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

[alsa-devel] Applied "ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a0c426fe143328760c9fd565cd203a37a7b4fde8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:42 +0900 Subject: [PATCH] ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() We will get DAI ID from "reg" property if it has on DT, otherwise get it by counting port/endpoint. But in below case, we need to get DAI ID = 0 via port reg = <0>, but current implementation returns ID = 1, because it can't judge ID = 0 was from "non reg" or "reg = <0>". Thus, it will count port/endpoint number as "non reg" case. of_graph_parse_endpoint() implementation itself is not a problem, but because asoc_simple_card_get_dai_id() need to count port/endpoint number when "non reg" case, it need to know ID = 0 was from "non reg" or "reg = <0>". This patch fix this issue. port { reg = <0>; xxxx: endpoint@0 { }; => xxxx: endpoint@1 { }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b807a47515eb..336895f7fd1e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -283,12 +283,20 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) /* use endpoint/port reg if exist */ ret = of_graph_parse_endpoint(ep, &info); if (ret == 0) { - if (info.id) + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_get_property(ep, "reg", NULL)) return info.id; - if (info.port) + + node = of_get_parent(ep); + of_node_put(node); + if (of_get_property(node, "reg", NULL)) return info.port; } - node = of_graph_get_port_parent(ep); /* -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add asoc_graph_card_get_conversion()" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-card: add asoc_graph_card_get_conversion() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 40dfae169ad047535d566a4791daae3b08f71c0c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:48 +0900 Subject: [PATCH] ASoC: audio-graph-card: add asoc_graph_card_get_conversion() audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_graph_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0d6144560a1e..c3e80bc27e80 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -169,6 +169,22 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_graph_card_get_conversion(struct device *dev, + struct device_node *ep, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); + struct device_node *node = of_graph_get_port_parent(ep); + + asoc_simple_card_parse_convert(dev, top, NULL, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, ports, NULL, adata); + asoc_simple_card_parse_convert(dev, port, NULL, adata); + asoc_simple_card_parse_convert(dev, ep, NULL, adata); +} + static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, struct device_node *cpu_ep, struct device_node *codec_ep, @@ -194,11 +210,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - asoc_simple_card_parse_convert(dev, top, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ports, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, port, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ep, NULL, &dai_props->adata); + asoc_graph_card_get_conversion(dev, ep, &dai_props->adata); of_node_put(ports); of_node_put(port); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e4f4fdfc57d9c846862ea6109e356b3a4542df5b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:27 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/audio-graph-scu-card.txt | 123 ------------------ 1 file changed, 123 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt deleted file mode 100644 index 62d42768a00b..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ /dev/null @@ -1,123 +0,0 @@ -Audio-Graph-SCU-Card: - -Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM". - -It is based on common bindings for device graphs. -see ${LINUX}/Documentation/devicetree/bindings/graph.txt - -Basically, Audio-Graph-SCU-Card property is same as -Simple-Card / Simple-SCU-Card / Audio-Graph-Card. -see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt - -Below are same as Simple-Card / Audio-Graph-Card. - -- label -- dai-format -- frame-master -- bitclock-master -- bitclock-inversion -- frame-inversion -- dai-tdm-slot-num -- dai-tdm-slot-width -- clocks / system-clock-frequency - -Below are same as Simple-SCU-Card. - -- convert-rate -- convert-channels -- prefix -- routing - -Required properties: - -- compatible : "audio-graph-scu-card"; -- dais : list of CPU DAI port{s} - -Example 1. Sampling Rate Conversion - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - prefix = "codec"; - routing = "codec Playback", "DAI0 Playback", - "DAI0 Capture", "codec Capture"; - convert-rate = <48000>; - - dais = <&cpu_port>; - }; - - audio-codec { - ... - - port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - }; - -Example 2. 2 CPU 1 Codec (Mixing) - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - routing = "codec Playback", "DAI0 Playback", - "codec Playback", "DAI1 Playback", - "DAI0 Capture", "codec Capture"; - - dais = <&cpu_port0 - &cpu_port1>; - }; - - audio-codec { - ... - - audio-graph-card,prefix = "codec"; - audio-graph-card,convert-rate = <48000>; - port { - codec_endpoint0: endpoint { - remote-endpoint = <&cpu_endpoint0>; - }; - codec_endpoint1: endpoint { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - dai-controller { - ... - ports { - cpu_port0: port { - cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint0>; - - dai-format = "left_j"; - ... - }; - }; - cpu_port1: port { - cpu_endpoint1: endpoint { - remote-endpoint = <&codec_endpoint1>; - - dai-format = "left_j"; - ... - }; - }; - }; - }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 61c263ac27a307cdf7f46aaee4810619103effad Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:32 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/Kconfig | 9 - sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-scu-card.c | 501 ----------------------- 3 files changed, 512 deletions(-) delete mode 100644 sound/soc/generic/audio-graph-scu-card.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 92c2cf06f40a..59190f42fc08 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -24,12 +24,3 @@ config SND_AUDIO_GRAPH_CARD This option enables generic simple sound card support with OF-graph DT bindings. It also support DPCM of multi CPU single Codec ststem. - -config SND_AUDIO_GRAPH_SCU_CARD - tristate "ASoC Audio Graph SCU sound card support" - depends on OF - select SND_SIMPLE_CARD_UTILS - help - This option enables generic simple SCU sound card support - with OF-graph DT bindings. - It supports DPCM of multi CPU single Codec ststem. diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9dec293a4c4d..9fbfdd524b24 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -3,10 +3,8 @@ snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o snd-soc-simple-scu-card-objs := simple-scu-card.o snd-soc-audio-graph-card-objs := audio-graph-card.o -snd-soc-audio-graph-scu-card-objs := audio-graph-scu-card.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o obj-$(CONFIG_SND_SIMPLE_SCU_CARD) += snd-soc-simple-scu-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_CARD) += snd-soc-audio-graph-card.o -obj-$(CONFIG_SND_AUDIO_GRAPH_SCU_CARD) += snd-soc-audio-graph-scu-card.o diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c deleted file mode 100644 index e1b192ea147b..000000000000 --- a/sound/soc/generic/audio-graph-scu-card.c +++ /dev/null @@ -1,501 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// ASoC audio graph SCU sound card support -// -// Copyright (C) 2017 Renesas Solutions Corp. -// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> -// -// based on -// ${LINUX}/sound/soc/generic/simple-scu-card.c -// ${LINUX}/sound/soc/generic/audio-graph-card.c - -#include <linux/clk.h> -#include <linux/device.h> -#include <linux/gpio.h> -#include <linux/module.h> -#include <linux/of.h> -#include <linux/of_device.h> -#include <linux/of_gpio.h> -#include <linux/of_graph.h> -#include <linux/platform_device.h> -#include <linux/string.h> -#include <sound/jack.h> -#include <sound/simple_card_utils.h> - -struct graph_card_data { - struct snd_soc_card snd_card; - struct graph_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; - struct snd_soc_dai_link_component platform; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - } *dai_props; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; -}; - -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - -#define PREFIX "audio-graph-card," - -static int asoc_graph_card_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - -static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - -static const struct snd_soc_ops asoc_graph_card_ops = { - .startup = asoc_graph_card_startup, - .shutdown = asoc_graph_card_shutdown, -}; - -static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - -static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - /* overwrite by top level adata if exist */ - asoc_simple_card_convert_fixup(&priv->adata, params); - - return 0; -} - -static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, - struct device_node *codec_ep, - struct graph_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_fe) -{ - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *ep = is_fe ? cpu_ep : codec_ep; - struct device_node *node = of_graph_get_port_parent(ep); - struct asoc_simple_dai *dai; - int ret; - - if (is_fe) { - struct snd_soc_dai_link_component *codecs; - - /* BE is dummy */ - codecs = dai_link->codecs; - codecs->of_node = NULL; - codecs->dai_name = "snd-soc-dummy-dai"; - codecs->name = "snd-soc-dummy"; - - /* FE settings */ - dai_link->dynamic = 1; - dai_link->dpcm_merged_format = 1; - - dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; - - ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); - if (ret) - return ret; - - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); - if (ret < 0) - return ret; - - /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); - } else { - struct snd_soc_codec_conf *cconf; - - /* FE is dummy */ - dai_link->cpu_of_node = NULL; - dai_link->cpu_dai_name = "snd-soc-dummy-dai"; - dai_link->cpu_name = "snd-soc-dummy"; - - /* BE settings */ - dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; - - dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; - - cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; - - ret = asoc_simple_card_parse_graph_codec(ep, dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - dai_link->codecs->dai_name); - if (ret < 0) - return ret; - - /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, cconf, - dai_link->codecs->of_node, - "prefix"); - /* check "prefix" from each node if top doesn't have */ - if (!cconf->of_node) - snd_soc_of_parse_node_prefix(node, cconf, - dai_link->codecs->of_node, - PREFIX "prefix"); - } - - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - - ret = asoc_simple_card_of_parse_tdm(ep, dai); - if (ret) - return ret; - - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); - if (ret < 0) - return ret; - - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_graph_card_ops; - dai_link->init = asoc_graph_card_dai_init; - - return 0; -} - -static int asoc_graph_card_parse_of(struct graph_card_data *priv) -{ - struct of_phandle_iterator it; - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - int dai_idx, link_idx, conf_idx, ret; - int rc, codec; - - if (!node) - return -EINVAL; - - /* - * we need to consider "widgets", "mclk-fs" around here - * see simple-card - */ - - ret = asoc_simple_card_of_parse_routing(card, NULL); - if (ret < 0) - return ret; - - asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); - - /* - * it supports multi CPU, single CODEC only here - * see asoc_graph_get_dais_count - */ - - link_idx = 0; - dai_idx = 0; - conf_idx = 0; - codec_port_old = NULL; - for (codec = 0; codec < 2; codec++) { - /* - * To listup valid sounds continuously, - * detect all CPU-dummy first, and - * detect all dummy-Codec second - */ - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(cpu_port); - of_node_put(codec_port); - it.node = NULL; - - if (codec) { - if (codec_port_old == codec_port) - continue; - - codec_port_old = codec_port; - } - - ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, - priv, &dai_idx, - link_idx++, &conf_idx, - !codec); - if (ret < 0) - goto parse_of_err; - } - } - - ret = asoc_simple_card_parse_card_name(card, NULL); - if (ret) - goto parse_of_err; - - if ((card->num_links != link_idx) || - (card->num_configs != conf_idx)) { - dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", - card->num_links, link_idx, card->num_configs, conf_idx); - ret = -EINVAL; - goto parse_of_err; - } - - ret = 0; - -parse_of_err: - return ret; -} - -static void asoc_graph_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) -{ - struct of_phandle_iterator it; - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - struct device_node *codec_port_old2; - int rc; - - /* - * link_num : number of links. - * CPU-Codec / CPU-dummy / dummy-Codec - * dais_num : number of DAIs - * ccnf_num : number of codec_conf - * same number for dummy-Codec - * - * ex1) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 7 - * CPU2 -/ ccnf : 1 - * CPU3 --- Codec2 - * - * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec - * => 7 DAIs = 4xCPU + 3xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex2) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 6 - * CPU2 -/ ccnf : 1 - * CPU3 -/ - * - * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex3) - * CPU0 --- Codec0 link : 6 - * CPU1 -/ dais : 6 - * CPU2 --- Codec1 ccnf : 2 - * CPU3 -/ - * - * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 2 ccnf = 2xdummy-Codec - */ - codec_port_old = NULL; - codec_port_old2 = NULL; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - - (*link_num)++; - (*dais_num)++; - - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } - - codec_port_old2 = codec_port_old; - continue; - } - - (*dais_num)++; - codec_port_old = codec_port; - } -} - -static int asoc_graph_card_probe(struct platform_device *pdev) -{ - struct graph_card_data *priv; - struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; - struct asoc_simple_dai *dais; - struct device *dev = &pdev->dev; - struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; - int lnum = 0, dnum = 0, cnum = 0; - int ret, i; - - /* Allocate the private data and the DAI link array */ - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) - return -EINVAL; - - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < lnum; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; - } - - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - /* Init snd_soc_card */ - card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = priv->dai_link; - card->num_links = lnum; - card->codec_conf = cconf; - card->num_configs = cnum; - - ret = asoc_graph_card_parse_of(priv); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - goto err; - } - - snd_soc_card_set_drvdata(card, priv); - - ret = devm_snd_soc_register_card(dev, card); - if (ret < 0) - goto err; - - return 0; -err: - asoc_simple_card_clean_reference(card); - - return ret; -} - -static int asoc_graph_card_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - return asoc_simple_card_clean_reference(card); -} - -static const struct of_device_id asoc_graph_of_match[] = { - { .compatible = "audio-graph-scu-card", }, - {}, -}; -MODULE_DEVICE_TABLE(of, asoc_graph_of_match); - -static struct platform_driver asoc_graph_card = { - .driver = { - .name = "asoc-audio-graph-scu-card", - .pm = &snd_soc_pm_ops, - .of_match_table = asoc_graph_of_match, - }, - .probe = asoc_graph_card_probe, - .remove = asoc_graph_card_remove, -}; -module_platform_driver(asoc_graph_card); - -MODULE_ALIAS("platform:asoc-audio-graph-scu-card"); -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("ASoC Audio Graph SCU Sound Card"); -MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 2.20.1

1 0

[alsa-devel] Applied "dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e3e12ec09a18ad779b637f4a006a908cb6045aa7 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:27 +0530 Subject: [PATCH] dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings Added documentation for audio formatter IP core DT bindings. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/xlnx,audio-formatter.txt | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt new file mode 100644 index 000000000000..cbc93c8f4963 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt @@ -0,0 +1,29 @@ +Device-Tree bindings for Xilinx PL audio formatter + +The IP core supports DMA, data formatting(AES<->PCM conversion) +of audio samples. + +Required properties: + - compatible: "xlnx,audio-formatter-1.0" + - interrupt-names: Names specified to list of interrupts in same + order mentioned under "interrupts". + List of supported interrupt names are: + "irq_mm2s" : interrupt from MM2S block + "irq_s2mm" : interrupt from S2MM block + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - reg: Base address and size of the IP core instance. + - clock-names: List of input clocks. + Required elements: "s_axi_lite_aclk", "aud_mclk" + - clocks: Input clock specifier. Refer to common clock bindings. + +Example: + audio_ss_0_audio_formatter_0: audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + reg = <0x0 0x80010000 0x0 0x1000>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: xlnx: enable audio formatter driver build" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: xlnx: enable audio formatter driver build has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b31daa15af760747b91dbb76c80306d77d8ae05f Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:29 +0530 Subject: [PATCH] ASoC: xlnx: enable audio formatter driver build Enable audio formatter driver build. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/xilinx/Kconfig | 7 +++++++ sound/soc/xilinx/Makefile | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 723a583a8d57..ac48d6a00c36 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -6,3 +6,10 @@ config SND_SOC_XILINX_I2S mode, IP receives audio in AES format, extracts PCM and sends PCM data. In receiver mode, IP receives PCM audio and encapsulates PCM in AES format and sends AES data. + +config SND_SOC_XILINX_AUDIO_FORMATTER + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index 6c1209b9ee75..432693b1cc79 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,2 +1,4 @@ snd-soc-xlnx-i2s-objs := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o +snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o +obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-card: add 1 CPU : 1 Codec support again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From de2949fe262197298036989924d05f5de6b9815a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:54 +0900 Subject: [PATCH] ASoC: audio-graph-card: add 1 CPU : 1 Codec support again audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 44 ++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c3e80bc27e80..638333cdac66 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -408,6 +408,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep = NULL; struct device_node *codec_port = NULL; struct device_node *codec_port_old = NULL; + struct asoc_simple_card_data adata; int rc, ret; int link_idx, dai_idx, conf_idx; int cpu; @@ -453,7 +454,13 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); - if (of_get_child_count(codec_port) > 1) { + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); + + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || + adata.convert_channels) { /* * for DPCM sound */ @@ -495,7 +502,7 @@ static void asoc_graph_get_dais_count(struct device *dev, struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - struct device_node *codec_port_old2; + struct asoc_simple_card_data adata; int rc; /* @@ -534,9 +541,17 @@ static void asoc_graph_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ codec_port_old = NULL; - codec_port_old2 = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = NULL; @@ -554,17 +569,22 @@ static void asoc_graph_get_dais_count(struct device *dev, (*link_num)++; (*dais_num)++; - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); - codec_port_old2 = codec_port_old; - continue; - } + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels) { - (*dais_num)++; + if (codec_port_old == codec_port) + continue; + + (*link_num)++; + (*ccnf_num)++; + (*dais_num)++; + } else { + (*dais_num)++; + } codec_port_old = codec_port; } } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: add asoc_simple_card_get_conversion()" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card: add asoc_simple_card_get_conversion() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7e5e1f8bbaa82e6877d8bc121cb1b44cb1ce7ddf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:42 +0900 Subject: [PATCH] ASoC: simple-card: add asoc_simple_card_get_conversion() simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_simple_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001cf9cd1..52048069b25a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,6 +165,21 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_simple_card_get_conversion(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *node = of_get_parent(np); + + asoc_simple_card_parse_convert(dev, top, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, NULL, adata); + asoc_simple_card_parse_convert(dev, np, NULL, adata); + + of_node_put(node); +} + static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, struct device_node *node, struct device_node *np, @@ -260,9 +275,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, "prefix"); } - asoc_simple_card_parse_convert(dev, top, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, prefix, &dai_props->adata); - asoc_simple_card_parse_convert(dev, np, NULL, &dai_props->adata); + asoc_simple_card_get_conversion(dev, np, &dai_props->adata); ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card: add 1 CPU : 1 Codec support again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7adee60ee2732f23f703ff83ee35caad561490ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:47 +0900 Subject: [PATCH] ASoC: simple-card: add 1 CPU : 1 Codec support again simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 52048069b25a..b15651409c7f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -453,6 +453,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *node; struct device_node *np; struct device_node *codec; + struct asoc_simple_card_data adata; bool is_fe; int ret, loop; int dai_idx, link_idx, conf_idx; @@ -480,8 +481,13 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + /* DPCM */ - if (of_get_child_count(node) > 2) { + if (of_get_child_count(node) > 2 || + adata.convert_rate || adata.convert_channels) { for_each_child_of_node(node, np) { codec = of_get_child_by_name(node, loop ? "codec" : @@ -495,14 +501,16 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) top, node, np, codec, priv, &dai_idx, link_idx++, &conf_idx, is_fe, !loop); + if (ret < 0) + return ret; } } else { ret = asoc_simple_card_dai_link_of( top, node, priv, &dai_idx, link_idx++, !loop); + if (ret < 0) + return ret; } - if (ret < 0) - return ret; node = of_get_next_child(top, node); } while (loop && node); @@ -523,6 +531,8 @@ static void asoc_simple_card_get_dais_count(struct device *dev, { struct device_node *top = dev->of_node; struct device_node *node; + struct device_node *np; + struct asoc_simple_card_data adata; int loop; int num; @@ -562,6 +572,15 @@ static void asoc_simple_card_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ if (!top) { (*link_num) = 1; @@ -578,9 +597,14 @@ static void asoc_simple_card_get_dais_count(struct device *dev, } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + num = of_get_child_count(node); (*dais_num) += num; - if (num > 2) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { (*link_num) += num; (*ccnf_num)++; } else { -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C)" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 51a13e401a83ef37aa98c049c2c30bba885676c2 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold <stephan(a)gerhold.net> Date: Wed, 2 Jan 2019 20:39:08 +0100 Subject: [PATCH] ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) Add quirks to select the correct input map, jack-detect options and channel map to make sound work on the ASUS MeMO Pad 7 (ME176C). Note: Although sound works out of the box, jack detection currently requires overriding the ACPI DSDT table. This is necessary because the rt5640 ACPI device (10EC5640) has the wrong GPIO listed as interrupt (one of the Bluetooth GPIOs). The correct GPIO is GPO2 0x0004 (listed as the first GPIO in the Intel(R) Audio Machine Driver - AMCR0F28 device). Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a22366ce33c4..ca8b4d5ff70f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -428,6 +428,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ME176C"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: Intel: Baytrail: remove unneeded variable" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: Intel: Baytrail: remove unneeded variable has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 060d35be2dfa9202d37f967fd20f133c530505d2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Sun, 16 Dec 2018 16:49:06 -0600 Subject: [PATCH] ASoC: Intel: Baytrail: remove unneeded variable Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 260447da32b8..2cd8f9668b50 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -278,7 +278,6 @@ static int sst_byt_process_notification(struct sst_byt *byt, struct sst_byt_stream *stream; u64 header; u8 msg_id, stream_id; - int handled = 1; header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); msg_id = sst_byt_header_msg_id(header); @@ -298,7 +297,7 @@ static int sst_byt_process_notification(struct sst_byt *byt, break; } - return handled; + return 1; } static irqreturn_t sst_byt_irq_thread(int irq, void *context) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a0c426fe143328760c9fd565cd203a37a7b4fde8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:42 +0900 Subject: [PATCH] ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() We will get DAI ID from "reg" property if it has on DT, otherwise get it by counting port/endpoint. But in below case, we need to get DAI ID = 0 via port reg = <0>, but current implementation returns ID = 1, because it can't judge ID = 0 was from "non reg" or "reg = <0>". Thus, it will count port/endpoint number as "non reg" case. of_graph_parse_endpoint() implementation itself is not a problem, but because asoc_simple_card_get_dai_id() need to count port/endpoint number when "non reg" case, it need to know ID = 0 was from "non reg" or "reg = <0>". This patch fix this issue. port { reg = <0>; xxxx: endpoint@0 { }; => xxxx: endpoint@1 { }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b807a47515eb..336895f7fd1e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -283,12 +283,20 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) /* use endpoint/port reg if exist */ ret = of_graph_parse_endpoint(ep, &info); if (ret == 0) { - if (info.id) + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_get_property(ep, "reg", NULL)) return info.id; - if (info.port) + + node = of_get_parent(ep); + of_node_put(node); + if (of_get_property(node, "reg", NULL)) return info.port; } - node = of_graph_get_port_parent(ep); /* -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add asoc_graph_card_get_conversion()" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-card: add asoc_graph_card_get_conversion() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 40dfae169ad047535d566a4791daae3b08f71c0c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:48 +0900 Subject: [PATCH] ASoC: audio-graph-card: add asoc_graph_card_get_conversion() audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_graph_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0d6144560a1e..c3e80bc27e80 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -169,6 +169,22 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_graph_card_get_conversion(struct device *dev, + struct device_node *ep, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); + struct device_node *node = of_graph_get_port_parent(ep); + + asoc_simple_card_parse_convert(dev, top, NULL, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, ports, NULL, adata); + asoc_simple_card_parse_convert(dev, port, NULL, adata); + asoc_simple_card_parse_convert(dev, ep, NULL, adata); +} + static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, struct device_node *cpu_ep, struct device_node *codec_ep, @@ -194,11 +210,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - asoc_simple_card_parse_convert(dev, top, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ports, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, port, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ep, NULL, &dai_props->adata); + asoc_graph_card_get_conversion(dev, ep, &dai_props->adata); of_node_put(ports); of_node_put(port); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e4f4fdfc57d9c846862ea6109e356b3a4542df5b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:27 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/audio-graph-scu-card.txt | 123 ------------------ 1 file changed, 123 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt deleted file mode 100644 index 62d42768a00b..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ /dev/null @@ -1,123 +0,0 @@ -Audio-Graph-SCU-Card: - -Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM". - -It is based on common bindings for device graphs. -see ${LINUX}/Documentation/devicetree/bindings/graph.txt - -Basically, Audio-Graph-SCU-Card property is same as -Simple-Card / Simple-SCU-Card / Audio-Graph-Card. -see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt - -Below are same as Simple-Card / Audio-Graph-Card. - -- label -- dai-format -- frame-master -- bitclock-master -- bitclock-inversion -- frame-inversion -- dai-tdm-slot-num -- dai-tdm-slot-width -- clocks / system-clock-frequency - -Below are same as Simple-SCU-Card. - -- convert-rate -- convert-channels -- prefix -- routing - -Required properties: - -- compatible : "audio-graph-scu-card"; -- dais : list of CPU DAI port{s} - -Example 1. Sampling Rate Conversion - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - prefix = "codec"; - routing = "codec Playback", "DAI0 Playback", - "DAI0 Capture", "codec Capture"; - convert-rate = <48000>; - - dais = <&cpu_port>; - }; - - audio-codec { - ... - - port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - }; - -Example 2. 2 CPU 1 Codec (Mixing) - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - routing = "codec Playback", "DAI0 Playback", - "codec Playback", "DAI1 Playback", - "DAI0 Capture", "codec Capture"; - - dais = <&cpu_port0 - &cpu_port1>; - }; - - audio-codec { - ... - - audio-graph-card,prefix = "codec"; - audio-graph-card,convert-rate = <48000>; - port { - codec_endpoint0: endpoint { - remote-endpoint = <&cpu_endpoint0>; - }; - codec_endpoint1: endpoint { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - dai-controller { - ... - ports { - cpu_port0: port { - cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint0>; - - dai-format = "left_j"; - ... - }; - }; - cpu_port1: port { - cpu_endpoint1: endpoint { - remote-endpoint = <&codec_endpoint1>; - - dai-format = "left_j"; - ... - }; - }; - }; - }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 61c263ac27a307cdf7f46aaee4810619103effad Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:32 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/Kconfig | 9 - sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-scu-card.c | 501 ----------------------- 3 files changed, 512 deletions(-) delete mode 100644 sound/soc/generic/audio-graph-scu-card.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 92c2cf06f40a..59190f42fc08 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -24,12 +24,3 @@ config SND_AUDIO_GRAPH_CARD This option enables generic simple sound card support with OF-graph DT bindings. It also support DPCM of multi CPU single Codec ststem. - -config SND_AUDIO_GRAPH_SCU_CARD - tristate "ASoC Audio Graph SCU sound card support" - depends on OF - select SND_SIMPLE_CARD_UTILS - help - This option enables generic simple SCU sound card support - with OF-graph DT bindings. - It supports DPCM of multi CPU single Codec ststem. diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9dec293a4c4d..9fbfdd524b24 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -3,10 +3,8 @@ snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o snd-soc-simple-scu-card-objs := simple-scu-card.o snd-soc-audio-graph-card-objs := audio-graph-card.o -snd-soc-audio-graph-scu-card-objs := audio-graph-scu-card.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o obj-$(CONFIG_SND_SIMPLE_SCU_CARD) += snd-soc-simple-scu-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_CARD) += snd-soc-audio-graph-card.o -obj-$(CONFIG_SND_AUDIO_GRAPH_SCU_CARD) += snd-soc-audio-graph-scu-card.o diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c deleted file mode 100644 index e1b192ea147b..000000000000 --- a/sound/soc/generic/audio-graph-scu-card.c +++ /dev/null @@ -1,501 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// ASoC audio graph SCU sound card support -// -// Copyright (C) 2017 Renesas Solutions Corp. -// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> -// -// based on -// ${LINUX}/sound/soc/generic/simple-scu-card.c -// ${LINUX}/sound/soc/generic/audio-graph-card.c - -#include <linux/clk.h> -#include <linux/device.h> -#include <linux/gpio.h> -#include <linux/module.h> -#include <linux/of.h> -#include <linux/of_device.h> -#include <linux/of_gpio.h> -#include <linux/of_graph.h> -#include <linux/platform_device.h> -#include <linux/string.h> -#include <sound/jack.h> -#include <sound/simple_card_utils.h> - -struct graph_card_data { - struct snd_soc_card snd_card; - struct graph_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; - struct snd_soc_dai_link_component platform; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - } *dai_props; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; -}; - -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - -#define PREFIX "audio-graph-card," - -static int asoc_graph_card_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - -static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - -static const struct snd_soc_ops asoc_graph_card_ops = { - .startup = asoc_graph_card_startup, - .shutdown = asoc_graph_card_shutdown, -}; - -static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - -static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - /* overwrite by top level adata if exist */ - asoc_simple_card_convert_fixup(&priv->adata, params); - - return 0; -} - -static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, - struct device_node *codec_ep, - struct graph_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_fe) -{ - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *ep = is_fe ? cpu_ep : codec_ep; - struct device_node *node = of_graph_get_port_parent(ep); - struct asoc_simple_dai *dai; - int ret; - - if (is_fe) { - struct snd_soc_dai_link_component *codecs; - - /* BE is dummy */ - codecs = dai_link->codecs; - codecs->of_node = NULL; - codecs->dai_name = "snd-soc-dummy-dai"; - codecs->name = "snd-soc-dummy"; - - /* FE settings */ - dai_link->dynamic = 1; - dai_link->dpcm_merged_format = 1; - - dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; - - ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); - if (ret) - return ret; - - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); - if (ret < 0) - return ret; - - /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); - } else { - struct snd_soc_codec_conf *cconf; - - /* FE is dummy */ - dai_link->cpu_of_node = NULL; - dai_link->cpu_dai_name = "snd-soc-dummy-dai"; - dai_link->cpu_name = "snd-soc-dummy"; - - /* BE settings */ - dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; - - dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; - - cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; - - ret = asoc_simple_card_parse_graph_codec(ep, dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - dai_link->codecs->dai_name); - if (ret < 0) - return ret; - - /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, cconf, - dai_link->codecs->of_node, - "prefix"); - /* check "prefix" from each node if top doesn't have */ - if (!cconf->of_node) - snd_soc_of_parse_node_prefix(node, cconf, - dai_link->codecs->of_node, - PREFIX "prefix"); - } - - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - - ret = asoc_simple_card_of_parse_tdm(ep, dai); - if (ret) - return ret; - - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); - if (ret < 0) - return ret; - - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_graph_card_ops; - dai_link->init = asoc_graph_card_dai_init; - - return 0; -} - -static int asoc_graph_card_parse_of(struct graph_card_data *priv) -{ - struct of_phandle_iterator it; - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - int dai_idx, link_idx, conf_idx, ret; - int rc, codec; - - if (!node) - return -EINVAL; - - /* - * we need to consider "widgets", "mclk-fs" around here - * see simple-card - */ - - ret = asoc_simple_card_of_parse_routing(card, NULL); - if (ret < 0) - return ret; - - asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); - - /* - * it supports multi CPU, single CODEC only here - * see asoc_graph_get_dais_count - */ - - link_idx = 0; - dai_idx = 0; - conf_idx = 0; - codec_port_old = NULL; - for (codec = 0; codec < 2; codec++) { - /* - * To listup valid sounds continuously, - * detect all CPU-dummy first, and - * detect all dummy-Codec second - */ - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(cpu_port); - of_node_put(codec_port); - it.node = NULL; - - if (codec) { - if (codec_port_old == codec_port) - continue; - - codec_port_old = codec_port; - } - - ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, - priv, &dai_idx, - link_idx++, &conf_idx, - !codec); - if (ret < 0) - goto parse_of_err; - } - } - - ret = asoc_simple_card_parse_card_name(card, NULL); - if (ret) - goto parse_of_err; - - if ((card->num_links != link_idx) || - (card->num_configs != conf_idx)) { - dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", - card->num_links, link_idx, card->num_configs, conf_idx); - ret = -EINVAL; - goto parse_of_err; - } - - ret = 0; - -parse_of_err: - return ret; -} - -static void asoc_graph_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) -{ - struct of_phandle_iterator it; - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - struct device_node *codec_port_old2; - int rc; - - /* - * link_num : number of links. - * CPU-Codec / CPU-dummy / dummy-Codec - * dais_num : number of DAIs - * ccnf_num : number of codec_conf - * same number for dummy-Codec - * - * ex1) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 7 - * CPU2 -/ ccnf : 1 - * CPU3 --- Codec2 - * - * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec - * => 7 DAIs = 4xCPU + 3xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex2) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 6 - * CPU2 -/ ccnf : 1 - * CPU3 -/ - * - * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex3) - * CPU0 --- Codec0 link : 6 - * CPU1 -/ dais : 6 - * CPU2 --- Codec1 ccnf : 2 - * CPU3 -/ - * - * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 2 ccnf = 2xdummy-Codec - */ - codec_port_old = NULL; - codec_port_old2 = NULL; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - - (*link_num)++; - (*dais_num)++; - - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } - - codec_port_old2 = codec_port_old; - continue; - } - - (*dais_num)++; - codec_port_old = codec_port; - } -} - -static int asoc_graph_card_probe(struct platform_device *pdev) -{ - struct graph_card_data *priv; - struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; - struct asoc_simple_dai *dais; - struct device *dev = &pdev->dev; - struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; - int lnum = 0, dnum = 0, cnum = 0; - int ret, i; - - /* Allocate the private data and the DAI link array */ - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) - return -EINVAL; - - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < lnum; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; - } - - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - /* Init snd_soc_card */ - card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = priv->dai_link; - card->num_links = lnum; - card->codec_conf = cconf; - card->num_configs = cnum; - - ret = asoc_graph_card_parse_of(priv); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - goto err; - } - - snd_soc_card_set_drvdata(card, priv); - - ret = devm_snd_soc_register_card(dev, card); - if (ret < 0) - goto err; - - return 0; -err: - asoc_simple_card_clean_reference(card); - - return ret; -} - -static int asoc_graph_card_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - return asoc_simple_card_clean_reference(card); -} - -static const struct of_device_id asoc_graph_of_match[] = { - { .compatible = "audio-graph-scu-card", }, - {}, -}; -MODULE_DEVICE_TABLE(of, asoc_graph_of_match); - -static struct platform_driver asoc_graph_card = { - .driver = { - .name = "asoc-audio-graph-scu-card", - .pm = &snd_soc_pm_ops, - .of_match_table = asoc_graph_of_match, - }, - .probe = asoc_graph_card_probe, - .remove = asoc_graph_card_remove, -}; -module_platform_driver(asoc_graph_card); - -MODULE_ALIAS("platform:asoc-audio-graph-scu-card"); -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("ASoC Audio Graph SCU Sound Card"); -MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 2.20.1

1 0

[alsa-devel] Applied "dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e3e12ec09a18ad779b637f4a006a908cb6045aa7 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:27 +0530 Subject: [PATCH] dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings Added documentation for audio formatter IP core DT bindings. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/xlnx,audio-formatter.txt | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt new file mode 100644 index 000000000000..cbc93c8f4963 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt @@ -0,0 +1,29 @@ +Device-Tree bindings for Xilinx PL audio formatter + +The IP core supports DMA, data formatting(AES<->PCM conversion) +of audio samples. + +Required properties: + - compatible: "xlnx,audio-formatter-1.0" + - interrupt-names: Names specified to list of interrupts in same + order mentioned under "interrupts". + List of supported interrupt names are: + "irq_mm2s" : interrupt from MM2S block + "irq_s2mm" : interrupt from S2MM block + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - reg: Base address and size of the IP core instance. + - clock-names: List of input clocks. + Required elements: "s_axi_lite_aclk", "aud_mclk" + - clocks: Input clock specifier. Refer to common clock bindings. + +Example: + audio_ss_0_audio_formatter_0: audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + reg = <0x0 0x80010000 0x0 0x1000>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: xlnx: enable audio formatter driver build" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: xlnx: enable audio formatter driver build has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b31daa15af760747b91dbb76c80306d77d8ae05f Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:29 +0530 Subject: [PATCH] ASoC: xlnx: enable audio formatter driver build Enable audio formatter driver build. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/xilinx/Kconfig | 7 +++++++ sound/soc/xilinx/Makefile | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 723a583a8d57..ac48d6a00c36 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -6,3 +6,10 @@ config SND_SOC_XILINX_I2S mode, IP receives audio in AES format, extracts PCM and sends PCM data. In receiver mode, IP receives PCM audio and encapsulates PCM in AES format and sends AES data. + +config SND_SOC_XILINX_AUDIO_FORMATTER + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index 6c1209b9ee75..432693b1cc79 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,2 +1,4 @@ snd-soc-xlnx-i2s-objs := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o +snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o +obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: audio-graph-card: add 1 CPU : 1 Codec support again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From de2949fe262197298036989924d05f5de6b9815a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:54 +0900 Subject: [PATCH] ASoC: audio-graph-card: add 1 CPU : 1 Codec support again audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 44 ++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c3e80bc27e80..638333cdac66 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -408,6 +408,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep = NULL; struct device_node *codec_port = NULL; struct device_node *codec_port_old = NULL; + struct asoc_simple_card_data adata; int rc, ret; int link_idx, dai_idx, conf_idx; int cpu; @@ -453,7 +454,13 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); - if (of_get_child_count(codec_port) > 1) { + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); + + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || + adata.convert_channels) { /* * for DPCM sound */ @@ -495,7 +502,7 @@ static void asoc_graph_get_dais_count(struct device *dev, struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - struct device_node *codec_port_old2; + struct asoc_simple_card_data adata; int rc; /* @@ -534,9 +541,17 @@ static void asoc_graph_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ codec_port_old = NULL; - codec_port_old2 = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = NULL; @@ -554,17 +569,22 @@ static void asoc_graph_get_dais_count(struct device *dev, (*link_num)++; (*dais_num)++; - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); - codec_port_old2 = codec_port_old; - continue; - } + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels) { - (*dais_num)++; + if (codec_port_old == codec_port) + continue; + + (*link_num)++; + (*ccnf_num)++; + (*dais_num)++; + } else { + (*dais_num)++; + } codec_port_old = codec_port; } } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: add asoc_simple_card_get_conversion()" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card: add asoc_simple_card_get_conversion() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7e5e1f8bbaa82e6877d8bc121cb1b44cb1ce7ddf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:42 +0900 Subject: [PATCH] ASoC: simple-card: add asoc_simple_card_get_conversion() simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_simple_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001cf9cd1..52048069b25a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,6 +165,21 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_simple_card_get_conversion(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *node = of_get_parent(np); + + asoc_simple_card_parse_convert(dev, top, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, NULL, adata); + asoc_simple_card_parse_convert(dev, np, NULL, adata); + + of_node_put(node); +} + static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, struct device_node *node, struct device_node *np, @@ -260,9 +275,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, "prefix"); } - asoc_simple_card_parse_convert(dev, top, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, prefix, &dai_props->adata); - asoc_simple_card_parse_convert(dev, np, NULL, &dai_props->adata); + asoc_simple_card_get_conversion(dev, np, &dai_props->adata); ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card: add 1 CPU : 1 Codec support again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7adee60ee2732f23f703ff83ee35caad561490ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:47 +0900 Subject: [PATCH] ASoC: simple-card: add 1 CPU : 1 Codec support again simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 52048069b25a..b15651409c7f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -453,6 +453,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *node; struct device_node *np; struct device_node *codec; + struct asoc_simple_card_data adata; bool is_fe; int ret, loop; int dai_idx, link_idx, conf_idx; @@ -480,8 +481,13 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + /* DPCM */ - if (of_get_child_count(node) > 2) { + if (of_get_child_count(node) > 2 || + adata.convert_rate || adata.convert_channels) { for_each_child_of_node(node, np) { codec = of_get_child_by_name(node, loop ? "codec" : @@ -495,14 +501,16 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) top, node, np, codec, priv, &dai_idx, link_idx++, &conf_idx, is_fe, !loop); + if (ret < 0) + return ret; } } else { ret = asoc_simple_card_dai_link_of( top, node, priv, &dai_idx, link_idx++, !loop); + if (ret < 0) + return ret; } - if (ret < 0) - return ret; node = of_get_next_child(top, node); } while (loop && node); @@ -523,6 +531,8 @@ static void asoc_simple_card_get_dais_count(struct device *dev, { struct device_node *top = dev->of_node; struct device_node *node; + struct device_node *np; + struct asoc_simple_card_data adata; int loop; int num; @@ -562,6 +572,15 @@ static void asoc_simple_card_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ if (!top) { (*link_num) = 1; @@ -578,9 +597,14 @@ static void asoc_simple_card_get_dais_count(struct device *dev, } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + num = of_get_child_count(node); (*dais_num) += num; - if (num > 2) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { (*link_num) += num; (*ccnf_num)++; } else { -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8c3590de0a378c2449fc1aec127cc693632458e4 Mon Sep 17 00:00:00 2001 From: Yizhuo <yzhai003(a)ucr.edu> Date: Thu, 3 Jan 2019 13:59:12 -0800 Subject: [PATCH] ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized Inside function rt274_i2c_probe(), if regmap_read() function returns -EINVAL, then local variable "val" leaves uninitialized but used in if statement. This is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt274.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 0ef966d56bac..e2855ab9a2c6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); -- 2.20.1

1 0

[alsa-devel] [PATCH] Variable "val" in function rt274_i2c_probe() could be uninitialized
by Yizhuo 04 Jan '19

04 Jan '19

Inside function rt274_i2c_probe(), if regmap_read() function returns -EINVAL, then local variable "val" leaves uninitialized but used in if statement. This is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> --- sound/soc/codecs/rt274.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 8f92e5c4dd9d..cd048df76232 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); -- 2.17.1

2 1

Re: [alsa-devel] [PATCH v5 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Rafael J. Wysocki 04 Jan '19

04 Jan '19

On Thu, Jan 3, 2019 at 5:08 AM Chandan Rajendra <chandan(a)linux.ibm.com> wrote: > > On Thursday, January 3, 2019 5:20:27 AM IST Pierre-Louis Bossart wrote: > > > > On 1/2/19 4:58 PM, Sinan Kaya wrote: > > > On Wed, Jan 2, 2019 at 10:50 PM Pierre-Louis Bossart > > > <pierre-louis.bossart(a)linux.intel.com> wrote: > > >> > > >>> This is pointing to a kconfig issue on ia64 arch. > > >>> > > >>> arch/ia64/Kconfig:128:error: recursive dependency detected! > > >>> arch/ia64/Kconfig:128: choice <choice> contains symbol IA64_HP_SIM > > >>> arch/ia64/Kconfig:202: symbol IA64_HP_SIM is part of choice PM > > >>> > > >>> IA64_HP_SIM is both a choice and is selected. > > >>> > > >>> I did allyesconfig and disabled PCI afterwards to find all the issues > > >>> on this patchset. > > >> Are you saying there's a newer series that fixes this issue for both > > >> allyesconfig and allmodconfig? > > >> > > >> if yes, then we're good. > > > > > > No, I haven't fixed ia64 kconfig issue. That's somebody else's job. I > > > used allyesconfig to find out all compilation issues on x86 arch to > > > come up with this patchset. > > > > Nothing makes me cringe more than "somebody else's job" statements. In > > this case, there is obviously a correlation with your ACPI changes since > > the circular dependency happens because of the ACPI symbol. > > > > arch/ia64/Kconfig:126:error: recursive dependency detected! > > arch/ia64/Kconfig:126: choice <choice> contains symbol IA64_HP_SIM > > arch/ia64/Kconfig:200: symbol IA64_HP_SIM is part of choice PM > > kernel/power/Kconfig:144: symbol PM is selected by PM_SLEEP > > kernel/power/Kconfig:104: symbol PM_SLEEP depends on HIBERNATE_CALLBACKS > > kernel/power/Kconfig:31: symbol HIBERNATE_CALLBACKS is selected by > > HIBERNATION > > kernel/power/Kconfig:34: symbol HIBERNATION depends on SWAP > > init/Kconfig:250: symbol SWAP depends on BLOCK > > block/Kconfig:5: symbol BLOCK is selected by UBIFS_FS > > fs/ubifs/Kconfig:1: symbol UBIFS_FS depends on MISC_FILESYSTEMS > > fs/Kconfig:220: symbol MISC_FILESYSTEMS is selected by ACPI_APEI > > drivers/acpi/apei/Kconfig:8: symbol ACPI_APEI depends on ACPI > > drivers/acpi/Kconfig:9: symbol ACPI depends on ARCH_SUPPORTS_ACPI > > <<<< LOOK HERE > > drivers/acpi/Kconfig:6: symbol ARCH_SUPPORTS_ACPI is selected by > > IA64_HP_SIM > > arch/ia64/Kconfig:200: symbol IA64_HP_SIM is part of choice <choice> > > > > At any rate, a 3 mn git bisect tells me the circular dependency is > > exposed by this change: > > > > f3fd6cd74fedf99b6060f75df00943fda13b65f2 is the first bad commit > > commit f3fd6cd74fedf99b6060f75df00943fda13b65f2 > > Author: Chandan Rajendra <chandan(a)linux.vnet.ibm.com> > > Date: Sat Dec 8 12:21:38 2018 +0530 > > > > fscrypt: remove filesystem specific build config option > > > > In order to have a common code base for fscrypt "post read" processing > > for all filesystems which support encryption, this commit removes > > filesystem specific build config option (e.g. > > CONFIG_EXT4_FS_ENCRYPTION) > > and replaces it with a build option (i.e. CONFIG_FS_ENCRYPTION) whose > > value affects all the filesystems making use of fscrypt. > > > > Signed-off-by: Chandan Rajendra <chandan(a)linux.vnet.ibm.com> > > Signed-off-by: Theodore Ts'o <tytso(a)mit.edu> > > > > FWIW, The patch at https://patchwork.kernel.org/patch/10725883/ fixes this > problem by removing "select BLOCK if FS_ENCRYPTION" from fs/ubifs/Kconfig. OK Pierre-Louis, can you check if this patch makes the issue go away for you, please?

3 2

[alsa-devel] [RFC PATCH] ASoC: fsl: Add Audio Mixer CPU DAI driver
by Viorel Suman 04 Jan '19

