April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dapm: Enable autodisable on SOC_DAPM_SINGLE_TLV_AUTODISABLE
by Charles Keepax 22 Apr '15

22 Apr '15

Correct small copy and paste error where autodisable was not being enabled for the SOC_DAPM_SINGLE_TLV_AUTODISABLE control. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0bc8364..1065095 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -287,7 +287,7 @@ struct device; .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } #define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ -- 1.7.2.5

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Re: [alsa-devel] Playback - Overwrite buffer with silence
by Clemens Ladisch 22 Apr '15

22 Apr '15

Nilhcraiv wrote: > To set the Buffer size and Periord size I have use the > snd_pcm_hw_params_set_period_size_near() and > snd_pcm_hw_params_set_buffer_size_near() functions. If I do not use > the ...near() functions, I return errors. The allowed values of the hardware parameters depend on the hardware's capabilities. Regards, Clemens

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[alsa-devel] [PATCH v8 0/7] ALSA: jack: Refactoring for jack kctls
by Jie Yang 22 Apr '15

22 Apr '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For HDA, With snd_jack_new(), we can use phantom_jack = true for phantom jack creating, and the jack kctl pointer will be filled with the new created kcontrol. Changes in v8: 1. remove snd_jack_kctl from HDA part, change snd_jack_new param to use bool for creating kcontrol at jack creating stage; 2. create jack for phantom jack, with no input device; 3. remove HDA jack get_unique_index() and index related code. Changes in v7: 1. move name generating and index getting part into ctljack.c; 2. remove exposing private_data and destructor to hda, then jack core can handle kctls by itself totally. 3. some small code fixing. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. *** BLURB HERE *** Jie Yang (7): ALSA: jack: implement kctl creating for jack device ALSA: Jack: refactoring snd_kctl_jack_new to support embedded kctl ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 48 ++++++++++ include/sound/control.h | 2 +- include/sound/jack.h | 19 +++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 41 +++++++-- sound/core/jack.c | 142 ++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_jack.c | 90 +++++------------- sound/pci/hda/hda_jack.h | 5 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 13 files changed, 260 insertions(+), 101 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

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[alsa-devel] [RFC 1/4] ASoC: topology: Add topology UAPI header.
by Liam Girdwood 22 Apr '15

