Merge tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've got many code additions at this cycle as a result of quite a few
  new drivers. Below are highlights:

  Core stuff:
   - Fix the long-standing issue with the device registration order; the
     control device is now registered at last
   - PCM locking code cleanups for RT kernels
   - Fixes for possible races in ALSA timer resolution accesses
   - TLV offset definitions in uapi

  ASoC:
   - Many fixes for the topology stuff, including fixes for v4 ABI
     compatibility
   - Lots of cleanups / quirks for Intel platforms based on Realtek
     CODECs
   - Continued componentization works, removing legacy CODEC stuff
   - Conversion of OMAP DMA to the new, more standard SDMA-PCM driver
   - Fixes and updates to Cirrus Logic SoC drivers
   - New Qualcomm DSP support
   - New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek
     MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306
     and RT5668 and TI TSCS454

  HD-audio:
   - Finally better support for some CA0132 boards, allowing Windows
     firmware
   - HP Spectre x360 support along with a bulk of COEF stuff
   - Blacklisting power save default some known boards reported on
     Fedora

  USB-audio:
   - Continued improvements on UAC3 support; now BADD is supported
   - Fixes / improvements for Dell WD15 dock
   - Allow DMA coherent pages for PCM buffers for ARCH, MIPS & co

  Others:
   - New Xen sound frontend driver support
   - Cache implementation and other improvements for FireWire DICE
   - Conversions to octal permissions in allover places"

* tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (386 commits)
  ASoC: dapm: delete dapm_kcontrol_data paths list before freeing it
  ALSA: usb-audio: remove redundant check on err
  ASoC: topology: Move skl-tplg-interface.h to uapi
  ASoC: topology: Move v4 manifest header data structures to uapi
  ASoC: topology: Improve backwards compatibility with v4 topology files
  ALSA: pci/hda: Remove unused, broken, header file
  ASoC: TSCS454: Add Support
  ASoC: Intel: kbl: Move codec sysclk config to codec_init function
  ASoC: simple-card: set cpu dai clk in hw_params
  ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
  ALSA: oxygen: use match_string() helper
  ASoC: dapm: use match_string() helper
  ASoC: max98095: use match_string() helper
  ASoC: max98088: use match_string() helper
  ASoC: Intel: bytcr_rt5651: Set card long_name based on quirks
  ASoC: mt6797-mt6351: add hostless phone call path
  ASoC: mt6797: add Hostless DAI
  ASoC: mt6797: add PCM interface
  ASoC: mediatek: export mtk-afe symbols as needed
  ASoC: codecs: PCM1789: include gpio/consumer.h
  ...

4 files changed, 351 insertions, 7 deletions

diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h
index 3a78e7145689..13d98e6e0db1 100644
--- a/include/uapi/linux/usb/audio.h
+++ b/include/uapi/linux/usb/audio.h
@@ -285,9 +285,22 @@ static inline __u8 uac_mixer_unit_iChannelNames(struct uac_mixer_unit_descriptor
 static inline __u8 *uac_mixer_unit_bmControls(struct uac_mixer_unit_descriptor *desc,
 					      int protocol)
 {
-	return (protocol == UAC_VERSION_1) ?
-		&desc->baSourceID[desc->bNrInPins + 4] :
-		&desc->baSourceID[desc->bNrInPins + 6];
+	switch (protocol) {
+	case UAC_VERSION_1:
+		return &desc->baSourceID[desc->bNrInPins + 4];
+	case UAC_VERSION_2:
+		return &desc->baSourceID[desc->bNrInPins + 6];
+	case UAC_VERSION_3:
+		return &desc->baSourceID[desc->bNrInPins + 2];
+	default:
+		return NULL;
+	}
+}
+
+static inline __u16 uac3_mixer_unit_wClusterDescrID(struct uac_mixer_unit_descriptor *desc)
+{
+	return (desc->baSourceID[desc->bNrInPins + 1] << 8) |
+		desc->baSourceID[desc->bNrInPins];
 }
 
 static inline __u8 uac_mixer_unit_iMixer(struct uac_mixer_unit_descriptor *desc)
diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h
index 69c37ecbff7e..a74ca232f1fc 100644
--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@@ -139,6 +139,15 @@
 #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS      (1 << 1)
 #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS    (1 << 2)
 
