From 1689333f8311f5952ee69d64adf242028dc7e6c6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Apr 2017 01:35:18 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() simple-card already has asoc_simple_card_parse_dai(), but graph base parsing needs graph specific version of it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'include/sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index af58d2362975..efab584af11b 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -60,6 +60,16 @@ int asoc_simple_card_parse_dai(struct device_node *node, const char *cells_name, int *is_single_links); +#define asoc_simple_card_parse_graph_cpu(ep, dai_link) \ + asoc_simple_card_parse_graph_dai(ep, &dai_link->cpu_of_node, \ + &dai_link->cpu_dai_name) +#define asoc_simple_card_parse_graph_codec(ep, dai_link) \ + asoc_simple_card_parse_graph_dai(ep, &dai_link->codec_of_node, \ + &dai_link->codec_dai_name) +int asoc_simple_card_parse_graph_dai(struct device_node *ep, + struct device_node **endpoint_np, + const char **dai_name); + int asoc_simple_card_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai); -- cgit v1.2.3 From b7c752d68aee9c066cb0bd2f24ee73aed64575e8 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 18 May 2017 16:32:36 +0100 Subject: ASoC: cs35l35: Add Boost Inductor Calculation Add the Boost Inductor parameters based off the size of the inductor on the HW setup Signed-off-by: Brian Austin Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/cs35l35.h | 2 ++ sound/soc/codecs/cs35l35.c | 82 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l35.h | 6 ++++ 3 files changed, 90 insertions(+) (limited to 'include/sound') diff --git a/include/sound/cs35l35.h b/include/sound/cs35l35.h index 29da899e17e4..d69cd7847afd 100644 --- a/include/sound/cs35l35.h +++ b/include/sound/cs35l35.h @@ -99,6 +99,8 @@ struct cs35l35_platform_data { bool shared_bst; /* Specifies this amp is using an external boost supply */ bool ext_bst; + /* Inductor Value */ + int boost_ind; /* ClassH Algorithm */ struct classh_cfg classh_algo; /* Monitor Config */ diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index f8aef5869b03..c6eabb8610f0 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -756,6 +756,76 @@ static int cs35l35_codec_set_sysclk(struct snd_soc_codec *codec, return ret; } +static int cs35l35_boost_inductor(struct cs35l35_private *cs35l35, + int inductor) +{ + struct regmap *regmap = cs35l35->regmap; + unsigned int bst_ipk = 0; + + /* + * Digital Boost Converter Configuration for feedback, + * ramping, switching frequency, and estimation block seeding. + */ + + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_SWFREQ_MASK, 0x00); + + regmap_read(regmap, CS35L35_BST_PEAK_I, &bst_ipk); + bst_ipk &= CS35L35_BST_IPK_MASK; + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x24); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x24); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x00); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x4E); + break; + case 1200: /* 1.2 uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x20); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x20); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x01); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x47); + break; + case 1500: /* 1.5uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x20); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x20); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x02); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x3C); + break; + case 2200: /* 2.2uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x19); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x25); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x03); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x23); + break; + default: + dev_err(cs35l35->dev, "Invalid Inductor Value %d uH\n", + inductor); + return -EINVAL; + } + return 0; +} + static int cs35l35_codec_probe(struct snd_soc_codec *codec) { struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); @@ -775,6 +845,10 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec) cs35l35->pdata.bst_ipk << CS35L35_BST_IPK_SHIFT); + ret = cs35l35_boost_inductor(cs35l35, cs35l35->pdata.boost_ind); + if (ret) + return ret; + if (cs35l35->pdata.gain_zc) regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, CS35L35_AMP_GAIN_ZC_MASK, @@ -1198,6 +1272,14 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, pdata->bst_ipk = (val32 - 1680) / 110; } + ret = of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val32); + if (ret >= 0) { + pdata->boost_ind = val32; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + if (of_property_read_u32(np, "cirrus,sp-drv-strength", &val32) >= 0) pdata->sp_drv_str = val32; if (of_property_read_u32(np, "cirrus,sp-drv-unused", &val32) >= 0) diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index 5a6e43a87c4d..621bfef70d03 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -200,6 +200,12 @@ #define CS35L35_SP_I2S_DRV_MASK 0x03 #define CS35L35_SP_I2S_DRV_SHIFT 0 +/* Boost Converter Config */ +#define CS35L35_BST_CONV_COEFF_MASK 0xFF +#define CS35L35_BST_CONV_SLOPE_MASK 0xFF +#define CS35L35_BST_CONV_LBST_MASK 0x03 +#define CS35L35_BST_CONV_SWFREQ_MASK 0xF0 + /* Class H Algorithm Control */ #define CS35L35_CH_STEREO_MASK 0x40 #define CS35L35_CH_STEREO_SHIFT 6 -- cgit v1.2.3 From a180e8b988437b3e84a1b501ac4d073467602ca6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 18 May 2017 01:39:25 +0000 Subject: ASoC: add snd_soc_get_dai_id() function ALSA SoC needs to know connected DAI ID for detecting. It is not a big problem if device/driver was only for sound, but getting DAI ID will be difficult if device includes both Video/Sound, like HDMI. To solve this issue, this patch adds new snd_soc_get_dai_id() and its related .of_xlate_dai_id callback on component driver. In below case, we can handle Sound port (= port@2) as ID = 0 if .of_xlate_dai_id has its support. hdmi { port@0 { /* VIDEO */ }; port@1 { /* VIDEO */ }; port@2 { /* SOUND */ }; }; Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 37 +++++++++++++++++++++++++++++++++++++ 2 files changed, 40 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 5170fd81e1fd..9c94b97c17f8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -803,6 +803,8 @@ struct snd_soc_component_driver { int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name); + int (*of_xlate_dai_id)(struct snd_soc_component *comment, + struct device_node *endpoint); void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, int subseq); int (*stream_event)(struct snd_soc_component *, int event); @@ -1676,6 +1678,7 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, struct device_node **framemaster); +int snd_soc_get_dai_id(struct device_node *ep); int snd_soc_get_dai_name(struct of_phandle_args *args, const char **dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index aae099c0e502..b0fb17082691 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -34,6 +34,7 @@ #include #include #include +#include #include #include #include @@ -4044,6 +4045,42 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); +int snd_soc_get_dai_id(struct device_node *ep) +{ + struct snd_soc_component *pos; + struct device_node *node; + int ret; + + node = of_graph_get_port_parent(ep); + + /* + * For example HDMI case, HDMI has video/sound port, + * but ALSA SoC needs sound port number only. + * Thus counting HDMI DT port/endpoint doesn't work. + * Then, it should have .of_xlate_dai_id + */ + ret = -ENOTSUPP; + mutex_lock(&client_mutex); + list_for_each_entry(pos, &component_list, list) { + struct device_node *component_of_node = pos->dev->of_node; + + if (!component_of_node && pos->dev->parent) + component_of_node = pos->dev->parent->of_node; + + if (component_of_node != node) + continue; + + if (pos->driver->of_xlate_dai_id) + ret = pos->driver->of_xlate_dai_id(pos, ep); + + break; + } + mutex_unlock(&client_mutex); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_get_dai_id); + int snd_soc_get_dai_name(struct of_phandle_args *args, const char **dai_name) { -- cgit v1.2.3 From 96203fb4237bf70f0fd0fa307ca2975077db3ceb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 18 May 2017 01:40:20 +0000 Subject: ASoC: hdmi-codec: add .get_dai_id support ALSA SoC needs to know connected DAI ID for probing. It is not a big problem if device/driver was only for sound, but getting DAI ID will be difficult if device includes both Video/Sound, like HDMI. To solve this issue, this patch adds new .get_dai_id callback on hdmi_codec_ops Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/hdmi-codec.h | 9 +++++++++ sound/soc/codecs/hdmi-codec.c | 13 +++++++++++++ 2 files changed, 22 insertions(+) (limited to 'include/sound') diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 915c4357945c..9483c55f871b 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -18,9 +18,11 @@ #ifndef __HDMI_CODEC_H__ #define __HDMI_CODEC_H__ +#include #include #include #include +#include #include /* @@ -87,6 +89,13 @@ struct hdmi_codec_ops { */ int (*get_eld)(struct device *dev, void *data, uint8_t *buf, size_t len); + + /* + * Getting DAI ID + * Optional + */ + int (*get_dai_id)(struct snd_soc_component *comment, + struct device_node *endpoint); }; /* HDMI codec initalization data */ diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8659b76b066a..6d05161b625d 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -719,6 +719,18 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .pcm_new = hdmi_codec_pcm_new, }; +static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + int ret = -ENOTSUPP; /* see snd_soc_get_dai_id() */ + + if (hcp->hcd.ops->get_dai_id) + ret = hcp->hcd.ops->get_dai_id(component, endpoint); + + return ret; +} + static struct snd_soc_codec_driver hdmi_codec = { .component_driver = { .controls = hdmi_controls, @@ -727,6 +739,7 @@ static struct snd_soc_codec_driver hdmi_codec = { .