITS reg
*/
#define UNIPERIF_ITS_OFFSET(ip) 0x000C
@@ -143,7 +143,7 @@
0 : (BIT(UNIPERIF_ITS_UNDERFLOW_REC_FAILED_SHIFT(ip))))
/*
- * AUD_UNIPERIF_ITS_BCLR reg
+ * UNIPERIF_ITS_BCLR reg
*/
/* FIFO_ERROR */
@@ -160,7 +160,7 @@
writel_relaxed(value, ip->base + UNIPERIF_ITS_BCLR_OFFSET(ip))
/*
- * AUD_UNIPERIF_ITM reg
+ * UNIPERIF_ITM reg
*/
#define UNIPERIF_ITM_OFFSET(ip) 0x0018
@@ -188,7 +188,7 @@
0 : (BIT(UNIPERIF_ITM_UNDERFLOW_REC_FAILED_SHIFT(ip))))
/*
- * AUD_UNIPERIF_ITM_BCLR reg
+ * UNIPERIF_ITM_BCLR reg
*/
#define UNIPERIF_ITM_BCLR_OFFSET(ip) 0x001c
@@ -213,7 +213,7 @@
UNIPERIF_ITM_BCLR_DMA_ERROR_MASK(ip))
/*
- * AUD_UNIPERIF_ITM_BSET reg
+ * UNIPERIF_ITM_BSET reg
*/
#define UNIPERIF_ITM_BSET_OFFSET(ip) 0x0020
@@ -767,7 +767,7 @@
SET_UNIPERIF_REG(ip, \
UNIPERIF_CTRL_OFFSET(ip), \
UNIPERIF_CTRL_READER_OUT_SEL_SHIFT(ip), \
- CORAUD_UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1)
+ UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1)
/* UNDERFLOW_REC_WINDOW */
#define UNIPERIF_CTRL_UNDERFLOW_REC_WINDOW_SHIFT(ip) 20
@@ -1046,7 +1046,7 @@
UNIPERIF_STATUS_1_UNDERFLOW_DURATION_MASK(ip), value)
/*
- * AUD_UNIPERIF_CHANNEL_STA_REGN reg
+ * UNIPERIF_CHANNEL_STA_REGN reg
*/
#define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n))
@@ -1057,7 +1057,7 @@
UNIPERIF_CHANNEL_STA_REGN(ip, n))
/*
- * AUD_UNIPERIF_USER_VALIDITY reg
+ * UNIPERIF_USER_VALIDITY reg
*/
#define UNIPERIF_USER_VALIDITY_OFFSET(ip) 0x0090
@@ -1101,12 +1101,136 @@
UNIPERIF_DBG_STANDBY_LEFT_SP_MASK(ip), value)
/*
+ * UNIPERIF_TDM_ENABLE
+ */
+#define UNIPERIF_TDM_ENABLE_OFFSET(ip) 0x0118
+#define GET_UNIPERIF_TDM_ENABLE(ip) \
+ readl_relaxed(ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip))
+#define SET_UNIPERIF_TDM_ENABLE(ip, value) \
+ writel_relaxed(value, ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip))
+
+/* TDM_ENABLE */
+#define UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip) 0x0
+#define UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip) 0x1
+#define GET_UNIPERIF_TDM_ENABLE_EN_TDM(ip) \
+ GET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_ENABLE_OFFSET(ip), \
+ UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \
+ UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip))
+#define SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(ip) \
+ SET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_ENABLE_OFFSET(ip), \
+ UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \
+ UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 1)
+#define SET_UNIPERIF_TDM_ENABLE_TDM_DISABLE(ip) \
+ SET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_ENABLE_OFFSET(ip), \
+ UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \
+ UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 0)
+
+/*
+ * UNIPERIF_TDM_FS_REF_FREQ
+ */
+#define UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip) 0x011c
+#define GET_UNIPERIF_TDM_FS_REF_FREQ(ip) \
+ readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ(ip, value) \
+ writel_relaxed(value, ip->base + \
+ UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip))
+
+/* REF_FREQ */
+#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip) 0x0
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) 0
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) 1
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) 2
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) 3
+#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip) 0x3