04 Jan '19

This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs. The Audio Mixer is a on-chip functional module that allows mixing of two audio streams into a single audio stream. Audio Mixer datasheet is available here: https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- .../devicetree/bindings/sound/fsl,amix.txt | 45 ++ sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/fsl_amix.c | 554 +++++++++++++++++++++ sound/soc/fsl/fsl_amix.h | 101 ++++ 5 files changed, 710 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,amix.txt create mode 100644 sound/soc/fsl/fsl_amix.c create mode 100644 sound/soc/fsl/fsl_amix.h diff --git a/Documentation/devicetree/bindings/sound/fsl,amix.txt b/Documentation/devicetree/bindings/sound/fsl,amix.txt new file mode 100644 index 0000000..049144f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,amix.txt @@ -0,0 +1,45 @@ +NXP Audio Mixer (AMIX). + +The Audio Mixer is a on-chip functional module that allows mixing of two +audio streams into a single audio stream. Audio Mixer has two input serial +audio interfaces. These are driven by two Synchronous Audio interface +modules (SAI). Each input serial interface carries 8 audio channels in its +frame in TDM manner. Mixer mixes audio samples of corresponding channels +from two interfaces into a single sample. Before mixing, audio samples of +two inputs can be attenuated based on configuration. The output of the +Audio Mixer is also a serial audio interface. Like input interfaces it has +the same TDM frame format. This output is used to drive the serial DAC TDM +interface of audio codec and also sent to the external pins along with the +receive path of normal audio SAI module for readback by the CPU. + +The output of Audio mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - Mixed audio + +Mixing operation is independent of audio sample rate but the two audio +input streams must have same audio sample rate with same number of channels +in TDM frame to be eligible for mixing. + +Device driver required properties: +================================= + - compatible : Compatible list, contains "fsl,imx8qm-amix" + + - reg : Offset and length of the register set for the device. + + - clocks : Must contain an entry for each entry in clock-names. + + - clock-names : Must include the "ipg" for register access. + + - power-domains : Must contain the phandle to the AMIX power domain node + +Device driver configuration example: +====================================== + amix: amix@59840000 { + compatible = "fsl,imx8qm-amix"; + reg = <0x0 0x59840000 0x0 0x10000>; + clocks = <&clk IMX8QXP_AUD_AMIX_IPG>; + clock-names = "ipg"; + power-domains = <&pd_amix>; + }; + diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 2e75b5bc..0b08eb2 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_AMIX + tristate "Audio Mixer (AMIX) module support" + select REGMAP_MMIO + help + Say Y if you want to add Audio Mixer (AMIX) + support for the NXP iMX CPUs. + config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index de94fa0..3263634 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,6 +12,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-amix-objs := fsl_amix.o snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o @@ -21,6 +22,8 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o + +obj-$(CONFIG_SND_SOC_FSL_AMIX) += snd-soc-fsl-amix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o diff --git a/sound/soc/fsl/fsl_amix.c b/sound/soc/fsl/fsl_amix.c new file mode 100644 index 0000000..d752029 --- /dev/null +++ b/sound/soc/fsl/fsl_amix.c @@ -0,0 +1,554 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * NXP AMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "fsl_amix.h" + +#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \ + SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts) + +static const char + *tdm_sel[] = { "TDM1", "TDM2", }, + *mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", }, + *width_sel[] = { "16b", "18b", "20b", "24b", "32b", }, + *pol_sel[] = { "Positive edge", "Negative edge", }, + *endis_sel[] = { "Disabled", "Enabled", }, + *updn_sel[] = { "Downward", "Upward", }, + *mask_sel[] = { "Unmask", "Mask", }; + +static const struct soc_enum fsl_amix_enum[] = { +/* FSL_AMIX_CTR enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MIXCLK_SHIFT, tdm_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTSRC_SHIFT, mode_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTWIDTH_SHIFT, width_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTCKPOL_SHIFT, pol_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MASKRTDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MASKCKDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_SYNCMODE_SHIFT, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_SYNCSRC_SHIFT, tdm_sel), +/* FSL_AMIX_ATCR0 enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR0, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR0, 1, updn_sel), +/* FSL_AMIX_ATCR1 enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR1, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR1, 1, updn_sel), +}; + +struct fsl_amix_state { + u8 tdms; + u8 clk; + char msg[64]; +}; + +static const struct fsl_amix_state prms[4][4] = {{ + /* DIS->DIS, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* DIS->TDM1*/ + { .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" }, + /* DIS->TDM2*/ + { .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" }, + /* DIS->MIX */ + { .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" } +}, { /* TDM1->DIS */ + { .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" }, + /* TDM1->TDM1, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM1->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" }, + /* TDM1->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" } +}, { /* TDM2->DIS */ + { .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" }, + /* TDM2->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" }, + /* TDM2->TDM2, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM2->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" } +}, { /* MIX->DIS */ + { .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" }, + /* MIX->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" }, + /* MIX->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" }, + /* MIX->MIX, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" } +}, }; + +static int fsl_amix_state_trans(struct snd_soc_component *comp, + unsigned int *mask, unsigned int *ctr, + const struct fsl_amix_state prm) +{ + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + /* Enforce all required TDMs are started */ + if ((priv->tdms & prm.tdms) != prm.tdms) { + dev_dbg(comp->dev, prm.msg); + return -EINVAL; + } + + switch (prm.clk) { + case 1: + case 2: + /* Set mix clock */ + (*mask) |= FSL_AMIX_CTR_MIXCLK_MASK; + (*ctr) |= FSL_AMIX_CTR_MIXCLK(prm.clk - 1); + break; + default: + break; + } + + return 0; +} + +static int fsl_amix_put_mix_clk_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int reg_val, val, mix_clk; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AMIX_CTR, &reg_val); + if (ret) + return ret; + + mix_clk = ((reg_val & FSL_AMIX_CTR_MIXCLK_MASK) + >> FSL_AMIX_CTR_MIXCLK_SHIFT); + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + if (!(priv->tdms & BIT(val))) { + dev_err(comp->dev, + "The selected clock source has no TDM%d enabled!\n", + val + 1); + return -EINVAL; + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static int fsl_amix_put_out_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + u32 out_src, mix_clk; + unsigned int reg_val, val, mask = 0, ctr = 0; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AMIX_CTR, &reg_val); + if (ret) + return ret; + + /* "From" state */ + out_src = ((reg_val & FSL_AMIX_CTR_OUTSRC_MASK) + >> FSL_AMIX_CTR_OUTSRC_SHIFT); + mix_clk = ((reg_val & FSL_AMIX_CTR_MIXCLK_MASK) + >> FSL_AMIX_CTR_MIXCLK_SHIFT); + + /* "To" state */ + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /* Check if state is changing ... */ + if (out_src == val) + return 0; + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + /* Check state transition constraints */ + ret = fsl_amix_state_trans(comp, &mask, &ctr, prms[out_src][val]); + if (ret) + return ret; + + /* Complete transition to new state */ + mask |= FSL_AMIX_CTR_OUTSRC_MASK; + ctr |= FSL_AMIX_CTR_OUTSRC(val); + + return snd_soc_component_update_bits(comp, FSL_AMIX_CTR, mask, ctr); +} + +static const struct snd_kcontrol_new fsl_amix_snd_controls[] = { + /* FSL_AMIX_CTR controls */ + SOC_ENUM_EXT("Mixing Clock Source", fsl_amix_enum[0], + snd_soc_get_enum_double, fsl_amix_put_mix_clk_src), + SOC_ENUM_EXT("Output Source", fsl_amix_enum[1], + snd_soc_get_enum_double, fsl_amix_put_out_src), + SOC_ENUM("Output Width", fsl_amix_enum[2]), + SOC_ENUM("Output Clock Polarity", fsl_amix_enum[3]), + SOC_ENUM("Frame Rate Diff Error", fsl_amix_enum[4]), + SOC_ENUM("Clock Freq Diff Error", fsl_amix_enum[5]), + SOC_ENUM("Sync Mode Config", fsl_amix_enum[6]), + SOC_ENUM("Sync Mode Clk Source", fsl_amix_enum[7]), + /* TDM1 Attenuation controls */ + SOC_ENUM("TDM1 Attenuation", fsl_amix_enum[8]), + SOC_ENUM("TDM1 Attenuation Direction", fsl_amix_enum[9]), + SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AMIX_ATCR0, + 2, 0x00fff, 0), + SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AMIX_ATIVAL0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AMIX_ATSTPUP0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AMIX_ATSTPDN0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AMIX_ATSTPTGT0, + 0, 0x3ffff, 0), + /* TDM2 Attenuation controls */ + SOC_ENUM("TDM2 Attenuation", fsl_amix_enum[10]), + SOC_ENUM("TDM2 Attenuation Direction", fsl_amix_enum[11]), + SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AMIX_ATCR1, + 2, 0x00fff, 0), + SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AMIX_ATIVAL1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AMIX_ATSTPUP1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AMIX_ATSTPDN1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AMIX_ATSTPTGT1, + 0, 0x3ffff, 0), +}; + +static int fsl_amix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + u32 mask = 0, ctr = 0; + + /* AMIX is working in DSP_A format only */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + /* For playback the AMIX is slave, and for record is master */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + /* Output data will be written on positive edge of the clock */ + ctr |= FSL_AMIX_CTR_OUTCKPOL(0); + break; + case SND_SOC_DAIFMT_NB_NF: + /* Output data will be written on negative edge of the clock */ + ctr |= FSL_AMIX_CTR_OUTCKPOL(1); + break; + default: + return -EINVAL; + } + + mask |= FSL_AMIX_CTR_OUTCKPOL_MASK; + + return snd_soc_component_update_bits(comp, FSL_AMIX_CTR, mask, ctr); +} + +static int fsl_amix_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_amix *priv = snd_soc_dai_get_drvdata(dai); + + /* Capture stream shall not be handled */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->tdms |= BIT(dai->driver->id); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->tdms &= ~BIT(dai->driver->id); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops fsl_amix_dai_ops = { + .set_fmt = fsl_amix_dai_set_fmt, + .trigger = fsl_amix_dai_trigger, +}; + +static struct snd_soc_dai_driver fsl_amix_dai[] = { + { + .id = 0, + .name = "amix-0", + .playback = { + .stream_name = "AMIX-Playback-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .capture = { + .stream_name = "AMIX-Capture-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .ops = &fsl_amix_dai_ops, + }, + { + .id = 1, + .name = "amix-1", + .playback = { + .stream_name = "AMIX-Playback-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .capture = { + .stream_name = "AMIX-Capture-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .ops = &fsl_amix_dai_ops, + }, +}; + +static const struct snd_soc_component_driver fsl_amix_component = { + .name = "fsl-amix-dai", + .controls = fsl_amix_snd_controls, + .num_controls = ARRAY_SIZE(fsl_amix_snd_controls), +}; + +static bool fsl_amix_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AMIX_CTR: + case FSL_AMIX_STR: + case FSL_AMIX_ATCR0: + case FSL_AMIX_ATIVAL0: + case FSL_AMIX_ATSTPUP0: + case FSL_AMIX_ATSTPDN0: + case FSL_AMIX_ATSTPTGT0: + case FSL_AMIX_ATTNVAL0: + case FSL_AMIX_ATSTP0: + case FSL_AMIX_ATCR1: + case FSL_AMIX_ATIVAL1: + case FSL_AMIX_ATSTPUP1: + case FSL_AMIX_ATSTPDN1: + case FSL_AMIX_ATSTPTGT1: + case FSL_AMIX_ATTNVAL1: + case FSL_AMIX_ATSTP1: + return true; + default: + return false; + } +} + +static bool fsl_amix_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AMIX_CTR: + case FSL_AMIX_ATCR0: + case FSL_AMIX_ATIVAL0: + case FSL_AMIX_ATSTPUP0: + case FSL_AMIX_ATSTPDN0: + case FSL_AMIX_ATSTPTGT0: + case FSL_AMIX_ATCR1: + case FSL_AMIX_ATIVAL1: + case FSL_AMIX_ATSTPUP1: + case FSL_AMIX_ATSTPDN1: + case FSL_AMIX_ATSTPTGT1: + return true; + default: + return false; + } +} + +static const struct reg_default fsl_amix_reg[] = { + { FSL_AMIX_CTR, 0x00060 }, + { FSL_AMIX_STR, 0x00003 }, + { FSL_AMIX_ATCR0, 0x00000 }, + { FSL_AMIX_ATIVAL0, 0x3FFFF }, + { FSL_AMIX_ATSTPUP0, 0x2AAAA }, + { FSL_AMIX_ATSTPDN0, 0x30000 }, + { FSL_AMIX_ATSTPTGT0, 0x00010 }, + { FSL_AMIX_ATTNVAL0, 0x00000 }, + { FSL_AMIX_ATSTP0, 0x00000 }, + { FSL_AMIX_ATCR1, 0x00000 }, + { FSL_AMIX_ATIVAL1, 0x3FFFF }, + { FSL_AMIX_ATSTPUP1, 0x2AAAA }, + { FSL_AMIX_ATSTPDN1, 0x30000 }, + { FSL_AMIX_ATSTPTGT1, 0x00010 }, + { FSL_AMIX_ATTNVAL1, 0x00000 }, + { FSL_AMIX_ATSTP1, 0x00000 }, +}; + +static const struct regmap_config fsl_amix_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = FSL_AMIX_ATSTP1, + .reg_defaults = fsl_amix_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_amix_reg), + .readable_reg = fsl_amix_readable_reg, + .writeable_reg = fsl_amix_writeable_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int fsl_amix_probe(struct platform_device *pdev) +{ + struct fsl_amix *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->pdev = pdev; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs, + &fsl_amix_regmap_config); + if (IS_ERR(priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(priv->regmap); + } + + priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(priv->ipg_clk); + } + + platform_set_drvdata(pdev, priv); + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_amix_component, + fsl_amix_dai, + ARRAY_SIZE(fsl_amix_dai)); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + return 0; +} + +#ifdef CONFIG_PM +static int fsl_amix_runtime_resume(struct device *dev) +{ + struct fsl_amix *priv = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(priv->ipg_clk); + if (ret) { + dev_err(dev, "Failed to enable IPG clock: %d\n", ret); + return ret; + } + + regcache_cache_only(priv->regmap, false); + regcache_mark_dirty(priv->regmap); + + return regcache_sync(priv->regmap); +} + +static int fsl_amix_runtime_suspend(struct device *dev) +{ + struct fsl_amix *priv = dev_get_drvdata(dev); + + regcache_cache_only(priv->regmap, true); + + clk_disable_unprepare(priv->ipg_clk); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_amix_pm = { + SET_RUNTIME_PM_OPS(fsl_amix_runtime_suspend, + fsl_amix_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static const struct of_device_id fsl_amix_ids[] = { + { .compatible = "fsl,imx8qm-amix", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_amix_ids); + +static struct platform_driver fsl_amix_driver = { + .probe = fsl_amix_probe, + .driver = { + .name = "fsl-amix", + .of_match_table = fsl_amix_ids, + .pm = &fsl_amix_pm, + }, +}; +module_platform_driver(fsl_amix_driver); + +MODULE_DESCRIPTION("NXP AMIX ASoC DAI driver"); +MODULE_AUTHOR("Viorel Suman <viorel.suman(a)nxp.com>"); +MODULE_ALIAS("platform:fsl-amix"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_amix.h b/sound/soc/fsl/fsl_amix.h new file mode 100644 index 0000000..5312683 --- /dev/null +++ b/sound/soc/fsl/fsl_amix.h @@ -0,0 +1,101 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * NXP AMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#ifndef __FSL_AMIX_H +#define __FSL_AMIX_H + +#define FSL_AMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +/* AMIX Registers */ +#define FSL_AMIX_CTR 0x200 /* Control */ +#define FSL_AMIX_STR 0x204 /* Status */ + +#define FSL_AMIX_ATCR0 0x208 /* Attenuation Control */ +#define FSL_AMIX_ATIVAL0 0x20c /* Attenuation Initial Value */ +#define FSL_AMIX_ATSTPUP0 0x210 /* Attenuation step up factor */ +#define FSL_AMIX_ATSTPDN0 0x214 /* Attenuation step down factor */ +#define FSL_AMIX_ATSTPTGT0 0x218 /* Attenuation step target */ +#define FSL_AMIX_ATTNVAL0 0x21c /* Attenuation Value */ +#define FSL_AMIX_ATSTP0 0x220 /* Attenuation step number */ + +#define FSL_AMIX_ATCR1 0x228 /* Attenuation Control */ +#define FSL_AMIX_ATIVAL1 0x22c /* Attenuation Initial Value */ +#define FSL_AMIX_ATSTPUP1 0x230 /* Attenuation step up factor */ +#define FSL_AMIX_ATSTPDN1 0x234 /* Attenuation step down factor */ +#define FSL_AMIX_ATSTPTGT1 0x238 /* Attenuation step target */ +#define FSL_AMIX_ATTNVAL1 0x23c /* Attenuation Value */ +#define FSL_AMIX_ATSTP1 0x240 /* Attenuation step number */ + +/* AMIX Control Register */ +#define FSL_AMIX_CTR_MIXCLK_SHIFT 0 +#define FSL_AMIX_CTR_MIXCLK_MASK BIT(FSL_AMIX_CTR_MIXCLK_SHIFT) +#define FSL_AMIX_CTR_MIXCLK(i) ((i) << FSL_AMIX_CTR_MIXCLK_SHIFT) +#define FSL_AMIX_CTR_OUTSRC_SHIFT 1 +#define FSL_AMIX_CTR_OUTSRC_MASK (0x3 << FSL_AMIX_CTR_OUTSRC_SHIFT) +#define FSL_AMIX_CTR_OUTSRC(i) (((i) << FSL_AMIX_CTR_OUTSRC_SHIFT) \ + & FSL_AMIX_CTR_OUTSRC_MASK) +#define FSL_AMIX_CTR_OUTWIDTH_SHIFT 3 +#define FSL_AMIX_CTR_OUTWIDTH_MASK (0x7 << FSL_AMIX_CTR_OUTWIDTH_SHIFT) +#define FSL_AMIX_CTR_OUTWIDTH(i) (((i) << FSL_AMIX_CTR_OUTWIDTH_SHIFT) \ + & FSL_AMIX_CTR_OUTWIDTH_MASK) +#define FSL_AMIX_CTR_OUTCKPOL_SHIFT 6 +#define FSL_AMIX_CTR_OUTCKPOL_MASK BIT(FSL_AMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AMIX_CTR_OUTCKPOL(i) ((i) << FSL_AMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AMIX_CTR_MASKRTDF_SHIFT 7 +#define FSL_AMIX_CTR_MASKRTDF_MASK BIT(FSL_AMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AMIX_CTR_MASKRTDF(i) ((i) << FSL_AMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AMIX_CTR_MASKCKDF_SHIFT 8 +#define FSL_AMIX_CTR_MASKCKDF_MASK BIT(FSL_AMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AMIX_CTR_MASKCKDF(i) ((i) << FSL_AMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AMIX_CTR_SYNCMODE_SHIFT 9 +#define FSL_AMIX_CTR_SYNCMODE_MASK BIT(FSL_AMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AMIX_CTR_SYNCMODE(i) ((i) << FSL_AMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AMIX_CTR_SYNCSRC_SHIFT 10 +#define FSL_AMIX_CTR_SYNCSRC_MASK BIT(FSL_AMIX_CTR_SYNCSRC_SHIFT) +#define FSL_AMIX_CTR_SYNCSRC(i) ((i) << FSL_AMIX_CTR_SYNCSRC_SHIFT) + +/* AMIX Status Register */ +#define FSL_AMIX_STR_RATEDIFF BIT(0) +#define FSL_AMIX_STR_CLKDIFF BIT(1) +#define FSL_AMIX_STR_MIXSTAT_SHIFT 2 +#define FSL_AMIX_STR_MIXSTAT_MASK (0x3 << FSL_AMIX_STR_MIXSTAT_SHIFT) +#define FSL_AMIX_STR_MIXSTAT(i) ((i & FSL_AMIX_STR_MIXSTAT_MASK) \ + >> FSL_AMIX_STR_MIXSTAT_SHIFT) +/* AMIX Attenuation Control Register */ +#define FSL_AMIX_ATCR_AT_EN BIT(0) +#define FSL_AMIX_ATCR_AT_UPDN BIT(1) +#define FSL_AMIX_ATCR_ATSTPDIF_SHIFT 2 +#define FSL_AMIX_ATCR_ATSTPDFI_MASK (0xfff << FSL_AMIX_ATCR_ATSTPDIF_SHIFT) + +/* AMIX Attenuation Initial Value Register */ +#define FSL_AMIX_ATIVAL_ATINVAL_MASK 0x3FFFF + +/* AMIX Attenuation Step Up Factor Register */ +#define FSL_AMIX_ATSTPUP_ATSTEPUP_MASK 0x3FFFF + +/* AMIX Attenuation Step Down Factor Register */ +#define FSL_AMIX_ATSTPDN_ATSTEPDN_MASK 0x3FFFF + +/* AMIX Attenuation Step Target Register */ +#define FSL_AMIX_ATSTPTGT_ATSTPTG_MASK 0x3FFFF + +/* AMIX Attenuation Value Register */ +#define FSL_AMIX_ATTNVAL_ATCURVAL_MASK 0x3FFFF + +/* AMIX Attenuation Step Number Register */ +#define FSL_AMIX_ATSTP_STPCTR_MASK 0x3FFFF + +#define FSL_AMIX_MAX_DAIS 2 +struct fsl_amix { + struct platform_device *pdev; + struct regmap *regmap; + struct clk *ipg_clk; + u8 tdms; +}; + +#endif /* __FSL_AMIX_H */ -- 2.7.4

4 8

[alsa-devel] [PATCH] ASoC: rt5682: Fix recording no sound issue
by shumingf＠realtek.com 03 Jan '19

03 Jan '19

From: Shuming Fan <shumingf(a)realtek.com> The ADC mixer setting needs to restore to default value after calibration. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 34cfaf8..89c43b2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2512,6 +2512,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); mutex_unlock(&rt5682->calibrate_mutex); -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: virtuoso: add de-emphasis control
by tom.ty89＠gmail.com 03 Jan '19

03 Jan '19

From: Tom Yan <tom.ty89(a)gmail.com> Add control for the de-emphasis filter in the PCM179x DACs Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> diff --git a/sound/pci/oxygen/pcm1796.h b/sound/pci/oxygen/pcm1796.h index 34d07dd2d22e..d5dcb09e44cd 100644 --- a/sound/pci/oxygen/pcm1796.h +++ b/sound/pci/oxygen/pcm1796.h @@ -10,7 +10,6 @@ #define PCM1796_MUTE 0x01 #define PCM1796_DME 0x02 #define PCM1796_DMF_MASK 0x0c -#define PCM1796_DMF_DISABLED 0x00 #define PCM1796_DMF_48 0x04 #define PCM1796_DMF_441 0x08 #define PCM1796_DMF_32 0x0c diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 24109d37ca09..a1c6b98b191e 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -331,7 +331,7 @@ static void pcm1796_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = - PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; + PCM1796_FMT_24_I2S | PCM1796_ATLD; if (!data->broken_i2c) data->pcm1796_regs[0][18 - PCM1796_REG_BASE] |= PCM1796_MUTE; data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = @@ -621,6 +621,23 @@ static void update_pcm1796_oversampling(struct oxygen *chip) pcm1796_write_cached(chip, i, 20, reg); } +static void update_pcm1796_deemph(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + u8 reg; + + reg = data->pcm1796_regs[0][18 - PCM1796_REG_BASE] & ~PCM1796_DMF_MASK; + if (data->current_rate == 48000) + reg |= PCM1796_DMF_48; + else if (data->current_rate == 44100) + reg |= PCM1796_DMF_441; + else if (data->current_rate == 32000) + reg |= PCM1796_DMF_32; + for (i = 0; i < data->dacs; ++i) + pcm1796_write_cached(chip, i, 18, reg); +} + static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -629,6 +646,7 @@ static void set_pcm1796_params(struct oxygen *chip, msleep(1); data->current_rate = params_rate(params); update_pcm1796_oversampling(chip); + update_pcm1796_deemph(chip); } static void update_pcm1796_volume(struct oxygen *chip) @@ -653,9 +671,11 @@ static void update_pcm1796_mute(struct oxygen *chip) unsigned int i; u8 value; - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; + value = data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; if (chip->dac_mute) value |= PCM1796_MUTE; + else + value &= ~PCM1796_MUTE; for (i = 0; i < data->dacs; ++i) pcm1796_write_cached(chip, i, 18, value); } @@ -777,6 +797,49 @@ static const struct snd_kcontrol_new rolloff_control = { .put = rolloff_put, }; +static int deemph_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + value->value.integer.value[0] = + !!(data->pcm1796_regs[0][18 - PCM1796_REG_BASE] & PCM1796_DME); + return 0; +} + +static int deemph_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; + if (!value->value.integer.value[0]) + reg &= ~PCM1796_DME; + else + reg |= PCM1796_DME; + changed = reg != data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; + if (changed) { + for (i = 0; i < data->dacs; ++i) + pcm1796_write(chip, i, 18, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new deemph_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "De-emphasis Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = deemph_get, + .put = deemph_put, +}; + static const struct snd_kcontrol_new hdav_hdmi_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "HDMI Playback Switch", @@ -1011,6 +1074,10 @@ static int add_pcm1796_controls(struct oxygen *chip) snd_ctl_new1(&rolloff_control, chip)); if (err < 0) return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&deemph_control, chip)); + if (err < 0) + return err; } return 0; } -- 2.20.1

3 2

[alsa-devel] Applied "ASoC: Intel: atom: Make PCI dependency explicit" to the asoc tree
by Mark Brown 03 Jan '19

03 Jan '19

The patch ASoC: Intel: atom: Make PCI dependency explicit has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6a444da7ddc6e478637a189a9c2898f2c3e0d399 Mon Sep 17 00:00:00 2001 From: Sinan Kaya <okaya(a)kernel.org> Date: Wed, 2 Jan 2019 18:10:35 +0000 Subject: [PATCH] ASoC: Intel: atom: Make PCI dependency explicit After 'commit 5d32a66541c4 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set")' dependencies on CONFIG_PCI that previously were satisfied implicitly through dependencies on CONFIG_ACPI have to be specified directly. This code relies on IOSF_MBI and IOSF_MBI depends on PCI. For this reason, add a direct dependency on CONFIG_PCI to the IOSF_MBI driver. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya <okaya(a)kernel.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 99a62ba409df..bd9fd2035c55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH -- 2.20.1

1 0

Re: [alsa-devel] [PATCH] In function rt274_i2c_probe(), if the regmap_read fails. The variable "val" could leave uninitialized but used in if statement.
by Mark Brown 03 Jan '19

03 Jan '19

On Wed, Jan 02, 2019 at 06:38:39PM -0800, Yizhuo wrote: > int ret; > - unsigned int val; > + unsigned int val = ~0; > This just shuts warnings up which may mask further problem, it is better to just leave it and fix the users. > - regmap_read(rt274->regmap, > + ret = regmap_read(rt274->regmap, > RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); > - if (val != RT274_VENDOR_ID) { > + if (ret || val != RT274_VENDOR_ID) { > dev_err(&i2c->dev, > "Device with ID register %#x is not rt274\n", val); > return -ENODEV; If the read returned an error we should pass back the error code we got from it, not mask it with another error code. This is especially true here where the error message would be very misleading if there's a problem with I/O.

1 0

Re: [alsa-devel] [PATCH] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode
by Mark Brown 03 Jan '19

03 Jan '19

On Wed, Jan 02, 2019 at 10:36:33PM +0530, b-ak wrote: > During the bootup of the kernel, as soon as the DAPM framework kicks in > it pushes the codec into standy mode. > > The existing TVL320AIC32x4 codec driver doesn't prepare the clock in > the probe function. > This leads to an OOPS when the DAPM tries to put it into standy by calling > clk_disable_unprepare() > > This patch fixes that problem. This isn't the best way of fixing this because it makes it look like there's a missing disable in the removal process. What would be better would be to do what other drivers do and check to see what state we're transitioning from before we disable the clock in set_bias_level(). See drivers like wm8903.c for examples.

1 0

[alsa-devel] [PATCH v5 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Sinan Kaya 03 Jan '19

03 Jan '19

After 'commit 5d32a66541c4 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set")' dependencies on CONFIG_PCI that previously were satisfied implicitly through dependencies on CONFIG_ACPI have to be specified directly. This code relies on IOSF_MBI and IOSF_MBI depends on PCI. For this reason, add a direct dependency on CONFIG_PCI to the IOSF_MBI driver. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya <okaya(a)kernel.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..b0764b2fe001 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH -- 2.19.0

4 9

[alsa-devel] [PATCH 2/2] ALSA: oxygen: initialize spdif_playback_enable to 0
by tom.ty89＠gmail.com 03 Jan '19

03 Jan '19

From: Tom Yan <tom.ty89(a)gmail.com> oxygen_init() has been clearing the OXYGEN_SPDIF_OUT_ENABLE bit and initializing dac_mute to 1. Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index b4ef5804212d..6a743c878415 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -373,7 +373,7 @@ static void oxygen_init(struct oxygen *chip) for (i = 0; i < 8; ++i) chip->dac_volume[i] = chip->model.dac_volume_min; chip->dac_mute = 1; - chip->spdif_playback_enable = 1; + chip->spdif_playback_enable = 0; chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL | (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT); chip->spdif_pcm_bits = chip->spdif_bits; -- 2.20.1

1 0

[alsa-devel] [Bug] [oxygen] Left/Right volumes de-sync when changed fast
by Tom Yan 03 Jan '19

03 Jan '19

Hi all, This is actually quite a long-time issue as far as I am aware of. My card is Xonar Essence STX. There is no problem when I change the volume in alsamixer. The problem occurs when I bind XF86Audio{Raise,Lower}Volume (a volume wheel on my keyboard) to amixer commands. Volumes of the two channel can easily de-sync from each other as long as I roll the wheel (up or down) fast. The problem doesn't occur if the commands target volume control of a onboard or USB sound card (even while their control allows different volumes for the two channels). Regards, Tom

1 0

[alsa-devel] [PATCH 1/2] ASoC: Intel: sst: Fallback to BYT-CR if IRQ 5 is missing
by Stephan Gerhold 02 Jan '19

02 Jan '19

Some devices detected as BYT-T by the PMIC-type based detection have only a single IRQ listed in the 80860F28 ACPI device. This causes -ENXIO later when attempting to get the IRQ at index 5. It turns out these devices behave more like BYT-CR devices, and using the IRQ at index 0 makes sound work correctly. This patch adds a fallback for these devices to is_byt_cr(): If there is no IRQ resource at index 5, treating the device as BYT-T is guaranteed to fail later, so we can safely treat these devices as BYT-CR without breaking any working device. Link: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143176.h… Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Moved the "Detected Baytrail-CR platform" message to is_byt_cr() so we can log a different message if the fallback is used. Tested this on my device as-is, and simulated a "normal" BYT-T and BYT-CR device (copied their IRQs to a custom DSDT). sound/soc/intel/atom/sst/sst_acpi.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3a95ebbfc45d..755a396121ff 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -255,10 +255,22 @@ static int is_byt(void) return status; } -static int is_byt_cr(struct device *dev, bool *bytcr) +static int is_byt_cr(struct platform_device *pdev, bool *bytcr) { + struct device *dev = &pdev->dev; int status = 0; + if (platform_get_resource(pdev, IORESOURCE_IRQ, 5) == NULL) { + /* + * Some devices detected as BYT-T have only a single IRQ listed, + * causing platform_get_irq with index 5 to return -ENXIO. + * The correct IRQ in this case is at index 0, as used on BYT-CR. + */ + dev_info(dev, "Falling back to Baytrail-CR platform\n"); + *bytcr = true; + return status; + } + if (IS_ENABLED(CONFIG_IOSF_MBI)) { u32 bios_status; @@ -278,10 +290,12 @@ static int is_byt_cr(struct device *dev, bool *bytcr) /* bits 26:27 mirror PMIC options */ bios_status = (bios_status >> 26) & 3; - if ((bios_status == 1) || (bios_status == 3)) + if ((bios_status == 1) || (bios_status == 3)) { + dev_info(dev, "Detected Baytrail-CR platform\n"); *bytcr = true; - else + } else { dev_info(dev, "BYT-CR not detected\n"); + } } } else { dev_info(dev, "IOSF_MBI not enabled, no BYT-CR detection\n"); @@ -333,10 +347,8 @@ static int sst_acpi_probe(struct platform_device *pdev) if (ret < 0) return ret; - ret = is_byt_cr(dev, &bytcr); + ret = is_byt_cr(pdev, &bytcr); if (!(ret < 0 || !bytcr)) { - dev_info(dev, "Detected Baytrail-CR platform\n"); - /* override resource info */ byt_rvp_platform_data.res_info = &bytcr_res_info; } -- 2.20.1

3 10

[alsa-devel] [PATCH v4 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Sinan Kaya 02 Jan '19

02 Jan '19

Code does unconditional select for IOSF_MBI. IOSF_MBI driver depends on CONFIG_PCI set but this is not specified anywhere. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya <okaya(a)kernel.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..b0764b2fe001 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH -- 2.19.0

4 11

[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Avoid access before bLength check in build_audio_procunit()
by Takashi Iwai 02 Jan '19

02 Jan '19

The parser for the processing unit reads bNrInPins field before the bLength sanity check, which may lead to an out-of-bound access when a malformed descriptor is given. Fix it by assignment after the bLength check. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c63c84b54969..0131de348cf6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2314,7 +2314,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, char *name) { struct uac_processing_unit_descriptor *desc = raw_desc; - int num_ins = desc->bNrInPins; + int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; int i, err, nameid, type, len; @@ -2329,7 +2329,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, 0, NULL, default_value_info }; - if (desc->bLength < 13 || desc->bLength < 13 + num_ins || + if (desc->bLength < 13) { + usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); + return -EINVAL; + } + + num_ins = desc->bNrInPins; + if (desc->bLength < 13 + num_ins || desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) { usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); return -EINVAL; -- 2.19.2

1 1

[alsa-devel] [PATCH 0/8] alsa-utils: axfer: fulfill help texts and manual sections for each of xfer backend
by GitHub pull_request - opened 02 Jan '19

02 Jan '19

alsa-project/alsa-utils pull request #8 was opened from takaswie: Hi, This patchset fulfills help texts and manual sections for each of xfer backend. All of options are described in this patchset, however I leave a placeholder for scheduling model of libasound. This section will be fulfilled later. (I note that this is a first attempt to explain about `slave` and `snoop` options of libffado.) ``` Takashi Sakamoto (8): axfer: fix no return statement in print_help() function axfer: fix typo in axfer(1) manual axfer: correct description about signal handling axfer: print help for libffado backend axfer: fulfill manual section for libffado backend axfer: print help for libasound backend axfer: fulfill manual section for libasound backend axfer: add text for compatibility loss of sw parameter in libasound backend axfer/axfer-transfer.1 | 383 ++++++++++++++++++++++++++++++++++++++++- axfer/axfer.1 | 2 +- axfer/xfer-libasound.c | 25 ++- axfer/xfer-libffado.c | 11 +- axfer/xfer-options.c | 2 +- 5 files changed, 414 insertions(+), 9 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/8 Patch URL : https://github.com/alsa-project/alsa-utils/pull/8.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] conf/ucm: bytcr-rt5651: Document mono speaker wiring
by Hans de Goede 01 Jan '19

01 Jan '19

During my recent work on the bytcht-es8316 UCM profile I realized that the bytcr-rt5651 devices with a single speaker use a differential setup just like the es8316 does. The tell-tale here is the speaker going quiet when playing the exact same sound on both channels when things are configured for stereo speakers. I've run some tests and the rt5651 does have a special mono balanced out mode for its line-out but using this results in the same sound (and the same loudness / volume of the sound) as our current solution, so adding support for this to the kernel buys us nothing. This commit makes no changes, it just documents my findings in a big comment for future reference. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/codecs/rt5651/MonoSpeaker.conf | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/src/conf/ucm/codecs/rt5651/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5651/MonoSpeaker.conf index 0c3492fb..f4181d10 100644 --- a/src/conf/ucm/codecs/rt5651/MonoSpeaker.conf +++ b/src/conf/ucm/codecs/rt5651/MonoSpeaker.conf @@ -9,7 +9,21 @@ SectionDevice."MonoSpeaker" { EnableSequence [ cdev "hw:bytcrrt5651" - # Map left and right input on left + # On most (all?) devices with a single / mono speaker an + # amplifier with a differential input connected between LOUT + # right and left is used. We deal with this by silencing the + # right output and mixing left + right to the left output. + # + # An alternative approach for devices using the differential + # setup would be to set bit 15 of register 5 in the codec, + # which turns the stereo unbalanced line out into a mono + # balanced line out. + # + # This possibility was discovered after we already shipped the + # current solution and the current solution has the advantage + # of also working on devices which only use LOUT left in + # unbalanced mode. The net result is the same for devices using + # the differential setup, so we just stick with what we have. cset "name='Stereo DAC MIXR DAC R1 Switch' off" cset "name='Stereo DAC MIXL DAC R1 Switch' on" -- 2.20.1

2 1

[alsa-devel] [PATCH 1/4] conf/ucm: bytcr-rt5651: Enable Stereo? ADC MIXL ADC? switches when enabling inputs
by Hans de Goede 01 Jan '19

01 Jan '19

Explicitly enable the right 'Stereo? ADC MIXL ADC?' when selecting an input, rather then just enabling them all in ucm/codecs/rt5651/EnableSeq.conf . This is a preparation patch for adding digital mic support. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/codecs/rt5651/EnableSeq.conf | 16 ++++++++-------- src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf | 4 ++++ 5 files changed, 24 insertions(+), 8 deletions(-) diff --git a/src/conf/ucm/codecs/rt5651/EnableSeq.conf b/src/conf/ucm/codecs/rt5651/EnableSeq.conf index 6fb3c1f2..db2fd328 100644 --- a/src/conf/ucm/codecs/rt5651/EnableSeq.conf +++ b/src/conf/ucm/codecs/rt5651/EnableSeq.conf @@ -14,12 +14,14 @@ cset "name='Stereo DAC MIXL DAC L1 Switch' on" cset "name='Stereo DAC MIXL DAC R1 Switch' off" cset "name='Stereo DAC MIXR DAC L1 Switch' off" cset "name='Stereo DAC MIXR DAC R1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC2 Switch' on" -cset "name='Stereo2 ADC MIXL ADC1 Switch' on" -cset "name='Stereo2 ADC MIXL ADC2 Switch' on" -cset "name='Stereo2 ADC MIXR ADC1 Switch' on" -cset "name='Stereo2 ADC MIXR ADC2 Switch' on" +cset "name='Stereo1 ADC MIXL ADC1 Switch' off" +cset "name='Stereo1 ADC MIXR ADC1 Switch' off" +cset "name='Stereo1 ADC MIXL ADC2 Switch' off" +cset "name='Stereo1 ADC MIXR ADC2 Switch' off" +cset "name='Stereo2 ADC MIXL ADC1 Switch' off" +cset "name='Stereo2 ADC MIXR ADC1 Switch' off" +cset "name='Stereo2 ADC MIXL ADC2 Switch' off" +cset "name='Stereo2 ADC MIXR ADC2 Switch' off" cset "name='Headphone Switch' off" cset "name='Speaker Switch' off" @@ -41,8 +43,6 @@ cset "name='IN3 Boost' 1" # input cset "name='Stereo1 ADC L1 Mux' ADC" cset "name='Stereo1 ADC R1 Mux' ADC" -cset "name='Stereo1 ADC MIXL ADC1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC1 Switch' on" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" cset "name='RECMIXL BST3 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf index 85e25a5c..b2ce17b6 100644 --- a/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf @@ -12,10 +12,14 @@ SectionDevice."InternalMic-IN1" { cset "name='Internal Mic Switch' on" cset "name='RECMIXL BST1 Switch' on" cset "name='RECMIXR BST1 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf index f1cc17fe..e8041912 100644 --- a/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf @@ -14,10 +14,14 @@ SectionDevice."InternalMic-IN12" { cset "name='RECMIXR BST1 Switch' on" cset "name='RECMIXL BST2 Switch' on" cset "name='RECMIXR BST2 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf index 18f74e1a..048fcc95 100644 --- a/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf @@ -12,10 +12,14 @@ SectionDevice."InternalMic-IN2" { cset "name='Internal Mic Switch' on" cset "name='RECMIXL BST2 Switch' on" cset "name='RECMIXR BST2 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST2 Switch' off" cset "name='RECMIXR BST2 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf index bfd0b009..712d14bb 100644 --- a/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf @@ -12,10 +12,14 @@ SectionDevice."HeadsetMic-IN3" { cset "name='Headset Mic Switch' on" cset "name='RECMIXL BST3 Switch' on" cset "name='RECMIXR BST3 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Headset Mic Switch' off" cset "name='RECMIXL BST3 Switch' off" cset "name='RECMIXR BST3 Switch' off" -- 2.20.1

2 6

[alsa-devel] [PATCH v2] staging: bcm2835-audio: interpolate audio delay
by Mike Brady 01 Jan '19

01 Jan '19

When the BCM2835 audio output is used, userspace sees a jitter up to 10ms in the audio position, aka "delay" -- the number of frames that must be output before a new frame would be played. Make this a bit nicer for userspace by interpolating the position using the CPU clock. The overhead is small -- an extra ktime_get() every time a GPU message is sent -- and another call and a few calculations whenever the delay is sought from userland. At 48,000 frames per second, i.e. approximately 20 microseconds per frame, it would take a clock inaccuracy of 20 microseconds in 10 milliseconds -- 2,000 parts per million -- to result in an inaccurate estimate, whereas crystal- or resonator-based clocks typically have an inaccuracy of 10s to 100s of parts per million. Signed-off-by: Mike Brady <mikebrady(a)eircom.net> --- Changes in v2 -- remove inappropriate addition of SNDRV_PCM_INFO_BATCH flag .../vc04_services/bcm2835-audio/bcm2835-pcm.c | 20 +++++++++++++++++++ .../vc04_services/bcm2835-audio/bcm2835.h | 1 + 2 files changed, 21 insertions(+) diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c index e66da11af5cf..9053b996cada 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c @@ -74,6 +74,7 @@ void bcm2835_playback_fifo(struct bcm2835_alsa_stream *alsa_stream, atomic_set(&alsa_stream->pos, pos); alsa_stream->period_offset += bytes; + alsa_stream->interpolate_start = ktime_get(); if (alsa_stream->period_offset >= alsa_stream->period_size) { alsa_stream->period_offset %= alsa_stream->period_size; snd_pcm_period_elapsed(substream); @@ -243,6 +244,7 @@ static int snd_bcm2835_pcm_prepare(struct snd_pcm_substream *substream) atomic_set(&alsa_stream->pos, 0); alsa_stream->period_offset = 0; alsa_stream->draining = false; + alsa_stream->interpolate_start = ktime_get(); return 0; } @@ -292,6 +294,24 @@ snd_bcm2835_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct bcm2835_alsa_stream *alsa_stream = runtime->private_data; + ktime_t now = ktime_get(); + + /* Give userspace better delay reporting by interpolating between GPU + * notifications, assuming audio speed is close enough to the clock + * used for ktime + */ + + if ((ktime_to_ns(alsa_stream->interpolate_start)) && + (ktime_compare(alsa_stream->interpolate_start, now) < 0)) { + u64 interval = + (ktime_to_ns(ktime_sub(now, + alsa_stream->interpolate_start))); + u64 frames_output_in_interval = + div_u64((interval * runtime->rate), 1000000000); + snd_pcm_sframes_t frames_output_in_interval_sized = + -frames_output_in_interval; + runtime->delay = frames_output_in_interval_sized; + } return snd_pcm_indirect_playback_pointer(substream, &alsa_stream->pcm_indirect, diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h index e13435d1c205..595ad584243f 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h @@ -78,6 +78,7 @@ struct bcm2835_alsa_stream { unsigned int period_offset; unsigned int buffer_size; unsigned int period_size; + ktime_t interpolate_start; struct bcm2835_audio_instance *instance; int idx; -- 2.17.1

4 17

[alsa-devel] [PATCH] aplay: support no period wakeup option in argument
by S.j. Wang 01 Jan '19

01 Jan '19

In the case that alsa driver can't support period wakeup, we need to set the no period wakeup flag Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- aplay/aplay.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/aplay/aplay.c b/aplay/aplay.c index efc1eb4cae3a..4f562bfe2884 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -137,6 +137,7 @@ static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; static int dump_hw_params = 0; +static int no_period_wakeup = 0; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -243,6 +244,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --use-strftime apply the strftime facility to the output file name\n" " --dump-hw-params dump hw_params of the device\n" " --fatal-errors treat all errors as fatal\n" +" --no-period-wakeup set no period wakeup flag if necessary\n" ) , command); printf(_("Recognized sample formats are:")); @@ -429,6 +431,7 @@ enum { OPT_USE_STRFTIME, OPT_DUMP_HWPARAMS, OPT_FATAL_ERRORS, + OPT_NO_PERIOD_WAKEUP, }; /* @@ -516,6 +519,7 @@ int main(int argc, char *argv[]) {"interactive", 0, 0, 'i'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, + {"no-period-wakeup", 0, 0, OPT_NO_PERIOD_WAKEUP}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, 0, 'm'}, #endif @@ -799,6 +803,9 @@ int main(int argc, char *argv[]) case OPT_FATAL_ERRORS: fatal_errors = 1; break; + case OPT_NO_PERIOD_WAKEUP: + no_period_wakeup = 1; + break; #ifdef CONFIG_SUPPORT_CHMAP case 'm': channel_map = snd_pcm_chmap_parse_string(optarg); @@ -1396,6 +1403,12 @@ static void set_params(void) &buffer_frames); } assert(err >= 0); + + if (no_period_wakeup) { + err = snd_pcm_hw_params_set_period_wakeup(handle, params, 0); + assert(err >= 0); + } + monotonic = snd_pcm_hw_params_is_monotonic(params); can_pause = snd_pcm_hw_params_can_pause(params); err = snd_pcm_hw_params(handle, params); -- 1.9.1

3 6

Re: [alsa-devel] [PATCH] pci: fix a missing check of snd_i2c_sendbytes
by Takashi Iwai 01 Jan '19

01 Jan '19

On Wed, 26 Dec 2018 06:12:04 +0100, Kangjie Lu wrote: > > snd_i2c_sendbytes could fail. The fix checks its return value: if it > fails, issues an error message and returns with its error code. > > Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> > --- > sound/pci/ice1712/ews.c | 5 ++++- > 1 file changed, 4 insertions(+), 1 deletion(-) > > diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c > index b8af747ecb43..b978cdb0fc20 100644 > --- a/sound/pci/ice1712/ews.c > +++ b/sound/pci/ice1712/ews.c > @@ -826,7 +826,10 @@ static int snd_ice1712_6fire_read_pca(struct snd_ice1712 *ice, unsigned char reg > > snd_i2c_lock(ice->i2c); > byte = reg; > - snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1); > + if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1)) { > + dev_err(ice->card->dev, "cannot send pca\n"); > + return -EIO; You forgot to unlock before returning. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH] line6: add a check for snd_card_register
by Takashi Iwai 01 Jan '19

01 Jan '19

On Wed, 26 Dec 2018 04:39:31 +0100, Kangjie Lu wrote: > > The fix checks if snd_card_register() fails, and if so, issues a > warning. > > Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> Hmm, you submitted multiple patches doing the similar things but the error messages are handled completely differently in both cases. Please make them consistent. ... and above all: > - snd_card_register(line6->card); > + if (snd_card_register(line6->card)) > + pr_warn("Failed to register ALSA card.\n"); This error message is almost useless. Better to use dev_*() variant. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH] sound: sb: fix a missing check of snd_ctl_add
by Takashi Iwai 01 Jan '19

01 Jan '19

On Wed, 26 Dec 2018 02:45:32 +0100, Kangjie Lu wrote: > > snd_ctl_add() could fail, so let's check its return value and return its > error code upstream upon failure. > > Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> > --- > sound/isa/sb/sb16_main.c | 10 ++++++++-- > 1 file changed, 8 insertions(+), 2 deletions(-) > > diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c > index 37e6ce7b0b13..c9b25dc1f78e 100644 > --- a/sound/isa/sb/sb16_main.c > +++ b/sound/isa/sb/sb16_main.c > @@ -879,8 +879,14 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device) > snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops); > snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops); > > - if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) > - snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip)); > + if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) { > + err = > + snd_ctl_add(card, > + snd_ctl_new1(&snd_sb16_dma_control, > + chip)); > + if (err < 0) > + return err; > + } > else > pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; The code change looks OK, but could you fix the indentation and avoid unneeded line breaks, as well as the missing braces for else-block? Also, we keep "ALSA:" prefix for the subject line (except for ASoC ones). Please align with it. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH] gus: add a check of the status of snd_ctl_add
by Takashi Iwai 01 Jan '19

01 Jan '19

On Wed, 26 Dec 2018 02:40:51 +0100, Kangjie Lu wrote: > > snd_ctl_add() could fail, so let's check its status and issue an error > message if it indeed fails. > > Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> The function itself prints the error in most of useful cases, so the extra error here would be superfluous. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH] sound:emu10k1:Use kmemdup instead of duplicating its function
by Takashi Iwai 01 Jan '19

01 Jan '19

On Mon, 24 Dec 2018 09:42:48 +0100, huang.zijiang wrote: > > From: "huang.zijiang" <huang.zijiang(a)zte.com.cn> > > kmemdup has implemented the function that kmalloc() and memcpy(). > > Signed-off-by: huang.zijiang <huang.zijiang(a)zte.com.cn> > --- > sound/pci/emu10k1/emufx.c | 3 +-- > 1 file changed, 1 insertion(+), 2 deletions(-) > > diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c > index 6ebe817..b4fe4c5 100644 > --- a/sound/pci/emu10k1/emufx.c > +++ b/sound/pci/emu10k1/emufx.c > @@ -671,10 +671,9 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv, bool in_kernel) > return NULL; > if (data[1] >= MAX_TLV_SIZE) > return NULL; > - tlv = kmalloc(data[1] + sizeof(data), GFP_KERNEL); > + tlv = kmemdup(data, data[1] + sizeof(data), GFP_KERNEL); > if (!tlv) > return NULL; > - memcpy(tlv, data, sizeof(data)); These changes are not equivalent, and rather dangerous, unfortunately. The memcpy() performs only for sizeof(data), and in this case, it's not the same size as the allocation above. thanks, Takashi

1 0

[alsa-devel] [alsa-lib][PATCH] conf/ucm: kblrt5660: Add ucm setting for Dell Edge IoT platform
by Hui Wang 01 Jan '19

01 Jan '19

The new generation of Dell Edge IoT platform is based on Intel Kabylake platform, and the audio codec is ALC3277 which is 100% compatible with RT5660 in I2S mode. The audio design on this IoT platform is as below: - Intel kabylake platform - connect the codec ALC3277 via SSP0 - line-out and line-in with Micbias jacks - line-out mute control and jack detection of line-out and line-in - two HDMI ports with audio capability Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 1 + src/conf/ucm/kblrt5660/Hdmi1 | 20 +++++++ src/conf/ucm/kblrt5660/Hdmi2 | 20 +++++++ src/conf/ucm/kblrt5660/HiFi | 85 +++++++++++++++++++++++++++ src/conf/ucm/kblrt5660/Makefile.am | 4 ++ src/conf/ucm/kblrt5660/kblrt5660.conf | 18 ++++++ 7 files changed, 149 insertions(+) create mode 100755 src/conf/ucm/kblrt5660/Hdmi1 create mode 100755 src/conf/ucm/kblrt5660/Hdmi2 create mode 100755 src/conf/ucm/kblrt5660/HiFi create mode 100755 src/conf/ucm/kblrt5660/Makefile.am create mode 100755 src/conf/ucm/kblrt5660/kblrt5660.conf diff --git a/configure.ac b/configure.ac index d0cab2d6..6185a7eb 100644 --- a/configure.ac +++ b/configure.ac @@ -745,6 +745,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/gpd-win-pocket-rt5645/Makefile \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ src/conf/ucm/HDAudio-Lenovo-DualCodecs/Makefile \ + src/conf/ucm/kblrt5660/Makefile \ src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 85c15090..993d02bb 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -29,6 +29,7 @@ GoogleNyan \ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ HDAudio-Lenovo-DualCodecs \ +kblrt5660 \ LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \ PandaBoard \ PandaBoardES \ diff --git a/src/conf/ucm/kblrt5660/Hdmi1 b/src/conf/ucm/kblrt5660/Hdmi1 new file mode 100755 index 00000000..95e6187a --- /dev/null +++ b/src/conf/ucm/kblrt5660/Hdmi1 @@ -0,0 +1,20 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi1" { + Comment "HDMI/Display Port 1 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:kblrt5660,2" + PlaybackChannels "2" + PlaybackPriority "2" + JackControl "HDMI/DP, pcm=4 Jack" + } +} + diff --git a/src/conf/ucm/kblrt5660/Hdmi2 b/src/conf/ucm/kblrt5660/Hdmi2 new file mode 100755 index 00000000..f6d2feee --- /dev/null +++ b/src/conf/ucm/kblrt5660/Hdmi2 @@ -0,0 +1,20 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi2" { + Comment "HDMI/Display Port 2 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:kblrt5660,3" + PlaybackChannels "2" + PlaybackPriority "3" + JackControl "HDMI/DP, pcm=5 Jack" + } +} + diff --git a/src/conf/ucm/kblrt5660/HiFi b/src/conf/ucm/kblrt5660/HiFi new file mode 100755 index 00000000..dbdb9006 --- /dev/null +++ b/src/conf/ucm/kblrt5660/HiFi @@ -0,0 +1,85 @@ +# UCM for Intel Kabylake platforms with RT5660 +# For Audio in I2S mode + +SectionVerb { + + EnableSequence [ + ] + + DisableSequence [ + ] + + # ALSA PCM + Value { + # ALSA PCM device for HiFi + PlaybackPCM "hw:kblrt5660,0" + CapturePCM "hw:kblrt5660,1" + } +} + +SectionDevice."LineOut" { + Comment "LineOut playback" + + EnableSequence [ + cdev "hw:kblrt5660" + cset "name='DAC1 Playback Volume' 30" + cset "name='OUT Playback Volume' 19" + cset "name='OUT Playback Switch' on" + cset "name='codec0_out mo media0_in mi Switch' on" + cset "name='DAC1 MIXL DAC1 Switch' on" + cset "name='DAC1 MIXR DAC1 Switch' on" + cset "name='Stereo DAC MIXL DAC L1 Switch' on" + cset "name='Stereo DAC MIXR DAC L1 Switch' off" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + cset "name='Stereo DAC MIXL DAC R1 Switch' off" + cset "name='LOUT MIX DAC Switch' on" + cset "name='Line Out Switch' on" + ] + + DisableSequence [ + cdev "hw:kblrt5660" + cset "name='Line Out Switch' off" + cset "name='DAC1 MIXL DAC1' off" + cset "name='DAC1 MIXR DAC1' off" + ] + + Value { + PlaybackChannels "2" + JackDev "rt5660-jack" + JackControl "Line Out Jack" + } +} + +SectionDevice."LineIn" { + Comment "LineIn capture" + + EnableSequence [ + cdev "hw:kblrt5660" + + cset "name='ADC Capture Volume' 30" + cset "name='IN1 Boost Volume' 30" + cset "name='IN2 Boost Volume' 30" + cset "name='STO1 ADC Boost Gain Volume' 1" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' on" + cset "name='RECMIXL BST2 Switch' on" + cset "name='RECMIXR BST2 Switch' off" + cset "name='Line In Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='media0_out mo codec0_in mi Switch' on" + ] + + DisableSequence [ + cdev "hw:kblrt5660" + cset "name='Line In Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + ] + + Value { + CaptureChannels "2" + JackDev "rt5660-jack" + JackControl "Line In Jack" + } +} diff --git a/src/conf/ucm/kblrt5660/Makefile.am b/src/conf/ucm/kblrt5660/Makefile.am new file mode 100755 index 00000000..83c3743a --- /dev/null +++ b/src/conf/ucm/kblrt5660/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/kblrt5660 +ucm_DATA = kblrt5660.conf HiFi Hdmi1 Hdmi2 +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/kblrt5660/kblrt5660.conf b/src/conf/ucm/kblrt5660/kblrt5660.conf new file mode 100755 index 00000000..c5e9420e --- /dev/null +++ b/src/conf/ucm/kblrt5660/kblrt5660.conf @@ -0,0 +1,18 @@ +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play HiFi quality Music" +} + +SectionUseCase."Hdmi1" { + File "Hdmi1" + Comment "Play on Hdmi/DP 1" +} + +SectionUseCase."Hdmi2" { + File "Hdmi2" + Comment "Play on Hdmi/DP 2" +} + +SectionDefaults [ + cdev "hw:kblrt5660" +] -- 2.17.1

2 1

Re: [alsa-devel] [PATCH] sound: atom: fix a missing check of snd_pcm_lib_malloc_pages
by Pierre-Louis Bossart 01 Jan '19

01 Jan '19

On 12/25/18 8:29 PM, Kangjie Lu wrote: > snd_pcm_lib_malloc_pages() may fail, so let's check its status and > return its error code upstream. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > > Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> > --- > sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 +++++++- > 1 file changed, 7 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c > index afc559866095..91a2436ce952 100644 > --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c > +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c > @@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, > struct snd_pcm_hw_params *params, > struct snd_soc_dai *dai) > { > - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); > + int ret; > + > + ret = > + snd_pcm_lib_malloc_pages(substream, > + params_buffer_bytes(params)); > + if (ret) > + return ret; > memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); > return 0; > }

2 1

[alsa-devel] [PATCH] ALSA: hda/tegra: clear pending irq handlers
by Sameer Pujar 01 Jan '19

01 Jan '19

Even after disabling interrupts on the module, it could be possible that irq handlers are still running. System hang is seen during suspend path. It was found that, there were pending writes on the HDA bus and clock was disabled by that time. Above mentioned issue is fixed by clearing any pending irq handlers before disabling clocks and returning from hda suspend. Suggested-by: Mohan Kumar <mkumard(a)nvidia.com> Suggested-by: Dara Ramesh <dramesh(a)nvidia.com> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 83befd8..97a176d 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -234,10 +234,12 @@ static int hda_tegra_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct hdac_bus *bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_stop_chip(chip); + synchronize_irq(bus->irq); azx_enter_link_reset(chip); hda_tegra_disable_clocks(hda); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable the headset mic auto detection for ASUS laptops
by Jian-Hong Pan 01 Jan '19

01 Jan '19

The headset mic of ASUS laptops like UX533FD, UX433FN and UX333FA, whose CODEC is Realtek ALC294 has jack auto detection feature. This patch enables the feature. Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a4f4a9dd488d..aee4cbd29d53 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6501,7 +6501,7 @@ static const struct hda_fixup alc269_fixups[] = { [ALC294_FIXUP_ASUS_HEADSET_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x19, 0x01a1113c }, /* use as headset mic, without its own jack detect */ + { 0x19, 0x01a1103c }, /* use as headset mic */ { } }, .chained = true, -- 2.11.0

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Revert DSP detection on legacy HD-audio driver
by Takashi Iwai 01 Jan '19