+/* DAI clock gating */
+#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED	0
+#define SND_SOC_TPLG_DAI_CLK_GATE_GATED	1
+#define SND_SOC_TPLG_DAI_CLK_GATE_CONT		2
+
+/* DAI mclk_direction */
+#define SND_SOC_TPLG_MCLK_CO            0 /* for codec, mclk is output */
+#define SND_SOC_TPLG_MCLK_CI            1 /* for codec, mclk is input */
+
 /* DAI physical PCM data formats.
  * Add new formats to the end of the list.
  */
@@ -160,6 +169,18 @@
 #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS    (1 << 2)
 #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP            (1 << 3)
 
+/* DAI topology BCLK parameter
+ * For the backwards capability, by default codec is bclk master
+ */
+#define SND_SOC_TPLG_BCLK_CM         0 /* codec is bclk master */
+#define SND_SOC_TPLG_BCLK_CS         1 /* codec is bclk slave */
+
+/* DAI topology FSYNC parameter
+ * For the backwards capability, by default codec is fsync master
+ */
+#define SND_SOC_TPLG_FSYNC_CM         0 /* codec is fsync master */
+#define SND_SOC_TPLG_FSYNC_CS         1 /* codec is fsync slave */
+
 /*
  * Block Header.
  * This header precedes all object and object arrays below.
@@ -312,12 +333,12 @@ struct snd_soc_tplg_hw_config {
 	__le32 size;            /* in bytes of this structure */
 	__le32 id;		/* unique ID - - used to match */
 	__le32 fmt;		/* SND_SOC_DAI_FORMAT_ format value */
-	__u8 clock_gated;	/* 1 if clock can be gated to save power */
+	__u8 clock_gated;	/* SND_SOC_TPLG_DAI_CLK_GATE_ value */
 	__u8 invert_bclk;	/* 1 for inverted BCLK, 0 for normal */
 	__u8 invert_fsync;	/* 1 for inverted frame clock, 0 for normal */
-	__u8 bclk_master;	/* 1 for master of BCLK, 0 for slave */
-	__u8 fsync_master;	/* 1 for master of FSYNC, 0 for slave */
-	__u8 mclk_direction;    /* 0 for input, 1 for output */
+	__u8 bclk_master;	/* SND_SOC_TPLG_BCLK_ value */
+	__u8 fsync_master;	/* SND_SOC_TPLG_FSYNC_ value */
+	__u8 mclk_direction;    /* SND_SOC_TPLG_MCLK_ value */
 	__le16 reserved;	/* for 32bit alignment */
 	__le32 mclk_rate;	/* MCLK or SYSCLK freqency in Hz */
 	__le32 bclk_rate;	/* BCLK freqency in Hz */
@@ -552,4 +573,61 @@ struct snd_soc_tplg_dai {
 	__le32 flags;           /* SND_SOC_TPLG_DAI_FLGBIT_* */
 	struct snd_soc_tplg_private priv;
 } __attribute__((packed));
+
+/*
+ * Old version of ABI structs, supported for backward compatibility.
+ */
+
+/* Manifest v4 */
+struct snd_soc_tplg_manifest_v4 {
+	__le32 size;		/* in bytes of this structure */
+	__le32 control_elems;	/* number of control elements */
+	__le32 widget_elems;	/* number of widget elements */
+	__le32 graph_elems;	/* number of graph elements */
+	__le32 pcm_elems;	/* number of PCM elements */
+	__le32 dai_link_elems;	/* number of DAI link elements */
+	struct snd_soc_tplg_private priv;
+} __packed;
+
+/* Stream Capabilities v4 */
+struct snd_soc_tplg_stream_caps_v4 {
+	__le32 size;		/* in bytes of this structure */
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	__le64 formats;	/* supported formats SNDRV_PCM_FMTBIT_* */
+	__le32 rates;		/* supported rates SNDRV_PCM_RATE_* */
+	__le32 rate_min;	/* min rate */
+	__le32 rate_max;	/* max rate */
+	__le32 channels_min;	/* min channels */
+	__le32 channels_max;	/* max channels */
+	__le32 periods_min;	/* min number of periods */
+	__le32 periods_max;	/* max number of periods */
+	__le32 period_size_min;	/* min period size bytes */