+#define GET_UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ(ip) \
+ GET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) \
+ SET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+ VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) \
+ SET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+ VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) \
+ SET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+ VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) \
+ SET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+ UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+ VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip))
+
+/*
+ * UNIPERIF_TDM_FS_REF_DIV
+ */
+#define UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip) 0x0120
+#define GET_UNIPERIF_TDM_FS_REF_DIV(ip) \
+ readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip))
+#define SET_UNIPERIF_TDM_FS_REF_DIV(ip, value) \
+ writel_relaxed(value, ip->base + \
+ UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip))
+
+/* NUM_TIMESLOT */
+#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip) 0x0
+#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip) 0xff
+#define GET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip) \
+ GET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \
+ UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \
+ UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip))
+#define SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip, value) \
+ SET_UNIPERIF_REG(ip, \
+ UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \
+ UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \
+ UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip), value)
+
+/*
+ * UNIPERIF_TDM_WORD_POS_X_Y
+ * 32 bits of UNIPERIF_TDM_WORD_POS_X_Y register shall be set in 1 shot
+ */
+#define UNIPERIF_TDM_WORD_POS_1_2_OFFSET(ip) 0x013c
+#define UNIPERIF_TDM_WORD_POS_3_4_OFFSET(ip) 0x0140
+#define UNIPERIF_TDM_WORD_POS_5_6_OFFSET(ip) 0x0144
+#define UNIPERIF_TDM_WORD_POS_7_8_OFFSET(ip) 0x0148
+#define GET_UNIPERIF_TDM_WORD_POS(ip, words) \
+ readl_relaxed(ip->base + UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip))
+#define SET_UNIPERIF_TDM_WORD_POS(ip, words, value) \
+ writel_relaxed(value, ip->base + \
+ UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip))
+/*
* uniperipheral IP capabilities
*/
#define UNIPERIF_FIFO_SIZE 70 /* FIFO is 70 cells deep */
#define UNIPERIF_FIFO_FRAMES 4 /* FDMA trigger limit in frames */
+#define UNIPERIF_TYPE_IS_HDMI(p) \
+ ((p)->info->type == SND_ST_UNIPERIF_TYPE_HDMI)
+#define UNIPERIF_TYPE_IS_PCM(p) \
+ ((p)->info->type == SND_ST_UNIPERIF_TYPE_PCM)
+#define UNIPERIF_TYPE_IS_SPDIF(p) \
+ ((p)->info->type == SND_ST_UNIPERIF_TYPE_SPDIF)
+#define UNIPERIF_TYPE_IS_IEC958(p) \
+ (UNIPERIF_TYPE_IS_HDMI(p) || \
+ UNIPERIF_TYPE_IS_SPDIF(p))
+#define UNIPERIF_TYPE_IS_TDM(p) \
+ ((p)->info->type == SND_ST_UNIPERIF_TYPE_TDM)
+
/*
* Uniperipheral IP revisions
*/
@@ -1125,10 +1249,11 @@ enum uniperif_version {
};
enum uniperif_type {
- SND_ST_UNIPERIF_PLAYER_TYPE_NONE,
- SND_ST_UNIPERIF_PLAYER_TYPE_HDMI,
- SND_ST_UNIPERIF_PLAYER_TYPE_PCM,
- SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF
+ SND_ST_UNIPERIF_TYPE_NONE,
+ SND_ST_UNIPERIF_TYPE_HDMI,
+ SND_ST_UNIPERIF_TYPE_PCM,
+ SND_ST_UNIPERIF_TYPE_SPDIF,
+ SND_ST_UNIPERIF_TYPE_TDM
};
enum uniperif_state {
@@ -1145,9 +1270,17 @@ enum uniperif_iec958_encoding_mode {
UNIPERIF_IEC958_ENCODING_MODE_ENCODED
};
+enum uniperif_word_pos {
+ WORD_1_2,
+ WORD_3_4,