01 Jan '19

This essentially reverts the commits c337104b1a16 ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected") and d82b51c855a2 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection") for the path of legacy HD-audio controller (snd-hda-intel). The automatic DSP detection and skip of binding with the legacy driver caused regressions on several machines like Dell XPS13. They give the PCI class 0x40380 indicating the availability of DSP while they don't work with ASoC SKL driver (yet). As the support of ASoC driver for such devices isn't available, it's better to revert the whole DSP-detection-and-skip behavior of the legacy driver, so that we can get the old good driver working on such devices. The pci_binding option for ASoC SKL driver is still kept so that it can work without blacklisting. Fixes: c337104b1a16 ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected") Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Reported-by: Hans de Goede <hdegoede(a)redhat.com> Reported-by: Azat Khuzhin <dohardgopro(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Fix build error in sound/soc/intel/skylake/ sound/pci/hda/Kconfig | 62 ---------------------------------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 54 ++++------------------------- sound/soc/intel/Kconfig | 6 ---- 4 files changed, 8 insertions(+), 116 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 0d38c006e182..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -226,68 +226,6 @@ config SND_HDA_POWER_SAVE_DEFAULT The default time-out value in seconds for HD-audio automatic power-save mode. 0 means to disable the power-save mode. -if SND_HDA_INTEL - -# The options below should not be enabled by distributions or -# users. They are selected by Intel/Skylake or SOF drivers when they -# register for a PCI ID which is also handled by the HDAudio legacy -# driver. When this option is selected and the DSP is detected based on -# the PCI class/subclass/prog-if, the probe of the HDAudio legacy -# aborts. This mechanism removes the need for distributions to use -# blacklists. It can be bypassed with module parameters should the -# Intel/Skylake or SOF drivers fail to handle a specific platform. - -config SND_HDA_INTEL_DSP_DETECTION_SKL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - Skylake machines. - -config SND_HDA_INTEL_DSP_DETECTION_APL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - Broxton/ApolloLake machines - -config SND_HDA_INTEL_DSP_DETECTION_KBL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - KabyLake machines - -config SND_HDA_INTEL_DSP_DETECTION_GLK - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - GeminiLake machines - -config SND_HDA_INTEL_DSP_DETECTION_CNL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - CannonLake machines - -config SND_HDA_INTEL_DSP_DETECTION_CFL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - CoffeeLake machines - -config SND_HDA_INTEL_DSP_DETECTION_ICL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - IceLake machines - -endif ## SND_HDA_INTEL - endif endmenu diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index e0c3fcbaa028..7185ed574b41 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -37,7 +37,7 @@ #else #define AZX_DCAPS_I915_COMPONENT 0 /* NOP */ #endif -#define AZX_DCAPS_INTEL_SHARED (1 << 14) /* shared with ASoC */ +/* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ /* 17 unused */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e42cc2230977..e784130ea4e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -172,9 +172,6 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -static int skl_pci_binding; -module_param_named(pci_binding, skl_pci_binding, int, 0444); -MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -360,7 +357,6 @@ enum { AZX_DCAPS_NO_64BIT |\ AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF) -#define AZX_DCAPS_INTEL_DSP_DETECTION(conf) (IS_ENABLED(CONFIG_SND_HDA_INTEL_DSP_DETECTION_##conf) ? AZX_DCAPS_INTEL_SHARED : 0) /* * vga_switcheroo support */ @@ -2052,28 +2048,6 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - /* check if this driver can be used on SKL+ Intel platforms */ - if (pci_id->driver_data & AZX_DCAPS_INTEL_SHARED) { - switch (skl_pci_binding) { - case SND_SKL_PCI_BIND_AUTO: - if (pci->class != 0x040300) { - dev_info(&pci->dev, "The DSP is enabled on this platform, aborting probe\n"); - return -ENODEV; - } - dev_info(&pci->dev, "No DSP detected, continuing HDaudio legacy probe\n"); - break; - case SND_SKL_PCI_BIND_LEGACY: - dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, bypassed detection logic\n"); - break; - case SND_SKL_PCI_BIND_ASOC: - dev_info(&pci->dev, "Module parameter forced binding with SKL+ ASoC driver, aborting probe\n"); - return -ENODEV; - default: - dev_err(&pci->dev, "invalid value for skl_pci_binding module parameter, ignored\n"); - break; - } - } - if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { @@ -2380,48 +2354,34 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(SKL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake */ { PCI_DEVICE(0x8086, 0xa171), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake-LP */ { PCI_DEVICE(0x8086, 0x9d71), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(KBL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake-H */ { PCI_DEVICE(0x8086, 0xa2f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Coffelake */ { PCI_DEVICE(0x8086, 0xa348), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(CFL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Cannonlake */ { PCI_DEVICE(0x8086, 0x9dc8), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(CNL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(ICL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON | - AZX_DCAPS_INTEL_DSP_DETECTION(APL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Broxton-T */ { PCI_DEVICE(0x8086, 0x1a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Gemini-Lake */ { PCI_DEVICE(0x8086, 0x3198), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON | - AZX_DCAPS_INTEL_DSP_DETECTION(GLK) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..99a62ba409df 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -188,12 +188,6 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC - select SND_HDA_INTEL_DSP_DETECTION_SKL if SND_SOC_INTEL_SKL - select SND_HDA_INTEL_DSP_DETECTION_APL if SND_SOC_INTEL_APL - select SND_HDA_INTEL_DSP_DETECTION_KBL if SND_SOC_INTEL_KBL - select SND_HDA_INTEL_DSP_DETECTION_GLK if SND_SOC_INTEL_GLK - select SND_HDA_INTEL_DSP_DETECTION_CNL if SND_SOC_INTEL_CNL - select SND_HDA_INTEL_DSP_DETECTION_CFL if SND_SOC_INTEL_CFL select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ -- 2.19.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Revert DSP detection on legacy HD-audio driver
by Takashi Iwai 01 Jan '19

01 Jan '19

This essentially reverts the commits c337104b1a16 ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected") and d82b51c855a2 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection") for the path of legacy HD-audio controller (snd-hda-intel). The automatic DSP detection and skip of binding with the legacy driver caused regressions on several machines like Dell XPS13. They give the PCI class 0x40380 indicating the availability of DSP while they don't work with ASoC SKL driver (yet). As the support of ASoC driver for such devices isn't available, it's better to revert the whole DSP-detection-and-skip behavior of the legacy driver, so that we can get the old good driver working on such devices. The pci_binding option for ASoC SKL driver is still kept so that it can work without blacklisting. Fixes: c337104b1a16 ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected") Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Reported-by: Hans de Goede <hdegoede(a)redhat.com> Reported-by: Azat Khuzhin <dohardgopro(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 6 ---- sound/pci/hda/Kconfig | 62 ---------------------------------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 54 ++++------------------------- sound/soc/intel/Kconfig | 6 ---- 5 files changed, 8 insertions(+), 122 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b4fa1c775251..f9b2b6330d27 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -98,12 +98,6 @@ enum { HDA_DEV_ASOC, }; -enum { - SND_SKL_PCI_BIND_AUTO, /* automatic selection based on pci class */ - SND_SKL_PCI_BIND_LEGACY,/* bind only with legacy driver */ - SND_SKL_PCI_BIND_ASOC /* bind only with ASoC driver */ -}; - /* direction */ enum { HDA_INPUT, HDA_OUTPUT diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 0d38c006e182..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -226,68 +226,6 @@ config SND_HDA_POWER_SAVE_DEFAULT The default time-out value in seconds for HD-audio automatic power-save mode. 0 means to disable the power-save mode. -if SND_HDA_INTEL - -# The options below should not be enabled by distributions or -# users. They are selected by Intel/Skylake or SOF drivers when they -# register for a PCI ID which is also handled by the HDAudio legacy -# driver. When this option is selected and the DSP is detected based on -# the PCI class/subclass/prog-if, the probe of the HDAudio legacy -# aborts. This mechanism removes the need for distributions to use -# blacklists. It can be bypassed with module parameters should the -# Intel/Skylake or SOF drivers fail to handle a specific platform. - -config SND_HDA_INTEL_DSP_DETECTION_SKL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - Skylake machines. - -config SND_HDA_INTEL_DSP_DETECTION_APL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - Broxton/ApolloLake machines - -config SND_HDA_INTEL_DSP_DETECTION_KBL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - KabyLake machines - -config SND_HDA_INTEL_DSP_DETECTION_GLK - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - GeminiLake machines - -config SND_HDA_INTEL_DSP_DETECTION_CNL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - CannonLake machines - -config SND_HDA_INTEL_DSP_DETECTION_CFL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - CoffeeLake machines - -config SND_HDA_INTEL_DSP_DETECTION_ICL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - IceLake machines - -endif ## SND_HDA_INTEL - endif endmenu diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index e0c3fcbaa028..7185ed574b41 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -37,7 +37,7 @@ #else #define AZX_DCAPS_I915_COMPONENT 0 /* NOP */ #endif -#define AZX_DCAPS_INTEL_SHARED (1 << 14) /* shared with ASoC */ +/* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ /* 17 unused */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e42cc2230977..e784130ea4e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -172,9 +172,6 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -static int skl_pci_binding; -module_param_named(pci_binding, skl_pci_binding, int, 0444); -MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -360,7 +357,6 @@ enum { AZX_DCAPS_NO_64BIT |\ AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF) -#define AZX_DCAPS_INTEL_DSP_DETECTION(conf) (IS_ENABLED(CONFIG_SND_HDA_INTEL_DSP_DETECTION_##conf) ? AZX_DCAPS_INTEL_SHARED : 0) /* * vga_switcheroo support */ @@ -2052,28 +2048,6 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - /* check if this driver can be used on SKL+ Intel platforms */ - if (pci_id->driver_data & AZX_DCAPS_INTEL_SHARED) { - switch (skl_pci_binding) { - case SND_SKL_PCI_BIND_AUTO: - if (pci->class != 0x040300) { - dev_info(&pci->dev, "The DSP is enabled on this platform, aborting probe\n"); - return -ENODEV; - } - dev_info(&pci->dev, "No DSP detected, continuing HDaudio legacy probe\n"); - break; - case SND_SKL_PCI_BIND_LEGACY: - dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, bypassed detection logic\n"); - break; - case SND_SKL_PCI_BIND_ASOC: - dev_info(&pci->dev, "Module parameter forced binding with SKL+ ASoC driver, aborting probe\n"); - return -ENODEV; - default: - dev_err(&pci->dev, "invalid value for skl_pci_binding module parameter, ignored\n"); - break; - } - } - if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { @@ -2380,48 +2354,34 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(SKL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake */ { PCI_DEVICE(0x8086, 0xa171), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake-LP */ { PCI_DEVICE(0x8086, 0x9d71), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(KBL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake-H */ { PCI_DEVICE(0x8086, 0xa2f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Coffelake */ { PCI_DEVICE(0x8086, 0xa348), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(CFL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Cannonlake */ { PCI_DEVICE(0x8086, 0x9dc8), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(CNL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(ICL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON | - AZX_DCAPS_INTEL_DSP_DETECTION(APL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Broxton-T */ { PCI_DEVICE(0x8086, 0x1a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Gemini-Lake */ { PCI_DEVICE(0x8086, 0x3198), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON | - AZX_DCAPS_INTEL_DSP_DETECTION(GLK) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..99a62ba409df 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -188,12 +188,6 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC - select SND_HDA_INTEL_DSP_DETECTION_SKL if SND_SOC_INTEL_SKL - select SND_HDA_INTEL_DSP_DETECTION_APL if SND_SOC_INTEL_APL - select SND_HDA_INTEL_DSP_DETECTION_KBL if SND_SOC_INTEL_KBL - select SND_HDA_INTEL_DSP_DETECTION_GLK if SND_SOC_INTEL_GLK - select SND_HDA_INTEL_DSP_DETECTION_CNL if SND_SOC_INTEL_CNL - select SND_HDA_INTEL_DSP_DETECTION_CFL if SND_SOC_INTEL_CFL select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ -- 2.19.2

3 5

[alsa-devel] "ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected" breaks sound, suspend on Apollo Lake
by Hans de Goede 31 Dec '18

31 Dec '18

Hi Pierre-Louis, Since I recently have been doing a bunch of i915 work I'm currently running drm-tip. This brought in this commit from next: https://cgit.freedesktop.org/drm-tip/commit/?id=c337104b1a16becc486fdc95d54… This causes the hda driver to not bind to the audio on my Apollo Lake based laptop, and the asoc driver errors out with: [ 19.176417] skl_hda_dsp_generic skl_hda_dsp_generic: Unsupported HDAudio/iDisp configuration found [ 19.176428] skl_hda_dsp_generic: probe of skl_hda_dsp_generic failed with error -22 After which I get no sound and worse, the machine hangs on suspend. It seems to never reach its fully suspended state (the power-led never starts to fade on/off as it normally does when suspended). I've been looking through the Kconfig option and it seems that currently selecting SND_SOC_INTEL_APL makes little sense since there are 0 machine drivers for this setup ? I guess I can fix this by unselecting SND_SOC_INTEL_SKYLAKE and then during the manual platform selection not selecting SND_SOC_INTEL_APL, but IMHO we need to put something in place where the hda driver actually checks if there is a machine driver and if there isn't one then we should keep using the hda driver on the machine in question. Regards, Hans p.s. Merry Christmas!

3 5

[alsa-devel] snd_pcm_open() taking very long, starting from kernel 4.17
by Arnaud Rebillout 31 Dec '18

31 Dec '18

Dear ALSA maintainers, I noticed big change in how long `snd_pcm_open()` takes, between kernel 4.16 and 4.17. With a kernel 4.16, this function consistently returns in less than 1 millisecond, with occasional, but rare, peaks that go up to 25ms. While with kernel 4.17 and above, I measure an average of 40ms, with frequent peaks to 1s. The maximum I measured was 3 seconds! Is this change known, intended? Or is it a regression? Please read below for more details, thanks. ---- The long story Starting with kernel 4.17, I started to experience frequent lags on mylaptop, which seemed to happen in the terminal, while hitting the TAB for completion. I clearly didn't experience this behavior with 4.16. Later on, 4.18 was released, I tried it: same issue. So I kept using the kernel 4.16. So it took me a while to investigate this issue. What happens is that the GNOME terminal, or Emacs, might generate event sounds when you hit tab, or something. This happen fairly often (you know, this kind of bell sound), it's very annoying, so usually the first thing I'd do is to disable these sounds. Turns out that, even when disabled, this sounds are still generated, because GNOME doesn't really disable it, it just mutes it [1]. After understanding that, I tracked down how these event sounds are generated. As incredible as it seems, it all happens synchronously, from the calling app, through GTK, then mutter, then libcanberra, and ultimately the call ends up in pulseaudio. Pulseaudio is the one calling `snd_pcm_open()`, so that's where I did my measurements. Like I said, it's all synchronous, so when the call takes 3 seconds to return, I wait 3 seconds in front of my screen :) Pulseaudio has a plugin (enabled by default in my distrib) `suspend-on-idle`, which means that it closes the soundcard connection on inactivity, and re-opens it again with `snd_pcm_open()` when a sound is played. So, to sum that up: during a normal workday on my laptop, PA will constantly close the soundcard connection (due to the suspend-on-idle plugin), and open it again (due to event sounds generated by GNOME, despite being disabled). And all of that synchronously, in the GTK mainloop. While this could go un-noticed for a while, starting with kernel 4.17, the call `snd_pcm_open()` might take up to 3 seconds to return, and it made my system almost unusable. I found out that it's possible to disable these event sounds finally, it's just not possible through the GNOME settings, it needs to be done from the command- line. So I live happy, no problem for me. I'm merely reporting the problem, as you might be interested. [1] https://gitlab.gnome.org/GNOME/gnome-control-center/issues/259 ---- Some hardware details: $ sudo dmidecode | grep 'System Information' -A 2 System Information Manufacturer: Dell Inc. Product Name: XPS 13 9360 $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xdc428000 irq 144 $ aplay --list-devices **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC3246 Analog [ALC3246 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 $ sudo lspci -v | grep -i audio -A 7 00:1f.3 Audio device: Intel Corporation Sunrise Point-LP HD Audio (rev 21) (prog-if 80) Subsystem: Dell Sunrise Point-LP HD Audio Flags: bus master, fast devsel, latency 32, IRQ 128 Memory at dc428000 (64-bit, non-prefetchable) [size=16K] Memory at dc400000 (64-bit, non-prefetchable) [size=64K] Capabilities: [50] Power Management version 3 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel, snd_soc_skl ---- Feel free to ask more details, I'd be glad to help. Best regards, Arnaud

2 3

[alsa-devel] libhinawa v1.1.0 release
by Takashi Sakamoto 30 Dec '18

30 Dec '18

Hi, Today I released libhinawa v1.1.0. https://github.com/takaswie/libhinawa/releases/tag/1.1.0 This library is designed to send asynchronous transaction from Linux userspace applications to units on IEEE 1394 bus by any language binding for GObject Introspection. This library also includes some functionalities to handle model specific features with helps of drivers in ALSA firewire stack. In this release, a minor part of version number is incremented from 0 to 1 to submit an addition of new GObject object; HinawaSndTscm. This object corresponds to a work in ALSA firewire stack in Linux sound subsystem[1]. In a development period toward v4.21, a new feature was added into ALSA firewire-tascam driver to support userspace interface for status and control information demultiplexed from isochronous packet from target device. The new object supports two features: * 'control' GObject signal to handle events of control surface on target device * 'hinawa_snd_tscm_get_state()' to read the latest status of target device Additionally, in this release, thread model is changed. Till former releases, this library internally starts a thread to handle both of Linux firewire character device (/dev/fw*) and ALSA hwdep character device (/dev/snd/hw*). Nowadays I recognize a need to poll /dev/fw* during operating /dev/snd/hw* for asynchronous transactions on IEEE 1394 bus. To achieve this, this library is changed to start two threads for the both. Finally, I note that this series of work for ALSA firewire-tascam driver was assisted by Scott Bahling. Without his help to investigate content of status and control information on transferred image, we might not have handled events of control surface. For detail, please refer to archived messages for Sep. 2018 to Nov. 2018 in alsa-devel(a)alsa-project.org[2]. I greatly appreciate his help. [1] [GIT PULL] sound updates for 4.21 http://lkml.iu.edu/hypermail/linux/kernel/1812.2/04303.html [2] Alsa-devel Info Page http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Happy new year. Takashi Sakamoto

1 0

[alsa-devel] KASAN: use-after-free Read in delete_and_unsubscribe_port
by syzbot 29 Dec '18

29 Dec '18

Hello, syzbot found the following crash on: HEAD commit: 8fe28cb58bcb Linux 4.20 git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=166859b3400000 kernel config: https://syzkaller.appspot.com/x/.config?x=7d581260bae0899a dashboard link: https://syzkaller.appspot.com/bug?extid=9917f000776fb73fa40f compiler: gcc (GCC) 8.0.1 20180413 (experimental) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+9917f000776fb73fa40f(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in __read_once_size include/linux/compiler.h:191 [inline] BUG: KASAN: use-after-free in list_empty include/linux/list.h:226 [inline] BUG: KASAN: use-after-free in delete_and_unsubscribe_port+0x5d7/0x6d0 sound/core/seq/seq_ports.c:548 Read of size 8 at addr ffff8881d1c05590 by task syz-executor1/6930 CPU: 0 PID: 6930 Comm: syz-executor1 Not tainted 4.20.0 #167 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1d3/0x2c6 lib/dump_stack.c:113 print_address_description.cold.8+0x9/0x1ff mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report.cold.9+0x242/0x309 mm/kasan/report.c:412 __asan_report_load8_noabort+0x14/0x20 mm/kasan/report.c:433 __read_once_size include/linux/compiler.h:191 [inline] list_empty include/linux/list.h:226 [inline] delete_and_unsubscribe_port+0x5d7/0x6d0 sound/core/seq/seq_ports.c:548 snd_seq_port_disconnect+0x17b/0x280 sound/core/seq/seq_ports.c:628 snd_seq_ioctl_unsubscribe_port+0x1d8/0x310 sound/core/seq/seq_clientmgr.c:1499 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0x1de/0x1790 fs/ioctl.c:696 ksys_ioctl+0xa9/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457759 Code: fd b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f36626cac78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 0000000000457759 RDX: 00000000200001c0 RSI: 0000000040505331 RDI: 0000000000000003 RBP: 000000000073bf00 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f36626cb6d4 R13: 00000000004ca358 R14: 00000000004d35d0 R15: 00000000ffffffff Allocated by task 6924: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] kasan_kmalloc+0xc7/0xe0 mm/kasan/kasan.c:553 kmem_cache_alloc_trace+0x152/0x750 mm/slab.c:3620 kmalloc include/linux/slab.h:546 [inline] kzalloc include/linux/slab.h:741 [inline] snd_seq_port_connect+0xe0/0x760 sound/core/seq/seq_ports.c:571 snd_seq_ioctl_subscribe_port+0x1d8/0x310 sound/core/seq/seq_clientmgr.c:1458 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0x1de/0x1790 fs/ioctl.c:696 ksys_ioctl+0xa9/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe Freed by task 6931: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] __kasan_slab_free+0x102/0x150 mm/kasan/kasan.c:521 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:528 __cache_free mm/slab.c:3498 [inline] kfree+0xcf/0x230 mm/slab.c:3817 snd_seq_port_disconnect+0x205/0x280 sound/core/seq/seq_ports.c:632 snd_seq_ioctl_unsubscribe_port+0x1d8/0x310 sound/core/seq/seq_clientmgr.c:1499 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0x1de/0x1790 fs/ioctl.c:696 ksys_ioctl+0xa9/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe The buggy address belongs to the object at ffff8881d1c05540 which belongs to the cache kmalloc-128 of size 128 The buggy address is located 80 bytes inside of 128-byte region [ffff8881d1c05540, ffff8881d1c055c0) The buggy address belongs to the page: page:ffffea0007470140 count:1 mapcount:0 mapping:ffff8881da800640 index:0xffff8881d1c05b40 flags: 0x2fffc0000000200(slab) raw: 02fffc0000000200 ffffea0006fb09c8 ffffea0006f38908 ffff8881da800640 raw: ffff8881d1c05b40 ffff8881d1c05000 000000010000000e 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d1c05480: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8881d1c05500: fc fc fc fc fc fc fc fc fb fb fb fb fb fb fb fb > ffff8881d1c05580: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ^ ffff8881d1c05600: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8881d1c05680: fc fc fc fc fc fc fc fc fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

1 0

[alsa-devel] Coherence of status structure after SYNC_PTR
by Ricardo Biehl Pasquali 28 Dec '18

28 Dec '18

In alsa-lib, pcm_hw, htimestamp(), avail is rechecked to ensure a valid timestamp. I guess this is for the case when status is mmaped, when hardware pointer is read, an interrupt occur, and timestamp is read. Does SYNC_PTR ioctl always fill a coherent status? Cheers! pasquali

2 3

[alsa-devel] Codec tlv320aic3401 and SoC im6ul (no i2c input/output, aplay hang up)
by tmptmp15 27 Dec '18

27 Dec '18

Hello ! I try make kernel with ALSA drivers for tlv320aic3401. Platform arm / imx6ul. CPU: Freescale i.MX6UL rev1.2 528 MHz (running at 396 MHz) Kernel config: $grep SND_ .config |grep -v ^# CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_DMAENGINE_PCM=y CONFIG_SND_HWDEP=y CONFIG_SND_RAWMIDI=y CONFIG_SND_COMPRESS_OFFLOAD=y CONFIG_SND_JACK=y CONFIG_SND_SUPPORT_OLD_API=y CONFIG_SND_VERBOSE_PROCFS=y CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y CONFIG_SND_ARM=y CONFIG_SND_SPI=y CONFIG_SND_USB=y CONFIG_SND_USB_AUDIO=y CONFIG_SND_SOC=y CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM=y CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_SOC_FSL_SAI=y CONFIG_SND_SOC_FSL_SSI=y CONFIG_SND_SOC_FSL_ESAI=y CONFIG_SND_SOC_IMX_PCM_DMA=y CONFIG_SND_SOC_IMX_AUDMUX=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_IMX_PCM_FIQ=y CONFIG_SND_SOC_FSL_ASOC_CARD=y CONFIG_SND_SOC_IMX_TLV320AIC3X=y CONFIG_SND_SOC_I2C_AND_SPI=y CONFIG_SND_SOC_TLV320AIC3X=y $uname -a Linux myd-y6ul14x14 4.1.15+ #20 SMP PREEMPT Thu Dec 27 16:00:37 +07 2018 armv7l GNU/Linux dmesg / i2c: imx-i2c 21a4000.i2c: <i2c_imx_probe> i2c i2c-1: adapter [21a4000.i2c] registered i2c i2c-1: of_i2c: walking child nodes i2c i2c-1: of_i2c: register /soc/aips-bus@02100000/i2c@021a4000/tlv320aic3104@18 i2c 1-0018: uevent i2c i2c-1: client [tlv320aic3104] registered with bus id 1-0018 i2c i2c-1: of_i2c: register /soc/aips-bus@02100000/i2c@021a4000/rtc@68 i2c 1-0068: uevent i2c i2c-1: client [ds1307] registered with bus id 1-0068 i2c i2c-1: claimed irq 228 i2c i2c-1: device resources: [mem 0x021a4000-0x021a7fff] i2c i2c-1: adapter name: "21a4000.i2c" i2c i2c-1: IMX I2C adapter registered i2c i2c-1: can't request DMA tx channel i2c i2c-1: can't use DMA I2C physic work fine: i can access to ds1307, and tlv320 may detect with i2cdetect (if disable tlv320 in DTS). On console, dmesg / tlv320: .... of_get_named_gpiod_flags: parsed 'gpio-reset' property of node '/soc/aips-bus@02100000/i2c@021a4000/tlv320aic3104@18[0]' - status (0) aic3x_i2c_probe: model=3 1-0018 supply IOVDD not found, using dummy regulator 1-0018 supply DVDD not found, using dummy regulator 1-0018 supply AVDD not found, using dummy regulator 1-0018 supply DRVDD not found, using dummy regulator i2c-core: driver [tlv320aic3x-codec] registered fsl-asrc 2034000.asrc: driver registered ... tlv320aic3x sound: tlv320aic3x-hifi <-> 202c000.sai mapping ok ... ALSA device list: #0: tlv320-audio-U .... Device Tree Source: &i2c2 { clock_frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay"; codec: tlv320aic3104@18 { /* ai3x-micbias-vg = <2>; */ compatible = "ti,tlv320aic3104"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_gpio_key>; reg = <0x18>; gpio-reset = <&gpio5 0 GPIO_ACTIVE_LOW>; clocks = <&clks IMX6UL_CLK_SAI2>; clock-names = "mclk"; /* clock-frequency = <19200000>; */ status = "okay"; }; ..... }; / { ..... sound { compatible = "fsl,imx-audio-tlv320aic3x"; model = "tlv320-audio-U"; cpu-dai = <&sai2>; audio-codec = <&codec>; ssi-controller = <&sai2>; mux-int-port = <2>; /* 1, 2, 7 - SSI */ mux-ext-port = <5>; /* 3-6, IOMUX */ audio-routing = "LINE1L", "Mic Jack", "Headphone Jack", "HPLOUT", "Headphone Jack", "HPROUT", "Mic Jack", "Mic Bias"; }; ..... } Machine driver imx-audio-tlv320aic3x.c not present in kernel sources, I try some version from Inet. At any case codec driver sound/soc/codec/tlv320aic3x.c call procedure snd_soc_write() for init codec registers, but no any messages from i2c for address 0x18 in dmesg or /var/log/messages. In kernel config i2c debug messages enable: CONFIG_I2C_DEBUG_CORE=y CONFIG_I2C_DEBUG_ALGO=y CONFIG_I2C_DEBUG_BUS=y But if I run aplay: #aplay -vvv x.wav Playing WAVE 'x.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 15876 period_size : 1764 period_time : 40000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1764 period_event : 0 start_threshold : 15876 stop_threshold : 15876 silence_threshold: 0 silence_size : 0 boundary : 2080899072 Hardware PCM card 0 'tlv320-audio-U' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 15876 period_size : 1764 period_time : 40000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 1764 period_event : 0 start_threshold : 1 stop_threshold : 2080899072 silence_threshold: 0 silence_size : 2080899072 boundary : 2080899072 appl_ptr : 0 hw_ptr : 0 Max peak (3528 samples): 0x00000000 # 0% Max peak (3528 samples): 0x00000285 # 1% Max peak (3528 samples): 0x000003dd # 3% Max peak (3528 samples): 0x000004b8 # 3% Max peak (3528 samples): 0x00001739 #### 18% Max peak (3528 samples): 0x0000068d ## 5% Max peak (3528 samples): 0x0000058f # 4% Max peak (3528 samples): 0x00000534 # 4% Max peak (3528 samples): 0x000004a3 # 3% Then in /var/log/messages append some messages, like this: Dec 27 09:29:39 myd-y6ul14x14 user.debug kernel: i2c i2c-1: master_xfer[0] W, addr=0x18, len=2 and in I2C wire I see some pulses. And then aplay is locked. :( No pulses at SAI. Questions: 1) It problem may occur due to my error in DTS ? 2) snd_soc_write() / snd_soc_read() do call direct to i2c or more complex way ? For what reason can request not always reach i2c? 3) Any advice ? Thank you!

1 0

[alsa-devel] [PATCH 1/7] ASoC: Intel: common: Add quirk for PoV P1006W tablet
by Hans de Goede 26 Dec '18

26 Dec '18

The Point of View TAB-P1006W-232 (v1.0) tablet uses 10EC5640 as ACPI HID, but it has a rt5651 codec add a quirk for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- .../intel/common/soc-acpi-intel-byt-match.c | 35 +++++++++++++++++-- 1 file changed, 33 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index 097dc06377ba..027dc27262b7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -21,6 +21,7 @@ static unsigned long byt_machine_id; #define BYT_THINKPAD_10 1 +#define BYT_POV_P1006W 2 static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) { @@ -28,6 +29,11 @@ static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) return 1; } +static int byt_pov_p1006w_quirk_cb(const struct dmi_system_id *id) +{ + byt_machine_id = BYT_POV_P1006W; + return 1; +} static const struct dmi_system_id byt_table[] = { { @@ -58,6 +64,17 @@ static const struct dmi_system_id byt_table[] = { DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"), }, }, + { + /* Point of View mobii wintab p1006w (v1.0) */ + .callback = byt_pov_p1006w_quirk_cb, + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "BayTrail"), + /* Note 105b is Foxcon's USB/PCI vendor id */ + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "105B"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "0E57"), + }, + }, { } }; @@ -71,16 +88,30 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .asoc_plat_name = "sst-mfld-platform", }; +static struct snd_soc_acpi_mach byt_pov_p1006w = { + .id = "10EC5640", + .drv_name = "bytcr_rt5651", + .fw_filename = "intel/fw_sst_0f28.bin", + .board = "bytcr_rt5651", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5651.tplg", + .asoc_plat_name = "sst-mfld-platform", +}; + static struct snd_soc_acpi_mach *byt_quirk(void *arg) { struct snd_soc_acpi_mach *mach = arg; dmi_check_system(byt_table); - if (byt_machine_id == BYT_THINKPAD_10) + switch (byt_machine_id) { + case BYT_THINKPAD_10: return &byt_thinkpad_10; - else + case BYT_POV_P1006W: + return &byt_pov_p1006w; + default: return mach; + } } struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_legacy_machines[] = { -- 2.20.1

2 7

Re: [alsa-devel] [PATCH v3 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Pierre-Louis Bossart 26 Dec '18

26 Dec '18

On 12/23/18 5:25 PM, Sinan Kaya wrote: > Code does unconditional select for IOSF_MBI. IOSF_MBI driver depends on > CONFIG_PCI set but this is not specified anywhere. Can you please share the .config that exposes this problem? This hasn't changed in a long time and I wonder why this pops up now. You have similar cases elsewhere, e.g. arch/x86/Kconfig config X86_INTEL_LPSS bool "Intel Low Power Subsystem Support" depends on X86 && ACPI select COMMON_CLK select PINCTRL select IOSF_MBI or for the MMC. config MMC_SDHCI_ACPI tristate "SDHCI support for ACPI enumerated SDHCI controllers" depends on MMC_SDHCI && ACPI select IOSF_MBI if X86 The use of IOSF_MBI is only for the Baytrail-CR detection and there are already in-flight patches to change the code and move it to a helper. My understanding is that for compilation you only need CONFIG_x86 since there are fall-back routines available in iosf_mbi.h Thanks! > > Signed-off-by: Sinan Kaya <okaya(a)kernel.org> > --- > sound/soc/intel/Kconfig | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig > index 2fd1b61e8331..b0764b2fe001 100644 > --- a/sound/soc/intel/Kconfig > +++ b/sound/soc/intel/Kconfig > @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI > config SND_SST_ATOM_HIFI2_PLATFORM_ACPI > tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" > default ACPI > - depends on X86 && ACPI > + depends on X86 && ACPI && PCI > select SND_SST_IPC_ACPI > select SND_SST_ATOM_HIFI2_PLATFORM > select SND_SOC_ACPI_INTEL_MATCH

1 0

[alsa-devel] [PATCH v2 1/3] mfd: cros_ec: Add commands to control codec
by Cheng-Yi Chiang 26 Dec '18

26 Dec '18

Add EC host commands to control codec on EC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- Note: This patch is merged to mfd tree for-mfd-next branch already. But this is still needed on sound tree for-next branch in order to compile codec driver. include/linux/mfd/cros_ec_commands.h | 94 ++++++++++++++++++++++++++++ 1 file changed, 94 insertions(+) diff --git a/include/linux/mfd/cros_ec_commands.h b/include/linux/mfd/cros_ec_commands.h index 9a9631f0559e2..fc91082d4c357 100644 --- a/include/linux/mfd/cros_ec_commands.h +++ b/include/linux/mfd/cros_ec_commands.h @@ -2790,6 +2790,100 @@ struct ec_response_battery_vendor_param { uint32_t value; } __packed; +/*****************************************************************************/ +/* Commands for I2S recording on audio codec. */ + +#define EC_CMD_CODEC_I2S 0x00BC + +enum ec_codec_i2s_subcmd { + EC_CODEC_SET_SAMPLE_DEPTH = 0x0, + EC_CODEC_SET_GAIN = 0x1, + EC_CODEC_GET_GAIN = 0x2, + EC_CODEC_I2S_ENABLE = 0x3, + EC_CODEC_I2S_SET_CONFIG = 0x4, + EC_CODEC_I2S_SET_TDM_CONFIG = 0x5, + EC_CODEC_I2S_SET_BCLK = 0x6, +}; + +enum ec_sample_depth_value { + EC_CODEC_SAMPLE_DEPTH_16 = 0, + EC_CODEC_SAMPLE_DEPTH_24 = 1, +}; + +enum ec_i2s_config { + EC_DAI_FMT_I2S = 0, + EC_DAI_FMT_RIGHT_J = 1, + EC_DAI_FMT_LEFT_J = 2, + EC_DAI_FMT_PCM_A = 3, + EC_DAI_FMT_PCM_B = 4, + EC_DAI_FMT_PCM_TDM = 5, +}; + +struct ec_param_codec_i2s { + /* + * enum ec_codec_i2s_subcmd + */ + uint8_t cmd; + union { + /* + * EC_CODEC_SET_SAMPLE_DEPTH + * Value should be one of ec_sample_depth_value. + */ + uint8_t depth; + + /* + * EC_CODEC_SET_GAIN + * Value should be 0~43 for both channels. + */ + struct ec_param_codec_i2s_set_gain { + uint8_t left; + uint8_t right; + } __packed gain; + + /* + * EC_CODEC_I2S_ENABLE + * 1 to enable, 0 to disable. + */ + uint8_t i2s_enable; + + /* + * EC_CODEC_I2S_SET_COFNIG + * Value should be one of ec_i2s_config. + */ + uint8_t i2s_config; + + /* + * EC_CODEC_I2S_SET_TDM_CONFIG + * Value should be one of ec_i2s_config. + */ + struct ec_param_codec_i2s_tdm { + /* + * 0 to 496 + */ + int16_t ch0_delay; + /* + * -1 to 496 + */ + int16_t ch1_delay; + uint8_t adjacent_to_ch0; + uint8_t adjacent_to_ch1; + } __packed tdm_param; + + /* + * EC_CODEC_I2S_SET_BCLK + */ + uint32_t bclk; + }; +} __packed; + +/* + * For subcommand EC_CODEC_GET_GAIN. + */ +struct ec_response_codec_gain { + uint8_t left; + uint8_t right; +} __packed; + /*****************************************************************************/ /* System commands */ -- 2.20.1.415.g653613c723-goog

2 6

[alsa-devel] fixing seq_midi: MIDI output buffer overrun
by Adam Goode 26 Dec '18

26 Dec '18

Hi, I am investigating why Chromium can't reliably send sysex messages on Linux. (See https://bugs.chromium.org/p/chromium/issues/detail?id=917708.) I've seen the message at https://alsa-devel.alsa-project.narkive.com/6w35hffF/sysex-overflow-when-us… Chromium does use seq and we do need sysex to be reliable since vendors such as Novation and Yamaha use Web MIDI to interface with hardware. The code in question predates git history (2005). It's event_process_midi() and dump_midi() in seq_midi.c. I see that dump_midi() returns either ENOMEM or EINVAL under certain conditions, but the only caller (event_process_midi()) throws these errors away and always returns 0 if everything but dump_midi() succeeds. A few questions: 1. Can I change dump_midi to block instead of return ENOMEM if I don't use SND_SEQ_NONBLOCK in snd_seq_open? 2. What should be returned in NONBLOCK mode? Probably EAGAIN? 3. The general blocking behavior of seq seems inconsistent. If I am using snd_seq_ev_set_direct, then I get EAGAIN even with a blocking open. Should I expect blocking from the pool, or only if using a queue? 4. Is there any obvious fix I am overlooking? Thanks, Adam

2 3

Re: [alsa-devel] [PATCH -next] ASoC: Intel: remove set but not used variable 'stage_type'
by kbuild test robot 25 Dec '18

25 Dec '18

Hi YueHaibing, Thank you for the patch! Perhaps something to improve: [auto build test WARNING on next-20181224] url: https://github.com/0day-ci/linux/commits/YueHaibing/ASoC-Intel-remove-set-b… config: x86_64-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): arch/x86/kvm/vmx/nested.c:58:6: warning: symbol 'init_vmcs_shadow_fields' was not declared. Should it be static? arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (110011 becomes 11) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100790 becomes 790) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100790 becomes 790) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a000a becomes a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100490 becomes 490) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100310 becomes 310) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80a08 becomes a08) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80688 becomes 688) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80608 becomes 608) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80108 becomes 108) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80388 becomes 388) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100390 becomes 390) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80208 becomes 208) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80288 becomes 288) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a000a becomes a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100010 becomes 10) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80488 becomes 488) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100710 becomes 710) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100090 becomes 90) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20402 becomes 402) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80508 becomes 508) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80b88 becomes b88) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100790 becomes 790) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100490 becomes 490) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100310 becomes 310) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20482 becomes 482) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100590 becomes 590) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100610 becomes 610) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100690 becomes 690) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120912 becomes 912) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100590 becomes 590) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20002 becomes 2) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20082 becomes 82) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20102 becomes 102) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20182 becomes 182) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20202 becomes 202) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20282 becomes 282) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20302 becomes 302) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (20382 becomes 382) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120012 becomes 12) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120092 becomes 92) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120112 becomes 112) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120192 becomes 192) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120212 becomes 212) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120292 becomes 292) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120312 becomes 312) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120392 becomes 392) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120412 becomes 412) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120492 becomes 492) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120512 becomes 512) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120692 becomes 692) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120712 becomes 712) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120792 becomes 792) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120812 becomes 812) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120892 becomes 892) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a019a becomes 19a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a021a becomes 21a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a029a becomes 29a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a031a becomes 31a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a039a becomes 39a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a041a becomes 41a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a049a becomes 49a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a051a becomes 51a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a059a becomes 59a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a061a becomes 61a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120a92 becomes a92) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a089a becomes 89a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a091a becomes 91a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a099a becomes 99a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a028a becomes 28a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a030a becomes 30a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a038a becomes 38a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a040a becomes 40a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80b08 becomes b08) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100190 becomes 190) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100210 becomes 210) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80708 becomes 708) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80788 becomes 788) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80808 becomes 808) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80888 becomes 888) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100410 becomes 410) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100510 becomes 510) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120592 becomes 592) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (120612 becomes 612) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a008a becomes 8a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a008a becomes 8a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (180018 becomes 18) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a010a becomes 10a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (a010a becomes 10a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80408 becomes 408) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (80c88 becomes c88) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (180118 becomes 118) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (180198 becomes 198) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100410 becomes 410) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (100510 becomes 510) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1a081a becomes 81a) arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1b009b becomes 9b) >> arch/x86/kvm/vmx/evmcs.h:80:30: warning: cast truncates bits from constant value (1b011b becomes 11b) arch/x86/kvm/vmx/evmcs.h:80:30: warning: too many warnings vim +80 arch/x86/kvm/vmx/evmcs.h 75edce8a arch/x86/kvm/vmx/evmcs.h Sean Christopherson 2018-12-03 76 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 77 static __always_inline int get_evmcs_offset(unsigned long field, 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 78 u16 *clean_field) 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 79 { 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 @80 unsigned int index = ROL16(field, 6); 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 81 const struct evmcs_field *evmcs_field; 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 82 75edce8a arch/x86/kvm/vmx/evmcs.h Sean Christopherson 2018-12-03 83 if (unlikely(index >= nr_evmcs_1_fields)) { 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 84 WARN_ONCE(1, "KVM: accessing unsupported EVMCS field %lx\n", 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 85 field); 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 86 return -ENOENT; 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 87 } 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 88 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 89 evmcs_field = &vmcs_field_to_evmcs_1[index]; 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 90 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 91 if (clean_field) 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 92 *clean_field = evmcs_field->clean_field; 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 93 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 94 return evmcs_field->offset; 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 95 } 773e8a04 arch/x86/kvm/vmx_evmcs.h Vitaly Kuznetsov 2018-03-20 96 :::::: The code at line 80 was first introduced by commit :::::: 773e8a0425c923bc02668a2d6534a5ef5a43cc69 x86/kvm: use Enlightened VMCS when running on Hyper-V :::::: TO: Vitaly Kuznetsov <vkuznets(a)redhat.com> :::::: CC: Radim Krčmář <rkrcmar(a)redhat.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Message from ALSA found in my kernel logs
by Philippe Lefèvre 24 Dec '18

24 Dec '18

Sorry, I forgot to join my file. Philippe

1 0

[alsa-devel] Message from ALSA found in my kernel logs
by Philippe Lefèvre 24 Dec '18

24 Dec '18

Hi, I found this message into my kernel logs this morning and it is said that I shoud report it to ALSA developers. Sorry, it is French. I'm going to try to translate it : 1800 Dec 24 08:22:20 phlsys pulseaudio[2291]: ALSA nous a réveillé pour écrire de nouvelles données à partir du périphérique, mais il n'y avait en fait rien à écrire ! 1801 Dec 24 08:22:20 phlsys pulseaudio[2291]: Il s'agit très probablement d'un bogue dans le pilote ALSA « snd_usb_audio ». Veuillez rapporter ce problème aux développeurs d'ALSA. 1802 Dec 24 08:22:20 phlsys pulseaudio[2291]: Nous avons été réveillés avec POLLOUT actif, cependant un snd_pcm_avail() ultérieur a retourné 0 ou une autre valeur < min_avail. 1800 ... ALSA wake us up to write datas from the dev but there was nothing to write ! 1801 ... This is probably a bug within the ALSA driver "snb_usb_audio". Please report this issue to the ALSA developers. 1802 ... We were waked up with an active POLLOUT however a previous snd_pcm_avail() returned 0 or an other value < min_avail. I'm running 3.16.0-7-amd64 #1 SMP Debian 3.16.59-1 (2018-10-03) x86_64 GNU/Linux Please find attached my hardawre list. While I'm at it, I also tried to register to www.*alsa*-project.org without success because I might be able to bring my computer skills to that project even if my skills regarding any sound server are very low. The box #wpCaptchaWord (@ http://www.alsa-project.org/main/index.php?title=Special:UserLogin&type=sig…) remains empty. (My browser (Firefox Dev) do not block any cookie neither from www.*alsa*-project.org nor accounts.google.com) Best regards ... and happy Christmas ;-) Philippe

1 0

[alsa-devel] [PATCH 0/6] ASoC: rt5651: Various fixes and quirks
by Hans de Goede 24 Dec '18

24 Dec '18

Hi All, Here is my last (big) round of (Intel) rt5651 support updates, with these patches combined with the updated, matching UCM profiles from here: https://github.com/jwrdegoede/alsa-lib/commits/master All 7 different devices I have work 100% OOTB, without requiring any manual config or mixer setup. After this there should be no more big changes. I will submit the matching alsa-lib UCM profile changes upstream once this set is in -next (and thus I can be sure the long-names will not change due to review). Note that unfortunately my previous round of cleanup patches removed 1 input mapping (of the 3 mappings in total it removed) which is actually necessary. So this series re-introduces that 1 mapping. I did not notice this before because UCM input profile switching does not work reliably for some reason, something which I still need to investigate. This time I manually veryfied that the correct / expected mixer settings where used while testing. Regards, Hans

4 23

[alsa-devel] [PATCH 1/3] mfd: cros_ec: Add commands to control codec
by Cheng-Yi Chiang 24 Dec '18

24 Dec '18

Add EC host commands to control codec on EC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- include/linux/mfd/cros_ec_commands.h | 94 ++++++++++++++++++++++++++++ 1 file changed, 94 insertions(+) diff --git a/include/linux/mfd/cros_ec_commands.h b/include/linux/mfd/cros_ec_commands.h index 07fe4ea0361d3..a3ac792f4915b 100644 --- a/include/linux/mfd/cros_ec_commands.h +++ b/include/linux/mfd/cros_ec_commands.h @@ -4077,6 +4077,100 @@ struct __ec_align1 ec_response_i2c_passthru_protect { uint8_t status; /* Status flags (0: unlocked, 1: locked) */ }; +/*****************************************************************************/ +/* Commands for I2S recording on audio codec. */ + +#define EC_CMD_CODEC_I2S 0x00BC + +enum ec_codec_i2s_subcmd { + EC_CODEC_SET_SAMPLE_DEPTH = 0x0, + EC_CODEC_SET_GAIN = 0x1, + EC_CODEC_GET_GAIN = 0x2, + EC_CODEC_I2S_ENABLE = 0x3, + EC_CODEC_I2S_SET_CONFIG = 0x4, + EC_CODEC_I2S_SET_TDM_CONFIG = 0x5, + EC_CODEC_I2S_SET_BCLK = 0x6, +}; + +enum ec_sample_depth_value { + EC_CODEC_SAMPLE_DEPTH_16 = 0, + EC_CODEC_SAMPLE_DEPTH_24 = 1, +}; + +enum ec_i2s_config { + EC_DAI_FMT_I2S = 0, + EC_DAI_FMT_RIGHT_J = 1, + EC_DAI_FMT_LEFT_J = 2, + EC_DAI_FMT_PCM_A = 3, + EC_DAI_FMT_PCM_B = 4, + EC_DAI_FMT_PCM_TDM = 5, +}; + +struct ec_param_codec_i2s { + /* + * enum ec_codec_i2s_subcmd + */ + uint8_t cmd; + union { + /* + * EC_CODEC_SET_SAMPLE_DEPTH + * Value should be one of ec_sample_depth_value. + */ + uint8_t depth; + + /* + * EC_CODEC_SET_GAIN + * Value should be 0~43 for both channels. + */ + struct ec_param_codec_i2s_set_gain { + uint8_t left; + uint8_t right; + } __packed gain; + + /* + * EC_CODEC_I2S_ENABLE + * 1 to enable, 0 to disable. + */ + uint8_t i2s_enable; + + /* + * EC_CODEC_I2S_SET_COFNIG + * Value should be one of ec_i2s_config. + */ + uint8_t i2s_config; + + /* + * EC_CODEC_I2S_SET_TDM_CONFIG + * Value should be one of ec_i2s_config. + */ + struct ec_param_codec_i2s_tdm { + /* + * 0 to 496 + */ + int16_t ch0_delay; + /* + * -1 to 496 + */ + int16_t ch1_delay; + uint8_t adjacent_to_ch0; + uint8_t adjacent_to_ch1; + } __packed tdm_param; + + /* + * EC_CODEC_I2S_SET_BCLK + */ + uint32_t bclk; + }; +} __packed; + +/* + * For subcommand EC_CODEC_GET_GAIN. + */ +struct ec_response_codec_gain { + uint8_t left; + uint8_t right; +} __packed; + /*****************************************************************************/ /* System commands */ -- 2.20.0.rc0.387.gc7a69e6b6c-goog

3 12

[alsa-devel] [PATCH v2] mfd: cros_ec: Add commands to control codec
by Cheng-Yi Chiang 22 Dec '18

22 Dec '18

Add EC host commands to control codec on EC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- V2 fixed the wrong ancestor issue so the patch can be applied cleanly on mfd tree. Codec driver that uses these commands will be sent separately. Thanks! include/linux/mfd/cros_ec_commands.h | 94 ++++++++++++++++++++++++++++ 1 file changed, 94 insertions(+) diff --git a/include/linux/mfd/cros_ec_commands.h b/include/linux/mfd/cros_ec_commands.h index 9a9631f0559e2..fc91082d4c357 100644 --- a/include/linux/mfd/cros_ec_commands.h +++ b/include/linux/mfd/cros_ec_commands.h @@ -2790,6 +2790,100 @@ struct ec_response_battery_vendor_param { uint32_t value; } __packed; +/*****************************************************************************/ +/* Commands for I2S recording on audio codec. */ + +#define EC_CMD_CODEC_I2S 0x00BC + +enum ec_codec_i2s_subcmd { + EC_CODEC_SET_SAMPLE_DEPTH = 0x0, + EC_CODEC_SET_GAIN = 0x1, + EC_CODEC_GET_GAIN = 0x2, + EC_CODEC_I2S_ENABLE = 0x3, + EC_CODEC_I2S_SET_CONFIG = 0x4, + EC_CODEC_I2S_SET_TDM_CONFIG = 0x5, + EC_CODEC_I2S_SET_BCLK = 0x6, +}; + +enum ec_sample_depth_value { + EC_CODEC_SAMPLE_DEPTH_16 = 0, + EC_CODEC_SAMPLE_DEPTH_24 = 1, +}; + +enum ec_i2s_config { + EC_DAI_FMT_I2S = 0, + EC_DAI_FMT_RIGHT_J = 1, + EC_DAI_FMT_LEFT_J = 2, + EC_DAI_FMT_PCM_A = 3, + EC_DAI_FMT_PCM_B = 4, + EC_DAI_FMT_PCM_TDM = 5, +}; + +struct ec_param_codec_i2s { + /* + * enum ec_codec_i2s_subcmd + */ + uint8_t cmd; + union { + /* + * EC_CODEC_SET_SAMPLE_DEPTH + * Value should be one of ec_sample_depth_value. + */ + uint8_t depth; + + /* + * EC_CODEC_SET_GAIN + * Value should be 0~43 for both channels. + */ + struct ec_param_codec_i2s_set_gain { + uint8_t left; + uint8_t right; + } __packed gain; + + /* + * EC_CODEC_I2S_ENABLE + * 1 to enable, 0 to disable. + */ + uint8_t i2s_enable; + + /* + * EC_CODEC_I2S_SET_COFNIG + * Value should be one of ec_i2s_config. + */ + uint8_t i2s_config; + + /* + * EC_CODEC_I2S_SET_TDM_CONFIG + * Value should be one of ec_i2s_config. + */ + struct ec_param_codec_i2s_tdm { + /* + * 0 to 496 + */ + int16_t ch0_delay; + /* + * -1 to 496 + */ + int16_t ch1_delay; + uint8_t adjacent_to_ch0; + uint8_t adjacent_to_ch1; + } __packed tdm_param; + + /* + * EC_CODEC_I2S_SET_BCLK + */ + uint32_t bclk; + }; +} __packed; + +/* + * For subcommand EC_CODEC_GET_GAIN. + */ +struct ec_response_codec_gain { + uint8_t left; + uint8_t right; +} __packed; + /*****************************************************************************/ /* System commands */ -- 2.20.0.rc1.387.gf8505762e3-goog

3 5

Re: [alsa-devel] [PATCH v1 08/11] ASoC: Intel: Make PCI dependency explicit
by Randy Dunlap 22 Dec '18

22 Dec '18

On 12/21/18 4:14 PM, Sinan Kaya wrote: > IOSF_MBI driver depends on CONFIG_PCI set but this is not specified > anywhere. wrong patch description...? > Signed-off-by: Sinan Kaya <okaya(a)kernel.org> > --- > sound/soc/intel/Kconfig | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig > index 2fd1b61e8331..b0764b2fe001 100644 > --- a/sound/soc/intel/Kconfig > +++ b/sound/soc/intel/Kconfig > @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI > config SND_SST_ATOM_HIFI2_PLATFORM_ACPI > tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" > default ACPI > - depends on X86 && ACPI > + depends on X86 && ACPI && PCI > select SND_SST_IPC_ACPI > select SND_SST_ATOM_HIFI2_PLATFORM > select SND_SOC_ACPI_INTEL_MATCH > -- ~Randy

1 0

[alsa-devel] [PATCH] ASoC: Intel: fix unmet dependencies in Kconfig
by Pierre-Louis Bossart 22 Dec '18

22 Dec '18

Remove warnings by selecting SND_HDA_INTEL_DSP_DETECTION_XYZ only when SND_HDA_INTEL is part of the config. Fixes: c337104b1a16 ('ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected') Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..555cdb1d87d4 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -188,12 +188,12 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC - select SND_HDA_INTEL_DSP_DETECTION_SKL if SND_SOC_INTEL_SKL - select SND_HDA_INTEL_DSP_DETECTION_APL if SND_SOC_INTEL_APL - select SND_HDA_INTEL_DSP_DETECTION_KBL if SND_SOC_INTEL_KBL - select SND_HDA_INTEL_DSP_DETECTION_GLK if SND_SOC_INTEL_GLK - select SND_HDA_INTEL_DSP_DETECTION_CNL if SND_SOC_INTEL_CNL - select SND_HDA_INTEL_DSP_DETECTION_CFL if SND_SOC_INTEL_CFL + select SND_HDA_INTEL_DSP_DETECTION_SKL if SND_SOC_INTEL_SKL && SND_HDA_INTEL + select SND_HDA_INTEL_DSP_DETECTION_APL if SND_SOC_INTEL_APL && SND_HDA_INTEL + select SND_HDA_INTEL_DSP_DETECTION_KBL if SND_SOC_INTEL_KBL && SND_HDA_INTEL + select SND_HDA_INTEL_DSP_DETECTION_GLK if SND_SOC_INTEL_GLK && SND_HDA_INTEL + select SND_HDA_INTEL_DSP_DETECTION_CNL if SND_SOC_INTEL_CNL && SND_HDA_INTEL + select SND_HDA_INTEL_DSP_DETECTION_CFL if SND_SOC_INTEL_CFL && SND_HDA_INTEL select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ -- 2.17.1

2 1

Re: [alsa-devel] linux-next: Tree for Dec 21 (SND_HDA_INTEL_DSP_DETECTION)
by Randy Dunlap 21 Dec '18