+	__le32 period_size_max;	/* max period size bytes */
+	__le32 buffer_size_min;	/* min buffer size bytes */
+	__le32 buffer_size_max;	/* max buffer size bytes */
+} __packed;
+
+/* PCM v4 */
+struct snd_soc_tplg_pcm_v4 {
+	__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 with DAI link */
+	__le32 dai_id;		/* unique ID - used to match */
+	__le32 playback;	/* supports playback mode */
+	__le32 capture;		/* supports capture mode */
+	__le32 compress;	/* 1 = compressed; 0 = PCM */
+	struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */
+	__le32 num_streams;	/* number of streams */
+	struct snd_soc_tplg_stream_caps_v4 caps[2]; /* playback and capture for DAI */
+} __packed;
+
+/* Physical link config v4 */
+struct snd_soc_tplg_link_config_v4 {
+	__le32 size;            /* in bytes of this structure */
+	__le32 id;              /* unique ID - used to match */
+	struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
+	__le32 num_streams;     /* number of streams */
+} __packed;
+
 #endif
diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h
new file mode 100644
index 000000000000..f58cafa42f18
--- /dev/null
+++ b/include/uapi/sound/skl-tplg-interface.h
@@ -0,0 +1,237 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * skl-tplg-interface.h - Intel DSP FW private data interface
+ *
+ * Copyright (C) 2015 Intel Corp
+ * Author: Jeeja KP <jeeja.kp@intel.com>
+ *	    Nilofer, Samreen <samreen.nilofer@intel.com>
+ */
+
+#ifndef __HDA_TPLG_INTERFACE_H__
+#define __HDA_TPLG_INTERFACE_H__
+
+/*
+ * Default types range from 0~12. type can range from 0 to 0xff
+ * SST types start at higher to avoid any overlapping in future
+ */
+#define SKL_CONTROL_TYPE_BYTE_TLV	0x100
+#define SKL_CONTROL_TYPE_MIC_SELECT	0x102
+
+#define HDA_SST_CFG_MAX	900 /* size of copier cfg*/
+#define MAX_IN_QUEUE 8
+#define MAX_OUT_QUEUE 8
+
+#define SKL_UUID_STR_SZ 40
+/* Event types goes here */
+/* Reserve event type 0 for no event handlers */
+enum skl_event_types {
+	SKL_EVENT_NONE = 0,
+	SKL_MIXER_EVENT,
+	SKL_MUX_EVENT,
+	SKL_VMIXER_EVENT,
+	SKL_PGA_EVENT
+};
+
+/**
+ * enum skl_ch_cfg - channel configuration
+ *
+ * @SKL_CH_CFG_MONO:	One channel only
+ * @SKL_CH_CFG_STEREO:	L & R
+ * @SKL_CH_CFG_2_1:	L, R & LFE
+ * @SKL_CH_CFG_3_0:	L, C & R
+ * @SKL_CH_CFG_3_1:	L, C, R & LFE
+ * @SKL_CH_CFG_QUATRO:	L, R, Ls & Rs
+ * @SKL_CH_CFG_4_0:	L, C, R & Cs
+ * @SKL_CH_CFG_5_0:	L, C, R, Ls & Rs
+ * @SKL_CH_CFG_5_1:	L, C, R, Ls, Rs & LFE
+ * @SKL_CH_CFG_DUAL_MONO: One channel replicated in two
+ * @SKL_CH_CFG_I2S_DUAL_STEREO_0: Stereo(L,R) in 4 slots, 1st stream:[ L, R, -, - ]
+ * @SKL_CH_CFG_I2S_DUAL_STEREO_1: Stereo(L,R) in 4 slots, 2nd stream:[ -, -, L, R ]
+ * @SKL_CH_CFG_INVALID:	Invalid
+ */
+enum skl_ch_cfg {
+	SKL_CH_CFG_MONO = 0,
+	SKL_CH_CFG_STEREO = 1,
+	SKL_CH_CFG_2_1 = 2,
+	SKL_CH_CFG_3_0 = 3,
+	SKL_CH_CFG_3_1 = 4,
+	SKL_CH_CFG_QUATRO = 5,
+	SKL_CH_CFG_4_0 = 6,
+	SKL_CH_CFG_5_0 = 7,
+	SKL_CH_CFG_5_1 = 8,
+	SKL_CH_CFG_DUAL_MONO = 9,
+	SKL_CH_CFG_I2S_DUAL_STEREO_0 = 10,
+	SKL_CH_CFG_I2S_DUAL_STEREO_1 = 11,
+	SKL_CH_CFG_4_CHANNEL = 12,
+	SKL_CH_CFG_INVALID
+};
+
+enum skl_module_type {
+	SKL_MODULE_TYPE_MIXER = 0,
+	SKL_MODULE_TYPE_COPIER,
+	SKL_MODULE_TYPE_UPDWMIX,
+	SKL_MODULE_TYPE_SRCINT,
+	SKL_MODULE_TYPE_ALGO,
+	SKL_MODULE_TYPE_BASE_OUTFMT,
+	SKL_MODULE_TYPE_KPB,
+	SKL_MODULE_TYPE_MIC_SELECT,