+ WORD_5_6,
+ WORD_7_8,
+ WORD_MAX
+};
+
struct uniperif_info {
int id; /* instance value of the uniperipheral IP */
- enum uniperif_type player_type;
+ enum uniperif_type type;
int underflow_enabled; /* Underflow recovery mode */
};
@@ -1156,12 +1289,20 @@ struct uniperif_iec958_settings {
struct snd_aes_iec958 iec958;
};
+struct dai_tdm_slot {
+ unsigned int mask;
+ int slots;
+ int slot_width;
+ unsigned int avail_slots;
+};
+
struct uniperif {
/* System information */
struct uniperif_info *info;
struct device *dev;
int ver; /* IP version, used by register access macros */
struct regmap_field *clk_sel;
+ struct regmap_field *valid_sel;
/* capabilities */
const struct snd_pcm_hardware *hw;
@@ -1192,6 +1333,7 @@ struct uniperif {
/* dai properties */
unsigned int daifmt;
+ struct dai_tdm_slot tdm_slot;
/* DAI callbacks */
const struct snd_soc_dai_ops *dai_ops;
@@ -1209,6 +1351,28 @@ struct sti_uniperiph_data {
struct sti_uniperiph_dai dai_data;
};
+static const struct snd_pcm_hardware uni_tdm_hw = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID,
+
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE,
+
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 8000,
+ .rate_max = 48000,
+
+ .channels_min = 1,
+ .channels_max = 32,
+
+ .periods_min = 2,
+ .periods_max = 10,
+
+ .period_bytes_min = 128,
+ .period_bytes_max = 64 * PAGE_SIZE,
+ .buffer_bytes_max = 256 * PAGE_SIZE
+};
+
/* uniperiph player*/
int uni_player_init(struct platform_device *pdev,
struct uniperif *uni_player);
@@ -1226,4 +1390,28 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai);
+static inline int sti_uniperiph_get_user_frame_size(
+ struct snd_pcm_runtime *runtime)
+{
+ return (runtime->channels * snd_pcm_format_width(runtime->format) / 8);
+}
+
+static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni)
+{
+ return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8);
+}
+
+int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots,
+ int slot_width);
+
+int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni,
+ unsigned int *word_pos);
+
+int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule);
+
+int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule);
+
#endif
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 7aca6b92f718..ee1c7c245bc7 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -21,23 +21,14 @@
/* sys config registers definitions */
#define SYS_CFG_AUDIO_GLUE 0xA4
-#define SYS_CFG_AUDI0_GLUE_PCM_CLKX 8
/*
* Driver specific types.
*/
-#define UNIPERIF_PLAYER_TYPE_IS_HDMI(p) \
- ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_HDMI)
-#define UNIPERIF_PLAYER_TYPE_IS_PCM(p) \
- ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_PCM)
-#define UNIPERIF_PLAYER_TYPE_IS_SPDIF(p) \
- ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF)
-#define UNIPERIF_PLAYER_TYPE_IS_IEC958(p) \
- (UNIPERIF_PLAYER_TYPE_IS_HDMI(p) || \
- UNIPERIF_PLAYER_TYPE_IS_SPDIF(p))
#define UNIPERIF_PLAYER_CLK_ADJ_MIN -999999
#define UNIPERIF_PLAYER_CLK_ADJ_MAX 1000000
+#define UNIPERIF_PLAYER_I2S_OUT 1 /* player id connected to I2S/TDM TX bus */
/*
* Note: snd_pcm_hardware is linked to DMA controller but is declared here to
@@ -444,18 +435,11 @@ static int uni_player_prepare_pcm(struct uniperif *player,
/* Force slot width to 32 in I2S mode (HW constraint) */
if ((player->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
- SND_SOC_DAIFMT_I2S) {
+ SND_SOC_DAIFMT_I2S)
slot_width = 32;
- } else {
- switch (runtime->format) {