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, .num_dapm_routes = ARRAY_SIZE(hdmi_routes), + .of_xlate_dai_id = hdmi_of_xlate_dai_id, }, }; -- cgit v1.2.3 From 8e16638256425faf74c5b9ffa40e5f0d9aa4413b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 5 Jun 2017 04:28:45 +0000 Subject: ASoC: simple-card-utils: share same dev_dbg() for sysclk Let's share same debug message for sysclk Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 9 ++++++--- sound/soc/generic/audio-graph-card.c | 7 ------- sound/soc/generic/audio-graph-scu-card.c | 5 ----- sound/soc/generic/simple-card-utils.c | 5 ++++- sound/soc/generic/simple-card.c | 7 ------- sound/soc/generic/simple-scu-card.c | 5 ----- 6 files changed, 10 insertions(+), 28 deletions(-) (limited to 'include/sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index efab584af11b..108cae459ed0 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -35,13 +35,16 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, char *prefix); #define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai) \ - asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai) + asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \ + dai_link->cpu_dai_name) #define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai) \ - asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai) + asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\ + dai_link->codec_dai_name) int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, - struct asoc_simple_dai *simple_dai); + struct asoc_simple_dai *simple_dai, + const char *name); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2c3a1cc01442..0180b286bee3 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -169,13 +169,6 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, dai_link->ops = &asoc_graph_card_ops; dai_link->init = asoc_graph_card_dai_init; - dev_dbg(dev, "\tcpu : %s / %d\n", - dai_link->cpu_dai_name, - cpu_dai->sysclk); - dev_dbg(dev, "\tcodec : %s / %d\n", - dai_link->codec_dai_name, - codec_dai->sysclk); - asoc_simple_card_canonicalize_cpu(dai_link, card->num_links == 1); diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 1ce727b6bc21..0066102f5bc4 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -185,11 +185,6 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, dai_link->ops = &asoc_graph_card_ops; dai_link->init = asoc_graph_card_dai_init; - dev_dbg(dev, "\t%s / %04x / %d\n", - dai_link->name, - dai_link->dai_fmt, - dai_props->sysclk); - return 0; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 1f08064f65b1..d9d8b8a58348 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -113,7 +113,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, - struct asoc_simple_dai *simple_dai) + struct asoc_simple_dai *simple_dai, + const char *name) { struct clk *clk; u32 val; @@ -136,6 +137,8 @@ int asoc_simple_card_parse_clk(struct device *dev, simple_dai->sysclk = clk_get_rate(clk); } + dev_dbg(dev, "%s : sysclk = %d\n", name, simple_dai->sysclk); + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 1da0e2b068c3..e86c6e16146b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -301,13 +301,6 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_link->ops = &asoc_simple_card_ops; dai_link->init = asoc_simple_card_dai_init; - dev_dbg(dev, "\tcpu : %s / %d\n", - dai_link->cpu_dai_name, - dai_props->cpu_dai.sysclk); - dev_dbg(dev, "\tcodec : %s / %d\n", - dai_link->codec_dai_name, - dai_props->codec_dai.sysclk); - asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); dai_link_of_err: diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 5f4384f322c1..9a251400685e 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -189,11 +189,6 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, dai_link->ops = &asoc_simple_card_ops; dai_link->init = asoc_simple_card_dai_init; - dev_dbg(dev, "\t%s / %04x / %d\n", - dai_link->name, - dai_link->dai_fmt, - dai_props->sysclk); - return 0; } -- cgit v1.2.3 From 294de6e372673229432dc8bcd80964223bc1589d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Jun 2017 15:55:04 +0100 Subject: ASoC: topology: Fix potential build issues with undeclared structs We should be declaring snd_kcontrol_new and soc_dai_link as both are used within this header so need to be declared. [Reworded commit message to indicate this wasn't an immediate build failure -- broonie] Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f9cc7b9271ac..b8da221615e0 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -28,6 +28,8 @@ struct snd_soc_component; struct snd_soc_tplg_pcm_fe; struct snd_soc_dapm_context; struct snd_soc_card; +struct snd_kcontrol_new; +struct snd_soc_dai_link; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ -- cgit v1.2.3 From ebd259d33a900b28ef774c4c26e8ce6e2baea7e5 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 9 Jun 2017 15:43:23 +0100 Subject: ASoC: topology: Allow bespoke configuration post widget creation Current topology only allows for widget configuration before the widget is registered. This patch also allows further configuration and usage after registration is complete. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 3 +++ sound/soc/soc-topology.c | 19 +++++++++++++++++++ 2 files changed, 22 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index b8da221615e0..f552c3f56368 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -118,6 +118,9 @@ struct snd_soc_tplg_ops { int (*widget_load)(struct snd_soc_component *, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); + int (*widget_ready)(struct snd_soc_component *, + struct snd_soc_dapm_widget *, + struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f4ec236a418e..12e189701924 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -344,6 +344,17 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, return 0; } +/* optionally pass new dynamic widget to component driver. This is mainly for + * external widgets where we can assign private data/ops */ +static int soc_tplg_widget_ready(struct soc_tplg *tplg, + struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) +{ + if (tplg->comp && tplg->ops && tplg->ops->widget_ready) + return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + + return 0; +} + /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, struct snd_soc_dai_driver *dai_drv) @@ -1579,8 +1590,16 @@ widget: widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; list_add(&widget->dobj.list, &tplg->comp->dobj_list); + + ret = soc_tplg_widget_ready(tplg, widget, w); + if (ret < 0) + goto ready_err; + return 0; +ready_err: + snd_soc_tplg_widget_remove(widget); + snd_soc_dapm_free_widget(widget); hdr_err: kfree(template.sname); err: -- cgit v1.2.3 From 891caea417469b4efdf506b6be1ef461b759c999 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 9 Jun 2017 00:43:18 +0000 Subject: ASoC: simple_card_utils: add asoc_simple_card_clk_xxx() Current simple-card-utils sets asoc_simple_dai::clk via asoc_simple_card_parse_clk(). Current simple card drivers are using it directly for clk_enable/disable. Encapsulation is one of simple card util's purpose. Let's encapsulate it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 19 ++++++++++++++++++- 2 files changed, 20 insertions(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 108cae459ed0..840d624148df 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -45,6 +45,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai, const char *name); +int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai); +void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d9d8b8a58348..beb4e3817d22 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -110,6 +110,22 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); +static void asoc_simple_card_clk_register(struct asoc_simple_dai *dai, + struct clk *clk) +{ + dai->clk = clk; +} + +int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) +{ + return clk_prepare_enable(dai->clk); +} + +void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) +{ + clk_disable_unprepare(dai->clk); +} + int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, @@ -128,7 +144,8 @@ int asoc_simple_card_parse_clk(struct device *dev, clk = devm_get_clk_from_child(dev, node, NULL); if (!IS_ERR(clk)) { simple_dai->sysclk = clk_get_rate(clk); - simple_dai->clk = clk; + + asoc_simple_card_clk_register(simple_dai, clk); } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) { simple_dai->sysclk = val; } else { -- cgit v1.2.3 From e68ba207444354ddf295de2e7fbcc97c06cccc8b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 14 Jun 2017 00:34:35 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_of_parse_tdm() Current simple card drivers are using asoc_simple_dai's tx_slot_mask, rx_slot_mask, slots, slot_width directly to parse TDM. Encapsulation is one of simple card util's purpose. Let's add asoc_simple_card_of_parse_tdm for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include/sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 840d624148df..2679312228b3 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -75,6 +75,12 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, struct device_node **endpoint_np, const char **dai_name); +#define asoc_simple_card_of_parse_tdm(np, dai) \ + snd_soc_of_parse_tdm_slot(np, &(dai)->tx_slot_mask, \ + &(dai)->rx_slot_mask, \ + &(dai)->slots, \ + &(dai)->slot_width); + int