From a941e2fab3207cb0d57dc4ec47b1b12c8ea78b84 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Wed, 4 Apr 2018 06:19:37 +0200 Subject: ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. Signed-off-by: Kirill Marinushkin Reviewed-by: Pierre-Louis Bossart Tested-by: Pan Xiuli Tested-by: Pierre-Louis Bossart Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Mark Brown Cc: Liam Girdwood Cc: linux-kernel@vger.kernel.org Cc: alsa-devel@alsa-project.org Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) (limited to 'include/uapi') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69c37ecbff7e..f0e5e21efa54 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #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. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __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 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ -- cgit From 933e1c4a667103c4d10ebdc9505a0a6abd8c3fbd Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Wed, 4 Apr 2018 06:19:38 +0200 Subject: ASoC: topology: Add missing clock gating parameter when parsing hw_configs Clock gating parameter is a part of `dai_fmt`. It is supported by `alsa-lib` when creating a topology binary file, but ignored by kernel when loading this topology file. After applying this commit, the clock gating parameter is not ignored any more. This solution is backwards compatible. The existing behaviour is not broken, because by default the parameter value is 0 and is ignored. snd_soc_tplg_hw_config.clock_gated = 0 => no effect snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. pm_gate_clocks "true" # clock can be gated } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" ] default_hw_conf_id "1" } ~~~~ Signed-off-by: Kirill Marinushkin Reviewed-by: Pierre-Louis Bossart Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Mark Brown Cc: Pan Xiuli Cc: Liam Girdwood Cc: linux-kernel@vger.kernel.org Cc: alsa-devel@alsa-project.org Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'include/uapi') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index f0e5e21efa54..f3c4b46e39d8 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -139,6 +139,11 @@ #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 physical PCM data formats. * Add new formats to the end of the list. */ @@ -324,7 +329,7 @@ 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; /* SND_SOC_TPLG_BCLK_ value */ -- cgit From e590522a06adce8ca2eb47e77d80616cd1542d91 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Wed, 4 Apr 2018 06:19:39 +0200 Subject: ASoC: topology: Add definitions for mclk_direction values Current comment makes not clear the direction of mclk. Previously, similar description caused a misunderstanding for bclk_master and fsync_master. This commit solves the potential confusion the same way it is solved for bclk_master and fsync_master. Signed-off-by: Kirill Marinushkin Acked-by: Pierre-Louis Bossart Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Mark Brown Cc: Pan Xiuli Cc: Liam Girdwood Cc: linux-kernel@vger.kernel.org Cc: alsa-devel@alsa-project.org Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'include/uapi') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index f3c4b46e39d8..b901cdbe532a 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -144,6 +144,10 @@ #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. */ @@ -334,7 +338,7 @@ struct snd_soc_tplg_hw_config { __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ - __u8 mclk_direction; /* 0 for input, 1 for output */ + __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 */ -- cgit From 08f9f4485f2158de0fa77506687a073cb869e803 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 9 May 2018 17:53:04 -0700 Subject: ALSA: core api: define offsets for TLV items Currently, there are no pre-defined accessors for the elements in topology TLV data. In the absence of such offsets, the tlv data will have to be decoded using hardwired offset numbers 0-N depending on the type of TLV. This patch defines accessor offsets for the type, length, min and mute/step items in TLV data for DB_SCALE type tlv's. These will be used by drivers to decode the TLV data while loading topology thereby improving code readability. The type and len offsets are common for all TLV types. The min and step/mute offsets are specific to DB_SCALE tlv type. Signed-off-by: Ranjani Sridharan Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/tlv.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include/uapi') diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..e3437e96519a 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)) -- cgit From e6f32bf48fb1d3b7aedc0deb6e791362af71cb17 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 14 May 2018 07:09:50 +0900 Subject: ALSA: control: complement TLV macro for db-minmax and db-linear types A commit 08f9f4485f21 ('ALSA: core api: define offsets for TLV items') introduced a series of macro for offset of db-scale type of TLV, however there are some types of TLV to add similar macros. This commit complements macros for offset of db-minmax and db-linear types of TLV data. Cc: Ranjani Sridharan Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/tlv.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include/uapi') diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index e3437e96519a..7d6d65f60a42 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -83,6 +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)) @@ -91,6 +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 -- cgit From 6cfd839ae78ec3fac5ddbf7148155898727e90c3 Mon Sep 17 00:00:00 2001 From: Jorge Sanjuan Date: Fri, 11 May 2018 16:25:34 +0100 Subject: ALSA: usb-audio: UAC3. Add support for mixer unit. This adds support for the MIXER UNIT in UAC3. All the information is obtained from the (HIGH CAPABILITY) Cluster's header. We don't read the rest of the logical cluster to obtain the channel config as that wont make any difference in the current mixer behaviour. The name of the mixer unit is not yet requested as there is not support for the UAC3 Class Specific String requests. Tested in an UAC3 device working as a HEADSET with a basic mixer unit (same as the one in the BADD spec) with no controls. Signed-off-by: Jorge Sanjuan Reviewed-by: Ruslan Bilovol Tested-by: Ruslan Bilovol Signed-off-by: Takashi Iwai --- include/uapi/linux/usb/audio.h | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'include/uapi') 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) -- cgit From 348f48220b97130817de4aa2058569133c5cc051 Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Thu, 24 May 2018 12:49:22 -0700 Subject: ASoC: topology: Move v4 manifest header data structures to uapi Topology manifest v4 is still part of the ABI. Move its data structures into the uapi header file. No functional change. Signed-off-by: Guenter Roeck Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 57 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'include/uapi') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index b901cdbe532a..a74ca232f1fc 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -573,4 +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 -- cgit From 0c24fdc00244cc08309e397e3783f2943221dc53 Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Thu, 24 May 2018 12:49:23 -0700 Subject: ASoC: topology: Move skl-tplg-interface.h to uapi skl-tplg-interface.h describes firmware format details for Skylake topology files. It is part of the ABI and should reside in the uapi directory. While moving the file, also replace the license boilerplate with the SPDX License Identifier. No functional change. Signed-off-by: Guenter Roeck Signed-off-by: Mark Brown --- include/uapi/sound/skl-tplg-interface.h | 237 ++++++++++++++++++++++++++++++++ 1 file changed, 237 insertions(+) create mode 100644 include/uapi/sound/skl-tplg-interface.h (limited to 'include/uapi') 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 + * Nilofer, Samreen + */ + +#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 -- cgit