- case SNDRV_PCM_FORMAT_S16_LE:
- slot_width = 16;
- break;
- default:
- slot_width = 32;
- break;
- }
- }
+ else
+ slot_width = snd_pcm_format_width(runtime->format);
+
output_frame_size = slot_width * runtime->channels;
clk_div = player->mclk / runtime->rate;
@@ -530,7 +514,6 @@ static int uni_player_prepare_pcm(struct uniperif *player,
SET_UNIPERIF_CONFIG_ONE_BIT_AUD_DISABLE(player);
SET_UNIPERIF_I2S_FMT_ORDER_MSB(player);
- SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(player);
/* No iec958 formatting as outputting to DAC */
SET_UNIPERIF_CTRL_SPDIF_FMT_OFF(player);
@@ -538,6 +521,55 @@ static int uni_player_prepare_pcm(struct uniperif *player,
return 0;
}
+static int uni_player_prepare_tdm(struct uniperif *player,
+ struct snd_pcm_runtime *runtime)
+{
+ int tdm_frame_size; /* unip tdm frame size in bytes */
+ int user_frame_size; /* user tdm frame size in bytes */
+ /* default unip TDM_WORD_POS_X_Y */
+ unsigned int word_pos[4] = {
+ 0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A};
+ int freq, ret;
+
+ tdm_frame_size =
+ sti_uniperiph_get_unip_tdm_frame_size(player);
+ user_frame_size =
+ sti_uniperiph_get_user_frame_size(runtime);
+
+ /* fix 16/0 format */
+ SET_UNIPERIF_CONFIG_MEM_FMT_16_0(player);
+ SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(player);
+
+ /* number of words inserted on the TDM line */
+ SET_UNIPERIF_I2S_FMT_NUM_CH(player, user_frame_size / 4 / 2);
+
+ SET_UNIPERIF_I2S_FMT_ORDER_MSB(player);
+ SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(player);
+
+ /* Enable the tdm functionality */
+ SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(player);
+
+ /* number of 8 bits timeslots avail in unip tdm frame */
+ SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(player, tdm_frame_size);
+
+ /* set the timeslot allocation for words in FIFO */
+ sti_uniperiph_get_tdm_word_pos(player, word_pos);
+ SET_UNIPERIF_TDM_WORD_POS(player, 1_2, word_pos[WORD_1_2]);
+ SET_UNIPERIF_TDM_WORD_POS(player, 3_4, word_pos[WORD_3_4]);
+ SET_UNIPERIF_TDM_WORD_POS(player, 5_6, word_pos[WORD_5_6]);
+ SET_UNIPERIF_TDM_WORD_POS(player, 7_8, word_pos[WORD_7_8]);
+
+ /* set unip clk rate (not done vai set_sysclk ops) */
+ freq = runtime->rate * tdm_frame_size * 8;
+ mutex_lock(&player->ctrl_lock);
+ ret = uni_player_clk_set_rate(player, freq);
+ if (!ret)
+ player->mclk = freq;
+ mutex_unlock(&player->ctrl_lock);
+
+ return 0;
+}
+
/*
* ALSA uniperipheral iec958 controls
*/
@@ -668,11 +700,29 @@ static int uni_player_startup(struct snd_pcm_substream *substream,
{
struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
struct uniperif *player = priv->dai_data.uni;
+ int ret;
+
player->substream = substream;
player->clk_adj = 0;
- return 0;
+ if (!UNIPERIF_TYPE_IS_TDM(player))
+ return 0;
+
+ /* refine hw constraint in tdm mode */
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ sti_uniperiph_fix_tdm_chan,
+ player, SNDRV_PCM_HW_PARAM_CHANNELS,
+ -1);
+ if (ret < 0)
+ return ret;
+
+ return snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ sti_uniperiph_fix_tdm_format,
+ player, SNDRV_PCM_HW_PARAM_FORMAT,
+ -1);
}
static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id,
@@ -682,7 +732,7 @@ static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id,
struct uniperif *player = priv->dai_data.uni;
int ret;
- if (dir == SND_SOC_CLOCK_IN)
+ if (UNIPERIF_TYPE_IS_TDM(player) || (dir == SND_SOC_CLOCK_IN))
return 0;
if (clk_id != 0)
@@ -714,7 +764,13 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