21 Dec '18

On 12/21/18 12:32 AM, Stephen Rothwell wrote: > Hi all, > > News: there will be no linux-next release until Jan 2. Have a good break. > > Changes since 20181220: > on x86_64: A new group of Kconfig "select"s is causing kconfig warnings: WARNING: unmet direct dependencies detected for SND_HDA_INTEL_DSP_DETECTION_CFL Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_HDA [=n] && SND_HDA_INTEL [=n] Selected by [y]: - SND_SOC_INTEL_SKYLAKE_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_SST_TOPLEVEL [=y] && SND_SOC_INTEL_SKYLAKE_FAMILY [=y] && SND_SOC_INTEL_CFL [=y] WARNING: unmet direct dependencies detected for SND_HDA_INTEL_DSP_DETECTION_APL Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_HDA [=n] && SND_HDA_INTEL [=n] Selected by [y]: - SND_SOC_INTEL_SKYLAKE_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_SST_TOPLEVEL [=y] && SND_SOC_INTEL_SKYLAKE_FAMILY [=y] && SND_SOC_INTEL_APL [=y] WARNING: unmet direct dependencies detected for SND_HDA_INTEL_DSP_DETECTION_KBL Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_HDA [=n] && SND_HDA_INTEL [=n] Selected by [y]: - SND_SOC_INTEL_SKYLAKE_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_SST_TOPLEVEL [=y] && SND_SOC_INTEL_SKYLAKE_FAMILY [=y] && SND_SOC_INTEL_KBL [=y] WARNING: unmet direct dependencies detected for SND_HDA_INTEL_DSP_DETECTION_CNL Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_HDA [=n] && SND_HDA_INTEL [=n] Selected by [y]: - SND_SOC_INTEL_SKYLAKE_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_SST_TOPLEVEL [=y] && SND_SOC_INTEL_SKYLAKE_FAMILY [=y] && SND_SOC_INTEL_CNL [=y] WARNING: unmet direct dependencies detected for SND_HDA_INTEL_DSP_DETECTION_GLK Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_HDA [=n] && SND_HDA_INTEL [=n] Selected by [y]: - SND_SOC_INTEL_SKYLAKE_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_SST_TOPLEVEL [=y] && SND_SOC_INTEL_SKYLAKE_FAMILY [=y] && SND_SOC_INTEL_GLK [=y] WARNING: unmet direct dependencies detected for SND_HDA_INTEL_DSP_DETECTION_SKL Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_HDA [=n] && SND_HDA_INTEL [=n] Selected by [y]: - SND_SOC_INTEL_SKYLAKE_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_SST_TOPLEVEL [=y] && SND_SOC_INTEL_SKYLAKE_FAMILY [=y] && SND_SOC_INTEL_SKL [=y] when SND_HDA=n and SND_HDA_INTEL=n. Full randconfig file is attached. -- ~Randy

2 1

[alsa-devel] [PATCH v4 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 21 Dec '18

21 Dec '18

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- include/sound/cs35l36.h | 56 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36.c | 1987 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 446 ++++++++++ 5 files changed, 2496 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..5e30ba5 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,56 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H +#define __CS35L36_H + +/* INT/GPIO pin configuration */ +struct cs35l36_irq_cfg { + bool is_present; + int irq_drv_sel; + int irq_pol; + int irq_gpio_sel; +}; + +struct cs35l36_vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool sclk_frc; + bool lrclk_frc; + bool multi_amp_mode; + bool dcm_mode; + int ldm_mode_sel; + bool amp_gain_zc; + bool amp_pcm_inv; + bool pdm_ldm_exit; + bool pdm_ldm_enter; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + struct cs35l36_vpbr_cfg vpbr_config; + struct cs35l36_irq_cfg irq_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f18..77cb861 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -465,6 +466,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c..7d8a372 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,6 +45,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -310,6 +311,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..6f0b831 --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1987 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int chip_version; + int rev_id; + struct gpio_desc *reset_gpio; +}; + +struct cs35l36_pll_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +struct reg_default cs35l36_reg[] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_TST_FS_MON0: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_TST_FS_MON0: + return true; + default: + return false; + } +} + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static int cs35l36_ldm_sel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = cs35l36->pdata.ldm_mode_sel; + + return 0; +} + +static int cs35l36_ldm_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int val = (ucontrol->value.integer.value[0]) ? CS35L36_NG_AMP_EN_MASK : + 0; + + cs35l36->pdata.ldm_mode_sel = val; + + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, val); + + return 0; +} + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("Analog PCM Volume", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), + SOC_SINGLE("Amp Gain Zero-Cross", CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Enter Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Exit Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_SHIFT, 1, 0), + SOC_SINGLE_BOOL_EXT("LDM Select", 0, cs35l36_ldm_sel_get, + cs35l36_ldm_sel_put), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + + if (WARN_ON_ONCE(reg & CS35L36_PLL_UNLOCK_MASK)) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux = + SOC_DAPM_ENUM("Input Mux", chansel_enum); + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const struct snd_kcontrol_new boost_ctrl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, &cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH_E("BOOST Enable", SND_SOC_NOPM, 0, 0, &boost_ctrl, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Enable", "Switch", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Enable"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + clk_frc = 1; + break; + case SND_SOC_DAIFMT_GATED: + clk_frc = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, clk_frc << + CS35L36_SCLK_FRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, clk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + unsigned int asp_width, global_fs = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1, fs2; + + if (freq > CS35L36_FS_NOM_6MHZ) { + fs1 = CS35L36_FS1_DEFAULT_VAL; + fs2 = CS35L36_FS2_DEFAULT_VAL; + } else { + fs1 = 3 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + fs2 = 5 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + } + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK | CS35L36_FS2_WINDOW_MASK, + fs1 | (fs2 << CS35L36_FS2_WINDOW_SHIFT)); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static const struct cs35l36_pll_config *cs35l36_get_clk_config( + struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) + return &cs35l36_pll_sysclk[i]; + } + + return NULL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + const struct cs35l36_pll_config *clk_cfg; + int prev_clksrc; + bool pdm_switch; + + prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + clk_cfg = cs35l36_get_clk_config(cs35l36, freq); + if (clk_cfg == NULL) { + dev_err(component->dev, "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + clk_cfg->clk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, + cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + clk_cfg->fll_igain); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc != CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } else { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc == CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if ((cs35l36->rev_id == CS35L36_REV_A0) && cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) { + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + if (ret < 0) { + dev_err(cs35l36->dev, + "Boost inductor config failed(%d)\n", ret); + return ret; + } + } + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .precious_reg = cs35l36_precious_reg, + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + int ret = IRQ_NONE; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, status, + ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, masks, + ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, CS35L36_TEMP_WARN); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, CS35L36_TEMP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, CS35L36_BST_OVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct cs35l36_irq_cfg *irq_gpio_config = &pdata->irq_config; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *irq_gpio, *vpbr_node; + struct irq_data *irq_d; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + /* INT/GPIO Pin Config */ + irq_gpio = of_get_child_by_name(np, "cirrus,irq-config"); + irq_gpio_config->is_present = irq_gpio ? true : false; + if (irq_gpio_config->is_present) { + if (of_property_read_u32(irq_gpio, "cirrus,irq-drive-select", + &val) >= 0) + irq_gpio_config->irq_drv_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-gpio-select", + &val) >= 0) + irq_gpio_config->irq_gpio_sel = val; + + irq_d = irq_get_irq_data(i2c_client->irq); + if (!irq_d) { + dev_err(&i2c_client->dev, "Invalid IRQ: %d\n", + i2c_client->irq); + return -EINVAL; + } + + irq_gpio_config->irq_pol = irqd_get_trigger_type(irq_d); + } + of_node_put(irq_gpio); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id != CS35L36_REV_B0) + return 0; + + /* + * Magic code for internal PAC + */ + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + + return 0; +} + +static void cs35l36_apply_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); +} + +static int cs35l36_irq_gpio_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct cs35l36_irq_cfg *irq_config = &pdata->irq_config; + int irq_polarity; + + if (!irq_config->is_present) { + dev_err(cs35l36->dev, "IRQ config not present\n"); + return -EINVAL; + } + + switch (irq_config->irq_pol) { + case IRQF_TRIGGER_FALLING: + case IRQF_TRIGGER_LOW: + irq_polarity = 0; + break; + case IRQF_TRIGGER_RISING: + case IRQF_TRIGGER_HIGH: + irq_polarity = 1; + break; + default: + dev_err(cs35l36->dev, "Invalid IRQ polarity: %d\n", + irq_config->irq_pol); + return -EINVAL; + } + + /* setup the Interrupt Pin (INT | GPIO) */ + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, 1); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + irq_config->irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + irq_polarity << CS35L36_INT_POL_SEL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + irq_config->irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + return 0; +} + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + /* Errata Writes */ + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + /* PAC Config */ + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + int ret, irq_pol, i; + u32 reg_id, reg_revid, l37_id_reg; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l36->pdata = *pdata; + } else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) { + dev_info(dev, "Reset line busy, assuming shared reset\n"); + } else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err_disable_regs; + } + } + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", reg_id, + CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_apply_vpbr_config(cs35l36); + + ret = cs35l36_irq_gpio_config(cs35l36); + if (ret != 0) { + dev_err(dev, "IRQ config invalid\n"); + goto err; + } + + irq_pol = pdata->irq_config.irq_pol; + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | irq_pol, "cs35l36", + cs35l36); + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + +err_disable_regs: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + snd_soc_unregister_component(&client->dev); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..f6e38c6 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,446 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_SHIFT 10 +#define CS35L36_PDM_LDM_ENTER_SHIFT 3 +#define CS35L36_PDM_LDM_EXIT_SHIFT 4 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_FS1_DEFAULT_VAL 16 +#define CS35L36_FS2_DEFAULT_VAL 36 +#define CS35L36_FS_NOM_6MHZ 6000000 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

3 5

[alsa-devel] Applied "ASoC: Intel: Baytrail: remove unneeded variable" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: Intel: Baytrail: remove unneeded variable has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1c66634069189b184ac6ea8ccbdbfefdb01ded07 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Sun, 16 Dec 2018 16:49:06 -0600 Subject: [PATCH] ASoC: Intel: Baytrail: remove unneeded variable Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 260447da32b8..2cd8f9668b50 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -278,7 +278,6 @@ static int sst_byt_process_notification(struct sst_byt *byt, struct sst_byt_stream *stream; u64 header; u8 msg_id, stream_id; - int handled = 1; header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); msg_id = sst_byt_header_msg_id(header); @@ -298,7 +297,7 @@ static int sst_byt_process_notification(struct sst_byt *byt, break; } - return handled; + return 1; } static irqreturn_t sst_byt_irq_thread(int irq, void *context) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a4ad6c42a7368b6e9ff5732c390b319a1b259134 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:42 +0900 Subject: [PATCH] ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() We will get DAI ID from "reg" property if it has on DT, otherwise get it by counting port/endpoint. But in below case, we need to get DAI ID = 0 via port reg = <0>, but current implementation returns ID = 1, because it can't judge ID = 0 was from "non reg" or "reg = <0>". Thus, it will count port/endpoint number as "non reg" case. of_graph_parse_endpoint() implementation itself is not a problem, but because asoc_simple_card_get_dai_id() need to count port/endpoint number when "non reg" case, it need to know ID = 0 was from "non reg" or "reg = <0>". This patch fix this issue. port { reg = <0>; xxxx: endpoint@0 { }; => xxxx: endpoint@1 { }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b807a47515eb..336895f7fd1e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -283,12 +283,20 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) /* use endpoint/port reg if exist */ ret = of_graph_parse_endpoint(ep, &info); if (ret == 0) { - if (info.id) + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_get_property(ep, "reg", NULL)) return info.id; - if (info.port) + + node = of_get_parent(ep); + of_node_put(node); + if (of_get_property(node, "reg", NULL)) return info.port; } - node = of_graph_get_port_parent(ep); /* -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add asoc_graph_card_get_conversion()" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: audio-graph-card: add asoc_graph_card_get_conversion() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 946fbcc005d4b7f9c9232d49cf03ccec3289b437 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:48 +0900 Subject: [PATCH] ASoC: audio-graph-card: add asoc_graph_card_get_conversion() audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_graph_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0d6144560a1e..c3e80bc27e80 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -169,6 +169,22 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_graph_card_get_conversion(struct device *dev, + struct device_node *ep, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); + struct device_node *node = of_graph_get_port_parent(ep); + + asoc_simple_card_parse_convert(dev, top, NULL, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, ports, NULL, adata); + asoc_simple_card_parse_convert(dev, port, NULL, adata); + asoc_simple_card_parse_convert(dev, ep, NULL, adata); +} + static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, struct device_node *cpu_ep, struct device_node *codec_ep, @@ -194,11 +210,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - asoc_simple_card_parse_convert(dev, top, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ports, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, port, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ep, NULL, &dai_props->adata); + asoc_graph_card_get_conversion(dev, ep, &dai_props->adata); of_node_put(ports); of_node_put(port); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 09ceab9439b87825872fbde453be3fd8a376e38a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:27 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/audio-graph-scu-card.txt | 123 ------------------ 1 file changed, 123 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt deleted file mode 100644 index 62d42768a00b..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ /dev/null @@ -1,123 +0,0 @@ -Audio-Graph-SCU-Card: - -Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM". - -It is based on common bindings for device graphs. -see ${LINUX}/Documentation/devicetree/bindings/graph.txt - -Basically, Audio-Graph-SCU-Card property is same as -Simple-Card / Simple-SCU-Card / Audio-Graph-Card. -see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt - -Below are same as Simple-Card / Audio-Graph-Card. - -- label -- dai-format -- frame-master -- bitclock-master -- bitclock-inversion -- frame-inversion -- dai-tdm-slot-num -- dai-tdm-slot-width -- clocks / system-clock-frequency - -Below are same as Simple-SCU-Card. - -- convert-rate -- convert-channels -- prefix -- routing - -Required properties: - -- compatible : "audio-graph-scu-card"; -- dais : list of CPU DAI port{s} - -Example 1. Sampling Rate Conversion - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - prefix = "codec"; - routing = "codec Playback", "DAI0 Playback", - "DAI0 Capture", "codec Capture"; - convert-rate = <48000>; - - dais = <&cpu_port>; - }; - - audio-codec { - ... - - port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - }; - -Example 2. 2 CPU 1 Codec (Mixing) - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - routing = "codec Playback", "DAI0 Playback", - "codec Playback", "DAI1 Playback", - "DAI0 Capture", "codec Capture"; - - dais = <&cpu_port0 - &cpu_port1>; - }; - - audio-codec { - ... - - audio-graph-card,prefix = "codec"; - audio-graph-card,convert-rate = <48000>; - port { - codec_endpoint0: endpoint { - remote-endpoint = <&cpu_endpoint0>; - }; - codec_endpoint1: endpoint { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - dai-controller { - ... - ports { - cpu_port0: port { - cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint0>; - - dai-format = "left_j"; - ... - }; - }; - cpu_port1: port { - cpu_endpoint1: endpoint { - remote-endpoint = <&codec_endpoint1>; - - dai-format = "left_j"; - ... - }; - }; - }; - }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0d6ed5cd2b7e53e086290358f36708813e4c0a40 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:32 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/Kconfig | 9 - sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-scu-card.c | 501 ----------------------- 3 files changed, 512 deletions(-) delete mode 100644 sound/soc/generic/audio-graph-scu-card.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 92c2cf06f40a..59190f42fc08 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -24,12 +24,3 @@ config SND_AUDIO_GRAPH_CARD This option enables generic simple sound card support with OF-graph DT bindings. It also support DPCM of multi CPU single Codec ststem. - -config SND_AUDIO_GRAPH_SCU_CARD - tristate "ASoC Audio Graph SCU sound card support" - depends on OF - select SND_SIMPLE_CARD_UTILS - help - This option enables generic simple SCU sound card support - with OF-graph DT bindings. - It supports DPCM of multi CPU single Codec ststem. diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9dec293a4c4d..9fbfdd524b24 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -3,10 +3,8 @@ snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o snd-soc-simple-scu-card-objs := simple-scu-card.o snd-soc-audio-graph-card-objs := audio-graph-card.o -snd-soc-audio-graph-scu-card-objs := audio-graph-scu-card.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o obj-$(CONFIG_SND_SIMPLE_SCU_CARD) += snd-soc-simple-scu-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_CARD) += snd-soc-audio-graph-card.o -obj-$(CONFIG_SND_AUDIO_GRAPH_SCU_CARD) += snd-soc-audio-graph-scu-card.o diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c deleted file mode 100644 index e1b192ea147b..000000000000 --- a/sound/soc/generic/audio-graph-scu-card.c +++ /dev/null @@ -1,501 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// ASoC audio graph SCU sound card support -// -// Copyright (C) 2017 Renesas Solutions Corp. -// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> -// -// based on -// ${LINUX}/sound/soc/generic/simple-scu-card.c -// ${LINUX}/sound/soc/generic/audio-graph-card.c - -#include <linux/clk.h> -#include <linux/device.h> -#include <linux/gpio.h> -#include <linux/module.h> -#include <linux/of.h> -#include <linux/of_device.h> -#include <linux/of_gpio.h> -#include <linux/of_graph.h> -#include <linux/platform_device.h> -#include <linux/string.h> -#include <sound/jack.h> -#include <sound/simple_card_utils.h> - -struct graph_card_data { - struct snd_soc_card snd_card; - struct graph_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; - struct snd_soc_dai_link_component platform; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - } *dai_props; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; -}; - -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - -#define PREFIX "audio-graph-card," - -static int asoc_graph_card_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - -static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - -static const struct snd_soc_ops asoc_graph_card_ops = { - .startup = asoc_graph_card_startup, - .shutdown = asoc_graph_card_shutdown, -}; - -static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - -static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - /* overwrite by top level adata if exist */ - asoc_simple_card_convert_fixup(&priv->adata, params); - - return 0; -} - -static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, - struct device_node *codec_ep, - struct graph_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_fe) -{ - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *ep = is_fe ? cpu_ep : codec_ep; - struct device_node *node = of_graph_get_port_parent(ep); - struct asoc_simple_dai *dai; - int ret; - - if (is_fe) { - struct snd_soc_dai_link_component *codecs; - - /* BE is dummy */ - codecs = dai_link->codecs; - codecs->of_node = NULL; - codecs->dai_name = "snd-soc-dummy-dai"; - codecs->name = "snd-soc-dummy"; - - /* FE settings */ - dai_link->dynamic = 1; - dai_link->dpcm_merged_format = 1; - - dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; - - ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); - if (ret) - return ret; - - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); - if (ret < 0) - return ret; - - /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); - } else { - struct snd_soc_codec_conf *cconf; - - /* FE is dummy */ - dai_link->cpu_of_node = NULL; - dai_link->cpu_dai_name = "snd-soc-dummy-dai"; - dai_link->cpu_name = "snd-soc-dummy"; - - /* BE settings */ - dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; - - dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; - - cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; - - ret = asoc_simple_card_parse_graph_codec(ep, dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - dai_link->codecs->dai_name); - if (ret < 0) - return ret; - - /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, cconf, - dai_link->codecs->of_node, - "prefix"); - /* check "prefix" from each node if top doesn't have */ - if (!cconf->of_node) - snd_soc_of_parse_node_prefix(node, cconf, - dai_link->codecs->of_node, - PREFIX "prefix"); - } - - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - - ret = asoc_simple_card_of_parse_tdm(ep, dai); - if (ret) - return ret; - - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); - if (ret < 0) - return ret; - - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_graph_card_ops; - dai_link->init = asoc_graph_card_dai_init; - - return 0; -} - -static int asoc_graph_card_parse_of(struct graph_card_data *priv) -{ - struct of_phandle_iterator it; - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - int dai_idx, link_idx, conf_idx, ret; - int rc, codec; - - if (!node) - return -EINVAL; - - /* - * we need to consider "widgets", "mclk-fs" around here - * see simple-card - */ - - ret = asoc_simple_card_of_parse_routing(card, NULL); - if (ret < 0) - return ret; - - asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); - - /* - * it supports multi CPU, single CODEC only here - * see asoc_graph_get_dais_count - */ - - link_idx = 0; - dai_idx = 0; - conf_idx = 0; - codec_port_old = NULL; - for (codec = 0; codec < 2; codec++) { - /* - * To listup valid sounds continuously, - * detect all CPU-dummy first, and - * detect all dummy-Codec second - */ - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(cpu_port); - of_node_put(codec_port); - it.node = NULL; - - if (codec) { - if (codec_port_old == codec_port) - continue; - - codec_port_old = codec_port; - } - - ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, - priv, &dai_idx, - link_idx++, &conf_idx, - !codec); - if (ret < 0) - goto parse_of_err; - } - } - - ret = asoc_simple_card_parse_card_name(card, NULL); - if (ret) - goto parse_of_err; - - if ((card->num_links != link_idx) || - (card->num_configs != conf_idx)) { - dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", - card->num_links, link_idx, card->num_configs, conf_idx); - ret = -EINVAL; - goto parse_of_err; - } - - ret = 0; - -parse_of_err: - return ret; -} - -static void asoc_graph_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) -{ - struct of_phandle_iterator it; - struct device_node *node = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ep; - struct device_node *codec_ep; - struct device_node *codec_port; - struct device_node *codec_port_old; - struct device_node *codec_port_old2; - int rc; - - /* - * link_num : number of links. - * CPU-Codec / CPU-dummy / dummy-Codec - * dais_num : number of DAIs - * ccnf_num : number of codec_conf - * same number for dummy-Codec - * - * ex1) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 7 - * CPU2 -/ ccnf : 1 - * CPU3 --- Codec2 - * - * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec - * => 7 DAIs = 4xCPU + 3xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex2) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 6 - * CPU2 -/ ccnf : 1 - * CPU3 -/ - * - * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex3) - * CPU0 --- Codec0 link : 6 - * CPU1 -/ dais : 6 - * CPU2 --- Codec1 ccnf : 2 - * CPU3 -/ - * - * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 2 ccnf = 2xdummy-Codec - */ - codec_port_old = NULL; - codec_port_old2 = NULL; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - - (*link_num)++; - (*dais_num)++; - - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } - - codec_port_old2 = codec_port_old; - continue; - } - - (*dais_num)++; - codec_port_old = codec_port; - } -} - -static int asoc_graph_card_probe(struct platform_device *pdev) -{ - struct graph_card_data *priv; - struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; - struct asoc_simple_dai *dais; - struct device *dev = &pdev->dev; - struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; - int lnum = 0, dnum = 0, cnum = 0; - int ret, i; - - /* Allocate the private data and the DAI link array */ - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) - return -EINVAL; - - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < lnum; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; - } - - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - /* Init snd_soc_card */ - card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = priv->dai_link; - card->num_links = lnum; - card->codec_conf = cconf; - card->num_configs = cnum; - - ret = asoc_graph_card_parse_of(priv); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - goto err; - } - - snd_soc_card_set_drvdata(card, priv); - - ret = devm_snd_soc_register_card(dev, card); - if (ret < 0) - goto err; - - return 0; -err: - asoc_simple_card_clean_reference(card); - - return ret; -} - -static int asoc_graph_card_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - return asoc_simple_card_clean_reference(card); -} - -static const struct of_device_id asoc_graph_of_match[] = { - { .compatible = "audio-graph-scu-card", }, - {}, -}; -MODULE_DEVICE_TABLE(of, asoc_graph_of_match); - -static struct platform_driver asoc_graph_card = { - .driver = { - .name = "asoc-audio-graph-scu-card", - .pm = &snd_soc_pm_ops, - .of_match_table = asoc_graph_of_match, - }, - .probe = asoc_graph_card_probe, - .remove = asoc_graph_card_remove, -}; -module_platform_driver(asoc_graph_card); - -MODULE_ALIAS("platform:asoc-audio-graph-scu-card"); -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("ASoC Audio Graph SCU Sound Card"); -MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 2.20.1

1 0

[alsa-devel] Applied "dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 380e39f8c1d1f88b436e4bf0cf14f2079f6d1d57 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:27 +0530 Subject: [PATCH] dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings Added documentation for audio formatter IP core DT bindings. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/xlnx,audio-formatter.txt | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt new file mode 100644 index 000000000000..cbc93c8f4963 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt @@ -0,0 +1,29 @@ +Device-Tree bindings for Xilinx PL audio formatter + +The IP core supports DMA, data formatting(AES<->PCM conversion) +of audio samples. + +Required properties: + - compatible: "xlnx,audio-formatter-1.0" + - interrupt-names: Names specified to list of interrupts in same + order mentioned under "interrupts". + List of supported interrupt names are: + "irq_mm2s" : interrupt from MM2S block + "irq_s2mm" : interrupt from S2MM block + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - reg: Base address and size of the IP core instance. + - clock-names: List of input clocks. + Required elements: "s_axi_lite_aclk", "aud_mclk" + - clocks: Input clock specifier. Refer to common clock bindings. + +Example: + audio_ss_0_audio_formatter_0: audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + reg = <0x0 0x80010000 0x0 0x1000>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: xlnx: enable audio formatter driver build" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: xlnx: enable audio formatter driver build has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8c2249fb84054f83b528171091a4b436ebb3e058 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:29 +0530 Subject: [PATCH] ASoC: xlnx: enable audio formatter driver build Enable audio formatter driver build. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/xilinx/Kconfig | 7 +++++++ sound/soc/xilinx/Makefile | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 723a583a8d57..ac48d6a00c36 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -6,3 +6,10 @@ config SND_SOC_XILINX_I2S mode, IP receives audio in AES format, extracts PCM and sends PCM data. In receiver mode, IP receives PCM audio and encapsulates PCM in AES format and sends AES data. + +config SND_SOC_XILINX_AUDIO_FORMATTER + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index 6c1209b9ee75..432693b1cc79 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,2 +1,4 @@ snd-soc-xlnx-i2s-objs := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o +snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o +obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: audio-graph-card: add 1 CPU : 1 Codec support again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c8b59d77c445dd7d1f3e1620bf030a7cc243ac28 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:54 +0900 Subject: [PATCH] ASoC: audio-graph-card: add 1 CPU : 1 Codec support again audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 44 ++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c3e80bc27e80..638333cdac66 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -408,6 +408,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep = NULL; struct device_node *codec_port = NULL; struct device_node *codec_port_old = NULL; + struct asoc_simple_card_data adata; int rc, ret; int link_idx, dai_idx, conf_idx; int cpu; @@ -453,7 +454,13 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); - if (of_get_child_count(codec_port) > 1) { + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); + + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || + adata.convert_channels) { /* * for DPCM sound */ @@ -495,7 +502,7 @@ static void asoc_graph_get_dais_count(struct device *dev, struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - struct device_node *codec_port_old2; + struct asoc_simple_card_data adata; int rc; /* @@ -534,9 +541,17 @@ static void asoc_graph_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ codec_port_old = NULL; - codec_port_old2 = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = NULL; @@ -554,17 +569,22 @@ static void asoc_graph_get_dais_count(struct device *dev, (*link_num)++; (*dais_num)++; - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); - codec_port_old2 = codec_port_old; - continue; - } + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels) { - (*dais_num)++; + if (codec_port_old == codec_port) + continue; + + (*link_num)++; + (*ccnf_num)++; + (*dais_num)++; + } else { + (*dais_num)++; + } codec_port_old = codec_port; } } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: add asoc_simple_card_get_conversion()" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: simple-card: add asoc_simple_card_get_conversion() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f0f48626e9130a43e690c8cb53ca75a4a825223b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:42 +0900 Subject: [PATCH] ASoC: simple-card: add asoc_simple_card_get_conversion() simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_simple_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001cf9cd1..52048069b25a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,6 +165,21 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_simple_card_get_conversion(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *node = of_get_parent(np); + + asoc_simple_card_parse_convert(dev, top, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, NULL, adata); + asoc_simple_card_parse_convert(dev, np, NULL, adata); + + of_node_put(node); +} + static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, struct device_node *node, struct device_node *np, @@ -260,9 +275,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, "prefix"); } - asoc_simple_card_parse_convert(dev, top, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, prefix, &dai_props->adata); - asoc_simple_card_parse_convert(dev, np, NULL, &dai_props->adata); + asoc_simple_card_get_conversion(dev, np, &dai_props->adata); ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 21 Dec '18

21 Dec '18

The patch ASoC: simple-card: add 1 CPU : 1 Codec support again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 68c8cc91f9d7387963083e1a794b888908416442 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:47 +0900 Subject: [PATCH] ASoC: simple-card: add 1 CPU : 1 Codec support again simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 52048069b25a..b15651409c7f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -453,6 +453,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *node; struct device_node *np; struct device_node *codec; + struct asoc_simple_card_data adata; bool is_fe; int ret, loop; int dai_idx, link_idx, conf_idx; @@ -480,8 +481,13 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + /* DPCM */ - if (of_get_child_count(node) > 2) { + if (of_get_child_count(node) > 2 || + adata.convert_rate || adata.convert_channels) { for_each_child_of_node(node, np) { codec = of_get_child_by_name(node, loop ? "codec" : @@ -495,14 +501,16 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) top, node, np, codec, priv, &dai_idx, link_idx++, &conf_idx, is_fe, !loop); + if (ret < 0) + return ret; } } else { ret = asoc_simple_card_dai_link_of( top, node, priv, &dai_idx, link_idx++, !loop); + if (ret < 0) + return ret; } - if (ret < 0) - return ret; node = of_get_next_child(top, node); } while (loop && node); @@ -523,6 +531,8 @@ static void asoc_simple_card_get_dais_count(struct device *dev, { struct device_node *top = dev->of_node; struct device_node *node; + struct device_node *np; + struct asoc_simple_card_data adata; int loop; int num; @@ -562,6 +572,15 @@ static void asoc_simple_card_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ if (!top) { (*link_num) = 1; @@ -578,9 +597,14 @@ static void asoc_simple_card_get_dais_count(struct device *dev, } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + num = of_get_child_count(node); (*dais_num) += num; - if (num > 2) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { (*link_num) += num; (*ccnf_num)++; } else { -- 2.20.1

1 0

Re: [alsa-devel] [PATCH v3 09/14] ASoC: SOF: Add firmware, loader support
by Bard liao 21 Dec '18

21 Dec '18

> Date: Wed, 12 Dec 2018 12:23:33 +0100 > From: Takashi Iwai <tiwai(a)suse.de> > To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Cc: alsa-devel(a)alsa-project.org, andriy.shevchenko(a)intel.com, Daniel > Baluta <daniel.baluta(a)gmail.com>, liam.r.girdwood(a)linux.intel.com, > vkoul(a)kernel.org, broonie(a)kernel.org, Alan Cox <alan(a)linux.intel.com>, > sound-open-firmware(a)alsa-project.org > Subject: Re: [alsa-devel] [PATCH v3 09/14] ASoC: SOF: Add firmware > loader support > Message-ID: <s5htvjjj8ei.wl-tiwai(a)suse.de> > Content-Type: text/plain; charset=US-ASCII > > On Tue, 11 Dec 2018 22:23:13 +0100, > Pierre-Louis Bossart wrote: > >> + >> + snd_sof_dsp_block_write(sdev, offset, >> + (void *)block + sizeof(*block), >> + block->size); >> + >> + /* next block */ >> + block = (void *)block + sizeof(*block) + block->size; > This may lead to an unaligned access. Did you mean we should double check the block->size to prevent access to an invalid address? > Also how is the endianess guaranteed? Did you mean we should guarantee the driver can work no matter what kernel's endianess is? ie. Use le32_to_cpu() to handle it? > > > thanks, > > Takashi > >

4 5

[alsa-devel] [PATCH] ASoC: pcm512x: Fix a double unlock in pcm512x_digital_mute()
by Dan Carpenter 21 Dec '18

21 Dec '18

We accidentally call mutex_unlock(&pcm512x->mutex); twice in a row. I re-wrote the error handling to use "goto unlock;" instead of returning directly. Hopefully, it makes the code a little simpler. Fixes: 3500f1c589e9 ("ASoC: pcm512x: Implement the digital_mute interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/pcm512x.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 6cb1653be804..4cc24a5d5c31 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1400,24 +1400,20 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) if (ret != 0) { dev_err(component->dev, "Failed to set digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, PCM512x_ANALOG_MUTE_DET, mute_det, (mute_det & 0x3) == 0, 200, 10000); - - mutex_unlock(&pcm512x->mutex); } else { pcm512x->mute &= ~0x1; ret = pcm512x_update_mute(pcm512x); if (ret != 0) { dev_err(component->dev, "Failed to update digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, @@ -1428,9 +1424,10 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) 200, 10000); } +unlock: mutex_unlock(&pcm512x->mutex); - return 0; + return ret; } static const struct snd_soc_dai_ops pcm512x_dai_ops = { -- 2.17.1

3 6

[alsa-devel] [PATCH 1/2] ASoC: qdsp6: q6asm-dai: Off by one in of_q6asm_parse_dai_data()
by Dan Carpenter 21 Dec '18

21 Dec '18

The q6asm_fe_dais[] array has MAX_SESSIONS (8) elements so the > comparison should be >= or we access one element beyond the end of the array. Fixes: 22930c79ac5c ("ASoC: qdsp6: q6asm-dai: Add support to compress offload") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 5b986b74dd36..9d738b4c1e05 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -874,7 +874,7 @@ static int of_q6asm_parse_dai_data(struct device *dev, for_each_child_of_node(dev->of_node, node) { ret = of_property_read_u32(node, "reg", &id); - if (ret || id > MAX_SESSIONS || id < 0) { + if (ret || id >= MAX_SESSIONS || id < 0) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } -- 2.17.1

3 10

[alsa-devel] [RESEND v2] mfd: cros_ec: Add commands to control codec
by Cheng-Yi Chiang 21 Dec '18

21 Dec '18

Add EC host commands to control codec on EC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- include/linux/mfd/cros_ec_commands.h | 94 ++++++++++++++++++++++++++++ 1 file changed, 94 insertions(+) diff --git a/include/linux/mfd/cros_ec_commands.h b/include/linux/mfd/cros_ec_commands.h index 9a9631f0559e2..fc91082d4c357 100644 --- a/include/linux/mfd/cros_ec_commands.h +++ b/include/linux/mfd/cros_ec_commands.h @@ -2790,6 +2790,100 @@ struct ec_response_battery_vendor_param { uint32_t value; } __packed; +/*****************************************************************************/ +/* Commands for I2S recording on audio codec. */ + +#define EC_CMD_CODEC_I2S 0x00BC + +enum ec_codec_i2s_subcmd { + EC_CODEC_SET_SAMPLE_DEPTH = 0x0, + EC_CODEC_SET_GAIN = 0x1, + EC_CODEC_GET_GAIN = 0x2, + EC_CODEC_I2S_ENABLE = 0x3, + EC_CODEC_I2S_SET_CONFIG = 0x4, + EC_CODEC_I2S_SET_TDM_CONFIG = 0x5, + EC_CODEC_I2S_SET_BCLK = 0x6, +}; + +enum ec_sample_depth_value { + EC_CODEC_SAMPLE_DEPTH_16 = 0, + EC_CODEC_SAMPLE_DEPTH_24 = 1, +}; + +enum ec_i2s_config { + EC_DAI_FMT_I2S = 0, + EC_DAI_FMT_RIGHT_J = 1, + EC_DAI_FMT_LEFT_J = 2, + EC_DAI_FMT_PCM_A = 3, + EC_DAI_FMT_PCM_B = 4, + EC_DAI_FMT_PCM_TDM = 5, +}; + +struct ec_param_codec_i2s { + /* + * enum ec_codec_i2s_subcmd + */ + uint8_t cmd; + union { + /* + * EC_CODEC_SET_SAMPLE_DEPTH + * Value should be one of ec_sample_depth_value. + */ + uint8_t depth; + + /* + * EC_CODEC_SET_GAIN + * Value should be 0~43 for both channels. + */ + struct ec_param_codec_i2s_set_gain { + uint8_t left; + uint8_t right; + } __packed gain; + + /* + * EC_CODEC_I2S_ENABLE + * 1 to enable, 0 to disable. + */ + uint8_t i2s_enable; + + /* + * EC_CODEC_I2S_SET_COFNIG + * Value should be one of ec_i2s_config. + */ + uint8_t i2s_config; + + /* + * EC_CODEC_I2S_SET_TDM_CONFIG + * Value should be one of ec_i2s_config. + */ + struct ec_param_codec_i2s_tdm { + /* + * 0 to 496 + */ + int16_t ch0_delay; + /* + * -1 to 496 + */ + int16_t ch1_delay; + uint8_t adjacent_to_ch0; + uint8_t adjacent_to_ch1; + } __packed tdm_param; + + /* + * EC_CODEC_I2S_SET_BCLK + */ + uint32_t bclk; + }; +} __packed; + +/* + * For subcommand EC_CODEC_GET_GAIN. + */ +struct ec_response_codec_gain { + uint8_t left; + uint8_t right; +} __packed; + /*****************************************************************************/ /* System commands */ -- 2.20.1.415.g653613c723-goog

2 1

[alsa-devel] [PATCH v3 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 20 Dec '18

20 Dec '18

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- include/sound/cs35l36.h | 56 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36.c | 1969 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 446 ++++++++++ 5 files changed, 2478 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..644dfb1 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,56 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H +#define __CS35L36_H + +/* INT/GPIO pin configuration */ +struct irq_cfg { + bool is_present; + int irq_drv_sel; + int irq_pol; + int irq_gpio_sel; +}; + +struct vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool sclk_frc; + bool lrclk_frc; + bool multi_amp_mode; + bool dcm_mode; + int ldm_mode_sel; + bool amp_gain_zc; + bool amp_pcm_inv; + bool pdm_ldm_exit; + bool pdm_ldm_enter; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + struct vpbr_cfg vpbr_config; + struct irq_cfg irq_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f18..77cb861 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -465,6 +466,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c..7d8a372 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,6 +45,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -310,6 +311,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..0da68f3 --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1969 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int chip_version; + int rev_id; + struct gpio_desc *reset_gpio; +}; + +struct cs35l36_pll_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +struct reg_default cs35l36_reg[] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_TST_FS_MON0: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_TST_FS_MON0: + return true; + default: + return false; + } +} + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static int cs35l36_ldm_sel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = cs35l36->pdata.ldm_mode_sel; + + return 0; +} + +static int cs35l36_ldm_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int val = (ucontrol->value.integer.value[0]) ? CS35L36_NG_AMP_EN_MASK : + 0; + + cs35l36->pdata.ldm_mode_sel = val; + + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, val); + + return 0; +} + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("Analog PCM Volume", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), + SOC_SINGLE("Amp Gain Zero-Cross", CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Enter Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Exit Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_SHIFT, 1, 0), + SOC_SINGLE_BOOL_EXT("LDM Select", 0, cs35l36_ldm_sel_get, + cs35l36_ldm_sel_put), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + + if (WARN_ON_ONCE(reg & CS35L36_PLL_UNLOCK_MASK)) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + + return ret; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + + return 0; +} + +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux = + SOC_DAPM_ENUM("Input Mux", chansel_enum); + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const struct snd_kcontrol_new boost_ctrl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, &cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH_E("BOOST Enable", SND_SOC_NOPM, 0, 0, &boost_ctrl, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Enable", "Switch", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Enable"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + clk_frc = 1; + break; + case SND_SOC_DAIFMT_GATED: + clk_frc = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, clk_frc << + CS35L36_SCLK_FRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, clk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + unsigned int asp_width, global_fs = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1, fs2; + + if (freq > CS35L36_FS_NOM_6MHZ) { + fs1 = CS35L36_FS1_DEFAULT_VAL; + fs2 = CS35L36_FS2_DEFAULT_VAL; + } else { + fs1 = 3 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + fs2 = 5 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + } + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK | CS35L36_FS2_WINDOW_MASK, + fs1 | (fs2 << CS35L36_FS2_WINDOW_SHIFT)); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static const struct cs35l36_pll_config *cs35l36_get_clk_config( + struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) + return &cs35l36_pll_sysclk[i]; + } + + return NULL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + const struct cs35l36_pll_config *clk_cfg; + int prev_clksrc; + bool pdm_switch; + + prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + clk_cfg = cs35l36_get_clk_config(cs35l36, freq); + if (clk_cfg == NULL) { + dev_err(component->dev, "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + clk_cfg->clk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, + cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + clk_cfg->fll_igain); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc != CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } else { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc == CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if ((cs35l36->rev_id == CS35L36_REV_A0) && cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) { + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + if (ret < 0) { + dev_err(cs35l36->dev, + "Boost inductor config failed(%d)\n", ret); + return ret; + } + } + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + int ret = IRQ_NONE; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, status, + ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, masks, + ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, CS35L36_TEMP_WARN); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, CS35L36_TEMP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, CS35L36_BST_OVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct irq_cfg *irq_gpio_config = &pdata->irq_config; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *irq_gpio, *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + /* INT/GPIO Pin Config */ + irq_gpio = of_get_child_by_name(np, "cirrus,irq-config"); + irq_gpio_config->is_present = irq_gpio ? true : false; + if (irq_gpio_config->is_present) { + if (of_property_read_u32(irq_gpio, "cirrus,irq-drive-select", + &val) >= 0) + irq_gpio_config->irq_drv_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-polarity", + &val) >= 0) + irq_gpio_config->irq_pol = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-gpio-select", + &val) >= 0) + irq_gpio_config->irq_gpio_sel = val; + } + of_node_put(irq_gpio); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id != CS35L36_REV_B0) + return 0; + + /* + * Magic code for internal PAC + */ + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + + return 0; +} + +static int cs35l36_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); + + return 0; +} + +static int cs35l36_irq_gpio_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct irq_cfg *irq_config = &pdata->irq_config; + int irq_polarity; + + if (!irq_config->is_present) { + dev_err(cs35l36->dev, "IRQ config not present\n"); + return -EINVAL; + } + + /* setup the Interrupt Pin (INT | GPIO) */ + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, 1); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + irq_config->irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + irq_config->irq_pol << + CS35L36_INT_POL_SEL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + irq_config->irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (irq_config->irq_pol) + irq_polarity = IRQF_TRIGGER_HIGH; + else + irq_polarity = IRQF_TRIGGER_LOW; + + return irq_polarity; +} + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + /* Errata Writes */ + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + /* PAC Config */ + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + + int i; + int ret; + u32 reg_id, reg_revid, l37_id_reg; + int irq_pol = IRQF_TRIGGER_HIGH; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l36->pdata = *pdata; + } else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) { + dev_info(dev, "Reset line busy, assuming shared reset\n"); + } else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err_disable_regs; + } + } + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", reg_id, + CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_vpbr_config(cs35l36); + + if (pdata->irq_config.is_present) + irq_pol = cs35l36_irq_gpio_config(cs35l36); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | irq_pol, "cs35l36", + cs35l36); + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + +err_disable_regs: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + snd_soc_unregister_component(&client->dev); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..f6e38c6 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,446 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_SHIFT 10 +#define CS35L36_PDM_LDM_ENTER_SHIFT 3 +#define CS35L36_PDM_LDM_EXIT_SHIFT 4 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_FS1_DEFAULT_VAL 16 +#define CS35L36_FS2_DEFAULT_VAL 36 +#define CS35L36_FS_NOM_6MHZ 6000000 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

5 5

[alsa-devel] [PATCH] ALSA: isa/wavefront: Fix potential Spectre v1 vulnerabilities
by Gustavo A. R. Silva 20 Dec '18

20 Dec '18

header->number is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/isa/wavefront/wavefront_synth.c:792 wavefront_send_patch() warn: potential spectre issue 'dev->patch_status' [w] (local cap) sound/isa/wavefront/wavefront_synth.c:819 wavefront_send_program() warn: potential spectre issue 'dev->prog_status' [w] (local cap) sound/isa/wavefront/wavefront_synth.c:1197 wavefront_send_alias() warn: potential spectre issue 'dev->sample_status' [w] sound/isa/wavefront/wavefront_synth.c:1248 wavefront_send_multisample() warn: potential spectre issue 'dev->sample_status' [w] sound/isa/wavefront/wavefront_synth.c:1548 wavefront_synth_control() warn: potential spectre issue 'dev->sample_status' [r] (local cap) Fix this by sanitizing header->number before using it to index dev->patch_status, dev->prog_status and dev->sample_status. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/isa/wavefront/wavefront_synth.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 0b1e4b34b299..80196a533a36 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -31,6 +31,7 @@ #include <linux/moduleparam.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/snd_wavefront.h> #include <sound/initval.h> @@ -788,7 +789,8 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) if (header->number >= ARRAY_SIZE(dev->patch_status)) return -EINVAL; - + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->patch_status)); dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -815,7 +817,8 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) if (header->number >= ARRAY_SIZE(dev->prog_status)) return -EINVAL; - + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->prog_status)); dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -1194,6 +1197,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) return -EIO; } + if (header->number >= ARRAY_SIZE(dev->sample_status)) + return -EINVAL; + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->sample_status)); dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_ALIAS); return (0); @@ -1245,6 +1252,10 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) return -EIO; } + if (header->number >= ARRAY_SIZE(dev->sample_status)) + return -EINVAL; + header->number = array_index_nospec(header->number, + ARRAY_SIZE(dev->sample_status)); dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_MULTISAMPLE); kfree(msample_hdr); @@ -1539,12 +1550,13 @@ wavefront_synth_control (snd_wavefront_card_t *acard, case WFC_IDENTIFY_SLOT_TYPE: i = wc->wbuf[0] | (wc->wbuf[1] << 7); - if (i <0 || i >= WF_MAX_SAMPLE) { + if (i < 0 || i >= WF_MAX_SAMPLE) { snd_printk ("invalid slot ID %d\n", i); wc->status = EINVAL; return -EINVAL; } + i = array_index_nospec(i, WF_MAX_SAMPLE); wc->rbuf[0] = dev->sample_status[i]; wc->status = 0; return 0; -- 2.20.1

2 4

[alsa-devel] Console downloaders give *The certificate of ‘www.alsa-project.org’ hasn't got a known issuer.*
by Paul Menzel 20 Dec '18

20 Dec '18

[Please CC, as I am not subscribed.] Dear ALSA folks, Despite working in the browser (Mozilla Firefox), GNU Wget and curl give the error below trying to download the script `alsa-info.sh`. $ wget https://www.alsa-project.org/alsa-info.sh --2018-12-18 17:27:57-- https://www.alsa-project.org/alsa-info.sh Resolving www.alsa-project.org (www.alsa-project.org)... 77.48.224.243 Connecting to www.alsa-project.org (www.alsa-project.org)|77.48.224.243|:443... connected. ERROR: The certificate of ‘www.alsa-project.org’ is not trusted. ERROR: The certificate of ‘www.alsa-project.org’ hasn't got a known issuer. Kind regards, Paul

2 5

[alsa-devel] [PATCH] ASoC: xlnx: change license header format style
by Maruthi Srinivas Bayyavarapu 20 Dec '18

20 Dec '18

Changed License header from C to C++ style comment block. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> --- sound/soc/xilinx/xlnx_i2s.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c index d4ae9ef..8b35316 100644 --- a/sound/soc/xilinx/xlnx_i2s.c +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -1,12 +1,11 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Xilinx ASoC I2S audio support - * - * Copyright (C) 2018 Xilinx, Inc. - * - * Author: Praveen Vuppala <praveenv(a)xilinx.com> - * Author: Maruthi Srinivas Bayyavarapu <maruthis(a)xilinx.com> - */ +// +// Xilinx ASoC I2S audio support +// +// Copyright (C) 2018 Xilinx, Inc. +// +// Author: Praveen Vuppala <praveenv(a)xilinx.com> +// Author: Maruthi Srinivas Bayyavarapu <maruthis(a)xilinx.com> #include <linux/io.h> #include <linux/module.h> -- 2.7.4

3 3

[alsa-devel] [PATCH 0/3] add Xilinx audio formatter driver
by Maruthi Srinivas Bayyavarapu 20 Dec '18

20 Dec '18

Audio formatter IP supports two streaming interfaces - MM2S for playback and S2MM for capture. The driver enables DMA functionality for both the interfaces. Patchset includes devicetree bindings documentation, driver and build enablement. Maruthi Srinivas Bayyavarapu (3): dt-bindings: ASoC: xlnx,audio-formatter: Document audio formatter bindings ASoC: xlnx: add pcm formatter platform driver ASoC: xlnx: enable audio formatter driver build .../bindings/sound/xlnx,audio-formatter.txt | 29 ++ sound/soc/xilinx/Kconfig | 7 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_formatter_pcm.c | 561 +++++++++++++++++++++ 4 files changed, 599 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt create mode 100644 sound/soc/xilinx/xlnx_formatter_pcm.c -- 2.7.4

2 4

[alsa-devel] [PATCH 0/9] ASoC/ARM: Merge the davinci and omap audio directories
by Peter Ujfalusi 20 Dec '18

20 Dec '18

Hi, Changes since RFC: - Added Jarkko's Acked-by/Tested-by for the {omap->ti} changes - Rebased on top of next-20181210 (20181214+ is not booting on any of my boards) For Texas Instruments audio support we have two directories at the moment: davinci omap In reality TI have the following SoC lines supported by Linux: daVinci OMAP Sitara Keystone We already needed to have cross directory 'links' for DRA7x/AM5 since they use McASP (davinci directory) and sDMA (omap directory). Keystone II line of SoCs are using drivers from davinci directory (McASP and EDMA). Which is kind of fine as they are in the same place. With the new AM654 (http://www.ti.com/lit/pdf/spruid7) we will have completely new DMA and there is no place to put that as it does not belong under davinci or omap, but it makes no sense to create yet another TI related directory just for the small udma-pcm. Another issue with the current support is how the Kconfigs are written. The DAI drivers are depending on the ASoC platform drivers and it is a bit of a mess at the moment. For simple-sound-card one need to know what DMA is used and then select the DAI driver (if the Kconfig is right). I have also cleaned up the Kconfig prompts and comments a bit to be more precise. As can be seen in the diffstat I tried to do the merger with minimal code change, resisting the urge to rename the davinci-i2c driver to davinci-asp. With the merger I have created a new Kconfig structure: - Submenu for TI related settings for clarity - Only the DAI drivers are selectable and the platform drivers are selected based on the available (enabled) DMA engines - The remaining ASoC machine drivers are selecting the DAIs they need - The davinci-evm is now a single entry, combining all the various daVinci EVMS it supports and selects the needed DAIs based on the enabled boards The series also includes patches to updated the arch/arm/ files (defconfigs, source files using audio Kconfig symbols). I think I have catched all (I hope) issues regarding to Kconfig thanks to the various autobuild machines out there kindly testing my linux-next-wip branch. Mark, Tony, Sekhar: This series - as I mentioned - is based on next-20181210. Not sure what tag, branch, commit I should base this, but I can resend v2 if needed. Regards, Peter --- Peter Ujfalusi (9): ASoC: ti: Merge davinci and omap directories MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option ARM: OMAP2: Update for new MCBSP Kconfig option ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options ARM: omap2plus_defconfig: Update the audio options ARM: omap1_defconfig: Do not select ASoC by default ARM: davinci_all_defconfig: Update the audio options ASoC: ti: Kconfig: Remove the deprecated options MAINTAINERS | 11 +- arch/arm/configs/davinci_all_defconfig | 5 +- arch/arm/configs/omap1_defconfig | 2 - arch/arm/configs/omap2plus_defconfig | 14 +- arch/arm/mach-davinci/board-dm365-evm.c | 4 +- arch/arm/mach-omap1/Makefile | 2 +- arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/pdata-quirks.c | 4 +- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 3 +- sound/soc/davinci/Kconfig | 106 --------- sound/soc/davinci/Makefile | 16 -- sound/soc/omap/Kconfig | 120 ---------- sound/soc/omap/Makefile | 30 --- sound/soc/ti/Kconfig | 209 ++++++++++++++++++ sound/soc/ti/Makefile | 44 ++++ sound/soc/{omap => ti}/ams-delta.c | 0 sound/soc/{davinci => ti}/davinci-evm.c | 4 +- sound/soc/{davinci => ti}/davinci-i2s.c | 0 sound/soc/{davinci => ti}/davinci-i2s.h | 0 sound/soc/{davinci => ti}/davinci-mcasp.c | 14 +- sound/soc/{davinci => ti}/davinci-mcasp.h | 0 sound/soc/{davinci => ti}/davinci-vcif.c | 0 sound/soc/{davinci => ti}/edma-pcm.c | 0 sound/soc/{davinci => ti}/edma-pcm.h | 4 +- sound/soc/{omap => ti}/n810.c | 0 sound/soc/{omap => ti}/omap-abe-twl6040.c | 0 sound/soc/{omap => ti}/omap-dmic.c | 0 sound/soc/{omap => ti}/omap-dmic.h | 0 .../omap-hdmi-audio.c => ti/omap-hdmi.c} | 0 sound/soc/{omap => ti}/omap-mcbsp-priv.h | 0 sound/soc/{omap => ti}/omap-mcbsp-st.c | 0 sound/soc/{omap => ti}/omap-mcbsp.c | 0 sound/soc/{omap => ti}/omap-mcbsp.h | 0 sound/soc/{omap => ti}/omap-mcpdm.c | 0 sound/soc/{omap => ti}/omap-mcpdm.h | 0 sound/soc/{omap => ti}/omap-twl4030.c | 0 sound/soc/{omap => ti}/omap3pandora.c | 0 sound/soc/{omap => ti}/osk5912.c | 0 sound/soc/{omap => ti}/rx51.c | 0 sound/soc/{omap => ti}/sdma-pcm.c | 0 sound/soc/{omap => ti}/sdma-pcm.h | 4 +- 42 files changed, 294 insertions(+), 307 deletions(-) delete mode 100644 sound/soc/davinci/Kconfig delete mode 100644 sound/soc/davinci/Makefile delete mode 100644 sound/soc/omap/Kconfig delete mode 100644 sound/soc/omap/Makefile create mode 100644 sound/soc/ti/Kconfig create mode 100644 sound/soc/ti/Makefile rename sound/soc/{omap => ti}/ams-delta.c (100%) rename sound/soc/{davinci => ti}/davinci-evm.c (99%) rename sound/soc/{davinci => ti}/davinci-i2s.c (100%) rename sound/soc/{davinci => ti}/davinci-i2s.h (100%) rename sound/soc/{davinci => ti}/davinci-mcasp.c (99%) rename sound/soc/{davinci => ti}/davinci-mcasp.h (100%) rename sound/soc/{davinci => ti}/davinci-vcif.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.h (91%) rename sound/soc/{omap => ti}/n810.c (100%) rename sound/soc/{omap => ti}/omap-abe-twl6040.c (100%) rename sound/soc/{omap => ti}/omap-dmic.c (100%) rename sound/soc/{omap => ti}/omap-dmic.h (100%) rename sound/soc/{omap/omap-hdmi-audio.c => ti/omap-hdmi.c} (100%) rename sound/soc/{omap => ti}/omap-mcbsp-priv.h (100%) rename sound/soc/{omap => ti}/omap-mcbsp-st.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.h (100%) rename sound/soc/{omap => ti}/omap-mcpdm.c (100%) rename sound/soc/{omap => ti}/omap-mcpdm.h (100%) rename sound/soc/{omap => ti}/omap-twl4030.c (100%) rename sound/soc/{omap => ti}/omap3pandora.c (100%) rename sound/soc/{omap => ti}/osk5912.c (100%) rename sound/soc/{omap => ti}/rx51.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.h (85%) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 11

[alsa-devel] alsa-lib: commit ce2095c41f2891c51f5dbd28e0317200314c5a75 breaks speaker-test/pulseaudio
by Diego Viola 19 Dec '18

19 Dec '18

Hello Takashi, I currently have an issue with speaker-test and pulseaudio, and the root of the problem seems to be a regression in alsa-lib. The issue occurs when I run speaker-test, it creates endless output with no sound, and the output does not stop until I hit ctrl-c. I've been investigating about this issue and concluded (through a git bisect) that this is a regression in alsa-lib, and this is the bad commit that introduces the bug: commit ce2095c41f2891c51f5dbd28e0317200314c5a75 Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Mar 29 09:51:46 2018 +0200 pcm: ioplug: Implement proper drain behavior This patch fixes the draining behavior of ioplug in the following ways: - When no draining ioplug callback is defined, implement the draining loop using snd_pcm_wait*() and sync with the drain finishes. This is equivalent with the implementation in the kernel write(). Similarly as in kernel code, for non-blocking mode, it returns immediately after setting DRAINING state. - The hw_ptr update function checks the PCM state and stops the stream if the draining finishes. - When draining ioplug callback is defined, leave the whole draining operation to it. The callback is supposed to return -EAGAIN for non-blocking case, too. - When an error happens during draining, it drops the stream, for a safety reason. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> I've sanity checked and can confirm that going back to the previous commit fixes the issue. My original bug report on the pulseaudio bug tracker: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/593 Thanks, and please let me know if you need more information. Regards, Diego

2 3

[alsa-devel] [PATCH trivial] ASoC: xlnx: Grammar s/the the/the/
by Geert Uytterhoeven 19 Dec '18

19 Dec '18

Fixes: 33f8db9a89200c18 ("ASoC: xlnx: enable i2s driver build") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/xilinx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 25e287feb58c33e8..723a583a8d570f62 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -1,5 +1,5 @@ config SND_SOC_XILINX_I2S - tristate "Audio support for the the Xilinx I2S" + tristate "Audio support for the Xilinx I2S" help Select this option to enable Xilinx I2S Audio. This enables I2S playback and capture using xilinx soft IP. In transmitter -- 2.17.1

3 3

[alsa-devel] [PATCH] ASoC: Intel: Haswell/Broadwell: fix setting for .dynamic field
by Rander Wang 19 Dec '18

19 Dec '18

For some reason this field was set to zero when all other drivers use .dynamic = 1 for front-ends. This change was tested on Dell XPS13 and has no impact with the existing legacy driver. The SOF driver also works with this change which enables it to override the fixed topology. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> --- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 68e6543e6cb0..99f2a0156ae8 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index eab1f439dd3f..a4022983a7ce 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, -- 2.14.1

3 2

[alsa-devel] Loading coefficients for Audio IP
by Daniel Baluta 19 Dec '18

19 Dec '18

Hello Pierre, Liam, Mark, What is the standard ALSA way of loading tables of coefficients for various audio IPs? We have IPs like PDM or ASRC which need large table of coefficients depending on configuration. This tables can reach up to 120K in size. Creating this kind of tables as arrays of integers in kernel doesn't seem to be a common used solution and we will end up with large files that contains only numbers. Another approach that we are thinking of is to create and compile this tables in userspace as raw binary files and then loading them in kernel using the firmware interface. Would this be an acceptable approach? thanks, Daniel.