+};
+
+enum skl_core_affinity {
+	SKL_AFFINITY_CORE_0 = 0,
+	SKL_AFFINITY_CORE_1,
+	SKL_AFFINITY_CORE_MAX
+};
+
+enum skl_pipe_conn_type {
+	SKL_PIPE_CONN_TYPE_NONE = 0,
+	SKL_PIPE_CONN_TYPE_FE,
+	SKL_PIPE_CONN_TYPE_BE
+};
+
+enum skl_hw_conn_type {
+	SKL_CONN_NONE = 0,
+	SKL_CONN_SOURCE = 1,
+	SKL_CONN_SINK = 2
+};
+
+enum skl_dev_type {
+	SKL_DEVICE_BT = 0x0,
+	SKL_DEVICE_DMIC = 0x1,
+	SKL_DEVICE_I2S = 0x2,
+	SKL_DEVICE_SLIMBUS = 0x3,
+	SKL_DEVICE_HDALINK = 0x4,
+	SKL_DEVICE_HDAHOST = 0x5,
+	SKL_DEVICE_NONE
+};
+
+/**
+ * enum skl_interleaving - interleaving style
+ *
+ * @SKL_INTERLEAVING_PER_CHANNEL: [s1_ch1...s1_chN,...,sM_ch1...sM_chN]
+ * @SKL_INTERLEAVING_PER_SAMPLE: [s1_ch1...sM_ch1,...,s1_chN...sM_chN]
+ */
+enum skl_interleaving {
+	SKL_INTERLEAVING_PER_CHANNEL = 0,
+	SKL_INTERLEAVING_PER_SAMPLE = 1,
+};
+
+enum skl_sample_type {
+	SKL_SAMPLE_TYPE_INT_MSB = 0,
+	SKL_SAMPLE_TYPE_INT_LSB = 1,
+	SKL_SAMPLE_TYPE_INT_SIGNED = 2,
+	SKL_SAMPLE_TYPE_INT_UNSIGNED = 3,
+	SKL_SAMPLE_TYPE_FLOAT = 4
+};
+
+enum module_pin_type {
+	/* All pins of the module takes same PCM inputs or outputs
+	* e.g. mixout
+	*/
+	SKL_PIN_TYPE_HOMOGENEOUS,
+	/* All pins of the module takes different PCM inputs or outputs
+	* e.g mux
+	*/
+	SKL_PIN_TYPE_HETEROGENEOUS,
+};
+
+enum skl_module_param_type {
+	SKL_PARAM_DEFAULT = 0,
+	SKL_PARAM_INIT,
+	SKL_PARAM_SET,
+	SKL_PARAM_BIND
+};
+
+struct skl_dfw_algo_data {
+	u32 set_params:2;
+	u32 rsvd:30;
+	u32 param_id;
+	u32 max;
+	char params[0];
+} __packed;
+
+enum skl_tkn_dir {
+	SKL_DIR_IN,
+	SKL_DIR_OUT
+};
+
+enum skl_tuple_type {
+	SKL_TYPE_TUPLE,
+	SKL_TYPE_DATA
+};
+
+/* v4 configuration data */
+
+struct skl_dfw_v4_module_pin {
+	u16 module_id;
+	u16 instance_id;
+} __packed;
+
+struct skl_dfw_v4_module_fmt {
+	u32 channels;
+	u32 freq;
+	u32 bit_depth;
+	u32 valid_bit_depth;
+	u32 ch_cfg;
+	u32 interleaving_style;
+	u32 sample_type;
+	u32 ch_map;
+} __packed;
+
+struct skl_dfw_v4_module_caps {
+	u32 set_params:2;
+	u32 rsvd:30;
+	u32 param_id;
+	u32 caps_size;
+	u32 caps[HDA_SST_CFG_MAX];
+} __packed;
+
+struct skl_dfw_v4_pipe {
+	u8 pipe_id;
+	u8 pipe_priority;
+	u16 conn_type:4;
+	u16 rsvd:4;
+	u16 memory_pages:8;
+} __packed;
+
+struct skl_dfw_v4_module {
+	char uuid[SKL_UUID_STR_SZ];
+
+	u16 module_id;
+	u16 instance_id;
+	u32 max_mcps;
+	u32 mem_pages;
+	u32 obs;
+	u32 ibs;
+	u32 vbus_id;
+
+	u32 max_in_queue:8;
+	u32 max_out_queue:8;
+	u32 time_slot:8;
+	u32 core_id:4;
+	u32 rsvd1:4;
+
+	u32 module_type:8;
+	u32 conn_type:4;
+	u32 dev_type:4;
+	u32 hw_conn_type:4;
+	u32 rsvd2:12;
+
+	u32 params_fixup:8;
+	u32 converter:8;
+	u32 input_pin_type:1;
+	u32 output_pin_type:1;
+	u32 is_dynamic_in_pin:1;
+	u32 is_dynamic_out_pin:1;
+	u32 is_loadable:1;
+	u32 rsvd3:11;
+
+	struct skl_dfw_v4_pipe pipe;
+	struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE];
+	struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE];
+	struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE];
+	struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE];
+	struct skl_dfw_v4_module_caps caps;
+} __packed;
+
+#endif
diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h
index be5371f09a62..7d6d65f60a42 100644
--- a/include/uapi/sound/tlv.h
+++ b/include/uapi/sound/tlv.h
@@ -42,6 +42,10 @@
 #define SNDRV_CTL_TLVD_LENGTH(...) \
 	((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
 