}
/* Calculate transfer size (in fifo cells and bytes) for frame count */
- transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
+ if (player->info->type == SND_ST_UNIPERIF_TYPE_TDM) {
+ /* transfer size = user frame size (in 32 bits FIFO cell) */
+ transfer_size =
+ sti_uniperiph_get_user_frame_size(runtime) / 4;
+ } else {
+ transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
+ }
/* Calculate number of empty cells available before asserting DREQ */
if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
@@ -738,16 +794,19 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(player, trigger_limit);
/* Uniperipheral setup depends on player type */
- switch (player->info->player_type) {
- case SND_ST_UNIPERIF_PLAYER_TYPE_HDMI:
+ switch (player->info->type) {
+ case SND_ST_UNIPERIF_TYPE_HDMI:
ret = uni_player_prepare_iec958(player, runtime);
break;
- case SND_ST_UNIPERIF_PLAYER_TYPE_PCM:
+ case SND_ST_UNIPERIF_TYPE_PCM:
ret = uni_player_prepare_pcm(player, runtime);
break;
- case SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF:
+ case SND_ST_UNIPERIF_TYPE_SPDIF:
ret = uni_player_prepare_iec958(player, runtime);
break;
+ case SND_ST_UNIPERIF_TYPE_TDM:
+ ret = uni_player_prepare_tdm(player, runtime);
+ break;
default:
dev_err(player->dev, "invalid player type");
return -EINVAL;
@@ -852,8 +911,8 @@ static int uni_player_start(struct uniperif *player)
* will not take affect and hang the player.
*/
if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
- if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player))
- SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player);
+ if (UNIPERIF_TYPE_IS_IEC958(player))
+ SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player);
/* Force channel status update (no update if clk disable) */
if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
@@ -954,27 +1013,30 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream,
player->substream = NULL;
}
-static int uni_player_parse_dt_clk_glue(struct platform_device *pdev,
- struct uniperif *player)
+static int uni_player_parse_dt_audio_glue(struct platform_device *pdev,
+ struct uniperif *player)
{
- int bit_offset;
struct device_node *node = pdev->dev.of_node;
struct regmap *regmap;
-
- bit_offset = SYS_CFG_AUDI0_GLUE_PCM_CLKX + player->info->id;
+ struct reg_field regfield[2] = {
+ /* PCM_CLK_SEL */
+ REG_FIELD(SYS_CFG_AUDIO_GLUE,
+ 8 + player->info->id,
+ 8 + player->info->id),
+ /* PCMP_VALID_SEL */
+ REG_FIELD(SYS_CFG_AUDIO_GLUE, 0, 1)
+ };
regmap = syscon_regmap_lookup_by_phandle(node, "st,syscfg");
- if (regmap) {
- struct reg_field regfield =
- REG_FIELD(SYS_CFG_AUDIO_GLUE, bit_offset, bit_offset);
-
- player->clk_sel = regmap_field_alloc(regmap, regfield);
- } else {
+ if (!regmap) {
dev_err(&pdev->dev, "sti-audio-clk-glue syscf not found\n");
return -EINVAL;
}
+ player->clk_sel = regmap_field_alloc(regmap, regfield[0]);
+ player->valid_sel = regmap_field_alloc(regmap, regfield[1]);
+
return 0;
}
@@ -1012,19 +1074,21 @@ static int uni_player_parse_dt(struct platform_device *pdev,
}
if (strcasecmp(mode, "hdmi") == 0)
- info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI;
+ info->type = SND_ST_UNIPERIF_TYPE_HDMI;
else if (strcasecmp(mode, "pcm") == 0)
- info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM;
+ info->type = SND_ST_UNIPERIF_TYPE_PCM;
else if (strcasecmp(mode, "spdif") == 0)
- info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF;
+ info->type = SND_ST_UNIPERIF_TYPE_SPDIF;