22 Apr '15

The ASoC topology UAPI header defines the structures required to define any DSP firmware audio topology and control objects from userspace. The following objects are supported :- o kcontrols including TLV controls. o DAPM widgets and graph elements o Vendor bespoke objects. o Coefficient data o FE PCM capabilities and config. o BE link capabilities and config. o Codec <-> codec link capabilities and config. o Topology object manifest. The file format is simple and divided into blocks for each object type and each block has a header that defines it's size and type. Blocks can be in any order of type and can either all be in a single file or spread across more than one file. Blocks also have a group identifier ID so that they can be loaded and unloaded by ID. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dapm.h | 33 +--- include/uapi/sound/asoc.h | 420 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 421 insertions(+), 32 deletions(-) create mode 100644 include/uapi/sound/asoc.h diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index b1ec05b..3263e36 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -15,6 +15,7 @@ #include <linux/types.h> #include <sound/control.h> +#include <sound/asoc.h> struct device; @@ -447,38 +448,6 @@ struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); -/* dapm widget types */ -enum snd_soc_dapm_type { - snd_soc_dapm_input = 0, /* input pin */ - snd_soc_dapm_output, /* output pin */ - snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ - snd_soc_dapm_mixer, /* mixes several analog signals together */ - snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ - snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ - snd_soc_dapm_out_drv, /* output driver */ - snd_soc_dapm_adc, /* analog to digital converter */ - snd_soc_dapm_dac, /* digital to analog converter */ - snd_soc_dapm_micbias, /* microphone bias (power) */ - snd_soc_dapm_mic, /* microphone */ - snd_soc_dapm_hp, /* headphones */ - snd_soc_dapm_spk, /* speaker */ - snd_soc_dapm_line, /* line input/output */ - snd_soc_dapm_switch, /* analog switch */ - snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ - snd_soc_dapm_pre, /* machine specific pre widget - exec first */ - snd_soc_dapm_post, /* machine specific post widget - exec last */ - snd_soc_dapm_supply, /* power/clock supply */ - snd_soc_dapm_regulator_supply, /* external regulator */ - snd_soc_dapm_clock_supply, /* external clock */ - snd_soc_dapm_aif_in, /* audio interface input */ - snd_soc_dapm_aif_out, /* audio interface output */ - snd_soc_dapm_siggen, /* signal generator */ - snd_soc_dapm_dai_in, /* link to DAI structure */ - snd_soc_dapm_dai_out, - snd_soc_dapm_dai_link, /* link between two DAI structures */ - snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ -}; - enum snd_soc_dapm_subclass { SND_SOC_DAPM_CLASS_INIT = 0, SND_SOC_DAPM_CLASS_RUNTIME = 1, diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h new file mode 100644 index 0000000..dde7133 --- /dev/null +++ b/include/uapi/sound/asoc.h @@ -0,0 +1,420 @@ +/* + * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * 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. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets etc. +*/ + +#ifndef __LINUX_UAPI_SND_ASOC_H +#define __LINUX_UAPI_SND_ASOC_H + +#include <linux/types.h> +#include <sound/asound.h> + +/* + * Numeric IDs for stock mixer types that are used to map and enumerate FW + * based mixers. + */ +#define SOC_CONTROL_ID_PUT(p) ((p & 0xff) << 16) +#define SOC_CONTROL_ID_GET(g) ((g & 0xff) << 8) +#define SOC_CONTROL_ID_INFO(i) ((i & 0xff) << 0) +#define SOC_CONTROL_ID(g, p, i) \ + (SOC_CONTROL_ID_PUT(p) | SOC_CONTROL_ID_GET(g) |\ + SOC_CONTROL_ID_INFO(i)) + +#define SOC_CONTROL_GET_ID_PUT(id) ((id & 0xff0000) >> 16) +#define SOC_CONTROL_GET_ID_GET(id) ((id & 0x00ff00) >> 8) +#define SOC_CONTROL_GET_ID_INFO(id) ((id & 0x0000ff) >> 0) + +/* individual kcontrol info types - can be mixed with other types */ +#define SOC_CONTROL_TYPE_EXT 0 /* driver defined */ +#define SOC_CONTROL_TYPE_VOLSW 1 +#define SOC_CONTROL_TYPE_VOLSW_SX 2 +#define SOC_CONTROL_TYPE_VOLSW_S8 3 +#define SOC_CONTROL_TYPE_VOLSW_XR_SX 4 +#define SOC_CONTROL_TYPE_ENUM 6 +#define SOC_CONTROL_TYPE_ENUM_EXT 7 +#define SOC_CONTROL_TYPE_BYTES 8 +#define SOC_CONTROL_TYPE_BOOL_EXT 9 +#define SOC_CONTROL_TYPE_ENUM_VALUE 10 +#define SOC_CONTROL_TYPE_RANGE 11 +#define SOC_CONTROL_TYPE_STROBE 12 +#define SOC_CONTROL_TYPE_BYTES_EXT 13 +#define SOC_CONTROL_TYPE_VOLSW_EXT 14 + +/* convenience macros for compound control type mappings */ +#define SOC_CONTROL_IO_VOLSW \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW, \ + SOC_CONTROL_TYPE_VOLSW, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_VOLSW_SX \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_SX, \ + SOC_CONTROL_TYPE_VOLSW_SX, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_VOLSW_S8 \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_S8, \ + SOC_CONTROL_TYPE_VOLSW_S8, \ + SOC_CONTROL_TYPE_VOLSW_S8) +#define SOC_CONTROL_IO_VOLSW_XR_SX \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_XR_SX, \ + SOC_CONTROL_TYPE_VOLSW_XR_SX, \ + SOC_CONTROL_TYPE_VOLSW_XR_SX) +#define SOC_CONTROL_IO_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_ENUM \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_ENUM, \ + SOC_CONTROL_TYPE_ENUM, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_CONTROL_IO_ENUM_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_ENUM_EXT) +#define SOC_CONTROL_IO_BYTES \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_BYTES, \ + SOC_CONTROL_TYPE_BYTES, \ + SOC_CONTROL_TYPE_BYTES) +#define SOC_CONTROL_IO_BOOL_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_BOOL_EXT) +#define SOC_CONTROL_IO_ENUM_VALUE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_CONTROL_IO_RANGE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_RANGE, \ + SOC_CONTROL_TYPE_RANGE, \ + SOC_CONTROL_TYPE_RANGE) +#define SOC_CONTROL_IO_STROBE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_STROBE, \ + SOC_CONTROL_TYPE_STROBE, \ + SOC_CONTROL_TYPE_STROBE) +#define SOC_CONTROL_IO_BYTES_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_BYTES_EXT) +#define SOC_CONTROL_IO_VOLSW_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_VOLSW) + +/* individual widget kcontrol info types - can be mixed with other types */ +#define SOC_DAPM_TYPE_VOLSW 64 +#define SOC_DAPM_TYPE_ENUM_DOUBLE 65 +#define SOC_DAPM_TYPE_ENUM_VIRT 66 +#define SOC_DAPM_TYPE_ENUM_VALUE 67 +#define SOC_DAPM_TYPE_PIN 68 +#define SOC_DAPM_TYPE_ENUM_EXT 69 + +/* convenience macros for compound widget kcontrol type mappings */ +#define SOC_DAPM_IO_VOLSW \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_VOLSW, \ + SOC_DAPM_TYPE_VOLSW, \ + SOC_DAPM_TYPE_VOLSW) +#define SOC_DAPM_IO_ENUM_DOUBLE \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_ENUM_DOUBLE, \ + SOC_DAPM_TYPE_ENUM_DOUBLE, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_DAPM_IO_ENUM_VIRT \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_ENUM_VIRT, \ + SOC_DAPM_TYPE_ENUM_VIRT, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_DAPM_IO_ENUM_VALUE \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_ENUM_VALUE, \ + SOC_DAPM_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_DAPM_IO_PIN \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_PIN, \ + SOC_DAPM_TYPE_PIN, \ + SOC_DAPM_TYPE_PIN) +#define SOC_DAPM_IO_ENUM_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_ENUM) + +/* dapm widget types */ +enum snd_soc_dapm_type { + snd_soc_dapm_input = 0, /* input pin */ + snd_soc_dapm_output, /* output pin */ + snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ + snd_soc_dapm_mixer, /* mixes several analog signals together */ + snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ + snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ + snd_soc_dapm_out_drv, /* output driver */ + snd_soc_dapm_adc, /* analog to digital converter */ + snd_soc_dapm_dac, /* digital to analog converter */ + snd_soc_dapm_micbias, /* microphone bias (power) */ + snd_soc_dapm_mic, /* microphone */ + snd_soc_dapm_hp, /* headphones */ + snd_soc_dapm_spk, /* speaker */ + snd_soc_dapm_line, /* line input/output */ + snd_soc_dapm_switch, /* analog switch */ + snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ + snd_soc_dapm_pre, /* machine specific pre widget - exec first */ + snd_soc_dapm_post, /* machine specific post widget - exec last */ + snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ + snd_soc_dapm_aif_in, /* audio interface input */ + snd_soc_dapm_aif_out, /* audio interface output */ + snd_soc_dapm_siggen, /* signal generator */ + snd_soc_dapm_dai_in, /* link to DAI structure */ + snd_soc_dapm_dai_out, + snd_soc_dapm_dai_link, /* link between two DAI structures */ + snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ +}; + +/* Header magic number and string sizes */ +#define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ + +/* string sizes */ +#define SND_SOC_TPLG_NUM_TEXTS 16 + +/* ABI version */ +#define SND_SOC_TPLG_ABI_VERSION 0x2 + +/* + * File and Block header data types. + * Add new generic and vendor types to end of list. + * Generic types are handled by the core whilst vendors types are passed + * to the component drivers for handling. + */ +#define SND_SOC_TPLG_MIXER 1 +#define SND_SOC_TPLG_DAPM_GRAPH 2 +#define SND_SOC_TPLG_DAPM_WIDGET 3 +#define SND_SOC_TPLG_DAI_LINK 4 +#define SND_SOC_TPLG_PCM 5 +#define SND_SOC_TPLG_MANIFEST 6 +#define SND_SOC_TPLG_CODEC_LINK 7 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_CODEC_LINK + +/* vendor block IDs - please add new vendor types to end */ +#define SND_SOC_TPLG_VENDOR_FW 1000 +#define SND_SOC_TPLG_VENDOR_CONFIG 1001 +#define SND_SOC_TPLG_VENDOR_COEFF 1002 +#define SND_SOC_TPLG_VENDOR_CODEC 1003 + +/* + * File and Block Header. + * This header preceeds all object and object arrays below. + */ +struct snd_soc_tplg_hdr { + __le32 magic; + __le32 abi; /* ABI version */ + __le32 version; /* optional vendor specific version details */ + __le32 type; /* SND_SOC_TPLG_ */ + __le32 