asoc_simple_card_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai); -- cgit v1.2.3 From 13bb1cc0ad205b2aeeb8d2ea5c790a396135283d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:24:09 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_convert_fixup() Current simple/audio scu card drivers are supporting same convert-rate/convert-channels on DT, but doesn't use same function for it. Encapsulation is one of simple card util's purpose. Let's add asoc_simple_card_parse_convert/asoc_simple_card_convert_fixup Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 10 +++++++++ sound/soc/generic/simple-card-utils.c | 40 +++++++++++++++++++++++++++++++++++ 2 files changed, 50 insertions(+) (limited to 'include/sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 2679312228b3..cc318ccd6a2d 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -22,6 +22,11 @@ struct asoc_simple_dai { struct clk *clk; }; +struct asoc_simple_card_data { + u32 convert_rate; + u32 convert_channels; +}; + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, @@ -90,4 +95,9 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int asoc_simple_card_clean_reference(struct snd_soc_card *card); +void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, + struct snd_pcm_hw_params *params); +void asoc_simple_card_parse_convert(struct device *dev, char *prefix, + struct asoc_simple_card_data *data); + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 2ad7633292bf..948a18842e64 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -13,6 +13,46 @@ #include #include +void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + if (data->convert_rate) + rate->min = + rate->max = data->convert_rate; + + if (data->convert_channels) + channels->min = + channels->max = data->convert_channels; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_convert_fixup); + +void asoc_simple_card_parse_convert(struct device *dev, char *prefix, + struct asoc_simple_card_data *data) +{ + struct device_node *np = dev->of_node; + char prop[128]; + + if (!prefix) + prefix = ""; + + /* sampling rate convert */ + snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-rate"); + of_property_read_u32(np, prop, &data->convert_rate); + + /* channels transfer */ + snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); + of_property_read_u32(np, prop, &data->convert_channels); + + dev_dbg(dev, "convert_rate %d\n", data->convert_rate); + dev_dbg(dev, "convert_channels %d\n", data->convert_channels); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_parse_convert); + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, -- cgit v1.2.3 From 3296d07826ebc698113832acb426f037e9b3b253 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 15 Jun 2017 00:25:02 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_of_parse_routing() Current simple card drivers are parsing routing on each own driver. Encapsulation is one of simple card util's purpose. Let's add asoc_simple_card_of_parse_routing for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 4 ++++ sound/soc/generic/simple-card-utils.c | 22 ++++++++++++++++++++++ 2 files changed, 26 insertions(+) (limited to 'include/sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index cc318ccd6a2d..889c8ff86369 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -100,4 +100,8 @@ void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, void asoc_simple_card_parse_convert(struct device *dev, char *prefix, struct asoc_simple_card_data *data); +int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, + char *prefix, + int optional); + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 948a18842e64..a2b6d95bc2f9 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -375,6 +375,28 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) } EXPORT_SYMBOL_GPL(asoc_simple_card_clean_reference); +int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, + char *prefix, + int optional) +{ + struct device_node *node = card->dev->of_node; + char prop[128]; + + if (!prefix) + prefix = ""; + + snprintf(prop, sizeof(prop), "%s%s", prefix, "routing"); + + if (!of_property_read_bool(node, prop)) { + if (optional) + return 0; + return -EINVAL; + } + + return snd_soc_of_parse_audio_routing(card, prop); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_routing); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From b31f11d036e689ba9e60d581ffe8e032a6305da9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 16 Jun 2017 01:38:50 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_of_parse_widgets() Current simple card drivers are parsing widgets on each own driver (only simple-card at this point, but will be supported on all drivers) Encapsulation is one of simple card util's purpose. Let's add asoc_simple_card_of_parse_widgets for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 2 files changed, 21 