+ else if (strcasecmp(mode, "tdm") == 0)
+ info->type = SND_ST_UNIPERIF_TYPE_TDM;
else
- info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE;
+ info->type = SND_ST_UNIPERIF_TYPE_NONE;
/* Save the info structure */
player->info = info;
- /* Get the PCM_CLK_SEL bit from audio-glue-ctrl SoC register */
- if (uni_player_parse_dt_clk_glue(pdev, player))
+ /* Get PCM_CLK_SEL & PCMP_VALID_SEL from audio-glue-ctrl SoC reg */
+ if (uni_player_parse_dt_audio_glue(pdev, player))
return -EINVAL;
return 0;
@@ -1037,7 +1101,8 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = {
.trigger = uni_player_trigger,
.hw_params = sti_uniperiph_dai_hw_params,
.set_fmt = sti_uniperiph_dai_set_fmt,
- .set_sysclk = uni_player_set_sysclk
+ .set_sysclk = uni_player_set_sysclk,
+ .set_tdm_slot = sti_uniperiph_set_tdm_slot
};
int uni_player_init(struct platform_device *pdev,
@@ -1047,7 +1112,6 @@ int uni_player_init(struct platform_device *pdev,
player->dev = &pdev->dev;
player->state = UNIPERIF_STATE_STOPPED;
- player->hw = &uni_player_pcm_hw;
player->dai_ops = &uni_player_dai_ops;
ret = uni_player_parse_dt(pdev, player);
@@ -1057,6 +1121,11 @@ int uni_player_init(struct platform_device *pdev,
return ret;
}
+ if (UNIPERIF_TYPE_IS_TDM(player))
+ player->hw = &uni_tdm_hw;
+ else
+ player->hw = &uni_player_pcm_hw;
+
/* Get uniperif resource */
player->clk = of_clk_get(pdev->dev.of_node, 0);
if (IS_ERR(player->clk))
@@ -1073,6 +1142,17 @@ int uni_player_init(struct platform_device *pdev,
}
}
+ /* connect to I2S/TDM TX bus */
+ if (player->valid_sel &&
+ (player->info->id == UNIPERIF_PLAYER_I2S_OUT)) {
+ ret = regmap_field_write(player->valid_sel, player->info->id);
+ if (ret) {
+ dev_err(player->dev,
+ "%s: unable to connect to tdm bus", __func__);
+ return ret;
+ }
+ }
+
ret = devm_request_irq(&pdev->dev, player->irq,
uni_player_irq_handler, IRQF_SHARED,
dev_name(&pdev->dev), player);
@@ -1087,7 +1167,7 @@ int uni_player_init(struct platform_device *pdev,
SET_UNIPERIF_CTRL_SPDIF_LAT_OFF(player);
SET_UNIPERIF_CONFIG_IDLE_MOD_DISABLE(player);
- if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) {
+ if (UNIPERIF_TYPE_IS_IEC958(player)) {
/* Set default iec958 status bits */
/* Consumer, PCM, copyright, 2ch, mode 0 */
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 8a0eb2050169..eb74a328c928 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -73,55 +73,10 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
return ret;
}
-static int uni_reader_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
+ struct uniperif *reader)
{
- struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
- struct uniperif *reader = priv->dai_data.uni;
- struct snd_pcm_runtime *runtime = substream->runtime;
- int transfer_size, trigger_limit;
int slot_width;
- int count = 10;
-
- /* The reader should be stopped */
- if (reader->state != UNIPERIF_STATE_STOPPED) {
- dev_err(reader->dev, "%s: invalid reader state %d", __func__,
- reader->state);
- return -EINVAL;
- }
-
- /* Calculate transfer size (in fifo cells and bytes) for frame count */
- transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
-
- /* Calculate number of empty cells available before asserting DREQ */
- if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
- trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size;
- else
- /*
- * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0
- * FDMA_TRIGGER_LIMIT also controls when the state switches
- * from OFF or STANDBY to AUDIO DATA.