vendor_type; /* optional vendor specific type info */ + __le32 size; /* data bytes, excluding this header */ + __le32 id; /* identifier for block */ + char reserved[128]; +} __attribute__((packed)); + +/* + * Manifest. List totals for each payload type. Not used in parsing, but will + * be passed to the component driver before any other objects in order for any + * global componnent resource allocations. + */ +struct snd_soc_tplg_manifest { + __le32 control_elems; /* number of control elements */ + __le32 widget_elems; /* number of widget elements */ + __le32 graph_elems; /* number of graph elements */ + __le32 dai_elems; /* number of DAI elements */ + __le32 dai_link_elems; /* number of DAI link elements */ + char reserved[128]; +} __attribute__((packed)); + +/* + * Element Header. Defines number of elements in an object array. + * This prceeeds al mixers, widgets, graph elems, BEs, FEs, CC links. + */ +struct snd_soc_tplg_elems { + __le32 count; /* number of elements */ + /* elements start here */ +} __attribute__((packed)); + +/* + * Private data. + * All topology objects may have private data that can be used by the driver or + * firmware. Core will ignore this data. + */ +struct snd_soc_tplg_private { + __le32 length; + char data[0]; +} __attribute__((packed)); + +/* + * Kcontrol TLV data. + */ +struct snd_soc_tplg_ctl_tlv { + __le32 numid; /* control element numeric identification */ + __le32 length; /* in bytes aligned to 4 */ + char reserved[16]; + /* tlv data starts here */ +} __attribute__((packed)); + +/* + * kcontrol header + */ +struct snd_soc_tplg_control_hdr { + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 index; + __le32 access; + __le32 tlv_size; + char reserved[16]; +} __attribute__((packed)); + +/* + * Mixer kcontrol. + */ +struct snd_soc_tplg_mixer_control { + struct snd_soc_tplg_control_hdr hdr; + __le32 min; + __le32 max; + __le32 platform_max; + __le32 reg; + __le32 rreg; + __le32 shift; + __le32 rshift; + __le32 invert; + char reserved[64]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Enumerated kcontrol + */ +struct snd_soc_tplg_enum_control { + struct snd_soc_tplg_control_hdr hdr; + __le32 reg; + __le32 shift_l; + __le32 shift_r; + __le32 items; + __le32 mask; + __le32 count; + char reserved[64]; + char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Bytes kcontrol + */ +struct snd_soc_tplg_bytes_ext { + struct snd_soc_tplg_control_hdr hdr; + __le32 max; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * DAPM Graph Element + */ +struct snd_soc_tplg_dapm_graph_elem { + char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +/* + * DAPM Widget. + */ +struct snd_soc_tplg_dapm_widget { + __le32 id; /* snd_soc_dapm_type */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + __le32 reg; /* negative reg = no direct dapm */ + __le32 shift; /* bits to shift */ + __le32 mask; /* non-shifted mask */ + __u32 invert; /* invert the power bit */ + __u32 ignore_suspend; /* kept enabled over suspend */ + __u16 event_flags; + __u16 event_type; + __u16 num_kcontrols; + char reserved[64]; + struct snd_soc_tplg_private priv; + /* + * kcontrols that relate to this widget + * follow here after widget private data + */ +} __attribute__((packed)); + +/* + * Coefficient File Data. + */ +struct snd_soc_file_coeff_data { + __le32 count; /* in elems */ + __le32 size; /* total data size */ + __le32 id; /* associated mixer ID */ + char reserved[16]; + /* data here */ +} __attribute__((packed)); + +/* + * Stream Capabilities + */ +struct snd_soc_tplg_stream_caps { + 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 */ + char reserved[64]; +} __attribute__((packed)); + +/* + * FE or BE Stream configuration supported by SW/FW + */ +struct snd_soc_tplg_stream { + __le64 format; /* SNDRV_PCM_FMTBIT_* */ + __le32 rate; /* SNDRV_PCM_RATE_* */ + __le32 period_bytes; /* size of period in bytes */ + __le32 buffer_bytes; /* size of buffer in bytes */ + __le32 channels; /* channels */ + __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ + __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ + char reserved[16]; +} __attribute__((packed)); + +/* + * Duplex stream configuration supported by SW/FW. + */ +struct snd_soc_tplg_stream_config { + struct snd_soc_tplg_stream playback; + struct snd_soc_tplg_stream capture; + char reserved[16]; +} __attribute__((packed)); + +/* + * Describes SW/FW specific features of PCM or DAI link. + */ +struct snd_soc_tplg_pcm_dai { + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 id; /* unique BE ID - used to match */ + __le32 playback; /* supports playback in DPCM mode */ + __le32 capture; /* supports capture in DPCM mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + __le32 num_configs; /* number of configs */ + char reserved[64]; + struct snd_soc_tplg_stream_caps caps[2]; /* capabilities */ + struct snd_soc_tplg_stream_config config[0]; /* supported SW/FW configs */ +}__attribute__((packed)); + +#endif -- 2.1.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: Adds push button support for rt5650
by Bard Liao 22 Apr '15