insertions(+) (limited to 'include/sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 889c8ff86369..42c6a6ac3ce6 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -103,5 +103,7 @@ void asoc_simple_card_parse_convert(struct device *dev, char *prefix, int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, char *prefix, int optional); +int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, + char *prefix); #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index a2b6d95bc2f9..26d64fa40c9c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -397,6 +397,25 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_routing); +int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, + char *prefix) +{ + struct device_node *node = card->dev->of_node; + char prop[128]; + + if (!prefix) + prefix = ""; + + snprintf(prop, sizeof(prop), "%s%s", prefix, "widgets"); + + if (of_property_read_bool(node, prop)) + return snd_soc_of_parse_audio_simple_widgets(card, prop); + + /* no widgets is not error */ + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From 895750228c9d3361ed82e9786322604de3232466 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 28 Jun 2017 14:49:37 +0200 Subject: ASoC: rt5645: rename jd_invert flag in platform data The jd_invert flag is actually used for level triggered IRQ. Rename it to let code more readable. Signed-off-by: Bard Liao Tested-by: Hans de Goede Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/rt5645.h | 4 ++-- sound/soc/codecs/rt5645.c | 8 ++++---- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'include/sound') diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf6152e778..c427f10a39ae 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -21,8 +21,8 @@ struct rt5645_platform_data { /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ unsigned int jd_mode; - /* Invert JD when jack insert */ - bool jd_invert; + /* Use level triggered irq */ + bool level_trigger_irq; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 8e419ea418e9..e0c09bbd3f12 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3151,7 +3151,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } - if (rt5645->pdata.jd_invert) + if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); } else { /* jack out */ @@ -3172,7 +3172,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_disable_pin(dapm, "LDO2"); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); - if (rt5645->pdata.jd_invert) + if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } @@ -3586,7 +3586,7 @@ static struct rt5645_platform_data buddy_platform_data = { .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, - .jd_invert = true, + .level_trigger_irq = true, }; static struct dmi_system_id dmi_platform_intel_broadwell[] = { @@ -3838,7 +3838,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); - if (rt5645->pdata.jd_invert) { + if (rt5645->pdata.level_trigger_irq) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } -- cgit v1.2.3 From aea086dda2d5df659a7c5d9efe85721e9442a133 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 28 Jun 2017 14:49:38 +0200 Subject: ASoC: rt5645: add inv_jd1_1 flag The flag will invert jd1_1 status. Which will be used if the jack connector is normal closed. Signed-off-by: Bard Liao Tested-by: Hans de Goede Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/rt5645.h | 2 ++ sound/soc/codecs/rt5645.c | 4 ++++ 2 files changed, 6 insertions(+) (limited to 'include/sound') diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index c427f10a39ae..d0c33a9972b9 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,8 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Use level triggered irq */ bool level_trigger_irq; + /* Invert JD1_1 status polarity */ + bool inv_jd1_1; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e0c09bbd3f12..162044d82632 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3833,6 +3833,10 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, default: break; } + if (rt5645->pdata.inv_jd1_1) { + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + } } regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, -- cgit v1.2.3 From 286345eef97ea8f4ea223410f025ed35f265e506 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 23 Jun 2017 12:35:00 -0400 Subject: ASoC: dwc: Added a quirk DW_I2S_QUIRK_16BIT_IDX_OVERRIDE to dwc driver Added quirk DW_I2S_QUIRK_16BIT_IDX_OVERRIDE to Designware driver. This quirk will set idx value to 1. By setting this quirk, it will override supported format as 16 bit resolution and bus width as 2 Bytes. Reviewed-by: Alex Deucher Signed-off-by: Vijendar Mukunda Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- include/sound/designware_i2s.h | 1 + sound/soc/dwc/dwc-i2s.c | 6 ++++++ 2 files changed, 7 insertions(+) (limited to 'include/sound') diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 