- */
- trigger_limit = transfer_size;
-
- /* Trigger limit must be an even number */
- if ((!trigger_limit % 2) ||
- (trigger_limit != 1 && transfer_size % 2) ||
- (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
- dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
- return -EINVAL;
- }
-
- SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit);
-
- switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) {
- case SND_SOC_DAIFMT_IB_IF:
- case SND_SOC_DAIFMT_NB_IF:
- SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader);
- break;
- default:
- SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader);
- }
/* Force slot width to 32 in I2S mode */
if ((reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK)
@@ -173,6 +128,109 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
return -EINVAL;
}
+ /* Number of channels must be even */
+ if ((runtime->channels % 2) || (runtime->channels < 2) ||
+ (runtime->channels > 10)) {
+ dev_err(reader->dev, "%s: invalid nb of channels", __func__);
+ return -EINVAL;
+ }
+
+ SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2);
+ SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader);
+
+ return 0;
+}
+
+static int uni_reader_prepare_tdm(struct snd_pcm_runtime *runtime,
+ struct uniperif *reader)
+{
+ int frame_size; /* user tdm frame size in bytes */
+ /* default unip TDM_WORD_POS_X_Y */
+ unsigned int word_pos[4] = {
+ 0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A};
+
+ frame_size = sti_uniperiph_get_user_frame_size(runtime);
+
+ /* fix 16/0 format */
+ SET_UNIPERIF_CONFIG_MEM_FMT_16_0(reader);
+ SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(reader);
+
+ /* number of words inserted on the TDM line */
+ SET_UNIPERIF_I2S_FMT_NUM_CH(reader, frame_size / 4 / 2);
+
+ SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader);
+ SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader);
+ SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(reader);
+
+ /*
+ * set the timeslots allocation for words in FIFO
+ *
+ * HW bug: (LSB word < MSB word) => this config is not possible
+ * So if we want (LSB word < MSB) word, then it shall be
+ * handled by user
+ */
+ sti_uniperiph_get_tdm_word_pos(reader, word_pos);
+ SET_UNIPERIF_TDM_WORD_POS(reader, 1_2, word_pos[WORD_1_2]);
+ SET_UNIPERIF_TDM_WORD_POS(reader, 3_4, word_pos[WORD_3_4]);
+ SET_UNIPERIF_TDM_WORD_POS(reader, 5_6, word_pos[WORD_5_6]);
+ SET_UNIPERIF_TDM_WORD_POS(reader, 7_8, word_pos[WORD_7_8]);
+
+ return 0;
+}
+
+static int uni_reader_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+ struct uniperif *reader = priv->dai_data.uni;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int transfer_size, trigger_limit, ret;
+ int count = 10;
+
+ /* The reader should be stopped */
+ if (reader->state != UNIPERIF_STATE_STOPPED) {
+ dev_err(reader->dev, "%s: invalid reader state %d", __func__,
+ reader->state);
+ return -EINVAL;
+ }
+
+ /* Calculate transfer size (in fifo cells and bytes) for frame count */
+ if (reader->info->type == SND_ST_UNIPERIF_TYPE_TDM) {
+ /* transfer size = unip frame size (in 32 bits FIFO cell) */
+ transfer_size =
+ sti_uniperiph_get_user_frame_size(runtime) / 4;
+ } else {
+ transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
+ }
+
+ /* Calculate number of empty cells available before asserting DREQ */
+ if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
+ trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size;
+ else
+ /*
+ * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0
+ * FDMA_TRIGGER_LIMIT also controls when the state switches
+ * from OFF or STANDBY to AUDIO DATA.