22 Apr '15

rt5650 support headset button detection. Currently, the button detection is only implemented for rt5650 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 262 +++++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5645.h | 8 +- 2 files changed, 231 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 69528ae..7c59f9c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -198,7 +198,6 @@ static const struct reg_default rt5645_reg[] = { { 0xda, 0x0000 }, { 0xdb, 0x0003 }, { 0xdc, 0x0049 }, - { 0xdd, 0x001b }, { 0xdf, 0x0008 }, { 0xe0, 0x4000 }, { 0xe6, 0x8000 }, @@ -2368,6 +2367,8 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2398,8 +2399,9 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_update_bits(codec, RT5645_GEN_CTRL1, - RT5645_DIG_GATE_CTRL, 0); + if (!rt5645->en_button_func) + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | @@ -2414,28 +2416,71 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5645_jack_detect(struct snd_soc_codec *codec) +static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - int gpio_state, jack_type = 0; - unsigned int val; - if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) { - dev_err(codec->dev, "invalid gpio\n"); - return -EINVAL; + if (enable) { + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_update_bits(codec, + RT5645_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, + RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); + snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1, + snd_soc_read(codec, RT5650_4BTN_IL_CMD1)); + } else { + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC R power"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); } - gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); +} - dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, - gpio_state); +static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int val; - if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || - (!rt5645->pdata.gpio_hp_det_active_high && !gpio_state)) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (jack_insert) { + if (codec->component.card->instantiated) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + /* Power up necessary bits for JD if dapm is + not ready yet */ + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, + RT5645_PWR_MB1 | RT5645_PWR_MB2); + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, RT5645_PWR_LDO2); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, RT5645_PWR_MIC_DET); + } snd_soc_write(codec, RT5645_IN1_CTRL1, 0x0006); snd_soc_write(codec, RT5645_JD_CTRL3, 0x00b0); @@ -2449,32 +2494,62 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; dev_dbg(codec->dev, "val = %d\n", val); - if (val == 1 || val == 2) - jack_type = SND_JACK_HEADSET; - else - jack_type = SND_JACK_HEADPHONE; + if (codec->component.card->instantiated) { + snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_disable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, 0); + if (rt5645->pdata.jd_mode == 0) + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, 0); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, 0); + } - snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (val == 1 || val == 2) { + rt5645->jack_type = SND_JACK_HEADSET; + if (rt5645->en_button_func) { + msleep(100); + rt5645_enable_push_button_irq(codec, true); + } + } else { + rt5645->jack_type = SND_JACK_HEADPHONE; + } + + } else { /* jack out */ + rt5645->jack_type = 0; + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, false); } - snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); - snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE); - return 0; + return rt5645->jack_type; } int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); rt5645->hp_jack = hp_jack; rt5645->mic_jack = mic_jack; - rt5645_jack_detect(codec); + rt5645->btn_jack = btn_jack; + if (rt5645->btn_jack && rt5645->codec_type == CODEC_TYPE_RT5650) { + rt5645->en_button_func = true; + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + } + rt5645_irq_detection(rt5645); return 0; } @@ -2485,7 +2560,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - rt5645_jack_detect(rt5645->codec); + rt5645_irq_detection(rt5645); } static irqreturn_t rt5645_irq(int irq, void *data) @@ -2498,6 +2573,120 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + +static int rt5645_irq_detection(struct rt5645_priv *rt5645) +{ + int val, btn_type, gpio_state = 0, report = 0; + + switch (rt5645->pdata.jd_mode) { + case 0: /* Not using rt5645 JD */ + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); + dev_dbg(rt5645->codec->dev, "gpio = %d(%d)\n", + rt5645->pdata.hp_det_gpio, gpio_state); + } + if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || + (!rt5645->pdata.gpio_hp_det_active_high && + !gpio_state)) { + report = rt5645_jack_detect(rt5645->codec, 1); + } else { + report = rt5645_jack_detect(rt5645->codec, 0); + } + snd_soc_jack_report(rt5645->hp_jack, + report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, + report, SND_JACK_MICROPHONE); + return report; + case 1: /* 2 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070; + break; + default: /* 1 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0020; + break; + + } + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5645->jack_type == 0) { + report = rt5645_jack_detect(rt5645->codec, 1); + /* for push button and jack out */ + break; + } + btn_type = 0; + if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5645_button_detect(rt5645->codec); + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5645->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5645->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5645->codec, + RT5645_INT_IRQ_ST, 0x1, 0x0); + rt5645_jack_detect(rt5645->codec, 0); + break; + default: + break; + } + + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); + if (rt5645->en_button_func) + snd_soc_jack_report(rt5645->btn_jack, + report, SND_JACK_MICROPHONE); + + return report; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2784,8 +2973,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.en_jd_func) { regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU); + RT5645_IRQ_CLK_GATE_CTRL, RT5645_IRQ_CLK_GATE_CTRL); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index db78e94..4473636 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2184,6 +2184,7 @@ struct rt5645_priv { struct i2c_client *i2c; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; + struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; int codec_type; @@ -2196,9 +2197,12 @@ struct rt5645_priv { int pll_src; int pll_in; int pll_out; + + int jack_type; + bool en_button_func; }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); - + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack); #endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/2] Calculate BCLK using TDM slots and remove channels rule
by Jyri Sarha 22 Apr '15