5 4

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with, ALC294
by Wandrille RONCE 19 Dec '18

19 Dec '18

From c87ce2119f010db78933c424b8ba6738c3056dff Mon Sep 17 00:00:00 2001 From: Wandrille RONCE <w(a)ndrille.fr> Date: Mon, 17 Dec 2018 19:00:00 +0100 Subject: [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with ALC294 --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 15021c839372..90f30e769a42 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6584,6 +6584,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 2.19.1

2 3

[alsa-devel] 4.21 merge window
by Takashi Iwai 19 Dec '18

19 Dec '18

Hi, we're approaching to 4.20 final release, and it means that the merge window of 4.21 will be closed soon, too. Actually, I'll be traveling from the next Saturday (more exactly, already leaving on Friday afternoon), so I'd like to close the merges of major changes in today, and prepare the pull request for tomorrow. If you still have anything really important that must be in 4.21, please let me know. Thanks! Takashi

1 0

[alsa-devel] [PATCH] ALSA: bebob: fix model-id of unit for Apogee Ensemble
by Takashi Sakamoto 19 Dec '18

19 Dec '18

This commit fixes hard-coded model-id for an unit of Apogee Ensemble with a correct value. This unit uses DM1500 ASIC produced ArchWave AG (formerly known as BridgeCo AG). I note that this model supports three modes in the number of data channels in tx/rx streams; 8 ch pairs, 10 ch pairs, 18 ch pairs. The mode is switched by Vendor-dependent AV/C command, like: $ cd linux-firewire-utils $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0600000000 (8ch pairs) $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0601000000 (10ch pairs) $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0602000000 (18ch pairs) When switching between different mode, the unit disappears from IEEE 1394 bus, then appears on the bus with different combination of stream formats. In a mode of 18 ch pairs, available sampling rate is up to 96.0 kHz, else up to 192.0 kHz. $ ./hinawa-config-rom-printer /dev/fw1 { 'bus-info': { 'adj': False, 'bmc': True, 'chip_ID': 21474898341, 'cmc': True, 'cyc_clk_acc': 100, 'generation': 2, 'imc': True, 'isc': True, 'link_spd': 2, 'max_ROM': 1, 'max_rec': 512, 'name': '1394', 'node_vendor_ID': 987, 'pmc': False}, 'root-directory': [ ['HARDWARE_VERSION', 19], [ 'NODE_CAPABILITIES', { 'addressing': {'64': True, 'fix': True, 'prv': False}, 'misc': {'int': False, 'ms': False, 'spt': True}, 'state': { 'atn': False, 'ded': False, 'drq': True, 'elo': False, 'init': False, 'lst': True, 'off': False}, 'testing': {'bas': False, 'ext': False}}], ['VENDOR', 987], ['DESCRIPTOR', 'Apogee Electronics'], ['MODEL', 126702], ['DESCRIPTOR', 'Ensemble'], ['VERSION', 5297], [ 'UNIT', [ ['SPECIFIER_ID', 41005], ['VERSION', 65537], ['MODEL', 126702], ['DESCRIPTOR', 'Ensemble']]], [ 'DEPENDENT_INFO', [ ['SPECIFIER_ID', 2037], ['VERSION', 1], [(58, 'IMMEDIATE'), 16777159], [(59, 'IMMEDIATE'), 1048576], [(60, 'IMMEDIATE'), 16777159], [(61, 'IMMEDIATE'), 6291456]]]]} Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 672d13488454..d91874275d2c 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -408,7 +408,7 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */ SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal), /* Apogee Electronics, Ensemble */ - SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal), /* ESI, Quatafire610 */ SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal), /* AcousticReality, eARMasterOne */ -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: emu10k1: Fix potential Spectre v1 vulnerabilities
by Gustavo A. R. Silva 19 Dec '18

19 Dec '18

ipcm->substream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/pci/emu10k1/emufx.c:1031 snd_emu10k1_ipcm_poke() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap) sound/pci/emu10k1/emufx.c:1075 snd_emu10k1_ipcm_peek() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap) Fix this by sanitizing ipcm->substream before using it to index emu->fx8010.pcm Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/emu10k1/emufx.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 6ebe817801ea..1f25e6d029d8 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -36,6 +36,7 @@ #include <linux/init.h> #include <linux/mutex.h> #include <linux/moduleparam.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/tlv.h> @@ -1026,6 +1027,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); if (ipcm->channels > 32) return -EINVAL; pcm = &emu->fx8010.pcm[ipcm->substream]; @@ -1072,6 +1075,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); pcm = &emu->fx8010.pcm[ipcm->substream]; mutex_lock(&emu->fx8010.lock); spin_lock_irq(&emu->reg_lock); -- 2.19.2

3 2

[alsa-devel] [PATCH] ALSA: rme9652: Fix potential Spectre v1 vulnerability
by Gustavo A. R. Silva 19 Dec '18

19 Dec '18

info->channel is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/pci/rme9652/hdsp.c:4100 snd_hdsp_channel_info() warn: potential spectre issue 'hdsp->channel_map' [r] (local cap) Fix this by sanitizing info->channel before using it to index hdsp->channel_map Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. Also, notice that I refactored the code a bit in order to get rid of the following checkpatch warning: ERROR: do not use assignment in if condition FILE: sound/pci/rme9652/hdsp.c:4103: if ((mapped_channel = hdsp->channel_map[info->channel]) < 0) [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/rme9652/hdsp.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 1bff4b1b39cd..ba99ff0e93e0 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -30,6 +30,7 @@ #include <linux/math64.h> #include <linux/vmalloc.h> #include <linux/io.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/control.h> @@ -4092,15 +4093,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream, struct snd_pcm_channel_info *info) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - int mapped_channel; + unsigned int channel = info->channel; - if (snd_BUG_ON(info->channel >= hdsp->max_channels)) + if (snd_BUG_ON(channel >= hdsp->max_channels)) return -EINVAL; + channel = array_index_nospec(channel, hdsp->max_channels); - if ((mapped_channel = hdsp->channel_map[info->channel]) < 0) + if (hdsp->channel_map[channel] < 0) return -EINVAL; - info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES; + info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES; info->first = 0; info->step = 32; return 0; -- 2.19.2

3 2

[alsa-devel] [PATCH v3] ASoC: AMD: Fix race condition between register access
by Agrawal, Akshu 19 Dec '18

19 Dec '18

During simultaneous running of playback and capture, we got hit by incorrect value write on common register. This was due to race condition between 2 streams. Fixing this by locking the common register access. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Added 2 helper functions, removed locking in ch enable/disable as they are separate registers. v3: Modified helper acp_reg_read_mod_write, moved pte locks to helper sound/soc/amd/acp-pcm-dma.c | 66 ++++++++++++++++++++++++++++++--------------- 1 file changed, 44 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 0ac4b5b..0d4c2eb 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -121,6 +121,9 @@ .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +/* Lock to protect access to registers */ +static DEFINE_SPINLOCK(lock); + static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -131,6 +134,31 @@ static void acp_reg_write(u32 val, void __iomem *acp_mmio, u32 reg) writel(val, acp_mmio + (reg * 4)); } +static void acp_reg_write_srbm_targ(void __iomem *acp_mmio, + u32 addr, u32 data) +{ + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + acp_reg_write(addr, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); + acp_reg_write(data, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + spin_unlock_irqrestore(&lock, flags); +} + +static void acp_reg_read_mod_write(void __iomem *acp_mmio, u32 addr, + u32 mask, u32 value) +{ + u32 val; + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, addr); + val &= ~mask; + val |= (mask & value); + acp_reg_write(val, acp_mmio, addr); + spin_unlock_irqrestore(&lock, flags); +} + /* * Configure a given dma channel parameters - enable/disable, * number of descriptors, priority @@ -172,15 +200,12 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, sram_offset = (descr_idx * sizeof(acp_dma_dscr_transfer_t)); /* program the source base address. */ - acp_reg_write(sram_offset, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->src, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset, descr_info->src); /* program the destination base address. */ - acp_reg_write(sram_offset + 4, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->dest, acp_mmio, mmACP_SRBM_Targ_Idx_Data); - + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 4, descr_info->dest); /* program the number of bytes to be transferred for this descriptor. */ - acp_reg_write(sram_offset + 8, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->xfer_val, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 8, + descr_info->xfer_val); } static void pre_config_reset(void __iomem *acp_mmio, u16 ch_num) @@ -311,24 +336,22 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, u32 offset; offset = ACP_DAGB_GRP_SRBM_SRAM_BASE_OFFSET + (pte_offset * 8); + for (page_idx = 0; page_idx < (num_of_pages); page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ - acp_reg_write((offset + (page_idx * 8)), - acp_mmio, mmACP_SRBM_Targ_Idx_Addr); addr = page_to_phys(pg); low = lower_32_bits(addr); high = upper_32_bits(addr); - acp_reg_write(low, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, (offset + (page_idx * 8)), + low); /* Load the High address of page int ACP SRAM through SRBM */ - acp_reg_write((offset + (page_idx * 8) + 4), - acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - /* page enable in ACP */ high |= BIT(31); - acp_reg_write(high, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, (offset + (page_idx * 8) + 4), + high); /* Move to next physically contiguos page */ pg++; @@ -870,29 +893,28 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } } if (adata->asic_type == CHIP_STONEY) { - val = acp_reg_read(adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + val = ACP_I2S_SP_16BIT_RESOLUTION_EN; } } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + val = ACP_I2S_MIC_16BIT_RESOLUTION_EN; } } - acp_reg_write(val, adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); + acp_reg_read_mod_write(adata->acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN, + val, val); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: eukrea-tlv320: Do not enable on ARM64
by Andrey Smirnov 19 Dec '18

19 Dec '18

ARCH_MXC is also used by i.MX8/AArch64 SoC, building the driver for that platform results in: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory #include <asm/mach-types.h> ^~~~~~~~~~~~~~~~~~ compilation terminated. Since the code in question doesn't seem to have any use on i.MX8, disable it by adding a !ARM64 as a dependecy. Signed-off-by: Andrey Smirnov <andrew.smirnov(a)gmail.com> --- Sorry for the noise if this is fixed already. Thanks, Andrey Smirnov sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb4a934..2e75b5bc5f1d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.19.1

2 1

[alsa-devel] [Xen-devel][PATCH v2 1/3] xen: Introduce shared buffer helpers for page directory...
by Oleksandr Andrushchenko 18 Dec '18

18 Dec '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> based frontends. Currently the frontends which implement similar code for sharing big buffers between frontend and backend are para-virtualized DRM and sound drivers. Both define the same way to share grant references of a data buffer with the corresponding backend with little differences. Move shared code into a helper module, so there is a single implementation of the same functionality for all. This patch introduces code which is used by sound and display frontend drivers without functional changes with the intention to remove shared code from the corresponding drivers. Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> --- drivers/xen/Kconfig | 3 + drivers/xen/Makefile | 1 + drivers/xen/xen-front-pgdir-shbuf.c | 553 ++++++++++++++++++++++++++++ include/xen/xen-front-pgdir-shbuf.h | 89 +++++ 4 files changed, 646 insertions(+) create mode 100644 drivers/xen/xen-front-pgdir-shbuf.c create mode 100644 include/xen/xen-front-pgdir-shbuf.h diff --git a/drivers/xen/Kconfig b/drivers/xen/Kconfig index 815b9e9bb975..838b66a9a0e7 100644 --- a/drivers/xen/Kconfig +++ b/drivers/xen/Kconfig @@ -340,4 +340,7 @@ config XEN_SYMS config XEN_HAVE_VPMU bool +config XEN_FRONT_PGDIR_SHBUF + tristate + endmenu diff --git a/drivers/xen/Makefile b/drivers/xen/Makefile index 3e542f60f29f..c48927a58e10 100644 --- a/drivers/xen/Makefile +++ b/drivers/xen/Makefile @@ -44,3 +44,4 @@ xen-gntdev-y := gntdev.o xen-gntdev-$(CONFIG_XEN_GNTDEV_DMABUF) += gntdev-dmabuf.o xen-gntalloc-y := gntalloc.o xen-privcmd-y := privcmd.o privcmd-buf.o +obj-$(CONFIG_XEN_FRONT_PGDIR_SHBUF) += xen-front-pgdir-shbuf.o diff --git a/drivers/xen/xen-front-pgdir-shbuf.c b/drivers/xen/xen-front-pgdir-shbuf.c new file mode 100644 index 000000000000..48a658dc7ccf --- /dev/null +++ b/drivers/xen/xen-front-pgdir-shbuf.c @@ -0,0 +1,553 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen frontend/backend page directory based shared buffer + * helper module. + * + * Copyright (C) 2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/mm.h> + +#include <asm/xen/hypervisor.h> +#include <xen/balloon.h> +#include <xen/xen.h> +#include <xen/xenbus.h> +#include <xen/interface/io/ring.h> + +#include <xen/xen-front-pgdir-shbuf.h> + +#ifndef GRANT_INVALID_REF +/* + * FIXME: usage of grant reference 0 as invalid grant reference: + * grant reference 0 is valid, but never exposed to a PV driver, + * because of the fact it is already in use/reserved by the PV console. + */ +#define GRANT_INVALID_REF 0 +#endif + +/** + * This structure represents the structure of a shared page + * that contains grant references to the pages of the shared + * buffer. This structure is common to many Xen para-virtualized + * protocols at include/xen/interface/io/ + */ +struct xen_page_directory { + grant_ref_t gref_dir_next_page; + grant_ref_t gref[1]; /* Variable length */ +}; + +/** + * Shared buffer ops which are differently implemented + * depending on the allocation mode, e.g. if the buffer + * is allocated by the corresponding backend or frontend. + * Some of the operations. + */ +struct xen_front_pgdir_shbuf_ops { + /* + * Calculate number of grefs required to handle this buffer, + * e.g. if grefs are required for page directory only or the buffer + * pages as well. + */ + void (*calc_num_grefs)(struct xen_front_pgdir_shbuf *buf); + + /* Fill page directory according to para-virtual display protocol. */ + void (*fill_page_dir)(struct xen_front_pgdir_shbuf *buf); + + /* Claim grant references for the pages of the buffer. */ + int (*grant_refs_for_buffer)(struct xen_front_pgdir_shbuf *buf, + grant_ref_t *priv_gref_head, int gref_idx); + + /* Map grant references of the buffer. */ + int (*map)(struct xen_front_pgdir_shbuf *buf); + + /* Unmap grant references of the buffer. */ + int (*unmap)(struct xen_front_pgdir_shbuf *buf); +}; + +/** + * Get granted reference to the very first page of the + * page directory. Usually this is passed to the backend, + * so it can find/fill the grant references to the buffer's + * pages. + * + * \param buf shared buffer which page directory is of interest. + * \return granted reference to the very first page of the + * page directory. + */ +grant_ref_t +xen_front_pgdir_shbuf_get_dir_start(struct xen_front_pgdir_shbuf *buf) +{ + if (!buf->grefs) + return GRANT_INVALID_REF; + + return buf->grefs[0]; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_get_dir_start); + +/** + * Map granted references of the shared buffer. + * + * Depending on the shared buffer mode of allocation + * (be_alloc flag) this can either do nothing (for buffers + * shared by the frontend itself) or map the provided granted + * references onto the backing storage (buf->pages). + * + * \param buf shared buffer which grants to be maped. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_map(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->ops && buf->ops->map) + return buf->ops->map(buf); + + /* No need to map own grant references. */ + return 0; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_map); + +/** + * Unmap granted references of the shared buffer. + * + * Depending on the shared buffer mode of allocation + * (be_alloc flag) this can either do nothing (for buffers + * shared by the frontend itself) or unmap the provided granted + * references. + * + * \param buf shared buffer which grants to be unmaped. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_unmap(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->ops && buf->ops->unmap) + return buf->ops->unmap(buf); + + /* No need to unmap own grant references. */ + return 0; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_unmap); + +/** + * Free all the resources of the shared buffer. + * + * \param buf shared buffer which resources to be freed. + */ +void xen_front_pgdir_shbuf_free(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->grefs) { + int i; + + for (i = 0; i < buf->num_grefs; i++) + if (buf->grefs[i] != GRANT_INVALID_REF) + gnttab_end_foreign_access(buf->grefs[i], + 0, 0UL); + } + kfree(buf->grefs); + kfree(buf->directory); +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_free); + +/* + * Number of grefs a page can hold with respect to the + * struct xen_page_directory header. + */ +#define XEN_NUM_GREFS_PER_PAGE ((PAGE_SIZE - \ + offsetof(struct xen_page_directory, \ + gref)) / sizeof(grant_ref_t)) + +/** + * Get the number of pages the page directory consumes itself. + * + * \param buf shared buffer. + */ +static int get_num_pages_dir(struct xen_front_pgdir_shbuf *buf) +{ + return DIV_ROUND_UP(buf->num_pages, XEN_NUM_GREFS_PER_PAGE); +} + +/** + * Calculate the number of grant references needed to share the buffer + * and its pages when backend allocates the buffer. + * + * \param buf shared buffer. + */ +static void backend_calc_num_grefs(struct xen_front_pgdir_shbuf *buf) +{ + /* Only for pages the page directory consumes itself. */ + buf->num_grefs = get_num_pages_dir(buf); +} + +/** + * Calculate the number of grant references needed to share the buffer + * and its pages when frontend allocates the buffer. + * + * \param buf shared buffer. + */ +static void guest_calc_num_grefs(struct xen_front_pgdir_shbuf *buf) +{ + /* + * Number of pages the page directory consumes itself + * plus grefs for the buffer pages. + */ + buf->num_grefs = get_num_pages_dir(buf) + buf->num_pages; +} + +#define xen_page_to_vaddr(page) \ + ((uintptr_t)pfn_to_kaddr(page_to_xen_pfn(page))) + +/** + * Unmap the buffer previously mapped with grant references + * provided by the backend. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int backend_unmap(struct xen_front_pgdir_shbuf *buf) +{ + struct gnttab_unmap_grant_ref *unmap_ops; + int i, ret; + + if (!buf->pages || !buf->backend_map_handles || !buf->grefs) + return 0; + + unmap_ops = kcalloc(buf->num_pages, sizeof(*unmap_ops), + GFP_KERNEL); + if (!unmap_ops) + return -ENOMEM; + + for (i = 0; i < buf->num_pages; i++) { + phys_addr_t addr; + + addr = xen_page_to_vaddr(buf->pages[i]); + gnttab_set_unmap_op(&unmap_ops[i], addr, GNTMAP_host_map, + buf->backend_map_handles[i]); + } + + ret = gnttab_unmap_refs(unmap_ops, NULL, buf->pages, + buf->num_pages); + + for (i = 0; i < buf->num_pages; i++) { + if (unlikely(unmap_ops[i].status != GNTST_okay)) + dev_err(&buf->xb_dev->dev, + "Failed to unmap page %d: %d\n", + i, unmap_ops[i].status); + } + + if (ret) + dev_err(&buf->xb_dev->dev, + "Failed to unmap grant references, ret %d", ret); + + kfree(unmap_ops); + kfree(buf->backend_map_handles); + buf->backend_map_handles = NULL; + return ret; +} + +/** + * Map the buffer with grant references provided by the backend. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int backend_map(struct xen_front_pgdir_shbuf *buf) +{ + struct gnttab_map_grant_ref *map_ops = NULL; + unsigned char *ptr; + int ret, cur_gref, cur_dir_page, cur_page, grefs_left; + + map_ops = kcalloc(buf->num_pages, sizeof(*map_ops), GFP_KERNEL); + if (!map_ops) + return -ENOMEM; + + buf->backend_map_handles = kcalloc(buf->num_pages, + sizeof(*buf->backend_map_handles), + GFP_KERNEL); + if (!buf->backend_map_handles) { + kfree(map_ops); + return -ENOMEM; + } + + /* + * Read page directory to get grefs from the backend: for external + * buffer we only allocate buf->grefs for the page directory, + * so buf->num_grefs has number of pages in the page directory itself. + */ + ptr = buf->directory; + grefs_left = buf->num_pages; + cur_page = 0; + for (cur_dir_page = 0; cur_dir_page < buf->num_grefs; cur_dir_page++) { + struct xen_page_directory *page_dir = + (struct xen_page_directory *)ptr; + int to_copy = XEN_NUM_GREFS_PER_PAGE; + + if (to_copy > grefs_left) + to_copy = grefs_left; + + for (cur_gref = 0; cur_gref < to_copy; cur_gref++) { + phys_addr_t addr; + + addr = xen_page_to_vaddr(buf->pages[cur_page]); + gnttab_set_map_op(&map_ops[cur_page], addr, + GNTMAP_host_map, + page_dir->gref[cur_gref], + buf->xb_dev->otherend_id); + cur_page++; + } + + grefs_left -= to_copy; + ptr += PAGE_SIZE; + } + ret = gnttab_map_refs(map_ops, NULL, buf->pages, buf->num_pages); + + /* Save handles even if error, so we can unmap. */ + for (cur_page = 0; cur_page < buf->num_pages; cur_page++) { + buf->backend_map_handles[cur_page] = map_ops[cur_page].handle; + if (unlikely(map_ops[cur_page].status != GNTST_okay)) + dev_err(&buf->xb_dev->dev, + "Failed to map page %d: %d\n", + cur_page, map_ops[cur_page].status); + } + + if (ret) { + dev_err(&buf->xb_dev->dev, + "Failed to map grant references, ret %d", ret); + backend_unmap(buf); + } + + kfree(map_ops); + return ret; +} + +/** + * Fill page directory with grant references to the pages of the + * page directory itself. + * + * The grant references to the buffer pages are provided by the + * backend in this case. + * + * \param buf shared buffer. + */ +static void backend_fill_page_dir(struct xen_front_pgdir_shbuf *buf) +{ + struct xen_page_directory *page_dir; + unsigned char *ptr; + int i, num_pages_dir; + + ptr = buf->directory; + num_pages_dir = get_num_pages_dir(buf); + + /* Fill only grefs for the page directory itself. */ + for (i = 0; i < num_pages_dir - 1; i++) { + page_dir = (struct xen_page_directory *)ptr; + + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + ptr += PAGE_SIZE; + } + /* Last page must say there is no more pages. */ + page_dir = (struct xen_page_directory *)ptr; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; +} + +/** + * Fill page directory with grant references to the pages of the + * page directory and the buffer we share with the backend. + * + * \param buf shared buffer. + */ +static void guest_fill_page_dir(struct xen_front_pgdir_shbuf *buf) +{ + unsigned char *ptr; + int cur_gref, grefs_left, to_copy, i, num_pages_dir; + + ptr = buf->directory; + num_pages_dir = get_num_pages_dir(buf); + + /* + * While copying, skip grefs at start, they are for pages + * granted for the page directory itself. + */ + cur_gref = num_pages_dir; + grefs_left = buf->num_pages; + for (i = 0; i < num_pages_dir; i++) { + struct xen_page_directory *page_dir = + (struct xen_page_directory *)ptr; + + if (grefs_left <= XEN_NUM_GREFS_PER_PAGE) { + to_copy = grefs_left; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; + } else { + to_copy = XEN_NUM_GREFS_PER_PAGE; + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + } + memcpy(&page_dir->gref, &buf->grefs[cur_gref], + to_copy * sizeof(grant_ref_t)); + ptr += PAGE_SIZE; + grefs_left -= to_copy; + cur_gref += to_copy; + } +} + +/** + * Grant references to the frontend's buffer pages. + * + * These will be shared with the backend, so it can + * access the buffer's data. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int guest_grant_refs_for_buffer(struct xen_front_pgdir_shbuf *buf, + grant_ref_t *priv_gref_head, + int gref_idx) +{ + int i, cur_ref, otherend_id; + + otherend_id = buf->xb_dev->otherend_id; + for (i = 0; i < buf->num_pages; i++) { + cur_ref = gnttab_claim_grant_reference(priv_gref_head); + if (cur_ref < 0) + return cur_ref; + + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, + xen_page_to_gfn(buf->pages[i]), + 0); + buf->grefs[gref_idx++] = cur_ref; + } + return 0; +} + +/** + * Grant all the references needed to share the buffer. + * + * Grant references to the page directory pages and, if + * needed, also to the pages of the shared buffer data. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int grant_references(struct xen_front_pgdir_shbuf *buf) +{ + grant_ref_t priv_gref_head; + int ret, i, j, cur_ref; + int otherend_id, num_pages_dir; + + ret = gnttab_alloc_grant_references(buf->num_grefs, &priv_gref_head); + if (ret < 0) { + dev_err(&buf->xb_dev->dev, + "Cannot allocate grant references\n"); + return ret; + } + + otherend_id = buf->xb_dev->otherend_id; + j = 0; + num_pages_dir = get_num_pages_dir(buf); + for (i = 0; i < num_pages_dir; i++) { + unsigned long frame; + + cur_ref = gnttab_claim_grant_reference(&priv_gref_head); + if (cur_ref < 0) + return cur_ref; + + frame = xen_page_to_gfn(virt_to_page(buf->directory + + PAGE_SIZE * i)); + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); + buf->grefs[j++] = cur_ref; + } + + if (buf->ops->grant_refs_for_buffer) { + ret = buf->ops->grant_refs_for_buffer(buf, &priv_gref_head, j); + if (ret) + return ret; + } + + gnttab_free_grant_references(priv_gref_head); + return 0; +} + +/** + * Allocate all required structures to mange shared buffer. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int alloc_storage(struct xen_front_pgdir_shbuf *buf) +{ + buf->grefs = kcalloc(buf->num_grefs, sizeof(*buf->grefs), GFP_KERNEL); + if (!buf->grefs) + return -ENOMEM; + + buf->directory = kcalloc(get_num_pages_dir(buf), PAGE_SIZE, GFP_KERNEL); + if (!buf->directory) + return -ENOMEM; + + return 0; +} + +/* + * For backend allocated buffers we don't need grant_refs_for_buffer + * as those grant references are allocated at backend side. + */ +static const struct xen_front_pgdir_shbuf_ops backend_ops = { + .calc_num_grefs = backend_calc_num_grefs, + .fill_page_dir = backend_fill_page_dir, + .map = backend_map, + .unmap = backend_unmap +}; + +/* + * For locally granted references we do not need to map/unmap + * the references. + */ +static const struct xen_front_pgdir_shbuf_ops local_ops = { + .calc_num_grefs = guest_calc_num_grefs, + .fill_page_dir = guest_fill_page_dir, + .grant_refs_for_buffer = guest_grant_refs_for_buffer, +}; + +/** + * Allocate a new instance of a shared buffer. + * + * \param cfg configuration to be used while allocating a new shared buffer. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_alloc(struct xen_front_pgdir_shbuf_cfg *cfg) +{ + struct xen_front_pgdir_shbuf *buf = cfg->pgdir; + int ret; + + if (cfg->be_alloc) + buf->ops = &backend_ops; + else + buf->ops = &local_ops; + buf->xb_dev = cfg->xb_dev; + buf->num_pages = cfg->num_pages; + buf->pages = cfg->pages; + + buf->ops->calc_num_grefs(buf); + + ret = alloc_storage(buf); + if (ret) + goto fail; + + ret = grant_references(buf); + if (ret) + goto fail; + + buf->ops->fill_page_dir(buf); + + return 0; + +fail: + xen_front_pgdir_shbuf_free(buf); + return ret; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_alloc); + +MODULE_DESCRIPTION("Xen frontend/backend page directory based " + "shared buffer handling"); +MODULE_AUTHOR("Oleksandr Andrushchenko"); +MODULE_LICENSE("GPL"); diff --git a/include/xen/xen-front-pgdir-shbuf.h b/include/xen/xen-front-pgdir-shbuf.h new file mode 100644 index 000000000000..150ef7ec51ec --- /dev/null +++ b/include/xen/xen-front-pgdir-shbuf.h @@ -0,0 +1,89 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen frontend/backend page directory based shared buffer + * helper module. + * + * Copyright (C) 2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> + */ + +#ifndef __XEN_FRONT_PGDIR_SHBUF_H_ +#define __XEN_FRONT_PGDIR_SHBUF_H_ + +#include <linux/kernel.h> + +#include <xen/grant_table.h> + +struct xen_front_pgdir_shbuf_ops; + +struct xen_front_pgdir_shbuf { + /* + * Number of references granted for the backend use: + * + * - for frontend allocated/imported buffers this holds the number + * of grant references for the page directory and the pages + * of the buffer + * + * - for the buffer provided by the backend this only holds the number + * of grant references for the page directory itself as grant + * references for the buffer will be provided by the backend. + */ + int num_grefs; + grant_ref_t *grefs; + /* Page directory backing storage. */ + u8 *directory; + + /* + * Number of pages for the shared buffer itself (excluding the page + * directory). + */ + int num_pages; + /* + * Backing storage of the shared buffer: these are the pages being + * shared. + */ + struct page **pages; + + struct xenbus_device *xb_dev; + + /* These are the ops used internally depending on be_alloc mode. */ + const struct xen_front_pgdir_shbuf_ops *ops; + + /* Xen map handles for the buffer allocated by the backend. */ + grant_handle_t *backend_map_handles; +}; + +struct xen_front_pgdir_shbuf_cfg { + struct xenbus_device *xb_dev; + + /* Number of pages of the buffer backing storage. */ + int num_pages; + /* Pages of the buffer to be shared. */ + struct page **pages; + + /* + * This is allocated outside because there are use-cases when + * the buffer structure is allocated as a part of a bigger one. + */ + struct xen_front_pgdir_shbuf *pgdir; + /* + * Mode of grant reference sharing: if set then backend will share + * grant references to the buffer with the frontend. + */ + int be_alloc; +}; + +int xen_front_pgdir_shbuf_alloc(struct xen_front_pgdir_shbuf_cfg *cfg); + +grant_ref_t +xen_front_pgdir_shbuf_get_dir_start(struct xen_front_pgdir_shbuf *buf); + +int xen_front_pgdir_shbuf_map(struct xen_front_pgdir_shbuf *buf); + +int xen_front_pgdir_shbuf_unmap(struct xen_front_pgdir_shbuf *buf); + +void xen_front_pgdir_shbuf_free(struct xen_front_pgdir_shbuf *buf); + +#endif /* __XEN_FRONT_PGDIR_SHBUF_H_ */ -- 2.19.1

4 16

[alsa-devel] [PATCH] ASoC: sdm845: set jack only for a specific backend
by Rohit kumar 18 Dec '18

18 Dec '18

Headset codec is connected over PRIMARY_MI2S interface. Call set_jack for codec associated with Primary Mi2s interface. Also, set_jack to NULL when jack is freed. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/sdm845.c | 31 ++++++++++++++++++++++--------- 1 file changed, 22 insertions(+), 9 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 1db8ef66..6f66a58 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -158,17 +158,24 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static void sdm845_jack_free(struct snd_jack *jack) +{ + struct snd_soc_component *component = jack->private_data; + + snd_soc_component_set_jack(component, NULL, NULL); +} + static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); - int i, rval; + struct snd_jack *jack; + int rval; if (!pdata->jack_setup) { - struct snd_jack *jack; - rval = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_HEADPHONE | @@ -190,16 +197,22 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + jack = pdata->jack.jack; + component = codec_dai->component; - component = dai->component; - rval = snd_soc_component_set_jack( - component, &pdata->jack, NULL); + jack->private_data = component; + jack->private_free = sdm845_jack_free; + rval = snd_soc_component_set_jack(component, + &pdata->jack, NULL); if (rval != 0 && rval != -ENOTSUPP) { dev_warn(card->dev, "Failed to set jack: %d\n", rval); return rval; } + break; + default: + break; } return 0; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[alsa-devel] Applied "ASoC: dma-sh7760: cleanup a debug printk" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ASoC: dma-sh7760: cleanup a debug printk has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 308424c9e0e17cff77f394d85ee0ff9b475397d4 Mon Sep 17 00:00:00 2001 From: Dan Carpenter <dan.carpenter(a)oracle.com> Date: Tue, 18 Dec 2018 11:18:10 +0300 Subject: [PATCH] ASoC: dma-sh7760: cleanup a debug printk The intent was to print the address as a hexadecimal but there is an extra "u" in the "0x%08ulx" format specification so it is displayed as decimal. Fixes: aef3b06ac697 ("[ALSA] SH7760 ASoC support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/dma-sh7760.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 2.19.0.rc2

1 0

[alsa-devel] Regarding snd_soc_googlevoicechat_soundcard drivers and arecord on Raspberry PI
by Mantas Gridinas 18 Dec '18

18 Dec '18

Hi! I've been looking into how to record audio using arecord and the following audio card extension for Raspberry PI https://pinout.xyz/pinout/raspiaudio_audio_speakers_mic#. The guide in it seems to install snd_soc_googlevoicechat_soundcard driver and after recording for some minutes it throws this error: Dec 18 15:32:17 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 7022640 bytes (18288 ms). Dec 18 15:32:17 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_soc_googlevoicehat_soundcard'. Please report this issue to the ALSA developers. Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: snd_pcm_dump(): Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: Hardware PCM card 0 'snd_rpi_googlevoicehat_soundcar' device 0 subdevice 0 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: Its setup is: Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: stream : PLAYBACK Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: access : MMAP_INTERLEAVED Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: format : S32_LE Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: subformat : STD Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: channels : 2 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: rate : 48000 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: exact rate : 48000 (48000/1) Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: msbits : 32 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: buffer_size : 65536 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: period_size : 32768 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: period_time : 682666 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: tstamp_mode : ENABLE Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: tstamp_type : MONOTONIC Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: period_step : 1 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: avail_min : 65345 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: period_event : 0 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: start_threshold : -1 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: stop_threshold : 1073741824 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: silence_threshold: 0 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: silence_size : 0 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: boundary : 1073741824 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: appl_ptr : 7938534 Dec 18 15:32:22 homepie pulseaudio[8048]: [alsa-sink-Google voiceHAT SoundCard HiFi voicehat-hifi-0] alsa-util.c: hw_ptr : 8750828 And I start the recording with the following command arecord -c 2 -r 48000 -f s32_le AlsaInfo output is available at http://www.alsa-project.org/db/?f=17a71c8e22f097d844906274a0bd78dbf124513c. Although after inspecting the provided log it seems that the Raspberry PI board is not included in the DMI, which is supposed to be BCM2835. I can't say anything else on on other details.

1 0

[alsa-devel] [GIT PULL] ASoC updates for v4.21
by Mark Brown 18 Dec '18

18 Dec '18

The following changes since commit 7566ec393f4161572ba6f11ad5171fd5d59b0fbd: Linux 4.20-rc7 (2018-12-16 15:46:55 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.21 for you to fetch changes up to a7a850dba82498a1e050d8d153cae67ce0edb3b2: Merge branch 'asoc-4.21' into asoc-next (2018-12-18 12:23:59 +0000) ---------------------------------------------------------------- ASoC: Updates for v4.21 Not much work on the core this time around but we've seen quite a bit of driver work, including on the generic DT drivers. There's also a large part of the diff from a merge of the DaVinci and OMAP directories, along with some active development there: - Preparatory work from Morimoto-san for merging the audio-graph and audio-graph-scu cards. - A merge of the TI OMAP and DaVinci directories, the OMAP product line has been merged into the DaVinci product line so there is now a lot of IP sharing which meant that the split directories just got in the way. This has pulled in a few architecture changes as well. - A big cleanup of the Maxim MAX9867 driver from Ladislav Michl. - Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S controllers. ---------------------------------------------------------------- Adrien Charruel (1): ASoC: ak4118: Add support for AK4118 S/PDIF transceiver Arnd Bergmann (5): ASoC: wm97xx: fix uninitialized regmap pointer problem ASoC: Intel: mrfld: fix uninitialized variable access ASoC: pxa: change ac97 dependencies ASoC: sdm845: add rt5663 codec select ASoC: simple-card-utils: fix build warning without CONFIG_OF Axel Lin (1): ASoC: ak5558: Remove redundant snd_soc_component_read32 calls Bard liao (2): ASoC: Intel: common: add SOF information for APL RVP ASoC: Intel: hdac_hdmi: add Icelake support Chen-Yu Tsai (2): ASoC: dt-bindings: sun50i-codec-analog: Add headphone amp regulator supply ASoC: sunxi: sun50i-codec-analog: Add support for cpvdd regulator supply Cheng-Yi Chiang (7): ASoC: rt5663: Add regulator support ASoC: rt5663: Add documentation for power supply support ASoC: rt5663: Fix error handling of regulator_set_load ASoC: qcom: sdm845: Add board specific dapm widgets ASoC: qcom: sdm845: Create and setup jack in init callback ASoC: sdm845: Add TDM configuration for speaker ASoC: sdm845: Add configuration for headset codec Clément Péron (1): ASoC: dt-bindings: add bindings for AK4118 transceiver Colin Ian King (7): ASoC: stm32: sai: fix less than zero comparison on unsigned int ASoC: amd: fix memory leak of i2s_data on error return ASoC: tlv320aic31xx: asihpi: clean up indentation, remove extraneous tab ASoC: tlv320dac33: clean up indentation, remove extraneous tab ASoC: arizona: fix indentation issue with return statement ASoC: qcom: clean up indentation, remove extraneous tab ASoC: amd: fix spelling mistake "Inavlid" -> "Invalid" Dan Carpenter (1): ASoC: amd: Fix a NULL vs IS_ERR() check in probe Daniel Mack (5): ASoC: pxa: remove raumfeld machine driver ASoC: dt-bindings: cs4270: use 'reset-gpios' rather than 'reset-gpio' ASoC: codecs: cs4270: move to GPIO consumer API ASoC: dt-bindings: ak4104: use 'reset-gpios' rather than 'reset-gpio' ASoC: codecs: ak4104: move to GPIO consumer API David Lin (2): ASoC: nau8822: convert to SPDX identifiers ASoC: nau8822: convert to SPDX identifiers Dimitris Papavasiliou (1): ASoC: pcm512x: Implement the digital_mute interface Fabio Estevam (2): ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M ASoC: fsl-sai: Fix typo in "transmitter" Fabrizio Castro (1): ASoC: rsnd: Add r8a774c0 support Hans de Goede (4): ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Clapper ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Gnawty ASoC: Intel: bytcr_rt5640: Add quirk for the Prowise PT301 tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Point of View Mobii TAB-P1005W-232 Hiroyuki Yokoyama (1): ASoC: rsnd: Add device tree support for r8a77995 Hui Wang (2): ASoC: rt5660: Add a new ACPI match ID ASoC: Intel: kbl_rt5660: Add a new machine driver for kbl with rt5660 Jenny TC (2): ASoC: dmic: introduce mode switch delay ASoC: dmic: introduce module_param wakeup_delay Jerome Brunet (5): ASoC: meson: axg-toddr: add support for spdifin backend ASoC: meson: add axg spdif input DT binding documentation ASoC: meson: add axg spdif input MAINTAINERS: Add Amlogic sound drivers entry ASoC: meson: fix do_div warning in spdifin Jiada Wang (1): ASoC: pcm3168a: Don't disable pcm3168a when CONFIG_PM defined Julia Lawall (4): ASoC: smd845: constify snd_soc_ops structure ASoC: qdsp6: q6afe-dai: constify snd_soc_dai_ops structures ASoC: codecs: constify snd_soc_dai_ops structures ASoC: AMD: constify regulator_desc structure Kirill Marinushkin (3): ASoC: pcm3060: Add DT property for single-ended output ASoC: Add documentation for pcm3060 property out-single-ended ASoC: pcm3060: Add powersaving widgets for DAC and ADC Kuninori Morimoto (65): ASoC: pcm3168a: cleanup hw_constraint for LEFT_J/I2S ASoC: rsnd: move .get_status under rsnd_mod_ops ASoC: rsnd: add .get_id/.get_id_sub ASoC: rsnd: rsnd_mod_name() handles both name and ID ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new ASoC: audio-graph-card: remove endpoint bidirectional check ASoC: audio-graph-scu-card: remove endpoint bidirectional check ASoC: audio-graph-scu-card: remove error check which never happen ASoC: rsnd: remove endpoint bidirectional check ASoC: rsnd: rename rsnd_runtime_is_ssi_xxx() ASoC: rsnd: use defined io_playback/io_capture ASoC: rsnd: move HDMI information from ssi.c to core.c ASoC: rsnd: handle DPCM converted rate/chan under core ASoC: rsnd: tidyup Document for DT binding ASoC: rsnd: add SSIU BUSIF support for Document ASoC: rsnd: add SSIU BUSIF support ASoC: rsnd: add TDM Split mode support for Document ASoC: rsnd: add TDM Split mode support ASoC: rsnd: use ring buffer for rsnd_mod_name() ASoC: rsnd: tidyup rsnd_parse_connect_ssiu_compatible() ASoC: rsnd: makes rsnd_ssi_is_dma_mode() static ASoC: simple-card-utils: remove asoc_simple_card_clk_register() ASoC: simple-card-utils: accept NULL parameter on asoc_simple_card_xxx() ASoC: simple-card-utils: tidyup asoc_simple_card_parse_convert() ASoC: simple_card_utils: remove "option" from asoc_simple_card_of_parse_routing() ASoC: soc-core: add snd_soc_of_parse_node_prefix() ASoC: simple-scu-card: tidyup "prefix" parsing on Doc ASoC: simple-scu-card: tidyup "prefix" parsing ASoC: audio-graph-scu-card: tidyup "prefix" parsing on Doc ASoC: audio-graph-scu-card: tidyup "prefix" parsing ASoC: audio-graph-scu-card: tidyup "convert-rate/channels" parsing on Doc ASoC: audio-graph-scu-card: tidyup "convert-rate/channels" parsing ASoC: audio-graph-scu-card: tidyup asoc_simple_card_parse_daifmt() timing ASoC: audio-graph-card: tidyup typo SND_AUDIO_GRAPH_CARD ASoC: audio-graph-card: tidyup define position ASoC: rsnd: tidyup for SSIU subnode ASoC: soc.h: makes snd_soc_of_parse_audio_prefix() inline ASoC: rsnd: tidyup registering method for rsnd_kctrl_new() ASoC: rsnd: indicates Channel and Mode for debug ASoC: rsnd: fixup mod ID for CTU regmap read/write ASoC: rsnd: add missing TDM Split mode support for simple-card ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting ASoC: audio-graph-scu-card: care link / dai count ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props ASoC: audio-graph-scu-card: care multi DPCM codec_conf ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props ASoC: simple-card: use cpu/codec pointer on simple_dai_props ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing ASoC: simple-scu-card: add dai-link support ASoC: simple-scu-card: care link / dai count ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props ASoC: simple-scu-card: care multi DPCM codec_conf ASoC: rsnd: remove RSND_REG_ from rsnd_reg ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID method ASoC: audio-graph-card: merge audio-graph-scu-card on Doc ASoC: audio-graph-card: merge audio-graph-scu-card ASoC: audio-graph-card: tidyup mclk-fs method ASoC: audio-graph-card: tidyup convert_rate/channel method ASoC: audio-graph-card: tidyup prefix for snd_soc_codec_conf ASoC: simple-card: merge simple-scu-card on Doc ASoC: simple-card: merge simple-scu-card ASoC: simple-card: tidyup mclk-fs method ASoC: simple-card: tidyup convert_rate/channel method ASoC: simple-card: tidyup prefix for snd_soc_codec_conf Ladislav Michl (6): ASoC: max9867: Fix whitespace ASoC: max9867: Remove useless assignment ASoC: max9867: Fix power management ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix signal paths ASoC: max9867: Add copyright and module author Lubomir Rintel (1): ASoC: dt-bindings: use a generic node name for rt5631 Mark Brown (3): Merge branch 'for-4.20' of https://git.kernel.org/.../broonie/sound into asoc-4.21 intel dep Merge branch 'asoc-4.20' into asoc-linus Merge branch 'asoc-4.21' into asoc-next Maruthi Srinivas Bayyavarapu (5): ASoC: AMD: add ACP 3.x IP register header ASoC: AMD: add ACP3.0 PCI driver dt-bindings: ASoC: xlnx, i2s: Document i2s bindings ASoC: xlnx: Add i2s driver ASoC: xlnx: enable i2s driver build Nathan Chancellor (1): ASoC: simple-card: Dereference pointer for memcpy sizeof in asoc_simple_card_probe Peter Ujfalusi (25): ASoC: omap-mcbsp: On OMAP1 and OMAP2420 only 16bit format is supported ASoC: omap: Remove unused machine driver for AM3517-evm ASoC: omap-mcbsp: Clean up dma_data addr initialization code ASoC: omap-mcbsp: Skip dma_data.maxburst initialization ASoC: omap-mcbsp: Clean up the interrupt handlers ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used ASoC: tlv320aic3x: Add support for CBM_CFS and CBS_CFM clocking modes ASoC: davinci-mcasp: Clear TXSTAT register before activating serializers ASoC: davinci-mcasp: Update PDIR (pin direction) register handling ASoC: davinci-mcasp: Document dismod optional property ASoC: davinci-mcasp: Implement configurable dismod handling dt-bindings: sound: omap-mcpdm: Update documentation for pdmclk ASoC: ti: Merge davinci and omap directories MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option ARM: OMAP2: Update for new MCBSP Kconfig option ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options ARM: omap2plus_defconfig: Update the audio options ARM: omap1_defconfig: Do not select ASoC by default ARM: davinci_all_defconfig: Update the audio options ASoC: ti: Kconfig: Remove the deprecated options Pierre-Louis Bossart (14): ASoC: acpi: define common interface for machine driver configuration ASoC: Intel: use standard interface for Hdaudio machine driver ASoC: Intel: use standard interface for Atom machine drivers ASoC: Intel: boards: fix Skylake typo ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL ASoC: Intel: common: add quirk for APL RVP boards ASoC: Intel: fix interface for Chromebook machine drivers ASoC: Intel: use platform_data for machine drivers ASoC: Intel: common: add ACPI matching tables for ICL ASoC: Intel: Skylake: Harden DSP detection with PCI class/subclass info ASoC: Intel: Skylake: stop probe if HDaudio capabilities don't exist ASoC: Intel: Skylake: remove useless tests on HDaudio capabilities ASoC: Intel: Skylake: add error logs on probe, remove dependency on NHLT ASoC: Intel: Skylake: Add more platform granularity Richard Fitzgerald (1): ASoC: wm_adsp: Factor out common init code Rob Herring (2): ASoC: Convert a few more users to using %pOFn instead of device_node.name ASoC: Use of_node_name_eq for node name comparisons Rohit kumar (7): ASoC: sdm845: Update slot_width for Quaternary TDM port ASoC: sdm845: Add support for Secondary MI2S interface ASoC: core: Invoke pcm_new() for all DAI-link ASoC: qdsp6: dt-bindings: Add q6afe display_port dt binding ASoC: qdsp6: qdafe: add support for display_port_rx ASoC: qdsp6: q6afe-dai: add support to Display port RX dais ASoC: qdsp6: q6routing: add display_port_rx port routing Ryan Lee (1): ASoC: max98373: Added max98373_reset for stable amp reset Srinivas Kandagatla (5): ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() ALSA: compress: make use of runtime buffer for copy ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload Subhransu S. Prusty (2): ASoC: Intel: Skylake: Make DSP replies more human readable ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC Takashi Iwai (1): ASoC: Intel: Skylake: Add CFL-S support Vasily Khoruzhick (2): ASoC: dt-bindings: add regulator property to simple amplifier ASoC: simple-amplifier: add VCC regulator widget Vijendar Mukunda (11): ASoC: amd: create ACP3x PCM platform device ASoC: amd: add ACP3x PCM platform driver ASoC: amd: Interrupt handler changes for ACP3x DMA driver ASoC: amd: add acp3x pcm driver dma ops ASoC: amd: add acp3x i2s ops ASoC: amd: add acp3x tdm mode support ASoC: amd: add acp3x runtime pm ops ASoC: amd: add acp3x system resume pm op ASoC: amd: enable acp3x drivers build ASoC: amd: Fixed build errors ASoC: amd: Switch to SPDX identifier Wei Yongjun (1): ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider() Yangtao Li (2): ASoC: nau8825: remove unnecessary unlikely() ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro Young_X (1): ASoC: au8540: use 64-bit arithmetic instead of 32-bit Yu Zhao (2): ASoC: use DMA addr rather than CPU pa for acp_audio_dma ASoC: use dma_ops of parent device for acp_audio_dma YueHaibing (1): ASoC: amd: remove set but not used variable 'dma_buffer' Zhuohao Lee (1): ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping Documentation/devicetree/bindings/sound/ak4104.txt | 4 +- Documentation/devicetree/bindings/sound/ak4118.txt | 22 + .../bindings/sound/amlogic,axg-spdifin.txt | 22 + .../devicetree/bindings/sound/audio-graph-card.txt | 205 ++++ .../bindings/sound/audio-graph-scu-card.txt | 4 +- Documentation/devicetree/bindings/sound/cs4270.txt | 4 +- .../bindings/sound/davinci-mcasp-audio.txt | 5 + Documentation/devicetree/bindings/sound/dmic.txt | 2 + .../devicetree/bindings/sound/fsl-sai.txt | 8 +- .../devicetree/bindings/sound/omap-mcpdm.txt | 10 + .../devicetree/bindings/sound/pcm3060.txt | 6 + .../devicetree/bindings/sound/qcom,q6asm.txt | 27 + .../devicetree/bindings/sound/renesas,rsnd.txt | 395 ++++--- Documentation/devicetree/bindings/sound/rt5631.txt | 4 +- Documentation/devicetree/bindings/sound/rt5663.txt | 6 + .../devicetree/bindings/sound/simple-amplifier.txt | 5 + .../devicetree/bindings/sound/simple-card.txt | 137 +++ .../devicetree/bindings/sound/simple-scu-card.txt | 2 +- .../bindings/sound/sun50i-codec-analog.txt | 2 + .../devicetree/bindings/sound/xlnx,i2s.txt | 28 + MAINTAINERS | 18 +- arch/arm/configs/davinci_all_defconfig | 5 +- arch/arm/configs/omap1_defconfig | 2 - arch/arm/configs/omap2plus_defconfig | 14 +- arch/arm/mach-davinci/board-dm365-evm.c | 4 +- arch/arm/mach-omap1/Makefile | 2 +- arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/pdata-quirks.c | 4 +- include/dt-bindings/sound/qcom,q6afe.h | 1 + include/linux/platform_data/davinci_asp.h | 1 + include/sound/compress_driver.h | 19 + include/sound/simple_card_utils.h | 6 +- include/sound/soc-acpi-intel-match.h | 1 + include/sound/soc-acpi.h | 15 + include/sound/soc.h | 14 +- sound/core/compress_offload.c | 18 +- sound/soc/Kconfig | 4 +- sound/soc/Makefile | 4 +- sound/soc/amd/Kconfig | 6 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-pcm-dma.c | 22 +- sound/soc/amd/acp.h | 2 +- sound/soc/amd/raven/Makefile | 6 + sound/soc/amd/raven/acp3x-pcm-dma.c | 777 ++++++++++++++ sound/soc/amd/raven/acp3x.h | 58 + sound/soc/amd/raven/chip_offset_byte.h | 639 +++++++++++ sound/soc/amd/raven/pci-acp3x.c | 156 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4104.c | 22 +- sound/soc/codecs/ak4118.c | 438 ++++++++ sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak5558.c | 19 +- sound/soc/codecs/cs4270.c | 23 +- sound/soc/codecs/dmic.c | 40 + sound/soc/codecs/hdac_hda.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 63 +- sound/soc/codecs/max98373.c | 35 +- sound/soc/codecs/max9867.c | 505 ++++----- sound/soc/codecs/max9867.h | 41 +- sound/soc/codecs/nau8540.c | 2 +- sound/soc/codecs/nau8822.c | 26 +- sound/soc/codecs/nau8822.h | 9 +- sound/soc/codecs/nau8825.c | 4 +- sound/soc/codecs/pcm3060.c | 28 +- sound/soc/codecs/pcm3060.h | 3 + sound/soc/codecs/pcm3168a.c | 40 +- sound/soc/codecs/pcm512x.c | 121 ++- sound/soc/codecs/pcm512x.h | 2 + sound/soc/codecs/rt5660.c | 1 + sound/soc/codecs/rt5663.c | 75 +- sound/soc/codecs/simple-amplifier.c | 4 + sound/soc/codecs/tas6424.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 10 + sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/codecs/wm9705.c | 10 +- sound/soc/codecs/wm9712.c | 10 +- sound/soc/codecs/wm9713.c | 10 +- sound/soc/codecs/wm_adsp.c | 14 +- sound/soc/davinci/Kconfig | 106 -- sound/soc/davinci/Makefile | 16 - sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl-asoc-card.c | 6 +- sound/soc/fsl/fsl_ssi_dbg.c | 14 +- sound/soc/generic/Kconfig | 4 +- sound/soc/generic/audio-graph-card.c | 465 +++++++-- sound/soc/generic/audio-graph-scu-card.c | 262 +++-- sound/soc/generic/simple-card-utils.c | 45 +- sound/soc/generic/simple-card.c | 402 +++++-- sound/soc/generic/simple-scu-card.c | 264 +++-- sound/soc/intel/Kconfig | 73 +- sound/soc/intel/atom/sst/sst_acpi.c | 4 + sound/soc/intel/atom/sst/sst_loader.c | 8 +- sound/soc/intel/atom/sst/sst_pvt.c | 4 +- sound/soc/intel/boards/Kconfig | 26 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcr_rt5640.c | 33 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 14 + sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 14 +- sound/soc/intel/boards/kbl_rt5660.c | 543 ++++++++++ sound/soc/intel/boards/kbl_rt5663_max98927.c | 14 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 22 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 14 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 14 +- sound/soc/intel/common/Makefile | 2 +- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 36 + sound/soc/intel/common/soc-acpi-intel-icl-match.c | 32 + sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 10 + sound/soc/intel/skylake/skl-messages.c | 8 + sound/soc/intel/skylake/skl-nhlt.c | 3 + sound/soc/intel/skylake/skl-sst-ipc.c | 50 +- sound/soc/intel/skylake/skl.c | 144 ++- sound/soc/intel/skylake/skl.h | 3 - sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/meson/Kconfig | 9 +- sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-fifo.h | 3 +- sound/soc/meson/axg-spdifin.c | 521 +++++++++ sound/soc/meson/axg-toddr.c | 15 +- sound/soc/omap/Kconfig | 129 --- sound/soc/omap/Makefile | 32 - sound/soc/omap/am3517evm.c | 141 --- sound/soc/omap/mcbsp.c | 1104 -------------------- sound/soc/pxa/Kconfig | 26 +- sound/soc/pxa/Makefile | 1 - sound/soc/pxa/raumfeld.c | 318 ------ sound/soc/qcom/Kconfig | 2 + sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 30 +- sound/soc/qcom/qdsp6/q6afe.c | 4 + sound/soc/qcom/qdsp6/q6asm-dai.c | 372 ++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/sdm845.c | 186 +++- sound/soc/sh/rcar/adg.c | 38 +- sound/soc/sh/rcar/cmd.c | 11 +- sound/soc/sh/rcar/core.c | 256 ++++- sound/soc/sh/rcar/ctu.c | 138 +-- sound/soc/sh/rcar/dma.c | 84 +- sound/soc/sh/rcar/dvc.c | 21 +- sound/soc/sh/rcar/gen.c | 49 +- sound/soc/sh/rcar/mix.c | 3 +- sound/soc/sh/rcar/rsnd.h | 382 +++---- sound/soc/sh/rcar/src.c | 67 +- sound/soc/sh/rcar/ssi.c | 269 ++--- sound/soc/sh/rcar/ssiu.c | 300 ++++-- sound/soc/soc-core.c | 13 +- sound/soc/stm/stm32_sai.c | 8 +- sound/soc/stm/stm32_sai_sub.c | 3 +- sound/soc/sunxi/sun50i-codec-analog.c | 2 + sound/soc/ti/Kconfig | 209 ++++ sound/soc/ti/Makefile | 44 + sound/soc/{omap => ti}/ams-delta.c | 0 sound/soc/{davinci => ti}/davinci-evm.c | 4 +- sound/soc/{davinci => ti}/davinci-i2s.c | 0 sound/soc/{davinci => ti}/davinci-i2s.h | 0 sound/soc/{davinci => ti}/davinci-mcasp.c | 123 ++- sound/soc/{davinci => ti}/davinci-mcasp.h | 30 +- sound/soc/{davinci => ti}/davinci-vcif.c | 0 sound/soc/{davinci => ti}/edma-pcm.c | 0 sound/soc/{davinci => ti}/edma-pcm.h | 4 +- sound/soc/{omap => ti}/n810.c | 0 sound/soc/{omap => ti}/omap-abe-twl6040.c | 0 sound/soc/{omap => ti}/omap-dmic.c | 0 sound/soc/{omap => ti}/omap-dmic.h | 0 .../soc/{omap/omap-hdmi-audio.c => ti/omap-hdmi.c} | 0 sound/soc/{omap/mcbsp.h => ti/omap-mcbsp-priv.h} | 126 +-- sound/soc/ti/omap-mcbsp-st.c | 516 +++++++++ sound/soc/{omap => ti}/omap-mcbsp.c | 857 ++++++++++++--- sound/soc/{omap => ti}/omap-mcbsp.h | 8 +- sound/soc/{omap => ti}/omap-mcpdm.c | 0 sound/soc/{omap => ti}/omap-mcpdm.h | 0 sound/soc/{omap => ti}/omap-twl4030.c | 0 sound/soc/{omap => ti}/omap3pandora.c | 0 sound/soc/{omap => ti}/osk5912.c | 0 sound/soc/{omap => ti}/rx51.c | 0 sound/soc/{omap => ti}/sdma-pcm.c | 0 sound/soc/{omap => ti}/sdma-pcm.h | 4 +- sound/soc/xilinx/Kconfig | 8 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_i2s.c | 185 ++++ 190 files changed, 9619 insertions(+), 4021 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4118.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,i2s.txt create mode 100644 sound/soc/amd/raven/Makefile create mode 100644 sound/soc/amd/raven/acp3x-pcm-dma.c create mode 100644 sound/soc/amd/raven/acp3x.h create mode 100644 sound/soc/amd/raven/chip_offset_byte.h create mode 100644 sound/soc/amd/raven/pci-acp3x.c create mode 100644 sound/soc/codecs/ak4118.c delete mode 100644 sound/soc/davinci/Kconfig delete mode 100644 sound/soc/davinci/Makefile create mode 100644 sound/soc/intel/boards/kbl_rt5660.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-icl-match.c create mode 100644 sound/soc/meson/axg-spdifin.c delete mode 100644 sound/soc/omap/Kconfig delete mode 100644 sound/soc/omap/Makefile delete mode 100644 sound/soc/omap/am3517evm.c delete mode 100644 sound/soc/omap/mcbsp.c delete mode 100644 sound/soc/pxa/raumfeld.c create mode 100644 sound/soc/ti/Kconfig create mode 100644 sound/soc/ti/Makefile rename sound/soc/{omap => ti}/ams-delta.c (100%) rename sound/soc/{davinci => ti}/davinci-evm.c (99%) rename sound/soc/{davinci => ti}/davinci-i2s.c (100%) rename sound/soc/{davinci => ti}/davinci-i2s.h (100%) rename sound/soc/{davinci => ti}/davinci-mcasp.c (94%) rename sound/soc/{davinci => ti}/davinci-mcasp.h (95%) rename sound/soc/{davinci => ti}/davinci-vcif.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.h (91%) rename sound/soc/{omap => ti}/n810.c (100%) rename sound/soc/{omap => ti}/omap-abe-twl6040.c (100%) rename sound/soc/{omap => ti}/omap-dmic.c (100%) rename sound/soc/{omap => ti}/omap-dmic.h (100%) rename sound/soc/{omap/omap-hdmi-audio.c => ti/omap-hdmi.c} (100%) rename sound/soc/{omap/mcbsp.h => ti/omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/ti/omap-mcbsp-st.c rename sound/soc/{omap => ti}/omap-mcbsp.c (51%) rename sound/soc/{omap => ti}/omap-mcbsp.h (92%) rename sound/soc/{omap => ti}/omap-mcpdm.c (100%) rename sound/soc/{omap => ti}/omap-mcpdm.h (100%) rename sound/soc/{omap => ti}/omap-twl4030.c (100%) rename sound/soc/{omap => ti}/omap3pandora.c (100%) rename sound/soc/{omap => ti}/osk5912.c (100%) rename sound/soc/{omap => ti}/rx51.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.h (85%) create mode 100644 sound/soc/xilinx/Kconfig create mode 100644 sound/soc/xilinx/Makefile create mode 100644 sound/soc/xilinx/xlnx_i2s.c