+ */
+ trigger_limit = transfer_size;
+
+ /* Trigger limit must be an even number */
+ if ((!trigger_limit % 2) ||
+ (trigger_limit != 1 && transfer_size % 2) ||
+ (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
+ dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
+ return -EINVAL;
+ }
+
+ SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit);
+
+ if (UNIPERIF_TYPE_IS_TDM(reader))
+ ret = uni_reader_prepare_tdm(runtime, reader);
+ else
+ ret = uni_reader_prepare_pcm(runtime, reader);
+ if (ret)
+ return ret;
+
switch (reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader);
@@ -191,21 +249,26 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
return -EINVAL;
}
- SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader);
-
- /* Data clocking (changing) on the rising edge */
- SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader);
-
- /* Number of channels must be even */
-
- if ((runtime->channels % 2) || (runtime->channels < 2) ||
- (runtime->channels > 10)) {
- dev_err(reader->dev, "%s: invalid nb of channels", __func__);
- return -EINVAL;
+ /* Data clocking (changing) on the rising/falling edge */
+ switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader);
+ SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader);
+ SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader);
+ SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader);
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader);
+ SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader);
+ break;
}
- SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2);
-
/* Clear any pending interrupts */
SET_UNIPERIF_ITS_BCLR(reader, GET_UNIPERIF_ITS(reader));
@@ -293,6 +356,32 @@ static int uni_reader_trigger(struct snd_pcm_substream *substream,
}
}
+static int uni_reader_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+ struct uniperif *reader = priv->dai_data.uni;
+ int ret;
+
+ if (!UNIPERIF_TYPE_IS_TDM(reader))
+ return 0;
+
+ /* refine hw constraint in tdm mode */
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ sti_uniperiph_fix_tdm_chan,
+ reader, SNDRV_PCM_HW_PARAM_CHANNELS,
+ -1);
+ if (ret < 0)
+ return ret;
+
+ return snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ sti_uniperiph_fix_tdm_format,
+ reader, SNDRV_PCM_HW_PARAM_FORMAT,
+ -1);
+}
+
static void uni_reader_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -310,6 +399,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
{
struct uniperif_info *info;
struct device_node *node = pdev->dev.of_node;
+ const char *mode;
/* Allocate memory for the info structure */
info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
@@ -322,6 +412,17 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
return -EINVAL;
}
+ /* Read the device mode property */
+ if (of_property_read_string(node, "st,mode", &mode)) {
+ dev_err(&pdev->dev, "uniperipheral mode not defined");
+ return -EINVAL;
+ }
+
+ if (strcasecmp(mode, "tdm") == 0)
+ info->type = SND_ST_UNIPERIF_TYPE_TDM;
+ else
+ info->type = SND_ST_UNIPERIF_TYPE_PCM;
+
/* Save the info structure */
reader->info = info;
@@ -329,11 +430,13 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
}
static const struct snd_soc_dai_ops uni_reader_dai_ops = {
+ .startup = uni_reader_startup,
.shutdown = uni_reader_shutdown,
.prepare = uni_reader_prepare,
.trigger = uni_reader_trigger,
.hw_params = sti_uniperiph_dai_hw_params,
.set_fmt = sti_uniperiph_dai_set_fmt,
+ .set_tdm_slot = sti_uniperiph_set_tdm_slot
};
int uni_reader_init(struct platform_device *pdev,
@@ -343,7 +446,6 @@ int uni_reader_init(struct platform_device *pdev,
reader->dev = &pdev->dev;
reader->state = UNIPERIF_STATE_STOPPED;
- reader->hw = &uni_reader_pcm_hw;
reader->dai_ops = &uni_reader_dai_ops;
ret = uni_reader_parse_dt(pdev, reader);
@@ -352,6 +454,11 @@ int uni_reader_init(struct platform_device *pdev,
return ret;
}
+ if (UNIPERIF_TYPE_IS_TDM(reader))
+ reader->hw = &uni_tdm_hw;
+ else
+ reader->hw = &uni_reader_pcm_hw;
+
ret = devm_request_irq(&pdev->dev, reader->irq,
uni_reader_irq_handler, IRQF_SHARED,
dev_name(&pdev->dev), reader);

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