22 Apr '15

The first patch is a bugfix. I did not have the HW see the problem myself, but reading from the code the problem is evident. This should also fix Dan Carpenter's concern about stack usage in davinci_mcasp_hw_rule_channels(), as the whole function is removed. The second patch is just an optimization of the sample-rate rule. In effect I put in use Takashi Iwai's suggestion for fixing the stack usage issue in the channels rule. Jyri Sarha (2): ASoC: davinci-mcasp: Calculate BCLK using TDM slots and remove channels rule ASoC: davinci-macsp: Optimize implicit BLCK sample-rate rule sound/soc/davinci/davinci-mcasp.c | 104 ++++++++++---------------------------- 1 file changed, 27 insertions(+), 77 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v7 0/7] ALSA: jack: Refactoring for jack kctls
by Jie Yang 22 Apr '15

22 Apr '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For HDA, With snd_jack_new(), we can use phantom_jack = true for phantom jack creating, and the jack kctl pointer will be filled with the new created kcontrol. Changes in v7: 1. move name generating and index getting part into ctljack.c; 2. remove exposing private_data and destructor to hda, then jack core can handle kctls by itself totally. 3. some small code fixing. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (7): ALSA: jack: implement kctl creating for jack device ALSA: Jack: refactoring snd_kctl_jack_new to support embedded kctl ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 48 +++++++++ include/sound/control.h | 2 +- include/sound/jack.h | 19 +++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 41 ++++++-- sound/core/jack.c | 156 +++++++++++++++++++++++++---- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_jack.c | 53 ++++------ sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 12 files changed, 259 insertions(+), 76 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

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[alsa-devel] [RFC 0/4] ASoC DSP topology
by Liam Girdwood 22 Apr '15