2 1

[alsa-devel] Applied "ASoC: ti: Merge davinci and omap directories" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ASoC: ti: Merge davinci and omap directories has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f2055e145f2975a75dace8e386fad9364828cdb4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:34 +0200 Subject: [PATCH] ASoC: ti: Merge davinci and omap directories Create new directory to contain all Texas Instruments specific DAI, platform and machine drivers instead of scattering them under davinci and omap directories. There is already inter dependency between the two directories becasue of McASP (on dra7x it is serviced by sDMA, not EDMA). With the upcoming AM654 we will need to introduce new platform driver for UDMA and it does not fit under davinci, nor under omap. With the move I have restructured the Kconfig to be more usable in the era of simple-sound-card: CPU DAIs can be selected individually and they will select the platform driver they can be served with. To avoid breakage, I have moved over deprecated Kconfig options so defconfig builds will work without regression. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> For sound/soc/{omap => ti}: Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 3 +- sound/soc/davinci/Kconfig | 106 ----- sound/soc/davinci/Makefile | 16 - sound/soc/omap/Kconfig | 120 ------ sound/soc/omap/Makefile | 30 -- sound/soc/ti/Kconfig | 368 ++++++++++++++++++ sound/soc/ti/Makefile | 44 +++ sound/soc/{omap => ti}/ams-delta.c | 0 sound/soc/{davinci => ti}/davinci-evm.c | 4 +- sound/soc/{davinci => ti}/davinci-i2s.c | 0 sound/soc/{davinci => ti}/davinci-i2s.h | 0 sound/soc/{davinci => ti}/davinci-mcasp.c | 14 +- sound/soc/{davinci => ti}/davinci-mcasp.h | 0 sound/soc/{davinci => ti}/davinci-vcif.c | 0 sound/soc/{davinci => ti}/edma-pcm.c | 0 sound/soc/{davinci => ti}/edma-pcm.h | 4 +- sound/soc/{omap => ti}/n810.c | 0 sound/soc/{omap => ti}/omap-abe-twl6040.c | 0 sound/soc/{omap => ti}/omap-dmic.c | 0 sound/soc/{omap => ti}/omap-dmic.h | 0 .../omap-hdmi-audio.c => ti/omap-hdmi.c} | 0 sound/soc/{omap => ti}/omap-mcbsp-priv.h | 0 sound/soc/{omap => ti}/omap-mcbsp-st.c | 0 sound/soc/{omap => ti}/omap-mcbsp.c | 0 sound/soc/{omap => ti}/omap-mcbsp.h | 0 sound/soc/{omap => ti}/omap-mcpdm.c | 0 sound/soc/{omap => ti}/omap-mcpdm.h | 0 sound/soc/{omap => ti}/omap-twl4030.c | 0 sound/soc/{omap => ti}/omap3pandora.c | 0 sound/soc/{omap => ti}/osk5912.c | 0 sound/soc/{omap => ti}/rx51.c | 0 sound/soc/{omap => ti}/sdma-pcm.c | 0 sound/soc/{omap => ti}/sdma-pcm.h | 4 +- 34 files changed, 427 insertions(+), 289 deletions(-) delete mode 100644 sound/soc/davinci/Kconfig delete mode 100644 sound/soc/davinci/Makefile delete mode 100644 sound/soc/omap/Kconfig delete mode 100644 sound/soc/omap/Makefile create mode 100644 sound/soc/ti/Kconfig create mode 100644 sound/soc/ti/Makefile rename sound/soc/{omap => ti}/ams-delta.c (100%) rename sound/soc/{davinci => ti}/davinci-evm.c (99%) rename sound/soc/{davinci => ti}/davinci-i2s.c (100%) rename sound/soc/{davinci => ti}/davinci-i2s.h (100%) rename sound/soc/{davinci => ti}/davinci-mcasp.c (99%) rename sound/soc/{davinci => ti}/davinci-mcasp.h (100%) rename sound/soc/{davinci => ti}/davinci-vcif.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.h (91%) rename sound/soc/{omap => ti}/n810.c (100%) rename sound/soc/{omap => ti}/omap-abe-twl6040.c (100%) rename sound/soc/{omap => ti}/omap-dmic.c (100%) rename sound/soc/{omap => ti}/omap-dmic.h (100%) rename sound/soc/{omap/omap-hdmi-audio.c => ti/omap-hdmi.c} (100%) rename sound/soc/{omap => ti}/omap-mcbsp-priv.h (100%) rename sound/soc/{omap => ti}/omap-mcbsp-st.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.h (100%) rename sound/soc/{omap => ti}/omap-mcpdm.c (100%) rename sound/soc/{omap => ti}/omap-mcpdm.h (100%) rename sound/soc/{omap => ti}/omap-twl4030.c (100%) rename sound/soc/{omap => ti}/omap3pandora.c (100%) rename sound/soc/{omap => ti}/osk5912.c (100%) rename sound/soc/{omap => ti}/rx51.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.h (85%) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index d21659f5a7e2..6592a422a047 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -46,13 +46,11 @@ source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" source "sound/soc/cirrus/Kconfig" -source "sound/soc/davinci/Kconfig" source "sound/soc/dwc/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/hisilicon/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" -source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" @@ -70,6 +68,7 @@ source "sound/soc/sti/Kconfig" source "sound/soc/stm/Kconfig" source "sound/soc/sunxi/Kconfig" source "sound/soc/tegra/Kconfig" +source "sound/soc/ti/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/uniphier/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index aff7b1989436..48c48c1c893c 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -30,7 +30,6 @@ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ obj-$(CONFIG_SND_SOC) += cirrus/ -obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += hisilicon/ @@ -41,7 +40,6 @@ obj-$(CONFIG_SND_SOC) += mediatek/ obj-$(CONFIG_SND_SOC) += meson/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ -obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += qcom/ @@ -54,6 +52,7 @@ obj-$(CONFIG_SND_SOC) += sti/ obj-$(CONFIG_SND_SOC) += stm/ obj-$(CONFIG_SND_SOC) += sunxi/ obj-$(CONFIG_SND_SOC) += tegra/ +obj-$(CONFIG_SND_SOC) += ti/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += uniphier/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig deleted file mode 100644 index 778faff28e0e..000000000000 --- a/sound/soc/davinci/Kconfig +++ /dev/null @@ -1,106 +0,0 @@ -config SND_DAVINCI_SOC - tristate - depends on ARCH_DAVINCI - select SND_EDMA_SOC - -config SND_EDMA_SOC - tristate "SoC Audio for Texas Instruments chips using eDMA" - depends on TI_EDMA - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M here if you want audio support for TI SoC which uses eDMA. - The following line of SoCs are supported by this platform driver: - - daVinci devices - - AM335x - - AM437x/AM438x - - DRA7xx family - -config SND_DAVINCI_SOC_I2S - tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support" - depends on SND_EDMA_SOC - help - Say Y or M here if you want to have support for McBSP IP found in - Texas Instruments DaVinci DA850 SoCs. - -config SND_DAVINCI_SOC_MCASP - tristate "Multichannel Audio Serial Port (McASP) support" - depends on SND_SDMA_SOC || SND_EDMA_SOC - help - Say Y or M here if you want to have support for McASP IP found in - various Texas Instruments SoCs like: - - daVinci devices - - Sitara line of SoCs (AM335x, AM438x, etc) - - DRA7x devices - -config SND_DAVINCI_SOC_VCIF - tristate - -config SND_DAVINCI_SOC_GENERIC_EVM - tristate - select SND_SOC_TLV320AIC3X - select SND_DAVINCI_SOC_MCASP - -config SND_AM33XX_SOC_EVM - tristate "SoC Audio for the AM33XX chip based boards" - depends on SND_EDMA_SOC && SOC_AM33XX && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - help - Say Y or M if you want to add support for SoC audio on AM33XX - boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, - AM335X-EVMSK, and BeagelBone with AudioCape boards have this - setup. - -config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on SND_EDMA_SOC && I2C - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM - select SND_DAVINCI_SOC_GENERIC_EVM - help - Say Y if you want to add support for SoC audio on TI - DaVinci DM6446, DM355 or DM365 EVM platforms. - -choice - prompt "DM365 codec select" - depends on SND_DAVINCI_SOC_EVM - depends on MACH_DAVINCI_DM365_EVM - -config SND_DM365_AIC3X_CODEC - tristate "Audio Codec - AIC3101" - help - Say Y if you want to add support for AIC3101 audio codec - -config SND_DM365_VOICE_CODEC - tristate "Voice Codec - CQ93VC" - select MFD_DAVINCI_VOICECODEC - select SND_DAVINCI_SOC_VCIF - select SND_SOC_CQ0093VC - help - Say Y if you want to add support for SoC On-chip voice codec -endchoice - -config SND_DM6467_SOC_EVM - tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - select SND_SOC_SPDIF - - help - Say Y if you want to add support for SoC audio on TI - -config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - - help - Say Y if you want to add support for SoC audio on TI - DA830/OMAP-L137 EVM - -config SND_DA850_SOC_EVM - tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - help - Say Y if you want to add support for SoC audio on TI - DA850/OMAP-L138 EVM - diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile deleted file mode 100644 index 23c6592eb31a..000000000000 --- a/sound/soc/davinci/Makefile +++ /dev/null @@ -1,16 +0,0 @@ -# SPDX-License-Identifier: GPL-2.0 -# DAVINCI Platform Support -snd-soc-edma-objs := edma-pcm.o -snd-soc-davinci-i2s-objs := davinci-i2s.o -snd-soc-davinci-mcasp-objs:= davinci-mcasp.o -snd-soc-davinci-vcif-objs:= davinci-vcif.o - -obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o -obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o -obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o -obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o - -# Generic DAVINCI/AM33xx Machine Support -snd-soc-evm-objs := davinci-evm.o - -obj-$(CONFIG_SND_DAVINCI_SOC_GENERIC_EVM) += snd-soc-evm.o diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig deleted file mode 100644 index 3d418482d242..000000000000 --- a/sound/soc/omap/Kconfig +++ /dev/null @@ -1,120 +0,0 @@ -config SND_OMAP_SOC - tristate "SoC Audio for Texas Instruments OMAP chips (deprecated)" - depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST) - select SND_SDMA_SOC - -config SND_SDMA_SOC - tristate "SoC Audio for Texas Instruments chips using sDMA" - depends on DMA_OMAP || COMPILE_TEST - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_OMAP_SOC_DMIC - tristate - -config SND_OMAP_SOC_MCBSP - tristate - -config SND_OMAP_SOC_MCPDM - tristate - -config SND_OMAP_SOC_HDMI_AUDIO - tristate "HDMI audio support for OMAP4+ based SoCs" - depends on SND_SDMA_SOC - help - For HDMI audio to work OMAPDSS HDMI support should be - enabled. - The hdmi audio driver implements cpu-dai component using the - callbacks provided by OMAPDSS and registers the component - under DSS HDMI device. Omap-pcm is registered for platform - component also under DSS HDMI device. Dummy codec is used as - as codec component. The hdmi audio driver implements also - the card and registers it under its own platform device. - The device for the driver is registered by OMAPDSS hdmi - driver. - -config SND_OMAP_SOC_N810 - tristate "SoC Audio support for Nokia N810" - depends on SND_SDMA_SOC && MACH_NOKIA_N810 && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC3X - help - Say Y if you want to add support for SoC audio on Nokia N810. - -config SND_OMAP_SOC_RX51 - tristate "SoC Audio support for Nokia N900 (RX-51)" - depends on SND_SDMA_SOC && ARM && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC3X - select SND_SOC_TPA6130A2 - depends on GPIOLIB - help - Say Y if you want to add support for SoC audio on Nokia N900 - cellphone. - -config SND_OMAP_SOC_AMS_DELTA - tristate "SoC Audio support for Amstrad E3 (Delta) videophone" - depends on SND_SDMA_SOC && MACH_AMS_DELTA && TTY - select SND_OMAP_SOC_MCBSP - select SND_SOC_CX20442 - help - Say Y if you want to add support for SoC audio device connected to - a handset and a speakerphone found on Amstrad E3 (Delta) videophone. - - Note that in order to get those devices fully supported, you have to - build the kernel with standard serial port driver included and - configured for at least 4 ports. Then, from userspace, you must load - a line discipline #19 on the modem (ttyS3) serial line. The simplest - way to achieve this is to install util-linux-ng and use the included - ldattach utility. This can be started automatically from udev, - a simple rule like this one should do the trick (it does for me): - ACTION=="add", KERNEL=="controlC0", \ - RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" - -config SND_OMAP_SOC_OSK5912 - tristate "SoC Audio support for omap osk5912" - depends on SND_SDMA_SOC && MACH_OMAP_OSK && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC23_I2C - help - Say Y if you want to add support for SoC audio on osk5912. - -config SND_OMAP_SOC_OMAP_TWL4030 - tristate "SoC Audio support for TI SoC based boards with twl4030 codec" - depends on TWL4030_CORE && SND_SDMA_SOC - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on TI SoC based boards - using twl4030 as c codec. This driver currently supports: - - Beagleboard or Devkit8000 - - Gumstix Overo or CompuLab CM-T35/CM-T3730 - - IGEP v2 - - OMAP3EVM - - SDP3430 - - Zoom2 - -config SND_OMAP_SOC_OMAP_ABE_TWL6040 - tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_SDMA_SOC && COMMON_CLK - depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST - select SND_OMAP_SOC_DMIC - select SND_OMAP_SOC_MCPDM - select SND_SOC_TWL6040 - select SND_SOC_DMIC - select COMMON_CLK_PALMAS if (SOC_OMAP5 && MFD_PALMAS) - select CLK_TWL6040 - help - Say Y if you want to add support for SoC audio on OMAP boards using - ABE and twl6040 codec. This driver currently supports: - - SDP4430/Blaze boards - - PandaBoard (4430) - - PandaBoardES (4460) - - omap5-uevm (5432) - -config SND_OMAP_SOC_OMAP3_PANDORA - tristate "SoC Audio support for OMAP3 Pandora" - depends on TWL4030_CORE && SND_SDMA_SOC && MACH_OMAP3_PANDORA - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile deleted file mode 100644 index d005338dd13c..000000000000 --- a/sound/soc/omap/Makefile +++ /dev/null @@ -1,30 +0,0 @@ -# SPDX-License-Identifier: GPL-2.0 -# OMAP Platform Support -snd-soc-sdma-objs := sdma-pcm.o -snd-soc-omap-dmic-objs := omap-dmic.o -snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o -snd-soc-omap-mcpdm-objs := omap-mcpdm.o -snd-soc-omap-hdmi-audio-objs := omap-hdmi-audio.o - -obj-$(CONFIG_SND_SDMA_SOC) += snd-soc-sdma.o -obj-$(CONFIG_SND_OMAP_SOC_DMIC) += snd-soc-omap-dmic.o -obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o -obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o -obj-$(CONFIG_SND_OMAP_SOC_HDMI_AUDIO) += snd-soc-omap-hdmi-audio.o - -# OMAP Machine Support -snd-soc-n810-objs := n810.o -snd-soc-rx51-objs := rx51.o -snd-soc-ams-delta-objs := ams-delta.o -snd-soc-osk5912-objs := osk5912.o -snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o -snd-soc-omap-twl4030-objs := omap-twl4030.o -snd-soc-omap3pandora-objs := omap3pandora.o - -obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o -obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o -obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o -obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig new file mode 100644 index 000000000000..698810c770f0 --- /dev/null +++ b/sound/soc/ti/Kconfig @@ -0,0 +1,368 @@ +menu "Audio support for Texas Instruments SoCs" +depends on DMA_OMAP || TI_EDMA || COMPILE_TEST + +config SND_SOC_TI_EDMA_PCM + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_TI_SDMA_PCM + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +comment "Texas Instruments DAI support for:" +config SND_SOC_DAVINCI_ASP + tristate "daVinci Audio Serial Port (ASP) or McBSP suport" + depends on ARCH_DAVINCI || COMPILE_TEST + select SND_SOC_TI_EDMA_PCM + help + Say Y or M here if you want audio support via daVinci ASP or McBSP. + The driver only implements the ASP support which is a subset of + daVinci McBSP (w/o the multichannel support). + +config SND_SOC_DAVINCI_MCASP + tristate "Multichannel Audio Serial Port (McASP) support" + select SND_SOC_TI_EDMA_PCM if TI_EDMA + select SND_SOC_TI_SDMA_PCM if DMA_OMAP + help + Say Y or M here if you want to have support for McASP IP found in + various Texas Instruments SoCs like: + - daVinci devices + - Sitara line of SoCs (AM335x, AM438x, etc) + - DRA7x devices + - Keystone devices + +config SND_SOC_DAVINCI_VCIF + tristate "daVinci Voice Interface (VCIF) suport" + depends on ARCH_DAVINCI || COMPILE_TEST + select SND_SOC_TI_EDMA_PCM + help + Say Y or M here if you want audio support via daVinci VCIF. + +config SND_SOC_OMAP_DMIC + tristate "Digital Microphone Module (DMIC) support" + depends on ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + help + Say Y or M here if you want to have support for DMIC IP found in + OMAP4 and OMAP5. + +config SND_OMAP_SOC_MCBSP + tristate + +config SND_SOC_OMAP_MCBSP + tristate "Multichannel Buffered Serial Port (McBSP) support" + depends on ARCH_OMAP || ARCH_OMAP1 || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + select SND_OMAP_SOC_MCBSP + help + Say Y or M here if you want to have support for McBSP IP found in + Texas Instruments OMAP1/2/3/4/5 SoCs. + +config SND_SOC_OMAP_MCPDM + tristate "Multichannel PDM Controller (McPDM) support" + depends on ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + help + Say Y or M here if you want to have support for McPDM IP found in + OMAP4 and OMAP5. + +comment "Audio support for boards with Texas Instruments SoCs" +config SND_SOC_NOKIA_N810 + tristate "SoC Audio support for Nokia N810" + depends on MACH_NOKIA_N810 && I2C + select SND_SOC_OMAP_MCBSP + select SND_SOC_TLV320AIC3X + help + Say Y or M if you want to add support for SoC audio on Nokia N810. + +config SND_SOC_NOKIA_RX51 + tristate "SoC Audio support for Nokia RX-51" + depends on ARCH_OMAP3 && I2C && GPIOLIB + select SND_SOC_OMAP_MCBSP + select SND_SOC_TLV320AIC3X + select SND_SOC_TPA6130A2 + help + Say Y or M if you want to add support for SoC audio on Nokia RX-51 + hardware. This is also known as Nokia N900 product. + +config SND_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on ARCH_OMAP3 + depends on TWL4030_CORE + select SND_SOC_OMAP_MCBSP + select SND_SOC_TWL4030 + help + Say Y or M if you want to add support for SoC audio on the OMAP3 Pandora. + +config SND_SOC_OMAP3_TWL4030 + tristate "SoC Audio support for OMAP3 based boards with twl4030 codec" + depends on ARCH_OMAP3 || COMPILE_TEST + depends on TWL4030_CORE + select SND_SOC_OMAP_MCBSP + select SND_SOC_TWL4030 + help + Say Y or M if you want to add support for SoC audio on OMAP3 based + boards using twl4030 as codec. This driver currently supports: + - Beagleboard or Devkit8000 + - Gumstix Overo or CompuLab CM-T35/CM-T3730 + - IGEP v2 + - OMAP3EVM + - SDP3430 + - Zoom2 + +config SND_SOC_OMAP_ABE_TWL6040 + tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" + depends on TWL6040_CORE && COMMON_CLK + depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST + select SND_SOC_OMAP_DMIC + select SND_SOC_OMAP_MCPDM + select SND_SOC_TWL6040 + help + Say Y or M if you want to add support for SoC audio on OMAP boards + using ABE and twl6040 codec. This driver currently supports: + - SDP4430/Blaze boards + - PandaBoard (4430) + - PandaBoardES (4460) + - OMAP5 uEVM + +config SND_SOC_OMAP_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on MACH_AMS_DELTA && TTY + select SND_SOC_OMAP_MCBSP + select SND_SOC_CX20442 + help + Say Y or M if you want to add support for SoC audio device + connected to a handset and a speakerphone found on Amstrad E3 (Delta) + videophone. + + Note that in order to get those devices fully supported, you have to + build the kernel with standard serial port driver included and + configured for at least 4 ports. Then, from userspace, you must load + a line discipline #19 on the modem (ttyS3) serial line. The simplest + way to achieve this is to install util-linux-ng and use the included + ldattach utility. This can be started automatically from udev, + a simple rule like this one should do the trick (it does for me): + ACTION=="add", KERNEL=="controlC0", \ + RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" + +config SND_SOC_OMAP_HDMI + tristate "OMAP4/5 HDMI audio support" + depends on OMAP4_DSS_HDMI || OMAP5_DSS_HDMI || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + help + For HDMI audio to work OMAPDSS HDMI support should be + enabled. + The hdmi audio driver implements cpu-dai component using the + callbacks provided by OMAPDSS and registers the component + under DSS HDMI device. Omap-pcm is registered for platform + component also under DSS HDMI device. Dummy codec is used as + as codec component. The hdmi audio driver implements also + the card and registers it under its own platform device. + The device for the driver is registered by OMAPDSS hdmi + driver. + +config SND_SOC_OMAP_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on MACH_OMAP_OSK && I2C + select SND_SOC_OMAP_MCBSP + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M if you want to add support for SoC audio on osk5912. + +config SND_SOC_DAVINCI_EVM + tristate "SoC Audio support for DaVinci EVMs" + depends on ARCH_DAVINCI && I2C + select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_DM355_EVM + select SND_SOC_DAVINCI_ASP if SND_SOC_DM365_AIC3X_CODEC + select SND_SOC_DAVINCI_VCIF if SND_SOC_DM365_VOICE_CODEC + select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_EVM # DM6446 + select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DM6467_EVM + select SND_SOC_SPDIF if MACH_DAVINCI_DM6467_EVM + select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA830_EVM + select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA850_EVM + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on the following TI + DaVinci EVM platforms: + - DM355 + - DM365 + - DM6446 + - DM6447 + - DM830 + - DM850 + +choice + prompt "DM365 codec select" + depends on SND_SOC_DAVINCI_EVM + depends on MACH_DAVINCI_DM365_EVM + +config SND_SOC_DM365_AIC3X_CODEC + bool "Audio Codec - AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec + +config SND_SOC_DM365_VOICE_CODEC + bool "Voice Codec - CQ93VC" + select MFD_DAVINCI_VOICECODEC + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec +endchoice + +menu "Deprecated kconfig options:" + +config SND_AM33XX_SOC_EVM + tristate "SoC Audio for the AM33XX chip based boards" + depends on SOC_AM33XX && I2C + select SND_SOC_TLV320AIC3X + select SND_SOC_DAVINCI_MCASP + help + Say Y or M if you want to add support for SoC audio on AM33XX + boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, + AM335X-EVMSK, and BeagelBone with AudioCape boards have this + setup. + +config SND_DAVINCI_SOC_EVM + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" + depends on ARCH_DAVINCI && I2C + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + select SND_SOC_DAVINCI_EVM + help + Say Y if you want to add support for SoC audio on TI + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + +config SND_DM365_AIC3X_CODEC + tristate "Audio Codec - AIC3101" + select SND_SOC_DM365_AIC3X_CODEC + help + Say Y if you want to add support for AIC3101 audio codec + +config SND_DM365_VOICE_CODEC + tristate "Voice Codec - CQ93VC" + select SND_SOC_DM365_VOICE_CODEC + help + Say Y if you want to add support for SoC On-chip voice codec +endchoice + +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on MACH_DAVINCI_DM6467_EVM && I2C + select SND_SOC_DAVINCI_EVM + + help + Say Y if you want to add support for SoC audio on TI + +config SND_DA830_SOC_EVM + tristate "SoC Audio support for DA830/OMAP-L137 EVM" + depends on MACH_DAVINCI_DA830_EVM && I2C + select SND_SOC_DAVINCI_EVM + + help + Say Y if you want to add support for SoC audio on TI + DA830/OMAP-L137 EVM + +config SND_DA850_SOC_EVM + tristate "SoC Audio support for DA850/OMAP-L138 EVM" + depends on MACH_DAVINCI_DA850_EVM && I2C + select SND_SOC_DAVINCI_EVM + help + Say Y if you want to add support for SoC audio on TI + DA850/OMAP-L138 EVM + +config SND_OMAP_SOC_HDMI_AUDIO + tristate "HDMI audio support for OMAP4+ based SoCs" + select SND_SOC_OMAP_HDMI + help + For HDMI audio to work OMAPDSS HDMI support should be + enabled. + The hdmi audio driver implements cpu-dai component using the + callbacks provided by OMAPDSS and registers the component + under DSS HDMI device. Omap-pcm is registered for platform + component also under DSS HDMI device. Dummy codec is used as + as codec component. The hdmi audio driver implements also + the card and registers it under its own platform device. + The device for the driver is registered by OMAPDSS hdmi + driver. + +config SND_OMAP_SOC_N810 + tristate "SoC Audio support for Nokia N810" + depends on MACH_NOKIA_N810 && I2C + select SND_SOC_NOKIA_N810 + help + Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_RX51 + tristate "SoC Audio support for Nokia N900 (RX-51)" + depends on ARM && I2C && GPIOLIB && ARCH_OMAP3 + select SND_SOC_NOKIA_RX51 + help + Say Y if you want to add support for SoC audio on Nokia N900 + cellphone. + +config SND_OMAP_SOC_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on MACH_AMS_DELTA && TTY + select SND_SOC_OMAP_AMS_DELTA + help + Say Y if you want to add support for SoC audio device connected to + a handset and a speakerphone found on Amstrad E3 (Delta) videophone. + + Note that in order to get those devices fully supported, you have to + build the kernel with standard serial port driver included and + configured for at least 4 ports. Then, from userspace, you must load + a line discipline #19 on the modem (ttyS3) serial line. The simplest + way to achieve this is to install util-linux-ng and use the included + ldattach utility. This can be started automatically from udev, + a simple rule like this one should do the trick (it does for me): + ACTION=="add", KERNEL=="controlC0", \ + RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" + +config SND_OMAP_SOC_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on MACH_OMAP_OSK && I2C + select SND_SOC_OMAP_OSK5912 + help + Say Y if you want to add support for SoC audio on osk5912. + +config SND_OMAP_SOC_OMAP_TWL4030 + tristate "SoC Audio support for TI SoC based boards with twl4030 codec" + depends on TWL4030_CORE + select SND_SOC_OMAP3_TWL4030 + help + Say Y if you want to add support for SoC audio on TI SoC based boards + using twl4030 as c codec. This driver currently supports: + - Beagleboard or Devkit8000 + - Gumstix Overo or CompuLab CM-T35/CM-T3730 + - IGEP v2 + - OMAP3EVM + - SDP3430 + - Zoom2 + +config SND_OMAP_SOC_OMAP_ABE_TWL6040 + tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" + depends on TWL6040_CORE && COMMON_CLK + depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST + select SND_SOC_OMAP_ABE_TWL6040 + help + Say Y if you want to add support for SoC audio on OMAP boards using + ABE and twl6040 codec. This driver currently supports: + - SDP4430/Blaze boards + - PandaBoard (4430) + - PandaBoardES (4460) + - omap5-uevm (5432) + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE + select SND_SOC_OMAP3_PANDORA + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. +endmenu + +endmenu + diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile new file mode 100644 index 000000000000..08c44d56ef3e --- /dev/null +++ b/sound/soc/ti/Makefile @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: GPL-2.0 + +# Platform drivers +snd-soc-ti-edma-objs := edma-pcm.o +snd-soc-ti-sdma-objs := sdma-pcm.o + +obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o +obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o + +# CPU DAI drivers +snd-soc-davinci-asp-objs := davinci-i2s.o +snd-soc-davinci-mcasp-objs := davinci-mcasp.o +snd-soc-davinci-vcif-objs := davinci-vcif.o +snd-soc-omap-dmic-objs := omap-dmic.o +snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o + +obj-$(CONFIG_SND_SOC_DAVINCI_ASP) += snd-soc-davinci-asp.o +obj-$(CONFIG_SND_SOC_DAVINCI_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_SOC_DAVINCI_VCIF) += snd-soc-davinci-vcif.o +obj-$(CONFIG_SND_SOC_OMAP_DMIC) += snd-soc-omap-dmic.o +obj-$(CONFIG_SND_SOC_OMAP_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_SOC_OMAP_MCPDM) += snd-soc-omap-mcpdm.o + +# Machine drivers +snd-soc-davinci-evm-objs := davinci-evm.o +snd-soc-n810-objs := n810.o +snd-soc-rx51-objs := rx51.o +snd-soc-omap3pandora-objs := omap3pandora.o +snd-soc-omap-twl4030-objs := omap-twl4030.o +snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o +snd-soc-ams-delta-objs := ams-delta.o +snd-soc-omap-hdmi-objs := omap-hdmi.o +snd-soc-osk5912-objs := osk5912.o + +obj-$(CONFIG_SND_SOC_DAVINCI_EVM) += snd-soc-davinci-evm.o +obj-$(CONFIG_SND_SOC_NOKIA_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_SOC_NOKIA_RX51) += snd-soc-rx51.o +obj-$(CONFIG_SND_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o +obj-$(CONFIG_SND_SOC_OMAP3_TWL4030) += snd-soc-omap-twl4030.o +obj-$(CONFIG_SND_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o +obj-$(CONFIG_SND_SOC_OMAP_AMS_DELTA) += snd-soc-ams-delta.o +obj-$(CONFIG_SND_SOC_OMAP_HDMI) += snd-soc-omap-hdmi.o +obj-$(CONFIG_SND_SOC_OMAP_OSK5912) += snd-soc-osk5912.o diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/ti/ams-delta.c similarity index 100% rename from sound/soc/omap/ams-delta.c rename to sound/soc/ti/ams-delta.c diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/ti/davinci-evm.c similarity index 99% rename from sound/soc/davinci/davinci-evm.c rename to sound/soc/ti/davinci-evm.c index 7a369e0f2093..4869d6311510 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -170,7 +170,7 @@ static struct snd_soc_dai_link dm355_evm_dai = { }; static struct snd_soc_dai_link dm365_evm_dai = { -#ifdef CONFIG_SND_DM365_AIC3X_CODEC +#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", @@ -181,7 +181,7 @@ static struct snd_soc_dai_link dm365_evm_dai = { .ops = &evm_ops, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, -#elif defined(CONFIG_SND_DM365_VOICE_CODEC) +#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", .stream_name = "CQ93", .cpu_dai_name = "davinci-vcif", diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c similarity index 100% rename from sound/soc/davinci/davinci-i2s.c rename to sound/soc/ti/davinci-i2s.c diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/ti/davinci-i2s.h similarity index 100% rename from sound/soc/davinci/davinci-i2s.h rename to sound/soc/ti/davinci-i2s.h diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c similarity index 99% rename from sound/soc/davinci/davinci-mcasp.c rename to sound/soc/ti/davinci-mcasp.c index 40d3a916fb74..eeda6d5565bc 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -39,7 +39,7 @@ #include <sound/dmaengine_pcm.h> #include "edma-pcm.h" -#include "../omap/sdma-pcm.h" +#include "sdma-pcm.h" #include "davinci-mcasp.h" #define MCASP_MAX_AFIFO_DEPTH 64 @@ -2149,9 +2149,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = davinci_mcasp_get_dma_type(mcasp); switch (ret) { case PCM_EDMA: -#if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_EDMA_SOC)) +#if IS_BUILTIN(CONFIG_SND_SOC_TI_EDMA_PCM) || \ + (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ + IS_MODULE(CONFIG_SND_SOC_TI_EDMA_PCM)) ret = edma_pcm_platform_register(&pdev->dev); #else dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); @@ -2160,9 +2160,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) #endif break; case PCM_SDMA: -#if IS_BUILTIN(CONFIG_SND_SDMA_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_SDMA_SOC)) +#if IS_BUILTIN(CONFIG_SND_SOC_TI_SDMA_PCM) || \ + (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ + IS_MODULE(CONFIG_SND_SOC_TI_SDMA_PCM)) ret = sdma_pcm_platform_register(&pdev->dev, NULL, NULL); #else dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/ti/davinci-mcasp.h similarity index 100% rename from sound/soc/davinci/davinci-mcasp.h rename to sound/soc/ti/davinci-mcasp.h diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c similarity index 100% rename from sound/soc/davinci/davinci-vcif.c rename to sound/soc/ti/davinci-vcif.c diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/ti/edma-pcm.c similarity index 100% rename from sound/soc/davinci/edma-pcm.c rename to sound/soc/ti/edma-pcm.c diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/ti/edma-pcm.h similarity index 91% rename from sound/soc/davinci/edma-pcm.h rename to sound/soc/ti/edma-pcm.h index b0957744851c..8058bdb0f032 100644 --- a/sound/soc/davinci/edma-pcm.h +++ b/sound/soc/ti/edma-pcm.h @@ -20,13 +20,13 @@ #ifndef __EDMA_PCM_H__ #define __EDMA_PCM_H__ -#if IS_ENABLED(CONFIG_SND_EDMA_SOC) +#if IS_ENABLED(CONFIG_SND_SOC_TI_EDMA_PCM) int edma_pcm_platform_register(struct device *dev); #else static inline int edma_pcm_platform_register(struct device *dev) { return 0; } -#endif /* CONFIG_SND_EDMA_SOC */ +#endif /* CONFIG_SND_SOC_TI_EDMA_PCM */ #endif /* __EDMA_PCM_H__ */ diff --git a/sound/soc/omap/n810.c b/sound/soc/ti/n810.c similarity index 100% rename from sound/soc/omap/n810.c rename to sound/soc/ti/n810.c diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c similarity index 100% rename from sound/soc/omap/omap-abe-twl6040.c rename to sound/soc/ti/omap-abe-twl6040.c diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/ti/omap-dmic.c similarity index 100% rename from sound/soc/omap/omap-dmic.c rename to sound/soc/ti/omap-dmic.c diff --git a/sound/soc/omap/omap-dmic.h b/sound/soc/ti/omap-dmic.h similarity index 100% rename from sound/soc/omap/omap-dmic.h rename to sound/soc/ti/omap-dmic.h diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/ti/omap-hdmi.c similarity index 100% rename from sound/soc/omap/omap-hdmi-audio.c rename to sound/soc/ti/omap-hdmi.c diff --git a/sound/soc/omap/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h similarity index 100% rename from sound/soc/omap/omap-mcbsp-priv.h rename to sound/soc/ti/omap-mcbsp-priv.h diff --git a/sound/soc/omap/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c similarity index 100% rename from sound/soc/omap/omap-mcbsp-st.c rename to sound/soc/ti/omap-mcbsp-st.c diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c similarity index 100% rename from sound/soc/omap/omap-mcbsp.c rename to sound/soc/ti/omap-mcbsp.c diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/ti/omap-mcbsp.h similarity index 100% rename from sound/soc/omap/omap-mcbsp.h rename to sound/soc/ti/omap-mcbsp.h diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c similarity index 100% rename from sound/soc/omap/omap-mcpdm.c rename to sound/soc/ti/omap-mcpdm.c diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/ti/omap-mcpdm.h similarity index 100% rename from sound/soc/omap/omap-mcpdm.h rename to sound/soc/ti/omap-mcpdm.h diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c similarity index 100% rename from sound/soc/omap/omap-twl4030.c rename to sound/soc/ti/omap-twl4030.c diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/ti/omap3pandora.c similarity index 100% rename from sound/soc/omap/omap3pandora.c rename to sound/soc/ti/omap3pandora.c diff --git a/sound/soc/omap/osk5912.c b/sound/soc/ti/osk5912.c similarity index 100% rename from sound/soc/omap/osk5912.c rename to sound/soc/ti/osk5912.c diff --git a/sound/soc/omap/rx51.c b/sound/soc/ti/rx51.c similarity index 100% rename from sound/soc/omap/rx51.c rename to sound/soc/ti/rx51.c diff --git a/sound/soc/omap/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c similarity index 100% rename from sound/soc/omap/sdma-pcm.c rename to sound/soc/ti/sdma-pcm.c diff --git a/sound/soc/omap/sdma-pcm.h b/sound/soc/ti/sdma-pcm.h similarity index 85% rename from sound/soc/omap/sdma-pcm.h rename to sound/soc/ti/sdma-pcm.h index 34a7f90b2587..cb0627c8dd34 100644 --- a/sound/soc/omap/sdma-pcm.h +++ b/sound/soc/ti/sdma-pcm.h @@ -7,7 +7,7 @@ #ifndef __SDMA_PCM_H__ #define __SDMA_PCM_H__ -#if IS_ENABLED(CONFIG_SND_SDMA_SOC) +#if IS_ENABLED(CONFIG_SND_SOC_TI_SDMA_PCM) int sdma_pcm_platform_register(struct device *dev, char *txdmachan, char *rxdmachan); #else @@ -16,6 +16,6 @@ static inline int sdma_pcm_platform_register(struct device *dev, { return -ENODEV; } -#endif /* CONFIG_SND_SDMA_SOC */ +#endif /* CONFIG_SND_SOC_TI_SDMA_PCM */ #endif /* __SDMA_PCM_H__ */ -- 2.19.0.rc2