+/* Accessor offsets for TLV data items */
+#define SNDRV_CTL_TLVO_TYPE		0
+#define SNDRV_CTL_TLVO_LEN		1
+
 #define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
 #define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \
@@ -61,6 +65,10 @@
 		SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
 	}
 
+/* Accessor offsets for min, mute and step items in dB scale type TLV */
+#define SNDRV_CTL_TLVO_DB_SCALE_MIN		2
+#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP	3
+
 /* dB scale specified with min/max values instead of step */
 #define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
@@ -75,6 +83,10 @@
 		SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
 	}
 
+/* Accessor offsets for min, max items in db-minmax types of TLV. */
+#define SNDRV_CTL_TLVO_DB_MINMAX_MIN	2
+#define SNDRV_CTL_TLVO_DB_MINMAX_MAX	3
+
 /* linear volume between min_dB and max_dB (.01dB unit) */
 #define SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
@@ -83,6 +95,10 @@
 		SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
 	}
 
+/* Accessor offsets for min, max items in db-linear type of TLV. */
+#define SNDRV_CTL_TLVO_DB_LINEAR_MIN	2
+#define SNDRV_CTL_TLVO_DB_LINEAR_MAX	3
+
 /* dB range container:
  * Items in dB range container must be ordered by their values and by their
  * dB values. This implies that larger values must correspond with larger