22 Apr '15

It's been about 2 years since posting V1 of the ASoC dynamic firmware patches and a lot has changed since then ... This RFC series is being posted instead of a V2 patch series atm since we are still completing some testing on the codebase and developing a new userspace tool. Expect V2 once more testing is completed shortly. ASoC topology (formerly called dynamic firmware) allows audio DSP topology to be defined as part of the firmware package and not hard coded into the driver. This allows DSP vendors to ship multiple different firmwares or firmwares with programmable topologies for different devices (i.e. phones, tablets, TVs) using a single DSP driver. The DSP topology can be contained as part of the firmware file or exist as separate files. The topology core and file format is generic and not tied to any single vendor or device. The topology core consists of kernel and userspace components :- Kernel. The kernel core reads in the topology data and builds the driver capabilities and audio topology based on this data. This includes building the DAPM graph, setting PCM and DAI capabilities, registering kcontrols and registering DAPM widgets. The userspace tool allows the creation of binary DSP FW topology files based on a topology text files. Changes since V1:- o Name changed to ASoC Topology o Structures from soc.h and soc-dapm.h no longer exposed as UAPIs o Moved to component model from codec/platform model. o Added reserved fields in each UAPI structure. o UAPI structures packed and endianess defined for each structure. o Added support for PCM, DAI and codec <-> codec link capabilities. o Added support for client drivers manifests. o Added support for private data for each dynamic object type. o Single list of dynamic objects rather than list of each object type. Liam

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[alsa-devel] [PATCH v2 0/7] ASoC: hda - add ASoC Skylake HD audio driver
by Vinod Koul 22 Apr '15

22 Apr '15

This version incorporates the comments from Takashi & Mark on last RFC series. This also clubs the ASoC HDA wrapper and controller driver to single series. This is based on for ASoC Skylake HD Audio driver and based on Takashi's topic/hda Dropped the based on comments ALSA: hda - add id table support for hdac device/driver ALSA: hda - Add generic helper function to register/unregister driver ALSA: hda - add generic function to unregister all devices on bus Dropped the below patch and add new patch ALSA: hda - add id table support for hdac device/driver Jeeja KP (6): ALSA: hda - add generic functions to set hdac stream params ASoC: hda - add soc hda codec driver wrapper ASoC: hda - add Skylake platform driver ASoC: hda - add Skylake HD audio driver ASoC: hda - enable ASoC Skylake HD audio driver ASoC: hda - added Skylake I2S machine driver Ramesh Babu (1): ALSA: hda - add ASoC device type for hda core include/sound/hdaudio.h | 4 + sound/hda/hdac_stream.c | 31 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/soc-hda-codec.c | 79 ++++ sound/soc/codecs/soc-hda-codec.h | 43 +++ sound/soc/hda/Kconfig | 43 +++ sound/soc/hda/Makefile | 6 + sound/soc/hda/boards/Makefile | 3 + sound/soc/hda/boards/skl_rt286.c | 313 ++++++++++++++++ sound/soc/hda/hda_skl.c | 748 ++++++++++++++++++++++++++++++++++++++ sound/soc/hda/hda_skl.h | 44 +++ sound/soc/hda/hda_skl_pcm.c | 498 +++++++++++++++++++++++++ 15 files changed, 1819 insertions(+) create mode 100644 sound/soc/codecs/soc-hda-codec.c create mode 100644 sound/soc/codecs/soc-hda-codec.h create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/boards/Makefile create mode 100644 sound/soc/hda/boards/skl_rt286.c create mode 100644 sound/soc/hda/hda_skl.c create mode 100644 sound/soc/hda/hda_skl.h create mode 100644 sound/soc/hda/hda_skl_pcm.c -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: asound.h - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN
by Vinod Koul 21 Apr '15

21 Apr '15

we have defined SNDRV_CTL_ELEM_ID_NAME_MAXLEN as size of name array so use this define instead of numeric value Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 46145a5277fe..a45be6bdcf5b 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -864,7 +864,7 @@ struct snd_ctl_elem_id { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - unsigned char name[44]; /* ASCII name of item */ + unsigned char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* ASCII name of item */ unsigned int index; /* index of item */ }; -- 1.7.9.5

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