1 0

[alsa-devel] Applied "MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ca1c4d6535249ad43068834687f78d0db82d9e2e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:35 +0200 Subject: [PATCH] MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support The sound/soc/davinci and omap directories are merged under sound/soc/ti. Add myself as maintainer and update the OMAP audio support section with the new path and file names. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index ef17863af710..a6068f1f4b81 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10710,7 +10710,10 @@ M: Jarkko Nikula <jarkko.nikula(a)bitmer.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linux-omap(a)vger.kernel.org S: Maintained -F: sound/soc/omap/ +F: sound/soc/ti/omap* +F: sound/soc/ti/rx51.c +F: sound/soc/ti/n810.c +F: sound/soc/ti/sdma-pcm.* OMAP CLOCK FRAMEWORK SUPPORT M: Paul Walmsley <paul(a)pwsan.com> @@ -14737,6 +14740,12 @@ F: Documentation/devicetree/bindings/clock/ti,sci-clk.txt F: drivers/clk/keystone/sci-clk.c F: drivers/reset/reset-ti-sci.c +Texas Instruments ASoC drivers +M: Peter Ujfalusi <peter.ujfalusi(a)ti.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/ti/ + THANKO'S RAREMONO AM/FM/SW RADIO RECEIVER USB DRIVER M: Hans Verkuil <hverkuil(a)xs4all.nl> L: linux-media(a)vger.kernel.org -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c27ace2e95a29c3b359123363d4f88864452917d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:36 +0200 Subject: [PATCH] ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option The MCBSP config option has been changed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/arm/mach-omap1/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-omap1/Makefile b/arch/arm/mach-omap1/Makefile index e8ccf51c6f29..a7e9c6d19fb5 100644 --- a/arch/arm/mach-omap1/Makefile +++ b/arch/arm/mach-omap1/Makefile @@ -8,7 +8,7 @@ obj-y := io.o id.o sram-init.o sram.o time.o irq.o mux.o flash.o \ serial.o devices.o dma.o fb.o obj-y += clock.o clock_data.o opp_data.o reset.o pm_bus.o timer.o -ifneq ($(CONFIG_SND_OMAP_SOC_MCBSP),) +ifneq ($(CONFIG_SND_SOC_OMAP_MCBSP),) obj-y += mcbsp.o endif -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ARM: OMAP2: Update for new MCBSP Kconfig option" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ARM: OMAP2: Update for new MCBSP Kconfig option has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 558eb0bfb271323cfd9f864f34d3253a65f352dd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:37 +0200 Subject: [PATCH] ARM: OMAP2: Update for new MCBSP Kconfig option The MCBSP config option has been changed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/pdata-quirks.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/arch/arm/mach-omap2/Makefile b/arch/arm/mach-omap2/Makefile index 01377c292db4..899c60fac159 100644 --- a/arch/arm/mach-omap2/Makefile +++ b/arch/arm/mach-omap2/Makefile @@ -24,7 +24,7 @@ obj-$(CONFIG_SOC_OMAP5) += $(hwmod-common) $(secure-common) obj-$(CONFIG_SOC_AM43XX) += $(hwmod-common) $(secure-common) obj-$(CONFIG_SOC_DRA7XX) += $(hwmod-common) $(secure-common) -ifneq ($(CONFIG_SND_OMAP_SOC_MCBSP),) +ifneq ($(CONFIG_SND_SOC_OMAP_MCBSP),) obj-y += mcbsp.o endif diff --git a/arch/arm/mach-omap2/pdata-quirks.c b/arch/arm/mach-omap2/pdata-quirks.c index 9fec5f84bf77..8a5b6ed4ec36 100644 --- a/arch/arm/mach-omap2/pdata-quirks.c +++ b/arch/arm/mach-omap2/pdata-quirks.c @@ -524,7 +524,7 @@ void omap_auxdata_legacy_init(struct device *dev) dev->platform_data = &twl_gpio_auxdata; } -#if IS_ENABLED(CONFIG_SND_OMAP_SOC_MCBSP) +#if IS_ENABLED(CONFIG_SND_SOC_OMAP_MCBSP) static struct omap_mcbsp_platform_data mcbsp_pdata; static void __init omap3_mcbsp_init(void) { @@ -572,7 +572,7 @@ static struct of_dev_auxdata omap_auxdata_lookup[] = { OF_DEV_AUXDATA("ti,am3517-emac", 0x5c000000, "davinci_emac.0", &am35xx_emac_pdata), /* McBSP modules with sidetone core */ -#if IS_ENABLED(CONFIG_SND_OMAP_SOC_MCBSP) +#if IS_ENABLED(CONFIG_SND_SOC_OMAP_MCBSP) OF_DEV_AUXDATA("ti,omap3-mcbsp", 0x49022000, "49022000.mcbsp", &mcbsp_pdata), OF_DEV_AUXDATA("ti,omap3-mcbsp", 0x49024000, "49024000.mcbsp", &mcbsp_pdata), #endif -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4d8c1e7efb9f915034c3dba2bc85ed1f8509ae0c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:38 +0200 Subject: [PATCH] ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options The codec selection config options for DMA365 has been changed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 8143756ff38b..09e439d4abf5 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -794,9 +794,9 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); -#ifdef CONFIG_SND_DM365_AIC3X_CODEC +#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC dm365_init_asp(); -#elif defined(CONFIG_SND_DM365_VOICE_CODEC) +#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC) dm365_init_vc(); #endif dm365_init_rtc(); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ARM: omap2plus_defconfig: Update the audio options" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ARM: omap2plus_defconfig: Update the audio options has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3162b05fb74cd107b2a66f20485c62514b20048f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:39 +0200 Subject: [PATCH] ARM: omap2plus_defconfig: Update the audio options Use the new options for default audio support and enable Nokia n900 audio as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/arm/configs/omap2plus_defconfig | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/arch/arm/configs/omap2plus_defconfig b/arch/arm/configs/omap2plus_defconfig index 6491419b1dad..2274e45623f9 100644 --- a/arch/arm/configs/omap2plus_defconfig +++ b/arch/arm/configs/omap2plus_defconfig @@ -381,13 +381,13 @@ CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_USB_AUDIO=m CONFIG_SND_SOC=m -CONFIG_SND_EDMA_SOC=m -CONFIG_SND_AM33XX_SOC_EVM=m -CONFIG_SND_OMAP_SOC=m -CONFIG_SND_OMAP_SOC_HDMI_AUDIO=m -CONFIG_SND_OMAP_SOC_OMAP_TWL4030=m -CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040=m -CONFIG_SND_OMAP_SOC_OMAP3_PANDORA=m +CONFIG_SND_SOC_TLV320AIC3X=m +CONFIG_SND_SOC_DAVINCI_MCASP=m +CONFIG_SND_SOC_NOKIA_RX51=m +CONFIG_SND_SOC_OMAP_HDMI=m +CONFIG_SND_SOC_OMAP_ABE_TWL6040=m +CONFIG_SND_SOC_OMAP3_PANDORA=m +CONFIG_SND_SOC_OMAP3_TWL4030=m CONFIG_SND_SOC_CPCAP=m CONFIG_SND_SIMPLE_CARD=m CONFIG_SND_AUDIO_GRAPH_CARD=m -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ARM: omap1_defconfig: Do not select ASoC by default" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ARM: omap1_defconfig: Do not select ASoC by default has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From eab5b50a13de4717cad5eda3d26d889f2f4ea37d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:40 +0200 Subject: [PATCH] ARM: omap1_defconfig: Do not select ASoC by default No audio support is enabled by default via ASoC, disable the code as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/arm/configs/omap1_defconfig | 2 -- 1 file changed, 2 deletions(-) diff --git a/arch/arm/configs/omap1_defconfig b/arch/arm/configs/omap1_defconfig index 72f4bc83f467..cfc00b0961ec 100644 --- a/arch/arm/configs/omap1_defconfig +++ b/arch/arm/configs/omap1_defconfig @@ -175,8 +175,6 @@ CONFIG_SND_PCM_OSS=y # CONFIG_SND_VERBOSE_PROCFS is not set CONFIG_SND_DUMMY=y CONFIG_SND_USB_AUDIO=y -CONFIG_SND_SOC=y -CONFIG_SND_OMAP_SOC=y # CONFIG_USB_HID is not set CONFIG_USB=y CONFIG_USB_PHY=y -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ARM: davinci_all_defconfig: Update the audio options" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ARM: davinci_all_defconfig: Update the audio options has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 41e95652ee225f121d38916d8d6849e3c15a924a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:41 +0200 Subject: [PATCH] ARM: davinci_all_defconfig: Update the audio options Use the new options for default audio support. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/arm/configs/davinci_all_defconfig | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/arch/arm/configs/davinci_all_defconfig b/arch/arm/configs/davinci_all_defconfig index 07b2eadac3dd..207962a656a2 100644 --- a/arch/arm/configs/davinci_all_defconfig +++ b/arch/arm/configs/davinci_all_defconfig @@ -167,8 +167,9 @@ CONFIG_SOUND=m CONFIG_SND=m CONFIG_SND_USB_AUDIO=m CONFIG_SND_SOC=m -CONFIG_SND_EDMA_SOC=m -CONFIG_SND_DA850_SOC_EVM=m +CONFIG_SND_SOC_TLV320AIC3X=m +CONFIG_SND_SOC_DAVINCI_MCASP=m +CONFIG_SND_SOC_DAVINCI_EVM=m CONFIG_SND_SIMPLE_CARD=m CONFIG_HID=m CONFIG_HID_A4TECH=m -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: ti: Kconfig: Remove the deprecated options" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ASoC: ti: Kconfig: Remove the deprecated options has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0f4967bc21de0bb9e107f9ad3ced852daf66acd2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:42 +0200 Subject: [PATCH] ASoC: ti: Kconfig: Remove the deprecated options We no longer have these options used anywhere. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/ti/Kconfig | 159 ------------------------------------------- 1 file changed, 159 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 698810c770f0..4bf3c15d4e51 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -46,14 +46,10 @@ config SND_SOC_OMAP_DMIC Say Y or M here if you want to have support for DMIC IP found in OMAP4 and OMAP5. -config SND_OMAP_SOC_MCBSP - tristate - config SND_SOC_OMAP_MCBSP tristate "Multichannel Buffered Serial Port (McBSP) support" depends on ARCH_OMAP || ARCH_OMAP1 || COMPILE_TEST select SND_SOC_TI_SDMA_PCM - select SND_OMAP_SOC_MCBSP help Say Y or M here if you want to have support for McBSP IP found in Texas Instruments OMAP1/2/3/4/5 SoCs. @@ -209,160 +205,5 @@ config SND_SOC_DM365_VOICE_CODEC Say Y if you want to add support for SoC On-chip voice codec endchoice -menu "Deprecated kconfig options:" - -config SND_AM33XX_SOC_EVM - tristate "SoC Audio for the AM33XX chip based boards" - depends on SOC_AM33XX && I2C - select SND_SOC_TLV320AIC3X - select SND_SOC_DAVINCI_MCASP - help - Say Y or M if you want to add support for SoC audio on AM33XX - boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, - AM335X-EVMSK, and BeagelBone with AudioCape boards have this - setup. - -config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on ARCH_DAVINCI && I2C - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM - select SND_SOC_DAVINCI_EVM - help - Say Y if you want to add support for SoC audio on TI - DaVinci DM6446, DM355 or DM365 EVM platforms. - -choice - prompt "DM365 codec select" - depends on SND_DAVINCI_SOC_EVM - depends on MACH_DAVINCI_DM365_EVM - -config SND_DM365_AIC3X_CODEC - tristate "Audio Codec - AIC3101" - select SND_SOC_DM365_AIC3X_CODEC - help - Say Y if you want to add support for AIC3101 audio codec - -config SND_DM365_VOICE_CODEC - tristate "Voice Codec - CQ93VC" - select SND_SOC_DM365_VOICE_CODEC - help - Say Y if you want to add support for SoC On-chip voice codec -endchoice - -config SND_DM6467_SOC_EVM - tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on MACH_DAVINCI_DM6467_EVM && I2C - select SND_SOC_DAVINCI_EVM - - help - Say Y if you want to add support for SoC audio on TI - -config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on MACH_DAVINCI_DA830_EVM && I2C - select SND_SOC_DAVINCI_EVM - - help - Say Y if you want to add support for SoC audio on TI - DA830/OMAP-L137 EVM - -config SND_DA850_SOC_EVM - tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on MACH_DAVINCI_DA850_EVM && I2C - select SND_SOC_DAVINCI_EVM - help - Say Y if you want to add support for SoC audio on TI - DA850/OMAP-L138 EVM - -config SND_OMAP_SOC_HDMI_AUDIO - tristate "HDMI audio support for OMAP4+ based SoCs" - select SND_SOC_OMAP_HDMI - help - For HDMI audio to work OMAPDSS HDMI support should be - enabled. - The hdmi audio driver implements cpu-dai component using the - callbacks provided by OMAPDSS and registers the component - under DSS HDMI device. Omap-pcm is registered for platform - component also under DSS HDMI device. Dummy codec is used as - as codec component. The hdmi audio driver implements also - the card and registers it under its own platform device. - The device for the driver is registered by OMAPDSS hdmi - driver. - -config SND_OMAP_SOC_N810 - tristate "SoC Audio support for Nokia N810" - depends on MACH_NOKIA_N810 && I2C - select SND_SOC_NOKIA_N810 - help - Say Y if you want to add support for SoC audio on Nokia N810. - -config SND_OMAP_SOC_RX51 - tristate "SoC Audio support for Nokia N900 (RX-51)" - depends on ARM && I2C && GPIOLIB && ARCH_OMAP3 - select SND_SOC_NOKIA_RX51 - help - Say Y if you want to add support for SoC audio on Nokia N900 - cellphone. - -config SND_OMAP_SOC_AMS_DELTA - tristate "SoC Audio support for Amstrad E3 (Delta) videophone" - depends on MACH_AMS_DELTA && TTY - select SND_SOC_OMAP_AMS_DELTA - help - Say Y if you want to add support for SoC audio device connected to - a handset and a speakerphone found on Amstrad E3 (Delta) videophone. - - Note that in order to get those devices fully supported, you have to - build the kernel with standard serial port driver included and - configured for at least 4 ports. Then, from userspace, you must load - a line discipline #19 on the modem (ttyS3) serial line. The simplest - way to achieve this is to install util-linux-ng and use the included - ldattach utility. This can be started automatically from udev, - a simple rule like this one should do the trick (it does for me): - ACTION=="add", KERNEL=="controlC0", \ - RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" - -config SND_OMAP_SOC_OSK5912 - tristate "SoC Audio support for omap osk5912" - depends on MACH_OMAP_OSK && I2C - select SND_SOC_OMAP_OSK5912 - help - Say Y if you want to add support for SoC audio on osk5912. - -config SND_OMAP_SOC_OMAP_TWL4030 - tristate "SoC Audio support for TI SoC based boards with twl4030 codec" - depends on TWL4030_CORE - select SND_SOC_OMAP3_TWL4030 - help - Say Y if you want to add support for SoC audio on TI SoC based boards - using twl4030 as c codec. This driver currently supports: - - Beagleboard or Devkit8000 - - Gumstix Overo or CompuLab CM-T35/CM-T3730 - - IGEP v2 - - OMAP3EVM - - SDP3430 - - Zoom2 - -config SND_OMAP_SOC_OMAP_ABE_TWL6040 - tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && COMMON_CLK - depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST - select SND_SOC_OMAP_ABE_TWL6040 - help - Say Y if you want to add support for SoC audio on OMAP boards using - ABE and twl6040 codec. This driver currently supports: - - SDP4430/Blaze boards - - PandaBoard (4430) - - PandaBoardES (4460) - - omap5-uevm (5432) - -config SND_OMAP_SOC_OMAP3_PANDORA - tristate "SoC Audio support for OMAP3 Pandora" - depends on TWL4030_CORE - select SND_SOC_OMAP3_PANDORA - help - Say Y if you want to add support for SoC audio on the OMAP3 Pandora. -endmenu - endmenu -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: dma-sh7760: cleanup a debug printk
by Dan Carpenter 18 Dec '18

18 Dec '18

The intent was to print the address as a hexadecimal but there is an extra "u" in the "0x%08ulx" format specification so it is displayed as decimal. Fixes: aef3b06ac697 ("[ALSA] SH7760 ASoC support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sh/dma-sh7760.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 2.17.1

1 0

[alsa-devel] [PATCH v2 1/5] ASoC: qcom: sdm845: Add board specific dapm widgets
by Cheng-Yi Chiang 18 Dec '18

18 Dec '18

Add board specific dapm widgets so these widgets can be used in the route. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab151..95d8d4422dae0 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); -- 2.20.0.rc0.387.gc7a69e6b6c-goog

4 15

[alsa-devel] [PATCH 0/9] ASoC: audio/simple card tidyup/cleanup
by Kuninori Morimoto 18 Dec '18

18 Dec '18

Hi Mark We could merged normal sound and DPCM sound on audio/simple card. This is nice, but I noticed few issues. 1) : for "reg" property issue, again. 2) - 5) : missing 1CPU-1Codec DPCM case, and cleanup for audio-graph 6) - 9) : missing 1CPU-1Codec DPCM case, and cleanup for simple-card Kuninori Morimoto (9): 1) ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() 2) ASoC: audio-graph-card: add asoc_graph_card_get_conversion() 3) ASoC: audio-graph-card: add 1 CPU : 1 Codec support again 4) ASoC: audio-graph-card: cleanup DAI link loop method 5) ASoC: audio-graph-card: reduce naming prefix 6) ASoC: simple-card: add asoc_simple_card_get_conversion() 7) ASoC: simple-card: add 1 CPU : 1 Codec support again 8) ASoC: simple-card: cleanup DAI link loop method 9) ASoC: simple-card: reduce naming prefix sound/soc/generic/audio-graph-card.c | 461 +++++++++++++++++++--------------- sound/soc/generic/simple-card-utils.c | 14 +- sound/soc/generic/simple-card.c | 453 +++++++++++++++++++-------------- 3 files changed, 544 insertions(+), 384 deletions(-) -- 2.7.4

1 9

[alsa-devel] PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN variant with ALC294
by Wandrille RONCE 18 Dec '18

18 Dec '18

From 5a244e278c7bb626c901d668d359f47681e40647 Mon Sep 17 00:00:00 2001 From: Wandrille RONCE <w(a)ndrille.fr> Date: Mon, 17 Dec 2018 19:30:00 +0100 Subject: [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN variant with ALC294 --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 90f30e769a42..6f34f81125e0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6584,6 +6584,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x16ce, "ASUS UX433FN", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), -- 2.19.1

1 0

Re: [alsa-devel] [Sound-open-firmware] [PATCH v3 13/14] ASoC: SOF: Add xtensa support
by rander.wang 17 Dec '18

17 Dec '18

在 12/13/2018 2:15 PM, rander.wang 写道: > > 在 12/12/2018 8:00 AM, Pierre-Louis Bossart 写道: >> >> On 12/11/18 5:08 PM, Andy Shevchenko wrote: >>> On Tue, Dec 11, 2018 at 03:23:17PM -0600, Pierre-Louis Bossart wrote: >>>> Add common directory for xtensa architecture >>>> + for (i = 0; i <= stack_words - 4; i += 4) { >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x 0x%8.8x >>>> 0x%8.8x\n", >>>> + stack_ptr + i, stack[i], stack[i + 1], stack[i + 2], >>>> + stack[i + 3]); >>>> + } >>>> + >>>> + /* deal with any remaining words */ >>>> + switch (stack_words - i) { >>>> + case 0: >>>> + break; >>>> + case 1: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x\n", >>>> + stack_ptr + stack_words - 1, stack[stack_words - 1]); >>>> + break; >>>> + case 2: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x\n", >>>> + stack_ptr + stack_words - 2, stack[stack_words - 2], >>>> + stack[stack_words - 1]); >>>> + break; >>>> + case 3: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x 0x%8.8x\n", >>>> + stack_ptr + stack_words - 3, stack[stack_words - 3], >>>> + stack[stack_words - 2], stack[stack_words - 1]); >>>> + break; >>>> + default: >>>> + break; >>>> + } >>> hex_dump_to_buffer(). >> I vaguely recall there was a reason not to do it, we'll relook at this. >> > we don't want to convert a blob of data to "hex ASCII" in memory > (hex_dump_to_buffer). > > Here, we just want to dump the content of call stack, like: > > [ 3.250628] sof-audio sof-audio: 0x0049fbb0: 0x8000f2d0 0x0049fc00 > 0x6f6c6c61 0x00632e63 > [ 3.250670] sof-audio sof-audio: 0x0049fbb4: 0x00ffffff 0x0010b5d4 > 0x00000000 0x2000004c > [ 3.250710] sof-audio sof-audio: 0x0049fbb8: 0x0040d000 0x00000000 > 0x0000024c 0x2000930c > [ 3.250747] sof-audio sof-audio: 0x0049fbbc: 0x8000454c 0x0049fc30 > 0x00401e20 0x00000001 > [ 3.250787] sof-audio sof-audio: 0x0049fbc0: 0x8000454c 0x0049fc30 > 0x00401e20 0x00000001 > [ 3.250827] sof-audio sof-audio: 0x0049fbc4: 0x00000000 0x00000001 > 0x00000010 0x0049fcb8 > [ 3.250865] sof-audio sof-audio: 0x0049fbc8: 0x0040a000 0x00401e98 > 0x00000001 0x00000002 > [ 3.250889] sof-audio sof-audio: 0x0049fbcc: 0x80005ac5 0x0049fc60 > 0x00000000 0x00000000 Add more details: we use another tool to analysis the hex numbers with a ELF file to get the correct information, so we do it by ourself. Thanks! Rander >> _______________________________________________ >> Sound-open-firmware mailing list >> Sound-open-firmware(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware

3 2

Re: [alsa-devel] [PATCH 01/21] ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support.
by Yang, Libin 17 Dec '18

17 Dec '18

Hi Sakamoto, > >-------- Forwarded Message -------- >Subject: Re: [alsa-devel] [PATCH 01/21] ASoC: SOF: Intel: Add BYT, CHT and >BSW DSP HW support. >Date: Wed, 12 Dec 2018 13:08:48 +0900 >From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> >To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>, >alsa-devel(a)alsa-project.org >CC: Daniel Baluta <daniel.baluta(a)gmail.com>, andriy.shevchenko(a)intel.com, >tiwai(a)suse.de, Pan Xiuli <xiuli.pan(a)linux.intel.com>, >liam.r.girdwood(a)linux.intel.com, vkoul(a)kernel.org, broonie(a)kernel.org, >sound-open-firmware(a)alsa-project.org, Rander Wang ><rander.wang(a)linux.intel.com>, Alan Cox <alan(a)linux.intel.com> > >Hi, > >I have small nitpicking. > >On 2018/12/12 6:30, Pierre-Louis Bossart wrote: >> From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> >> >> Add support for the audio DSP hardware found on Intel Baytrail, >> Cherrytrail and Braswell based devices. >> >> Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> >> Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> >> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> >> Signed-off-by: Pierre-Louis Bossart >> <pierre-louis.bossart(a)linux.intel.com> >> --- >> sound/soc/sof/intel/byt.c | 805 >+++++++++++++++++++++++++++++++++++++ >> sound/soc/sof/intel/shim.h | 159 ++++++++ >> 2 files changed, 964 insertions(+) >> create mode 100644 sound/soc/sof/intel/byt.c >> create mode 100644 sound/soc/sof/intel/shim.h .. >> diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h >> new file mode 100644 index 000000000000..2fdcb2d262f4 >> --- /dev/null >> +++ b/sound/soc/sof/intel/shim.h >> @@ -0,0 +1,159 @@ >> ... >> +extern struct snd_sof_dsp_ops sof_byt_ops; extern struct >> +snd_sof_dsp_ops sof_cht_ops; > >These two symbols are found in this patch. > >> +extern struct snd_sof_dsp_ops sof_hsw_ops; > >This symbol is added in second patch and this first patch has no reference to it. > >> +extern struct snd_sof_dsp_ops sof_bdw_ops; > >This symbol is added in third patch and this first patch has no reference to it. > >> + >> +#endif > >I think it better to add these extern declarations when adding corresponding >symbols (Of course no issues as is). > >Furthermore, if you have no plan to change contents of these symbols in >kernel run time, it's better to have 'const' qualifier to locate the symbol to >readonly section. However, user of these symbols is machine driver and you >have a plan to implement it later. They can get 'const' >when you work for arrangement of existent/new codes. > >In this meaning, it might be good to add a new member to 'struct >snd_soc_acpi_mach' for new drivers to refer to the const ops, then 'pdata' >member is used from the existent drivers. I made a patch based on your suggestion. Could you please help review to see if my understanding is right? https://github.com/thesofproject/linux/pull/457 Regards, Libin > > >Regards > >Takashi Sakamoto >_______________________________________________ >Alsa-devel mailing list >Alsa-devel(a)alsa-project.org >http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [PATCH] ASoC: pcm512x: Implement the digital_mute interface
by Dimitris Papavasiliou 17 Dec '18

17 Dec '18

Clicks and pops of various volumes can be produced while the device is opened, closed, put into and taken out of standby, or reconfigured. Fix this, by implementing the digital_mute interface, so that the output is muted during such operations. Signed-off-by: Dimitris Papavasiliou <dpapavas(a)gmail.com> --- Notes: Although the datasheet isn't very specific about it, PCM512x_ANALOG_MUTE_DET is only updated once ramp-down to digital mute is complete (followed by analog mute, assuming PCM512x_ANALOG_MUTE_CTRL is left with the default value, as is now the case). I've measured the time it takes for the polling to finish and it seems consistent with that assumption, including properly tracking changes in the ramp-down parameters. Thus, polling the register ensures that the device has been completely muted before returning. When unmuting, the time required for PCM512x_ANALOG_MUTE_DET to reflect the change, doesn't depend on the ramp-up parameters (as might be expected, since analog mute is likely lifted first). Nevertheless the value is consistently substantially larger than zero, so that it seemingly reflects a particular stage in the unmuting process. It also does no harm and seems to help minimize clicks, so I've kept the delay. I've tested this patch both in my day-to-day use of my DAC for more than a month, as well as by running a script that opens the device, performs several sample rate and format changes and closes it, every one second. Without the patch, several different kinds of clicks and pops are produced. After the patch, I've run the script for several hours with the DAC being consistently virtually silent. sound/soc/codecs/pcm512x.c | 121 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/pcm512x.h | 2 + 2 files changed, 121 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index f0f2d4fd3769..6cb1653be804 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -53,6 +53,8 @@ struct pcm512x_priv { unsigned long overclock_pll; unsigned long overclock_dac; unsigned long overclock_dsp; + int mute; + struct mutex mutex; }; /* @@ -384,6 +386,61 @@ static const struct soc_enum pcm512x_veds = SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, pcm512x_ramp_step_text); +static int pcm512x_update_mute(struct pcm512x_priv *pcm512x) +{ + return regmap_update_bits( + pcm512x->regmap, PCM512x_MUTE, PCM512x_RQML | PCM512x_RQMR, + (!!(pcm512x->mute & 0x5) << PCM512x_RQML_SHIFT) + | (!!(pcm512x->mute & 0x3) << PCM512x_RQMR_SHIFT)); +} + +static int pcm512x_digital_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + mutex_lock(&pcm512x->mutex); + ucontrol->value.integer.value[0] = !(pcm512x->mute & 0x4); + ucontrol->value.integer.value[1] = !(pcm512x->mute & 0x2); + mutex_unlock(&pcm512x->mutex); + + return 0; +} + +static int pcm512x_digital_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret, changed = 0; + + mutex_lock(&pcm512x->mutex); + + if ((pcm512x->mute & 0x4) == (ucontrol->value.integer.value[0] << 2)) { + pcm512x->mute ^= 0x4; + changed = 1; + } + if ((pcm512x->mute & 0x2) == (ucontrol->value.integer.value[1] << 1)) { + pcm512x->mute ^= 0x2; + changed = 1; + } + + if (changed) { + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + } + + mutex_unlock(&pcm512x->mutex); + + return changed; +} + static const struct snd_kcontrol_new pcm512x_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), @@ -391,8 +448,15 @@ SOC_DOUBLE_TLV("Analogue Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Analogue Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, - PCM512x_RQMR_SHIFT, 1, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = pcm512x_digital_playback_switch_get, + .put = pcm512x_digital_playback_switch_put +}, SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), SOC_ENUM("DSP Program", pcm512x_dsp_program), @@ -1319,10 +1383,61 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret; + unsigned int mute_det; + + mutex_lock(&pcm512x->mutex); + + if (mute) { + pcm512x->mute |= 0x1; + ret = regmap_update_bits(pcm512x->regmap, PCM512x_MUTE, + PCM512x_RQML | PCM512x_RQMR, + PCM512x_RQML | PCM512x_RQMR); + if (ret != 0) { + dev_err(component->dev, + "Failed to set digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, (mute_det & 0x3) == 0, + 200, 10000); + + mutex_unlock(&pcm512x->mutex); + } else { + pcm512x->mute &= ~0x1; + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, + (mute_det & 0x3) + == ((~pcm512x->mute >> 1) & 0x3), + 200, 10000); + } + + mutex_unlock(&pcm512x->mutex); + + return 0; +} + static const struct snd_soc_dai_ops pcm512x_dai_ops = { .startup = pcm512x_dai_startup, .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, + .digital_mute = pcm512x_digital_mute, }; static struct snd_soc_dai_driver pcm512x_dai = { @@ -1388,6 +1503,8 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (!pcm512x) return -ENOMEM; + mutex_init(&pcm512x->mutex); + dev_set_drvdata(dev, pcm512x); pcm512x->regmap = regmap; diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index d70d9c0c2088..9dda8693498e 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -112,7 +112,9 @@ #define PCM512x_RQST_SHIFT 4 /* Page 0, Register 3 - mute */ +#define PCM512x_RQMR (1 << 0) #define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML (1 << 4) #define PCM512x_RQML_SHIFT 4 /* Page 0, Register 4 - PLL */ -- 2.11.0

2 8

[alsa-devel] [PATCH 0/3] Add Xilinx I2S driver
by Maruthi Srinivas Bayyavarapu 17 Dec '18

17 Dec '18

Xilinx I2S IP enables I2S audio. Each IP can either work in playback or capture mode exclusively. The driver supports both by reading the configuration in device tree. Patchset includes devicetree bindings documentation, driver and build enablement. Maruthi Srinivas Bayyavarapu (3): dt-bindings: ASoC: xlnx,i2s: Document i2s bindings ASoC: xlnx: Add i2s driver ASoC: xlnx: enable i2s driver build .../devicetree/bindings/sound/xlnx,i2s.txt | 28 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xilinx/Kconfig | 8 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_i2s.c | 185 +++++++++++++++++++++ 6 files changed, 225 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,i2s.txt create mode 100644 sound/soc/xilinx/Kconfig create mode 100644 sound/soc/xilinx/Makefile create mode 100644 sound/soc/xilinx/xlnx_i2s.c -- 2.7.4

3 10

[alsa-devel] ASoC: About graph node issue without 'reg' property
by Kunihiko Hayashi 17 Dec '18

17 Dec '18

Hello Morimoto-san, Mark, I found that your commit b6f3fc005a2c ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") replaced with of_graph_parse_endpoint(), that needs 'reg' property in each port that has an endpoint. Currently devicetrees for UniPhier SoCs have also audio graph nodes [1], and these graph nodes include an endpoint for each port. Although the port doesn't have 'reg' property, it has worked fine before. And after applying the commit, it doesn't work. It is written in Documentation/devicetree/bindings/graph.txt that common scheme is to include the 'reg' property. Maybe I think that we'd better add 'reg' properties to each port, however, there might be same matter in other SoCs, so I'm not sure whether your patch breaks the backword compatibility or our devicetrees are wrong. I'd be glad if I could hear something about this issue from you. Thank you, [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… --- Best Regards, Kunihiko Hayashi

3 5

[alsa-devel] [PATCH v2] ALSA: hda: add mute LED support for HP EliteBook 840 G4
by Mantas Mikulėnas 16 Dec '18

16 Dec '18

Tested with 4.19.9. v2: Changed from CXT_FIXUP_MUTE_LED_GPIO to CXT_FIXUP_HP_DOCK because that's what the existing fixups for EliteBooks use. Signed-off-by: Mantas Mikulėnas <grawity(a)gmail.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 950e02e71766..51cc6589443f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -923,6 +923,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: hda: add mute LED support for HP EliteBook 840 G4
by Mantas Mikulėnas 16 Dec '18

16 Dec '18

Tested with 4.19.8. (Existing entries for similar models use CXT_FIXUP_HP_DOCK, which might be a better choice here as well, but that's far beyond my knowledge -- and I don't have a dock to test it with either.) Signed-off-by: Mantas Mikulėnas <grawity(a)gmail.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 950e02e71766..b04ff8f2cf5c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -922,6 +922,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), -- 2.20.0

2 2

[alsa-devel] [PATCH 0/5] ALSA: fireface: add support for Fireface 800 with streaming functionality
by Takashi Sakamoto 16 Dec '18

16 Dec '18

Hi, My former patchset adds support for Fireface 800 with MIDI functionality only[1]. This patchset updates driver with packet streaming functionality. As a result, both of PCM frames/MIDI messages are available. This patchset is based on my former patch to fix a bug to switch frame fetching mode: [alsa-devel] [PATCH] ALSA: fireface: fix for state to fetch PCM frames http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143150.h… Unfortunately, my initial commit of ALSA fireface driver in v4.12 kernel includes bugs to handle tx isochronous packets. When testing this patchset, please apply below patchset in advance: [alsa-devel] [PATCH 0/3] ALSA: firewire-lib: fixes for processing packets without CIP header http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143146.h… For testers, a remote branch including overall patches is available in my remote repository in github.com[2]. They can be backport to Linux kernel v4.17 or later. I note that this driver can generate hissing noise long-periodically as I reported for Fireface 400[3]. The cause is not cleared yet. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/142826.h… [2] https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-pcm [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102261.h… Takashi Sakamoto (5): ALSA: fireface: share helper function to switch fetching mode ALSA: fireface: code refactoring to handle multiplier mode ALSA: fireface: allocate isochronous resources in mode-specific implementation ALSA: fireface: add support for packet streaming on Fireface 800 ALSA: fireface: code refactoring to handle model-specific registers sound/firewire/fireface/ff-pcm.c | 33 ++++-- sound/firewire/fireface/ff-protocol-ff400.c | 81 +++++++------ sound/firewire/fireface/ff-protocol-ff800.c | 116 ++++++++++++++++++ sound/firewire/fireface/ff-stream.c | 124 +++++++++++--------- sound/firewire/fireface/ff-transaction.c | 4 +- sound/firewire/fireface/ff.c | 37 +++--- sound/firewire/fireface/ff.h | 21 ++-- 7 files changed, 283 insertions(+), 133 deletions(-) -- 2.19.1

2 6

[alsa-devel] [PATCH] ALSA: fireface: fix for state to fetch PCM frames
by Takashi Sakamoto 16 Dec '18

16 Dec '18

According to my memo at hand and saved records, writing 0x00000001 to SND_FF_REG_FETCH_PCM_FRAMES disables fetching PCM frames in corresponding channel, however current implement uses reversed logic. This results in muted volume in device side during playback. This commit corrects the bug. Cc: <stable(a)vger.kernel.org> # v4.12+ Fixes: 76fdb3a9e13a ('ALSA: fireface: add support for Fireface 400') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-ff400.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index 8f34174ee813..fd257051d4a4 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -86,7 +86,7 @@ static int ff400_switch_fetching_mode(struct snd_ff *ff, bool enable) if (reg == NULL) return -ENOMEM; - if (enable) { + if (!enable) { /* * Each quadlet is corresponding to data channels in a data * blocks in reverse order. Precisely, quadlets for available -- 2.19.1

2 1

[alsa-devel] [PATCH 0/3] ALSA: firewire-lib: fixes for processing packets without CIP header
by Takashi Sakamoto 16 Dec '18

16 Dec '18

Hi, In kernel v4.12, a series of patch was applied to IEC 61883-1/6 packet streaming engine in ALSA firewire stack to process isochronous packet without CIP header. But this included some bugs. Especially, for capture PCM substream, the engine handles PCM frames for applications as four times against real time. This patchset fixes the bugs. Takashi Sakamoto (3): ALSA: firewire-lib: fix wrong handling payload_length as payload_quadlet ALSA: firewire-lib: fix wrong assignment for 'out_packet_without_header' tracepoint ALSA: firewire-lib: use the same print format for 'without_header' tracepoints sound/firewire/amdtp-stream-trace.h | 4 ++-- sound/firewire/amdtp-stream.c | 4 +++- 2 files changed, 5 insertions(+), 3 deletions(-) -- 2.19.1

2 4

Re: [alsa-devel] [PATCH] ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID method
by Kunihiko Hayashi 15 Dec '18

15 Dec '18

Hi Morioto-san, On 2018/12/14 11:29, Kuninori Morimoto wrote: > From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup > asoc_simple_card_get_dai_id() counting") fixuped getting DAI ID method. > It will get DAI ID from OF graph "port", but, we want to consider about > "endpoint", too. > And, we also want to keep compatibility. > > This patch fixup it as > > if (driver has specified DAI ID) > use it as DAI ID > else if (OF graph endpoint has reg) > use it as DAI ID > else if (OF graph port has reg) > use it as DAI ID > else > use endpoint count as DAI ID > > Fixes: commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") > Reported-by: Tony Lindgren <tony(a)atomide.com> > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Tested-by: Tony Lindgren <tony(a)atomide.com> Sorry for late reply. It works fine with our devicetree that doesn't have 'reg' properties. Tested-by: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> Thank you, > --- > sound/soc/generic/simple-card-utils.c | 32 ++++++++++++++++++++++++++++---- > 1 file changed, 28 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c > index 17d8aee..b807a47 100644 > --- a/sound/soc/generic/simple-card-utils.c > +++ b/sound/soc/generic/simple-card-utils.c > @@ -269,22 +269,46 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_parse_dai); > > static int asoc_simple_card_get_dai_id(struct device_node *ep) > { > + struct device_node *node; > + struct device_node *endpoint; > struct of_endpoint info; > + int i, id; > int ret; > > + /* use driver specified DAI ID if exist */ > ret = snd_soc_get_dai_id(ep); > if (ret != -ENOTSUPP) > return ret; > > + /* use endpoint/port reg if exist */ > + ret = of_graph_parse_endpoint(ep, &info); > + if (ret == 0) { > + if (info.id) > + return info.id; > + if (info.port) > + return info.port; > + } > + > + node = of_graph_get_port_parent(ep); > + > /* > * Non HDMI sound case, counting port/endpoint on its DT > * is enough. Let's count it. > */ > - ret = of_graph_parse_endpoint(ep, &info); > - if (ret) > - return -ENXIO; > + i = 0; > + id = -1; > + for_each_endpoint_of_node(node, endpoint) { > + if (endpoint == ep) > + id = i; > + i++; > + } > + > + of_node_put(node); > + > + if (id < 0) > + return -ENODEV; > > - return info.port; > + return id; > } > > int asoc_simple_card_parse_graph_dai(struct device_node *ep, >

1 0

[alsa-devel] [PATCH 4/7] slimbus: ngd: mark PM functions as __maybe_unused
by Arnd Bergmann 15 Dec '18

15 Dec '18

Commit 2e6ae11dd0d1c37f44cec51a58fb2092e55ed0f5 upstream. qcom_slim_ngd_runtime_suspend is protected by an #ifdef, qcom_slim_ngd_runtime_idle is now, which causes a build time warning: drivers/slimbus/qcom-ngd-ctrl.c:1470:12: error: 'qcom_slim_ngd_runtime_idle' defined but not used [-Werror=unused-function] Marking both as __maybe_unused lets us get rid of the warning as well as the #ifdef. Fixes: 917809e2280b ("slimbus: ngd: Add qcom SLIMBus NGD driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 8be4d6786c61..14a9d18306cb 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1467,7 +1467,7 @@ static int qcom_slim_ngd_remove(struct platform_device *pdev) return 0; } -static int qcom_slim_ngd_runtime_idle(struct device *dev) +static int __maybe_unused qcom_slim_ngd_runtime_idle(struct device *dev) { struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev); @@ -1477,8 +1477,7 @@ static int qcom_slim_ngd_runtime_idle(struct device *dev) return -EAGAIN; } -#ifdef CONFIG_PM -static int qcom_slim_ngd_runtime_suspend(struct device *dev) +static int __maybe_unused qcom_slim_ngd_runtime_suspend(struct device *dev) { struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev); int ret = 0; @@ -1491,7 +1490,6 @@ static int qcom_slim_ngd_runtime_suspend(struct device *dev) return ret; } -#endif static const struct dev_pm_ops qcom_slim_ngd_dev_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, -- 2.20.0

1 0

[alsa-devel] linux-firmware: intel: Update Cannonlake audio firmware.
by Rojewski, Cezary 14 Dec '18

14 Dec '18

Hello, This change updates the firmware version from .1191 to .1858 for CNL and its child platforms. Kindly pull the new CNL ADSP firmware from the following URL. The following changes since commit 1baa34868b2c0a004dc595b20678145e3fff83e7: Merge branch 'nxp_mc' of https://github.com/NXP/linux-firmware (2018-10-26 08:13:19 -0400) are available in the Git repository at: https://github.com/crojewsk/linux-firmware.git for-upstream for you to fetch changes up to 82f6dbf82d976064db15f4e7f3e770322dc6ce14: linux-firmware: intel: Update Cannonlake audio firmware. (2018-12-03 03:43:31 +0100) ---------------------------------------------------------------- Cezary Rojewski (1): linux-firmware: intel: Update Cannonlake audio firmware. WHENCE | 4 +++- intel/dsp_fw_cnl.bin | 2 +- intel/dsp_fw_cnl_v1858.bin | Bin 0 -> 583852 bytes 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 intel/dsp_fw_cnl_v1858.bin -------------------------------------------------------------------- Intel Technology Poland sp. z o.o. ul. Slowackiego 173 | 80-298 Gdansk | Sad Rejonowy Gdansk Polnoc | VII Wydzial Gospodarczy Krajowego Rejestru Sadowego - KRS 101882 | NIP 957-07-52-316 | Kapital zakladowy 200.000 PLN. Ta wiadomosc wraz z zalacznikami jest przeznaczona dla okreslonego adresata i moze zawierac informacje poufne. W razie przypadkowego otrzymania tej wiadomosci, prosimy o powiadomienie nadawcy oraz trwale jej usuniecie; jakiekolwiek przegladanie lub rozpowszechnianie jest zabronione. This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). If you are not the intended recipient, please contact the sender and delete all copies; any review or distribution by others is strictly prohibited.

3 2

[alsa-devel] [GIT PULL] Additional firmware files for CA0132 HD-audio codec
by Connor McAdams 14 Dec '18

14 Dec '18

The following changes since commit c6b6265d718d118e28e1ce8f91769aa886b54c94: Merge tag 'iwlwifi-fw-2018-10-03' of git://git.kernel.org/pub/scm/linux/kernel/git/iwlwifi/linux-firmware (2018-10-08 09:23:53 -0400) are available in the git repository at: git@github.com:Conmanx360/linux-firmware.git for you to fetch changes up to cd8899323352e772c5989230f35b1c494e0ab196: linux-firmware: Add new firmware for Creative CA0132 HD-Audio Codec (2018-10-09 14:41:12 -0400) ---------------------------------------------------------------- These are additional firmware files for CA0132 based sound cards. Creative has given me permission through email to send these to the linux-firmware repository under the same license as the previous ca0132 firmware. I asked my contact if they would sign-off on it, but they said to go ahead and submit it and that if there was any issue to contact them. If you need to contact them, or if there's more info needed, let me know and I can get see what I can do. ---------------------------------------------------------------- Connor McAdams (1): linux-firmware: Add new firmware for Creative CA0132 HD-Audio Codec WHENCE | 2 ++ ctefx-desktop.bin | Bin 0 -> 655856 bytes ctefx-r3di.bin | Bin 0 -> 655816 bytes 3 files changed, 2 insertions(+) create mode 100644 ctefx-desktop.bin create mode 100644 ctefx-r3di.bin

3 6

[alsa-devel] [PATCH 0/4] ASoC: qdsp6: Add support for audio over DP
by Rohit kumar 14 Dec '18

14 Dec '18

Add support for Audio over display port AFE port Rohit kumar (4): ASoC: qdsp6: dt-bindings: Add q6afe display_port dt binding ASoC: qdsp6: qdafe: add support for display_port_rx ASoC: qdsp6: q6afe-dai: add support to Display port RX dais ASoC: qdsp6: q6routing: add display_port_rx port routing include/dt-bindings/sound/qcom,q6afe.h | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 22 ++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 4 ++++ sound/soc/qcom/qdsp6/q6routing.c | 9 +++++++++ 4 files changed, 36 insertions(+) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 9

[alsa-devel] Applied "ALSA: soc-compress: add support to snd_compr_set_runtime_buffer()" to the asoc tree
by Mark Brown 14 Dec '18

14 Dec '18

The patch ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ba02eed9f300b6512181d526311d4e11aaa9714f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 15 Nov 2018 18:13:20 +0000 Subject: [PATCH] ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() Existing compress offload code allocates data buffers using simple kmalloc, however there are situations where these buffers have to be mapped in smmu. So provide a way to set the runtime buffer by the driver itself, simillar to what we do with pcm. This patch adds support to set runtime dma buffer on compressed stream. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Acked-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/compress_driver.h | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index ea8c93bbb0e0..0cdc3999ecfa 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -23,6 +23,7 @@ struct snd_compr_ops; * struct snd_compr_runtime: runtime stream description * @state: stream state * @ops: pointer to DSP callbacks + * @dma_buffer_p: runtime dma buffer pointer * @buffer: pointer to kernel buffer, valid only when not in mmap mode or * DSP doesn't implement copy * @buffer_size: size of the above buffer @@ -37,6 +38,7 @@ struct snd_compr_ops; struct snd_compr_runtime { snd_pcm_state_t state; struct snd_compr_ops *ops; + struct snd_dma_buffer *dma_buffer_p; void *buffer; u64 buffer_size; u32 fragment_size; @@ -175,6 +177,23 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) wake_up(&stream->runtime->sleep); } +/** + * snd_compr_set_runtime_buffer - Set the Compress runtime buffer + * @substream: compress substream to set + * @bufp: the buffer information, NULL to clear + * + * Copy the buffer information to runtime buffer when @bufp is non-NULL. + * Otherwise it clears the current buffer information. + */ +static inline void snd_compr_set_runtime_buffer( + struct snd_compr_stream *substream, + struct snd_dma_buffer *bufp) +{ + struct snd_compr_runtime *runtime = substream->runtime; + + runtime->dma_buffer_p = bufp; +} + int snd_compr_stop_error(struct snd_compr_stream *stream, snd_pcm_state_t state); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ALSA: compress: make use of runtime buffer for copy" to the asoc tree
by Mark Brown 14 Dec '18

14 Dec '18

The patch ALSA: compress: make use of runtime buffer for copy has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d00f749b00f7802bf944688ad2971455f84fdacb Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 15 Nov 2018 18:13:21 +0000 Subject: [PATCH] ALSA: compress: make use of runtime buffer for copy Default copy function uses kmalloc to allocate buffers, lets check if the runtime buffers are setup before making this allocations. This can be useful if the buffers are dma buffers. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Acked-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/core/compress_offload.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 26b5e245b074..a5b09e75e787 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -171,7 +171,8 @@ static int snd_compr_free(struct inode *inode, struct file *f) } data->stream.ops->free(&data->stream); - kfree(data->stream.runtime->buffer); + if (!data->stream.runtime->dma_buffer_p) + kfree(data->stream.runtime->buffer); kfree(data->stream.runtime); kfree(data); return 0; @@ -505,7 +506,7 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, struct snd_compr_params *params) { unsigned int buffer_size; - void *buffer; + void *buffer = NULL; buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { @@ -514,7 +515,18 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, * the data from core */ } else { - buffer = kmalloc(buffer_size, GFP_KERNEL); + if (stream->runtime->dma_buffer_p) { + + if (buffer_size > stream->runtime->dma_buffer_p->bytes) + dev_err(&stream->device->dev, + "Not enough DMA buffer"); + else + buffer = stream->runtime->dma_buffer_p->area; + + } else { + buffer = kmalloc(buffer_size, GFP_KERNEL); + } + if (!buffer) return -ENOMEM; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: q6asm-dai: dt-bindings: Add support to compress dais" to the asoc tree
by Mark Brown 14 Dec '18

14 Dec '18

The patch ASoC: q6asm-dai: dt-bindings: Add support to compress dais has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ee23cf37fd586769febab95e2fbbbb50da5f9f2d Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Thu, 15 Nov 2018 18:13:22 +0000 Subject: [PATCH] ASoC: q6asm-dai: dt-bindings: Add support to compress dais This patch adds board specific bindings required for dais, In particular for compressed dais and dai direction. Board specific setup involves setting up some of dais as compressed dais and also specify direction of any dai. Some of the dais might only support capture/playback depending on the board level wiring. These two new dt properties will allow such flexibilty at board level dts. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Rob Herring <robh(a)kernel.org> Acked-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/qcom,q6asm.txt | 27 +++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index f9c7bd8c1bc0..9f5378c51686 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -27,6 +27,28 @@ used by the apr service device. Value type: <u32> Definition: Must be 1 +== ASM DAI is subnode of "dais" and represent a dai, it includes board specific +configuration of each dai. Must contain the following properties. + +- reg + Usage: required + Value type: <u32> + Definition: Must be dai id + +- direction: + Usage: Required for Compress offload dais + Value type: <u32> + Definition: Specifies the direction of the dai stream + 0 for both tx and rx + 1 for only tx (Capture/Encode) + 2 for only rx (Playback/Decode) + +- is-compress-dai: + Usage: Required for Compress offload dais + Value type: <boolean> + Definition: present for Compress offload dais + + = EXAMPLE q6asm@7 { @@ -35,5 +57,10 @@ q6asm@7 { q6asmdai: dais { compatible = "qcom,q6asm-dais"; #sound-dai-cells = <1>; + mm@0 { + reg = <0>; + direction = <2>; + is-compress-dai; + }; }; }; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v3 0/5] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 14 Dec '18

14 Dec '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Changes since v2 (//lkml.org/lkml/2018/9/26/461): - Removed some unnecessary empty line adds. - Added more info to snd_compr_set_runtime_buffer() commit message Srinivas Kandagatla (5): ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() ALSA: compress: make use of runtime buffer for copy ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ include/sound/compress_driver.h | 19 + sound/core/compress_offload.c | 18 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 372 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 6 files changed, 438 insertions(+), 4 deletions(-) -- 2.19.1

4 10

[alsa-devel] Applied "ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID method" to the asoc tree
by Mark Brown 14 Dec '18

14 Dec '18

The patch ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID method has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b8b89f5ee0b2555fa5a9d778685d2d6fb5122bb9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 14 Dec 2018 11:29:42 +0900 Subject: [PATCH] ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID method commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") fixuped getting DAI ID method. It will get DAI ID from OF graph "port", but, we want to consider about "endpoint", too. And, we also want to keep compatibility. This patch fixup it as if (driver has specified DAI ID) use it as DAI ID else if (OF graph endpoint has reg) use it as DAI ID else if (OF graph port has reg) use it as DAI ID else use endpoint count as DAI ID Fixes: commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") Reported-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 32 +++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 17d8aee43835..b807a47515eb 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -269,22 +269,46 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_parse_dai); static int asoc_simple_card_get_dai_id(struct device_node *ep) { + struct device_node *node; + struct device_node *endpoint; struct of_endpoint info; + int i, id; int ret; + /* use driver specified DAI ID if exist */ ret = snd_soc_get_dai_id(ep); if (ret != -ENOTSUPP) return ret; + /* use endpoint/port reg if exist */ + ret = of_graph_parse_endpoint(ep, &info); + if (ret == 0) { + if (info.id) + return info.id; + if (info.port) + return info.port; + } + + node = of_graph_get_port_parent(ep); + /* * Non HDMI sound case, counting port/endpoint on its DT * is enough. Let's count it. */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret) - return -ENXIO; + i = 0; + id = -1; + for_each_endpoint_of_node(node, endpoint) { + if (endpoint == ep) + id = i; + i++; + } + + of_node_put(node); + + if (id < 0) + return -ENODEV; - return info.port; + return id; } int asoc_simple_card_parse_graph_dai(struct device_node *ep, -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: merge audio-graph-scu / simple-scu-card
by Kuninori Morimoto 14 Dec '18

14 Dec '18

Hi Mark I'm posting these 2 series ASoC: merge audio-graph-scu into audio-graph ASoC: merge simple-scu-card into simple-card This time, I merged these into this patch-set. Previous patch-set included "remove xx-suc-card" patch, too. But, I think postpone it to next version is more safety for us. So, I dropped it. As you know, I and Tony discussed about DAI ID method. It is included to this patch-set, too I post 3 series. 0*) for DAI ID fixup 1*) for merging audio-graph-scu to audio-graph 2*) for merging simple-card-scu to simple-card Kuninori Morimoto (11): 01) ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID method 11) ASoC: audio-graph-card: merge audio-graph-scu-card on Doc 12) ASoC: audio-graph-card: merge audio-graph-scu-card 13) ASoC: audio-graph-card: tidyup mclk-fs method 14) ASoC: audio-graph-card: tidyup convert_rate/channel method 15) ASoC: audio-graph-card: tidyup prefix for snd_soc_codec_conf 21) ASoC: simple-card: merge simple-scu-card on Doc 22) ASoC: simple-card: merge simple-scu-card 23) ASoC: simple-card: tidyup mclk-fs method 24) ASoC: simple-card: tidyup convert_rate/channel method 25) ASoC: simple-card: tidyup prefix for snd_soc_codec_conf Documentation/devicetree/bindings/sound/audio-graph-card.txt | 205 ++++++++++++++++++++++++++++++++ Documentation/devicetree/bindings/sound/simple-card.txt | 137 +++++++++++++++++++++ sound/soc/generic/Kconfig | 2 + sound/soc/generic/audio-graph-card.c | 403 +++++++++++++++++++++++++++++++++++++++++++++++++++++--------- sound/soc/generic/simple-card-utils.c | 32 ++++- sound/soc/generic/simple-card.c | 342 +++++++++++++++++++++++++++++++++++++++++++--------- 6 files changed, 1005 insertions(+), 116 deletions(-) -- 2.7.4

2 20

[alsa-devel] Applied "ASoC: audio-graph-card: merge audio-graph-scu-card" to the asoc tree
by Mark Brown 14 Dec '18