5681855396c4..830f5caa915c 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -47,6 +47,7 @@ struct i2s_platform_data { #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) + #define DW_I2S_QUIRK_16BIT_IDX_OVERRIDE (1 << 2) unsigned int quirks; unsigned int i2s_reg_comp1; unsigned int i2s_reg_comp2; diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 9c46e4112026..916067638180 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -496,6 +496,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, idx = COMP1_TX_WORDSIZE_0(comp1); if (WARN_ON(idx >= ARRAY_SIZE(formats))) return -EINVAL; + if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) + idx = 1; dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->playback.channels_max = 1 << (COMP1_TX_CHANNELS(comp1) + 1); @@ -508,6 +510,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, idx = COMP2_RX_WORDSIZE_0(comp2); if (WARN_ON(idx >= ARRAY_SIZE(formats))) return -EINVAL; + if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) + idx = 1; dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->capture.channels_max = 1 << (COMP1_RX_CHANNELS(comp1) + 1); @@ -543,6 +547,8 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, if (ret < 0) return ret; + if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) + idx = 1; /* Set DMA slaves info */ dev->play_dma_data.pd.data = pdata->play_dma_data; dev->capture_dma_data.pd.data = pdata->capture_dma_data; -- cgit v1.2.3 From 8a70b4544ef4f094cc2c52734e097cc358f56603 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 29 Jun 2017 14:22:24 +0100 Subject: ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs. Add some DAPM widget types to better support the construction of DAPM graphs within DSPs. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 18 ++++++++++++++++++ include/sound/soc-dapm.h | 7 +++++++ include/uapi/sound/asoc.h | 10 +++++++++- sound/soc/soc-topology.c | 8 ++++++++ 4 files changed, 42 insertions(+), 1 deletion(-) (limited to 'include/sound') diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index a27f42befa4d..8e44107933ab 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -105,6 +105,24 @@ Pre Special PRE widget (exec before all others) Post Special POST widget (exec after all others) +Buffer + Inter widget audio data buffer within a DSP. +Scheduler + DSP internal scheduler that schedules component/pipeline processing + work. +Effect + Widget that performs an audio processing effect. +SRC + Sample Rate Converter within DSP or CODEC +ASRC + Asynchronous Sample Rate Converter within DSP or CODEC +Encoder + Widget that encodes audio data from one format (usually PCM) to another + usually more compressed format. +Decoder + Widget that decodes audio data from a compressed format to an + uncompressed format like PCM. + (Widgets are defined in include/sound/soc-dapm.h) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a466f4bdc835..344b96c206a3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,13 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ + snd_soc_dapm_buffer, /* DSP/CODEC internal buffer */ + snd_soc_dapm_scheduler, /* DSP/CODEC internal scheduler */ + snd_soc_dapm_effect, /* DSP/CODEC effect component */ + snd_soc_dapm_src, /* DSP/CODEC SRC component */ + snd_soc_dapm_asrc, /* DSP/CODEC ASRC component */ + snd_soc_dapm_encoder, /* FW/SW audio encoder component */ + snd_soc_dapm_decoder, /* FW/SW audio decoder component */ }; enum snd_soc_dapm_subclass { diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6702533c8bd8..78014ec56357 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -73,7 +73,15 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_BUFFER 16 +#define SND_SOC_TPLG_DAPM_SCHEDULER 17 +#define SND_SOC_TPLG_DAPM_EFFECT 18 +#define SND_SOC_TPLG_DAPM_SIGGEN 19 +#define SND_SOC_TPLG_DAPM_SRC 20 +#define SND_SOC_TPLG_DAPM_ASRC 21 +#define SND_SOC_TPLG_DAPM_ENCODER 22 +#define SND_SOC_TPLG_DAPM_DECODER 23 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DECODER /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 002772e3ba2c..dd3a391476ae 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -242,6 +242,14 @@ static const struct soc_tplg_map dapm_map[] = { {SND_SOC_TPLG_DAPM_DAI_IN, snd_soc_dapm_dai_in}, {SND_SOC_TPLG_DAPM_DAI_OUT, snd_soc_dapm_dai_out}, {SND_SOC_TPLG_DAPM_DAI_LINK, snd_soc_dapm_dai_link}, + {SND_SOC_TPLG_DAPM_BUFFER, snd_soc_dapm_buffer}, + {SND_SOC_TPLG_DAPM_SCHEDULER, snd_soc_dapm_scheduler}, + {SND_SOC_TPLG_DAPM_EFFECT, snd_soc_dapm_effect}, + {SND_SOC_TPLG_DAPM_SIGGEN, snd_soc_dapm_siggen}, + {SND_SOC_TPLG_DAPM_SRC, snd_soc_dapm_src}, + {SND_SOC_TPLG_DAPM_ASRC, snd_soc_dapm_asrc}, + {SND_SOC_TPLG_DAPM_ENCODER, snd_soc_dapm_encoder}, + {SND_SOC_TPLG_DAPM_DECODER, snd_soc_dapm_decoder}, }; static int tplc_chan_get_reg(struct soc_tplg *tplg, -- cgit v1.2.3