14 Dec '18

The patch ASoC: audio-graph-card: merge audio-graph-scu-card has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ae3cb5790906b5caf519f6f21792c30fb5ddf8db Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 14 Dec 2018 11:32:25 +0900 Subject: [PATCH] ASoC: audio-graph-card: merge audio-graph-scu-card audio-graph-card and audio-graph-scu-card are very similar driver, but the former is supporting normal sound card, the latter is supporting DPCM sound card. We couldn't use normal sound and DPCM sound in same sound card by audio-graph-card. This patch merges both sound card into it. Now we can use both feature on same driver. audio-grap-card is now supporting .compatible = "audio-graph-scu-card". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/Kconfig | 1 + sound/soc/generic/audio-graph-card.c | 366 +++++++++++++++++++++++---- 2 files changed, 320 insertions(+), 47 deletions(-) diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index fa5aff9c1041..5395782424b4 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -22,6 +22,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. + It also support DPCM of multi CPU single Codec ststem. config SND_AUDIO_GRAPH_SCU_CARD tristate "ASoC Audio Graph SCU sound card support" diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1da9532ef897..8ec09bc711f8 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -27,6 +27,8 @@ struct graph_card_data { struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; @@ -34,6 +36,8 @@ struct graph_card_data { struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; struct gpio_desc *pa_gpio; }; @@ -42,6 +46,8 @@ struct graph_card_data { #define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) #define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) +#define PREFIX "audio-graph-card," + static int asoc_graph_card_outdrv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) @@ -156,7 +162,140 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, +static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + /* overwrite by top level adata if exist */ + asoc_simple_card_convert_fixup(&priv->adata, params); + + return 0; +} + +static int asoc_graph_card_dai_link_of_dpcm(struct device_node *cpu_ep, + struct device_node *codec_ep, + struct graph_card_data *priv, + int *dai_idx, int link_idx, + int *conf_idx, int is_cpu) +{ + struct device *dev = graph_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); + struct snd_soc_card *card = graph_priv_to_card(priv); + struct device_node *ep = is_cpu ? cpu_ep : codec_ep; + struct device_node *node = of_graph_get_port_parent(ep); + struct asoc_simple_dai *dai; + int ret; + + dev_dbg(dev, "link_of DPCM (for %s)\n", is_cpu ? "CPU" : "Codec"); + + if (is_cpu) { + struct snd_soc_dai_link_component *codecs; + + /* BE is dummy */ + codecs = dai_link->codecs; + codecs->of_node = NULL; + codecs->dai_name = "snd-soc-dummy-dai"; + codecs->name = "snd-soc-dummy"; + + /* FE settings */ + dai_link->dynamic = 1; + dai_link->dpcm_merged_format = 1; + + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + + ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); + if (ret) + return ret; + + ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_set_dailink_name(dev, dai_link, + "fe.%s", + dai_link->cpu_dai_name); + if (ret < 0) + return ret; + + /* card->num_links includes Codec */ + asoc_simple_card_canonicalize_cpu(dai_link, + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); + } else { + struct snd_soc_codec_conf *cconf; + + /* FE is dummy */ + dai_link->cpu_of_node = NULL; + dai_link->cpu_dai_name = "snd-soc-dummy-dai"; + dai_link->cpu_name = "snd-soc-dummy"; + + /* BE settings */ + dai_link->no_pcm = 1; + dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; + + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + + ret = asoc_simple_card_parse_graph_codec(ep, dai_link); + if (ret < 0) + return ret; + + ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_set_dailink_name(dev, dai_link, + "be.%s", + dai_link->codecs->dai_name); + if (ret < 0) + return ret; + + /* check "prefix" from top node */ + snd_soc_of_parse_audio_prefix(card, cconf, + dai_link->codecs->of_node, + "prefix"); + /* check "prefix" from each node if top doesn't have */ + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(node, cconf, + dai_link->codecs->of_node, + PREFIX "prefix"); + } + + asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); + + ret = asoc_simple_card_of_parse_tdm(ep, dai); + if (ret) + return ret; + + ret = asoc_simple_card_canonicalize_dailink(dai_link); + if (ret < 0) + return ret; + + of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); + + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, + NULL, &dai_link->dai_fmt); + if (ret < 0) + return ret; + + dai_link->dpcm_playback = 1; + dai_link->dpcm_capture = 1; + dai_link->ops = &asoc_graph_card_ops; + dai_link->init = asoc_graph_card_dai_init; + + return 0; +} + +static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, + struct device_node *codec_ep, struct graph_card_data *priv, int *dai_idx, int link_idx) { @@ -165,10 +304,10 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; - struct device_node *cpu_ep = of_get_next_child(cpu_port, NULL); - struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; + dev_dbg(dev, "link_of\n"); + cpu_dai = dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; codec_dai = @@ -177,45 +316,45 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) - goto dai_link_of_err; + return ret; of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_parse_graph_codec(codec_ep, dai_link); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_of_parse_tdm(cpu_ep, cpu_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_of_parse_tdm(codec_ep, codec_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_parse_clk_cpu(dev, cpu_ep, dai_link, cpu_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_parse_clk_codec(dev, codec_ep, dai_link, codec_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_canonicalize_dailink(dai_link); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, dai_link->codecs->dai_name); if (ret < 0) - goto dai_link_of_err; + return ret; dai_link->ops = &asoc_graph_card_ops; dai_link->init = asoc_graph_card_dai_init; @@ -223,11 +362,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); -dai_link_of_err: - of_node_put(cpu_ep); - of_node_put(codec_ep); - - return ret; + return 0; } static int asoc_graph_card_parse_of(struct graph_card_data *priv) @@ -236,8 +371,14 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device *dev = graph_priv_to_dev(priv); struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *node = dev->of_node; + struct device_node *cpu_port; + struct device_node *cpu_ep = NULL; + struct device_node *codec_ep = NULL; + struct device_node *codec_port = NULL; + struct device_node *codec_port_old = NULL; int rc, ret; - int link_idx, dai_idx; + int link_idx, dai_idx, conf_idx; + int cpu; ret = asoc_simple_card_of_parse_widgets(card, NULL); if (ret < 0) @@ -247,35 +388,159 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret < 0) return ret; + asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); + /* Factor to mclk, used in hw_params() */ of_property_read_u32(node, "mclk-fs", &priv->mclk_fs); - link_idx = 0; - dai_idx = 0; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - ret = asoc_graph_card_dai_link_of(it.node, priv, - &dai_idx, link_idx++); - if (ret < 0) { - of_node_put(it.node); - - return ret; + link_idx = 0; + dai_idx = 0; + conf_idx = 0; + codec_port_old = NULL; + for (cpu = 1; cpu >= 0; cpu--) { + /* + * Detect all CPU first, and Detect all Codec 2nd. + * + * In Normal sound case, all DAIs are detected + * as "CPU-Codec". + * + * In DPCM sound case, + * all CPUs are detected as "CPU-dummy", and + * all Codecs are detected as "dummy-Codec". + * To avoid random sub-device numbering, + * detect "dummy-Codec" in last; + */ + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + cpu_port = it.node; + cpu_ep = NULL; + while (1) { + cpu_ep = of_get_next_child(cpu_port, cpu_ep); + if (!cpu_ep) + break; + + codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_get_parent(codec_ep); + + of_node_put(codec_ep); + of_node_put(codec_port); + + dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); + + if (of_get_child_count(codec_port) > 1) { + /* + * for DPCM sound + */ + if (!cpu) { + if (codec_port_old == codec_port) + continue; + codec_port_old = codec_port; + } + ret = asoc_graph_card_dai_link_of_dpcm( + cpu_ep, codec_ep, priv, + &dai_idx, link_idx++, + &conf_idx, cpu); + } else if (cpu) { + /* + * for Normal sound + */ + ret = asoc_graph_card_dai_link_of( + cpu_ep, codec_ep, priv, + &dai_idx, link_idx++); + } + if (ret < 0) + return ret; + } } } return asoc_simple_card_parse_card_name(card, NULL); } -static int asoc_graph_get_dais_count(struct device *dev) +static void asoc_graph_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) { struct of_phandle_iterator it; struct device_node *node = dev->of_node; - int count = 0; + struct device_node *cpu_port; + struct device_node *cpu_ep; + struct device_node *codec_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; + struct device_node *codec_port_old2; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) - count++; - - return count; + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for "dummy-Codec" + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ + codec_port_old = NULL; + codec_port_old2 = NULL; + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + cpu_port = it.node; + cpu_ep = NULL; + while (1) { + cpu_ep = of_get_next_child(cpu_port, cpu_ep); + if (!cpu_ep) + break; + + codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_get_parent(codec_ep); + + of_node_put(codec_ep); + of_node_put(codec_port); + + (*link_num)++; + (*dais_num)++; + + if (codec_port_old == codec_port) { + if (codec_port_old2 != codec_port_old) { + (*link_num)++; + (*ccnf_num)++; + } + + codec_port_old2 = codec_port_old; + continue; + } + + (*dais_num)++; + codec_port_old = codec_port; + } + } } static int asoc_graph_soc_card_probe(struct snd_soc_card *card) @@ -302,20 +567,23 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; - int num, ret, i; + struct snd_soc_codec_conf *cconf; + int lnum = 0, dnum = 0, cnum = 0; + int ret, i; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - num = asoc_graph_get_dais_count(dev); - if (num == 0) + asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) return -EINVAL; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, num * 2, sizeof(*dais), GFP_KERNEL); + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -325,7 +593,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) * see * soc-core.c :: snd_soc_init_multicodec() */ - for (i = 0; i < num; i++) { + for (i = 0; i < lnum; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; dai_link[i].platform = &dai_props[i].platform; @@ -341,16 +609,19 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = dai_link; - card->num_links = num; - card->dapm_widgets = asoc_graph_card_dapm_widgets; - card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); - card->probe = asoc_graph_soc_card_probe; + card->owner = THIS_MODULE; + card->dev = dev; + card->dai_link = dai_link; + card->num_links = lnum; + card->dapm_widgets = asoc_graph_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); + card->probe = asoc_graph_soc_card_probe; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { @@ -381,6 +652,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static const struct of_device_id asoc_graph_of_match[] = { { .compatible = "audio-graph-card", }, + { .compatible = "audio-graph-scu-card", }, {}, }; MODULE_DEVICE_TABLE(of, asoc_graph_of_match); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: tidyup mclk-fs method" to the asoc tree
by Mark Brown 14 Dec '18

14 Dec '18

The patch ASoC: audio-graph-card: tidyup mclk-fs method has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 56eb818191fbe4b93bbe6e884f96061013c9a6ab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 14 Dec 2018 11:32:30 +0900 Subject: [PATCH] ASoC: audio-graph-card: tidyup mclk-fs method Current audio-graph-card is handling "mclk-fs" by many way. But, it is not useful and readable. We want to do is that allow having mclk-fs everywere. This patch support it. It will be overwrote if lower node has it. sound { mclk-fs = <xxx>; // initial }; codec { ports { mclk-fs = <xxx>; // overwrite port { mclk-fs = <xxx>; // overwrite endpoint { mclk-fs = <xxx>; // overwrite }; }; }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 52 +++++++++++++++++++--------- 1 file changed, 35 insertions(+), 17 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 8ec09bc711f8..7e32380e7a6f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -31,7 +31,6 @@ struct graph_card_data { struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; } *dai_props; - unsigned int mclk_fs; struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; @@ -115,9 +114,7 @@ static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, unsigned int mclk, mclk_fs = 0; int ret = 0; - if (priv->mclk_fs) - mclk_fs = priv->mclk_fs; - else if (dai_props->mclk_fs) + if (dai_props->mclk_fs) mclk_fs = dai_props->mclk_fs; if (mclk_fs) { @@ -176,7 +173,8 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int asoc_graph_card_dai_link_of_dpcm(struct device_node *cpu_ep, +static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, + struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, int *dai_idx, int link_idx, @@ -187,12 +185,21 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *cpu_ep, struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *ep = is_cpu ? cpu_ep : codec_ep; + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); struct device_node *node = of_graph_get_port_parent(ep); struct asoc_simple_dai *dai; int ret; dev_dbg(dev, "link_of DPCM (for %s)\n", is_cpu ? "CPU" : "Codec"); + of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(ports, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); + of_node_put(ports); + of_node_put(port); + if (is_cpu) { struct snd_soc_dai_link_component *codecs; @@ -279,8 +286,6 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *cpu_ep, if (ret < 0) return ret; - of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) @@ -294,7 +299,8 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *cpu_ep, return 0; } -static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, +static int asoc_graph_card_dai_link_of(struct device_node *top, + struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, int *dai_idx, int link_idx) @@ -302,6 +308,10 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); + struct device_node *cpu_port = of_get_parent(cpu_ep); + struct device_node *codec_port = of_get_parent(codec_ep); + struct device_node *cpu_ports = of_get_parent(cpu_port); + struct device_node *codec_ports = of_get_parent(codec_port); struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; int ret; @@ -313,14 +323,24 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, codec_dai = dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + /* Factor to mclk, used in hw_params() */ + of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_ports, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_ports, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_port, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_port, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); + of_node_put(cpu_port); + of_node_put(cpu_ports); + of_node_put(codec_port); + of_node_put(codec_ports); + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) return ret; - of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); - ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); if (ret < 0) return ret; @@ -370,7 +390,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct of_phandle_iterator it; struct device *dev = graph_priv_to_dev(priv); struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *node = dev->of_node; + struct device_node *top = dev->of_node; + struct device_node *node = top; struct device_node *cpu_port; struct device_node *cpu_ep = NULL; struct device_node *codec_ep = NULL; @@ -390,9 +411,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); - /* Factor to mclk, used in hw_params() */ - of_property_read_u32(node, "mclk-fs", &priv->mclk_fs); - link_idx = 0; dai_idx = 0; conf_idx = 0; @@ -436,7 +454,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_port_old = codec_port; } ret = asoc_graph_card_dai_link_of_dpcm( - cpu_ep, codec_ep, priv, + top, cpu_ep, codec_ep, priv, &dai_idx, link_idx++, &conf_idx, cpu); } else if (cpu) { @@ -444,7 +462,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) * for Normal sound */ ret = asoc_graph_card_dai_link_of( - cpu_ep, codec_ep, priv, + top, cpu_ep, codec_ep, priv, &dai_idx, link_idx++); } if (ret < 0) -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: tidyup convert_rate/channel method" to the asoc tree
by Mark Brown 14 Dec '18

14 Dec '18

The patch ASoC: audio-graph-card: tidyup convert_rate/channel method has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7ced65ffb82fa58c6e314c16770288fe9ce3dfc8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 14 Dec 2018 11:32:36 +0900 Subject: [PATCH] ASoC: audio-graph-card: tidyup convert_rate/channel method Current audio-graph-card is handling "convert_rate/channel" by many ways. But, it is not useful and readable. We want to do is that allow having it everywere. This patch support it. It will be overwrote if lower node has it. sound { convert-channels = <xxx>; // initial }; codec { audio-graph-card,convert-channels = <xxx>; // overwrite ports { convert_channels = <xxx>; // overwrite port { convert_channels = <xxx>; // overwrite endpoint { convert_channels = <xxx>; // overwrite }; }; }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 7e32380e7a6f..368b8ae463af 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -35,7 +35,6 @@ struct graph_card_data { struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; - struct asoc_simple_card_data adata; struct snd_soc_codec_conf *codec_conf; struct gpio_desc *pa_gpio; }; @@ -167,9 +166,6 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, asoc_simple_card_convert_fixup(&dai_props->adata, params); - /* overwrite by top level adata if exist */ - asoc_simple_card_convert_fixup(&priv->adata, params); - return 0; } @@ -197,6 +193,13 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_property_read_u32(ports, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); + + asoc_simple_card_parse_convert(dev, top, NULL, &dai_props->adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); + asoc_simple_card_parse_convert(dev, ports, NULL, &dai_props->adata); + asoc_simple_card_parse_convert(dev, port, NULL, &dai_props->adata); + asoc_simple_card_parse_convert(dev, ep, NULL, &dai_props->adata); + of_node_put(ports); of_node_put(port); @@ -276,8 +279,6 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, PREFIX "prefix"); } - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - ret = asoc_simple_card_of_parse_tdm(ep, dai); if (ret) return ret; @@ -409,8 +410,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret < 0) return ret; - asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); - link_idx = 0; dai_idx = 0; conf_idx = 0; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: rsnd: Add r8a774c0 support
by Fabrizio Castro 14 Dec '18

14 Dec '18

Document RZ/G2E (R8A774C0) SoC bindings. Signed-off-by: Fabrizio Castro <fabrizio.castro(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index d92b705..f93f56a 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -346,6 +346,7 @@ Required properties: - "renesas,rcar_sound-r8a7744" (RZ/G1N) - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) + - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7779" (R-Car H1) - "renesas,rcar_sound-r8a7790" (R-Car H2) -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Add more platform granularity
by Pierre-Louis Bossart 14 Dec '18

14 Dec '18

The current SKYLAKE kconfig is a all-you-can-eat selection that will support all known plaforms. This is however not necessarily a good thing: most platforms for SKL and KBL don't support the DSP, but a number of CNL/WHL ones do. Selecting this driver in all cases isn't really smart and will require users to muck with blacklists. Partition the configs to allow distributions to select on which platform this driver is used. Keep the existing SND_SOC_INTEL_SKYLAKE config to select everything for backwards compatibility. This patch does not provide new functionality, only finer-grained choices in supported platforms. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: solve Kconfig conflict with Hui Wang's patchset for KBL/RT5660 support sound/soc/intel/Kconfig | 73 ++++++++++++++++++++++++++++++---- sound/soc/intel/boards/Kconfig | 16 +++++++- sound/soc/intel/skylake/skl.c | 12 ++++++ 3 files changed, 92 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 18e717703685..99a62ba409df 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -102,15 +102,74 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI recommended option config SND_SOC_INTEL_SKYLAKE - tristate "SKL/BXT/KBL/GLK/CNL... Platforms" + tristate "All Skylake/SST Platforms" depends on PCI && ACPI - select SND_SOC_INTEL_SKYLAKE_COMMON + select SND_SOC_INTEL_SKL + select SND_SOC_INTEL_APL + select SND_SOC_INTEL_KBL + select SND_SOC_INTEL_GLK + select SND_SOC_INTEL_CNL + select SND_SOC_INTEL_CFL help - If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ - GeminiLake or CannonLake platform with the DSP enabled in the BIOS - then enable this option by saying Y or m. + This is a backwards-compatible option to select all devices + supported by the Intel SST/Skylake driver. This option is no + longer recommended and will be deprecated when the SOF + driver is introduced. Distributions should explicitly + select which platform uses this driver. + +config SND_SOC_INTEL_SKL + tristate "Skylake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel Skylake platform with the DSP enabled + in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_APL + tristate "Broxton/ApolloLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel Broxton/ApolloLake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_KBL + tristate "Kabylake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel Kabylake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_GLK + tristate "GeminiLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel GeminiLake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_CNL + tristate "CannonLake/WhiskyLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel CNL/WHL platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_CFL + tristate "CoffeeLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel CoffeeLake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_SKYLAKE_FAMILY + tristate + select SND_SOC_INTEL_SKYLAKE_COMMON -if SND_SOC_INTEL_SKYLAKE +if SND_SOC_INTEL_SKYLAKE_FAMILY config SND_SOC_INTEL_SKYLAKE_SSP_CLK tristate @@ -135,7 +194,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON GeminiLake or CannonLake platform with the DSP enabled in the BIOS then enable this option by saying Y or m. -endif ## SND_SOC_INTEL_SKYLAKE +endif ## SND_SOC_INTEL_SKYLAKE_FAMILY config SND_SOC_ACPI_INTEL_MATCH tristate diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3839d6205fcf..0a7e40d06395 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -172,7 +172,7 @@ config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH endif ## SND_SST_ATOM_HIFI2_PLATFORM -if SND_SOC_INTEL_SKYLAKE +if SND_SOC_INTEL_SKL config SND_SOC_INTEL_SKL_RT286_MACH tristate "SKL with RT286 I2S mode" @@ -212,6 +212,10 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +endif ## SND_SOC_INTEL_SKL + +if SND_SOC_INTEL_APL + config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -239,6 +243,10 @@ config SND_SOC_INTEL_BXT_RT298_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +endif ## SND_SOC_INTEL_APL + +if SND_SOC_INTEL_KBL + config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH tristate "KBL with RT5663 and MAX98927 in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -303,6 +311,10 @@ config SND_SOC_INTEL_KBL_RT5660_MACH create an alsa sound card for RT5660 I2S audio codec. Say Y if you have such a device. +endif ## SND_SOC_INTEL_KBL + +if SND_SOC_INTEL_GLK + config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -317,7 +329,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". -endif ## SND_SOC_INTEL_SKYLAKE +endif ## SND_SOC_INTEL_GLK if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f9888b1d5521..5abd35ca4e41 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -1142,24 +1142,36 @@ static void skl_remove(struct pci_dev *pci) /* PCI IDs */ static const struct pci_device_id skl_ids[] = { +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKL) /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) /* BXT-P */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_KBL) /* KBL */ { PCI_DEVICE(0x8086, 0x9D71), .driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) /* GLK */ { PCI_DEVICE(0x8086, 0x3198), .driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CNL) /* CNL */ { PCI_DEVICE(0x8086, 0x9dc8), .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CFL) /* CFL */ { PCI_DEVICE(0x8086, 0xa348), .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, +#endif { 0, } }; MODULE_DEVICE_TABLE(pci, skl_ids); -- 2.17.1

2 1

[alsa-devel] [PATCH v3 0/1] Add a new machine driver kbl_rt5660
by Hui Wang 14 Dec '18

14 Dec '18

In the V3, I addressed all comments in the V2 except one, the one is about: runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); I tried to change it to: runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_LE; snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); But there is only noise when playing sound or recording sound, so I keep it as it is in the v2. Here I paste the diff between V3 and V2, this is the change I did in the v3: sound/soc/intel/boards/kbl_rt5660.c | 59 +++++++++++-------- .../intel/common/soc-acpi-intel-kbl-match.c | 5 ++ 2 files changed, 38 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 0482e6a7fc2a..1dd9b32d612c 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// Copyright(c) 2018-19 Intel Corporation. +// Copyright(c) 2018-19 Canonical Corporation. /* * Intel Kabylake I2S Machine Driver with RT5660 Codec @@ -61,7 +61,7 @@ static const struct acpi_gpio_params mic_det_gpio = { GPIO_LINEIN_DET_INDEX, 0, static const struct acpi_gpio_mapping acpi_rt5660_gpios[] = { - { "lineout-mute-gpios", &lineout_mute_gpio , 1 }, + { "lineout-mute-gpios", &lineout_mute_gpio, 1 }, { "lineout-det-gpios", &lineout_det_gpio, 1 }, { "mic-det-gpios", &mic_det_gpio, 1 }, { NULL }, @@ -109,8 +109,6 @@ static const struct snd_kcontrol_new kabylake_rt5660_controls[] = { static const struct snd_soc_dapm_widget kabylake_rt5660_widgets[] = { SND_SOC_DAPM_MIC("Line In", NULL), SND_SOC_DAPM_LINE("Line Out", kabylake_5660_event_lineout), - SND_SOC_DAPM_SPK("DP", NULL), - SND_SOC_DAPM_SPK("HDMI", NULL), }; static const struct snd_soc_dapm_route kabylake_rt5660_map[] = { @@ -127,10 +125,12 @@ static const struct snd_soc_dapm_route kabylake_rt5660_map[] = { { "codec0_in", NULL, "ssp0 Rx" }, { "ssp0 Rx", NULL, "AIF1 Capture" }, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, { "hifi1", NULL, "iDisp1 Tx"}, { "iDisp1 Tx", NULL, "iDisp1_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, }; static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, @@ -164,35 +164,42 @@ static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) if (ret) dev_warn(component->dev, "Failed to add driver gpios\n"); - /* Request rt5660 GPIO for lineout mute control */ + /* Request rt5660 GPIO for lineout mute control, return if fails */ ctx->gpio_lo_mute = devm_gpiod_get(component->dev, "lineout-mute", - GPIOD_OUT_HIGH); + GPIOD_OUT_HIGH); if (IS_ERR(ctx->gpio_lo_mute)) { dev_err(component->dev, "Can't find GPIO_MUTE# gpio\n"); return PTR_ERR(ctx->gpio_lo_mute); } - /* Create and initialize headphone jack */ - if (!snd_soc_card_jack_new(rtd->card, "Lineout Jack", - SND_JACK_LINEOUT, &lineout_jack, - &lineout_jack_pin, 1)) { + /* Create and initialize headphone jack, this jack is not mandatory, don't return if fails */ + ret = snd_soc_card_jack_new(rtd->card, "Lineout Jack", + SND_JACK_LINEOUT, &lineout_jack, + &lineout_jack_pin, 1); + if (ret) + dev_warn(component->dev, "Can't create Lineout jack\n"); + else { lineout_jack_gpio.gpiod_dev = component->dev; - if (snd_soc_jack_add_gpios(&lineout_jack, 1, - &lineout_jack_gpio)) - dev_err(component->dev, "Can't add Lineout jack gpio\n"); - } else - dev_err(component->dev, "Can't create Lineout jack\n"); - - /* Create and initialize mic jack */ - if (!snd_soc_card_jack_new(rtd->card, "Mic Jack", - SND_JACK_MICROPHONE, &mic_jack, - &mic_jack_pin, 1)) { + ret = snd_soc_jack_add_gpios(&lineout_jack, 1, + &lineout_jack_gpio); + if (ret) + dev_warn(component->dev, "Can't add Lineout jack gpio\n"); + } + + /* Create and initialize mic jack, this jack is not mandatory, don't return if fails */ + ret = snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, &mic_jack, + &mic_jack_pin, 1); + if (ret) + dev_warn(component->dev, "Can't create mic jack\n"); + else { mic_jack_gpio.gpiod_dev = component->dev; - if (snd_soc_jack_add_gpios(&mic_jack, 1, &mic_jack_gpio)) - dev_err(component->dev, "Can't add mic jack gpio\n"); - } else - dev_err(component->dev, "Can't create mic jack\n"); + ret = snd_soc_jack_add_gpios(&mic_jack, 1, &mic_jack_gpio); + if (ret) + dev_warn(component->dev, "Can't add mic jack gpio\n"); + } + /* Here we enable some dapms in advance to reduce the pop noise for recording via line-in */ snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); snd_soc_dapm_force_enable_pin(dapm, "BST1"); snd_soc_dapm_force_enable_pin(dapm, "BST2"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index 1e41c7ded9e9..e6fa6f470526 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -96,6 +96,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .quirk_data = &kbl_7219_98927_codecs, .pdata = &skl_dmic_data }, + { + .id = "10EC5660", + .drv_name = "kbl_rt5660", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, { .id = "10EC3277", .drv_name = "kbl_rt5660", Hui Wang (1): ASoC: Intel: kbl_rt5660: Add a new machine driver for kbl with rt5660 sound/soc/intel/boards/Kconfig | 10 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_rt5660.c | 543 ++++++++++++++++++ .../intel/common/soc-acpi-intel-kbl-match.c | 10 + 4 files changed, 565 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_rt5660.c -- 2.17.1

3 4

[alsa-devel] [PATCH][RFC] ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID method
by Kuninori Morimoto 14 Dec '18

14 Dec '18

commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") fixuped getting DAI ID method. It will get DAI ID from OF graph "port", but, we want to consider about "endpoint", too. And, we also want to keep compatibility. This patch fixup it as if (driver has specified DAI ID) use it as DAI ID else if (OF graph endpoint has reg) use it as DAI ID else if (OF graph port has reg) use it as DAI ID else use endpoint count as DAI ID Fixes: commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") Reported-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Tony Can you test this patch ? It can keep compatibility, I think. > Mark I added [RFC] on this patch. If this patch was OK by Tony's test, I will re-post it as v3 patch-set. sound/soc/generic/simple-card-utils.c | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 17d8aee..b807a47 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -269,22 +269,46 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_parse_dai); static int asoc_simple_card_get_dai_id(struct device_node *ep) { + struct device_node *node; + struct device_node *endpoint; struct of_endpoint info; + int i, id; int ret; + /* use driver specified DAI ID if exist */ ret = snd_soc_get_dai_id(ep); if (ret != -ENOTSUPP) return ret; + /* use endpoint/port reg if exist */ + ret = of_graph_parse_endpoint(ep, &info); + if (ret == 0) { + if (info.id) + return info.id; + if (info.port) + return info.port; + } + + node = of_graph_get_port_parent(ep); + /* * Non HDMI sound case, counting port/endpoint on its DT * is enough. Let's count it. */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret) - return -ENXIO; + i = 0; + id = -1; + for_each_endpoint_of_node(node, endpoint) { + if (endpoint == ep) + id = i; + i++; + } + + of_node_put(node); + + if (id < 0) + return -ENODEV; - return info.port; + return id; } int asoc_simple_card_parse_graph_dai(struct device_node *ep, -- 2.7.4

2 2

[alsa-devel] [PATCH v2 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 14 Dec '18

14 Dec '18

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- include/sound/cs35l36.h | 58 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36-tables.c | 312 +++++++ sound/soc/codecs/cs35l36.c | 1665 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 452 ++++++++++ 6 files changed, 2494 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36-tables.c create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..00470f9 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,58 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H +#define __CS35L36_H + +/* INT/GPIO pin configuration */ +struct irq_cfg { + bool is_present; + int irq_drv_sel; + int irq_pol; + int irq_gpio_sel; + int irq_out_en; + int irq_src_sel; +}; + +struct vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool sclk_frc; + bool lrclk_frc; + bool multi_amp_mode; + bool dcm_mode; + int ldm_mode_sel; + bool amp_gain_zc; + bool amp_pcm_inv; + bool pdm_ldm_exit; + bool pdm_ldm_enter; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + struct vpbr_cfg vpbr_config; + struct irq_cfg irq_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f18..77cb861 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -465,6 +466,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c..7a6009a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,6 +45,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o cs35l36-tables.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -310,6 +311,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36-tables.c b/sound/soc/codecs/cs35l36-tables.c new file mode 100644 index 0000000..cbfda81 --- /dev/null +++ b/sound/soc/codecs/cs35l36-tables.c @@ -0,0 +1,312 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36-tables.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#include "cs35l36.h" + +struct reg_default cs35l36_reg[sizeof(CS35L36_MAX_NUM_REGS)] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; +EXPORT_SYMBOL_GPL(cs35l36_reg); + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} +EXPORT_SYMBOL_GPL(cs35l36_readable_reg); + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} +EXPORT_SYMBOL_GPL(cs35l36_volatile_reg); diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..1c6134b --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1665 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int chip_version; + int rev_id; + struct gpio_desc *reset_gpio; +}; + +struct cs35l36_pll_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static int cs35l36_ldm_sel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = cs35l36->pdata.ldm_mode_sel; + + return 0; +} + +static int cs35l36_ldm_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int val = (ucontrol->value.integer.value[0]) ? CS35L36_NG_AMP_EN_MASK : + 0; + + cs35l36->pdata.ldm_mode_sel = val; + + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, val); + + return 0; +} + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("Analog PCM Volume", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), + SOC_SINGLE("Amp Gain Zero-Cross", CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Enter Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Exit Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_SHIFT, 1, 0), + SOC_SINGLE_BOOL_EXT("LDM Select", 0, cs35l36_ldm_sel_get, + cs35l36_ldm_sel_put), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + + if (WARN_ON_ONCE(reg & CS35L36_PLL_UNLOCK_MASK)) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + return ret; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + + return 0; +} + +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux = + SOC_DAPM_ENUM("Input Mux", chansel_enum); + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const struct snd_kcontrol_new boost_ctrl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, &cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH_E("BOOST Enable", SND_SOC_NOPM, 0, 0, &boost_ctrl, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Enable", "Switch", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Enable"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + clk_frc = 1; + break; + case SND_SOC_DAIFMT_GATED: + clk_frc = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, clk_frc << + CS35L36_SCLK_FRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, clk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + unsigned int asp_width, global_fs = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1, fs2, reg; + + if (freq > CS35L36_FS_NOM_6MHZ) { + fs1 = CS35L36_FS1_DEFAULT_VAL; + fs2 = CS35L36_FS2_DEFAULT_VAL; + } else { + fs1 = 3 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + fs2 = 5 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + } + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_read(cs35l36->regmap, CS35L36_TST_FS_MON0, &reg); + reg &= ~(CS35L36_FS1_WINDOW_MASK | CS35L36_FS2_WINDOW_MASK); + reg |= ((fs1 & CS35L36_FS1_WINDOW_MASK) | + (fs2 << CS35L36_FS2_WINDOW_SHIFT & CS35L36_FS2_WINDOW_MASK)); + regmap_write(cs35l36->regmap, CS35L36_TST_FS_MON0, reg); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static const struct cs35l36_pll_config *cs35l36_get_clk_config( + struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) + return &cs35l36_pll_sysclk[i]; + } + + return NULL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + const struct cs35l36_pll_config *clk_cfg; + int prev_clksrc; + bool pdm_switch; + + prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + clk_cfg = cs35l36_get_clk_config(cs35l36, freq); + if (clk_cfg == NULL) { + dev_err(component->dev, "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + clk_cfg->clk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, + cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + clk_cfg->fll_igain); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc != CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } else { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc == CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if ((cs35l36->rev_id == CS35L36_REV_A0) && cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) { + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + if (ret < 0) { + dev_err(cs35l36->dev, + "Boost inductor config failed(%d)\n", ret); + return ret; + } + } + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + int ret = IRQ_NONE; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, status, + ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, masks, + ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, CS35L36_TEMP_WARN); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, CS35L36_TEMP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, CS35L36_BST_OVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct irq_cfg *irq_gpio_config = &pdata->irq_config; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *irq_gpio, *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + /* INT/GPIO Pin Config */ + irq_gpio = of_get_child_by_name(np, "cirrus,irq-config"); + irq_gpio_config->is_present = irq_gpio ? true : false; + if (irq_gpio_config->is_present) { + if (of_property_read_u32(irq_gpio, "cirrus,irq-drive-select", + &val) >= 0) + irq_gpio_config->irq_drv_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-polarity", + &val) >= 0) + irq_gpio_config->irq_pol = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-gpio-select", + &val) >= 0) + irq_gpio_config->irq_gpio_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-output-enable", + &val) >= 0) + irq_gpio_config->irq_out_en = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-src-select", + &val) >= 0) + irq_gpio_config->irq_src_sel = val; + } + of_node_put(irq_gpio); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id != CS35L36_REV_B0) + return 0; + + /* + * Magic code for internal PAC + */ + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + + return 0; +} + +static int cs35l36_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); + + return 0; +} + +static int cs35l36_irq_gpio_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct irq_cfg *irq_config = &pdata->irq_config; + int irq_polarity; + + /* setup the Interrupt Pin (INT | GPIO) */ + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, + irq_config->irq_out_en); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + irq_config->irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + irq_config->irq_pol << + CS35L36_INT_POL_SEL_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_IRQ_SRC_MASK, + irq_config->irq_src_sel << + CS35L36_IRQ_SRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + irq_config->irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (irq_config->irq_pol) + irq_polarity = IRQF_TRIGGER_HIGH; + else + irq_polarity = IRQF_TRIGGER_LOW; + + return irq_polarity; +} + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + /* Errata Writes */ + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + /* PAC Config */ + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + + int i; + int ret; + u32 reg_id, reg_revid, l37_id_reg; + int irq_pol = IRQF_TRIGGER_HIGH; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l36->pdata = *pdata; + } else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) { + dev_info(dev, "Reset line busy, assuming shared reset\n"); + } else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err_disable_regs; + } + } + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", reg_id, + CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_vpbr_config(cs35l36); + + if (pdata->irq_config.is_present) + irq_pol = cs35l36_irq_gpio_config(cs35l36); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | irq_pol, "cs35l36", + cs35l36); + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + +err_disable_regs: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + snd_soc_unregister_component(&client->dev); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..9ab040f --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,452 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +typedef char CS35L36_MAX_NUM_REGS[__LINE__]; + +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_SHIFT 10 +#define CS35L36_PDM_LDM_ENTER_SHIFT 3 +#define CS35L36_PDM_LDM_EXIT_SHIFT 4 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_FS1_DEFAULT_VAL 16 +#define CS35L36_FS2_DEFAULT_VAL 36 +#define CS35L36_FS_NOM_6MHZ 6000000 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg); +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg); + +extern struct reg_default cs35l36_reg[sizeof(CS35L36_MAX_NUM_REGS)]; +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

3 4

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: probe and Kconfig simplifications
by Pierre-Louis Bossart 13 Dec '18

13 Dec '18

The patches add support for CoffeeLake, harden and simplify the probe for the Skylake driver, introduce more compile-time granularity so that platforms can be selected individually. The main difference since the RFC posted on November 20 is the addition of the pci class/subclass/prog-if information. This greatly simplifies the detection of the DSP and will allow for an automatic selection between HDaudio legacy and Skylake drivers. As suggested by Takashi this automatic selection will be posted separately (it will result on changes on both sound/pci/hda and sound/soc/intel sides) Pierre-Louis Bossart (5): ASoC: Intel: Skylake: Harden DSP detection with PCI class/subclass info ASoC: Intel: Skylake: stop probe if HDaudio capabilities don't exist ASoC: Intel: Skylake: remove useless tests on HDaudio capabilities ASoC: Intel: Skylake: add error logs on probe, remove dependency on NHLT ASoC: Intel: Skylake: Add more platform granularity Takashi Iwai (1): ASoC: Intel: Skylake: Add CFL-S support sound/soc/intel/Kconfig | 73 +++++++++++++-- sound/soc/intel/boards/Kconfig | 16 +++- sound/soc/intel/skylake/skl-messages.c | 8 ++ sound/soc/intel/skylake/skl-nhlt.c | 3 + sound/soc/intel/skylake/skl.c | 125 +++++++++++++++++-------- 5 files changed, 179 insertions(+), 46 deletions(-) -- 2.17.1

2 15

[alsa-devel] [PATCH] ASoC: pcm3060: Add powersaving widgets for DAC and ADC
by Kirill Marinushkin 13 Dec '18

13 Dec '18

Enable DAC/ADC only when playing/capturing Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- sound/soc/codecs/pcm3060.c | 14 ++++++++++---- sound/soc/codecs/pcm3060.h | 2 ++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 1dd851a7b43b..6714aa8d9026 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -198,19 +198,25 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", PCM3060_REG64, + PCM3060_REG_SHIFT_DAPSV, 1), + SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), + + SND_SOC_DAPM_ADC("ADC", "Capture", PCM3060_REG64, + PCM3060_REG_SHIFT_ADPSV, 1), }; static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { - { "OUTL", NULL, "Playback" }, - { "OUTR", NULL, "Playback" }, + { "OUTL", NULL, "DAC" }, + { "OUTR", NULL, "DAC" }, - { "Capture", NULL, "INL" }, - { "Capture", NULL, "INR" }, + { "ADC", NULL, "INL" }, + { "ADC", NULL, "INR" }, }; /* soc component */ diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index c895cf40ee10..6a027b4a845d 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -37,7 +37,9 @@ int pcm3060_remove(struct device *dev); #define PCM3060_REG_MRST 0x80 #define PCM3060_REG_SRST 0x40 #define PCM3060_REG_ADPSV 0x20 +#define PCM3060_REG_SHIFT_ADPSV 0x05 #define PCM3060_REG_DAPSV 0x10 +#define PCM3060_REG_SHIFT_DAPSV 0x04 #define PCM3060_REG_SE 0x01 #define PCM3060_REG65 0x41 -- 2.13.6

2 1

[alsa-devel] Applied "ASoC: Intel: Skylake: Add CFL-S support" to the asoc tree
by Mark Brown 13 Dec '18

13 Dec '18

The patch ASoC: Intel: Skylake: Add CFL-S support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e6b98db945124987b1ecec3f5f030877627e01a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 7 Dec 2018 17:50:03 -0600 Subject: [PATCH] ASoC: Intel: Skylake: Add CFL-S support It's with CNP, supposed to be equivalent with CNL entry. Keep the existing declaration style for now, at a later point we may transition and use PCI_DEVICE_DATA(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 8 ++++++++ sound/soc/intel/skylake/skl.c | 3 +++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 8bfb8b0fa3d5..b0e6fb93eaf8 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -247,6 +247,14 @@ static const struct skl_dsp_ops dsp_ops[] = { .init_fw = cnl_sst_init_fw, .cleanup = cnl_sst_dsp_cleanup }, + { + .id = 0xa348, + .num_cores = 4, + .loader_ops = bxt_get_loader_ops, + .init = cnl_sst_dsp_init, + .init_fw = cnl_sst_init_fw, + .cleanup = cnl_sst_dsp_cleanup + }, }; const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 3f0ac1312982..df36b8fe6d5e 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -1121,6 +1121,9 @@ static const struct pci_device_id skl_ids[] = { /* CNL */ { PCI_DEVICE(0x8086, 0x9dc8), .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, + /* CFL */ + { PCI_DEVICE(0x8086, 0xa348), + .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, { 0, } }; MODULE_DEVICE_TABLE(pci, skl_ids); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: remove useless tests on HDaudio capabilities" to the asoc tree
by Mark Brown 13 Dec '18

13 Dec '18

The patch ASoC: Intel: Skylake: remove useless tests on HDaudio capabilities has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f981bdcf55fda28a9a70c9e8151dd200771a0a8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Fri, 7 Dec 2018 17:50:06 -0600 Subject: [PATCH] ASoC: Intel: Skylake: remove useless tests on HDaudio capabilities bus->ppcap is now tested upfront, there is no need to re-check if the hardware is exposed as needed. Remove tests and remove indentation. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 40 ++++++++++++++++------------------- 1 file changed, 18 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 72e366bbba33..f04e9ec4432b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -826,12 +826,10 @@ static void skl_probe_work(struct work_struct *work) return; } - if (bus->ppcap) { - err = skl_machine_device_register(skl); - if (err < 0) { - dev_err(bus->dev, "machine register failed: %d\n", err); - goto out_err; - } + err = skl_machine_device_register(skl); + if (err < 0) { + dev_err(bus->dev, "machine register failed: %d\n", err); + goto out_err; } /* @@ -1034,25 +1032,23 @@ static int skl_probe(struct pci_dev *pci, pci_set_drvdata(skl->pci, bus); - /* check if dsp is there */ - if (bus->ppcap) { - /* create device for dsp clk */ - err = skl_clock_device_register(skl); - if (err < 0) - goto out_clk_free; + /* create device for dsp clk */ + err = skl_clock_device_register(skl); + if (err < 0) + goto out_clk_free; - err = skl_find_machine(skl, (void *)pci_id->driver_data); - if (err < 0) - goto out_nhlt_free; + err = skl_find_machine(skl, (void *)pci_id->driver_data); + if (err < 0) + goto out_nhlt_free; - err = skl_init_dsp(skl); - if (err < 0) { - dev_dbg(bus->dev, "error failed to register dsp\n"); - goto out_nhlt_free; - } - skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge; - skl->skl_sst->clock_power_gating = skl_clock_power_gating; + err = skl_init_dsp(skl); + if (err < 0) { + dev_dbg(bus->dev, "error failed to register dsp\n"); + goto out_nhlt_free; } + skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge; + skl->skl_sst->clock_power_gating = skl_clock_power_gating; + if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: meson: fix do_div warning in spdifin
by Jerome Brunet 13 Dec '18

13 Dec '18

Even if this spdif input driver is only supposed to be used on 64bits platform, there is possible problem with 32bits and do_div, as reported by the kbuild robot. Just fix it. Fixes: 5ce5658375e6 ("ASoC: meson: add axg spdif input") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Hi Mark, This fixes the problem reported by kbuild robot [0]. Sorry for not catching this earlier. Regards Jerome [0]: https://lkml.kernel.org/r/201812132245.lBYydcs3%fengguang.wu@intel.com sound/soc/meson/axg-spdifin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index 09f385a2bbba..01b2035fa841 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -146,8 +146,8 @@ static void axg_spdifin_write_mode_param(struct regmap *map, int mode, unsigned int base_reg, unsigned int width) { - unsigned int offset = mode, rem; - unsigned int reg, shift; + uint64_t offset = mode; + unsigned int reg, shift, rem; rem = do_div(offset, num_per_reg); -- 2.19.2

2 1

[alsa-devel] [PATCH] ASoC: core: Invoke pcm_new() for all DAI-link
by Rohit kumar 13 Dec '18

13 Dec '18

Remove no_pcm check to invoke pcm_new() for backend dai-links too. This fixes crash in hdmi codec driver during hdmi_codec_startup() while accessing chmap_info struct. chmap_info struct memory is allocated in pcm_new() of hdmi codec driver which is not invoked in case of DPCM when hdmi codec driver is part of backend dai-link. Below is the crash stack: [ 61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018 .. [ 61.666696] CM = 0, WnR = 1 [ 61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000 [ 61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000 [ 61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP [ 61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21 .. [ 61.740269] PC is at hdmi_codec_startup+0x124/0x164 [ 61.745308] LR is at hdmi_codec_startup+0xe4/0x164 Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ddcf12..abdc460 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1467,7 +1467,7 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, for (i = 0; i < num_dais; ++i) { struct snd_soc_dai_driver *drv = dais[i]->driver; - if (!rtd->dai_link->no_pcm && drv->pcm_new) + if (drv->pcm_new) ret = drv->pcm_new(rtd, dais[i]); if (ret < 0) { dev_err(dais[i]->dev, -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 8

[alsa-devel] [PATCH 0/2] Graph fixes for using multiple endpoints per port
by Tony Lindgren 13 Dec '18

13 Dec '18

Hi all, Here are two fixes that allow me to have multiple endpoints defined in the dts file for audio-graph-card. To do that, we need to fix up few issues as the graph binding Documentation/devicetree/bindings/graph.txt allows multiple endpoints per port. This allows configuring TDM codecs for I2S for example. Kuninori-san, can you please check if this makes sense to you and compare against the graph binding? Cheers, Tony Tony Lindgren (2): ASoC: simple-card-utils: revert port changes to follow graph binding ASoC: audio-graph-card: Fix parsing of multiple endpoints sound/soc/generic/audio-graph-card.c | 41 +++++++++++++++++++++++---- sound/soc/generic/simple-card-utils.c | 22 ++++++++++++-- 2 files changed, 55 insertions(+), 8 deletions(-) -- 2.19.2

3 28

[alsa-devel] [PATCH 0/7 v2] ASoC: merge audio-graph-scu into audio-graph - final step
by Kuninori Morimoto 13 Dec '18

13 Dec '18

Hi Mark, Tony audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. This is v2 of final step patches. Tony, I think 1) and 3) patch will solve your issue. Can you please test these ? Kuninori Morimoto (7): 1) ASoC: simple-card-utils: care endpoint reg for asoc_simple_card_get_dai_id() 2) ASoC: audio-graph-card: merge audio-graph-scu-card on Doc 3) ASoC: audio-graph-card: merge audio-graph-scu-card 4) ASoC: audio-graph-card: tidyup mclk-fs method 5) ASoC: audio-graph-card: tidyup convert_rate/channel method 6) ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc 7) ASoC: audio-graph-scu-card: remove audio-graph-scu-card .../devicetree/bindings/sound/audio-graph-card.txt | 217 +++++++++ .../bindings/sound/audio-graph-scu-card.txt | 123 ----- sound/soc/generic/Kconfig | 10 +- sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-card.c | 405 ++++++++++++++--- sound/soc/generic/audio-graph-scu-card.c | 501 --------------------- sound/soc/generic/simple-card-utils.c | 3 + 7 files changed, 568 insertions(+), 693 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt delete mode 100644 sound/soc/generic/audio-graph-scu-card.c -- 2.7.4

3 16

[alsa-devel] [asoc:for-4.21 167/168] sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div'
by kbuild test robot 13 Dec '18

13 Dec '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.21 head: 1b46ed96961db1a826b3e5cadac18c7c4857f054 commit: 5ce5658375e6de2468c4884f7ae474c4ed40a13f [167/168] ASoC: meson: add axg spdif input config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 5ce5658375e6de2468c4884f7ae474c4ed40a13f # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm All warnings (new ones prefixed by >>): In file included from arch/arm/include/asm/div64.h:127:0, from include/linux/kernel.h:207, from include/linux/clk.h:16, from sound/soc//meson/axg-spdifin.c:7: sound/soc//meson/axg-spdifin.c: In function 'axg_spdifin_write_mode_param': include/asm-generic/div64.h:222:28: warning: comparison of distinct pointer types lacks a cast (void)(((typeof((n)) *)0) == ((uint64_t *)0)); \ ^ >> sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div' rem = do_div(offset, num_per_reg); ^~~~~~ In file included from include/linux/build_bug.h:5:0, from include/linux/bitfield.h:18, from sound/soc//meson/axg-spdifin.c:6: include/asm-generic/div64.h:235:25: warning: right shift count >= width of type [-Wshift-count-overflow] } else if (likely(((n) >> 32) == 0)) { \ ^ include/linux/compiler.h:76:40: note: in definition of macro 'likely' # define likely(x) __builtin_expect(!!(x), 1) ^ >> sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div' rem = do_div(offset, num_per_reg); ^~~~~~ In file included from arch/arm/include/asm/div64.h:127:0, from include/linux/kernel.h:207, from include/linux/clk.h:16, from sound/soc//meson/axg-spdifin.c:7: include/asm-generic/div64.h:239:22: error: passing argument 1 of '__div64_32' from incompatible pointer type [-Werror=incompatible-pointer-types] __rem = __div64_32(&(n), __base); \ ^ >> sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div' rem = do_div(offset, num_per_reg); ^~~~~~ In file included from include/linux/kernel.h:207:0, from include/linux/clk.h:16, from sound/soc//meson/axg-spdifin.c:7: arch/arm/include/asm/div64.h:33:24: note: expected 'uint64_t * {aka long long unsigned int *}' but argument is of type 'unsigned int *' static inline uint32_t __div64_32(uint64_t *n, uint32_t base) ^~~~~~~~~~ cc1: some warnings being treated as errors vim +/do_div +152 sound/soc//meson/axg-spdifin.c 5 6 #include <linux/bitfield.h> > 7 #include <linux/clk.h> 8 #include <linux/module.h> 9 #include <linux/of_platform.h> 10 #include <linux/regmap.h> 11 #include <sound/soc.h> 12 #include <sound/soc-dai.h> 13 #include <sound/pcm_params.h> 14 15 #define SPDIFIN_CTRL0 0x00 16 #define SPDIFIN_CTRL0_EN BIT(31) 17 #define SPDIFIN_CTRL0_RST_OUT BIT(29) 18 #define SPDIFIN_CTRL0_RST_IN BIT(28) 19 #define SPDIFIN_CTRL0_WIDTH_SEL BIT(24) 20 #define SPDIFIN_CTRL0_STATUS_CH_SHIFT 11 21 #define SPDIFIN_CTRL0_STATUS_SEL GENMASK(10, 8) 22 #define SPDIFIN_CTRL0_SRC_SEL GENMASK(5, 4) 23 #define SPDIFIN_CTRL0_CHK_VALID BIT(3) 24 #define SPDIFIN_CTRL1 0x04 25 #define SPDIFIN_CTRL1_BASE_TIMER GENMASK(19, 0) 26 #define SPDIFIN_CTRL1_IRQ_MASK GENMASK(27, 20) 27 #define SPDIFIN_CTRL2 0x08 28 #define SPDIFIN_THRES_PER_REG 3 29 #define SPDIFIN_THRES_WIDTH 10 30 #define SPDIFIN_CTRL3 0x0c 31 #define SPDIFIN_CTRL4 0x10 32 #define SPDIFIN_TIMER_PER_REG 4 33 #define SPDIFIN_TIMER_WIDTH 8 34 #define SPDIFIN_CTRL5 0x14 35 #define SPDIFIN_CTRL6 0x18 36 #define SPDIFIN_STAT0 0x1c 37 #define SPDIFIN_STAT0_MODE GENMASK(30, 28) 38 #define SPDIFIN_STAT0_MAXW GENMASK(17, 8) 39 #define SPDIFIN_STAT0_IRQ GENMASK(7, 0) 40 #define SPDIFIN_IRQ_MODE_CHANGED BIT(2) 41 #define SPDIFIN_STAT1 0x20 42 #define SPDIFIN_STAT2 0x24 43 #define SPDIFIN_MUTE_VAL 0x28 44 45 #define SPDIFIN_MODE_NUM 7 46 47 struct axg_spdifin_cfg { 48 const unsigned int *mode_rates; 49 unsigned int ref_rate; 50 }; 51 52 struct axg_spdifin { 53 const struct axg_spdifin_cfg *conf; 54 struct regmap *map; 55 struct clk *refclk; 56 struct clk *pclk; 57 }; 58 59 /* 60 * TODO: 61 * It would have been nice to check the actual rate against the sample rate 62 * requested in hw_params(). Unfortunately, I was not able to make the mode 63 * detection and IRQ work reliably: 64 * 65 * 1. IRQs are generated on mode change only, so there is no notification 66 * on transition between no signal and mode 0 (32kHz). 67 * 2. Mode detection very often has glitches, and may detects the 68 * lowest or the highest mode before zeroing in on the actual mode. 69 * 70 * This makes calling snd_pcm_stop() difficult to get right. Even notifying 71 * the kcontrol would be very unreliable at this point. 72 * Let's keep things simple until the magic spell that makes this work is 73 * found. 74 */ 75 76 static unsigned int axg_spdifin_get_rate(struct axg_spdifin *priv) 77 { 78 unsigned int stat, mode, rate = 0; 79 80 regmap_read(priv->map, SPDIFIN_STAT0, &stat); 81 mode = FIELD_GET(SPDIFIN_STAT0_MODE, stat); 82 83 /* 84 * If max width is zero, we are not capturing anything. 85 * Also Sometimes, when the capture is on but there is no data, 86 * mode is SPDIFIN_MODE_NUM, but not always ... 87 */ 88 if (FIELD_GET(SPDIFIN_STAT0_MAXW, stat) && 89 mode < SPDIFIN_MODE_NUM) 90 rate = priv->conf->mode_rates[mode]; 91 92 return rate; 93 } 94 95 static int axg_spdifin_prepare(struct snd_pcm_substream *substream, 96 struct snd_soc_dai *dai) 97 { 98 struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); 99 100 /* Apply both reset */ 101 regmap_update_bits(priv->map, SPDIFIN_CTRL0, 102 SPDIFIN_CTRL0_RST_OUT | 103 SPDIFIN_CTRL0_RST_IN, 104 0); 105 106 /* Clear out reset before in reset */ 107 regmap_update_bits(priv->map, SPDIFIN_CTRL0, 108 SPDIFIN_CTRL0_RST_OUT, SPDIFIN_CTRL0_RST_OUT); 109 regmap_update_bits(priv->map, SPDIFIN_CTRL0, 110 SPDIFIN_CTRL0_RST_IN, SPDIFIN_CTRL0_RST_IN); 111 112 return 0; 113 } 114 115 static int axg_spdifin_startup(struct snd_pcm_substream *substream, 116 struct snd_soc_dai *dai) 117 { 118 struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); 119 int ret; 120 121 ret = clk_prepare_enable(priv->refclk); 122 if (ret) { 123 dev_err(dai->dev, 124 "failed to enable spdifin reference clock\n"); 125 return ret; 126 } 127 128 regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 129 SPDIFIN_CTRL0_EN); 130 131 return 0; 132 } 133 134 static void axg_spdifin_shutdown(struct snd_pcm_substream *substream, 135 struct snd_soc_dai *dai) 136 { 137 struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); 138 139 regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); 140 clk_disable_unprepare(priv->refclk); 141 } 142 143 static void axg_spdifin_write_mode_param(struct regmap *map, int mode, 144 unsigned int val, 145 unsigned int num_per_reg, 146 unsigned int base_reg, 147 unsigned int width) 148 { 149 unsigned int offset = mode, rem; 150 unsigned int reg, shift; 151 > 152 rem = do_div(offset, num_per_reg); 153 154 reg = offset * regmap_get_reg_stride(map) + base_reg; 155 shift = width * (num_per_reg - 1 - rem); 156 157 regmap_update_bits(map, reg, GENMASK(width - 1, 0) << shift, 158 val << shift); 159 } 160 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2 0/7] ALSA: HD-audio display power fixes
by Takashi Iwai 13 Dec '18

13 Dec '18

Hi, this patch a revised patchset for fixing possible HD-audio display power unbalance. Basically the "fix" is done by refactoring the whole relevant code. It starts from the Intel HD-audio runtime PM refactoring, followed by the display PM API change, and lots of code cleanups. Since it changes the display power API function, it hits both legacy and ASoC drivers. The patches are kept in topic/hda-pm-refactor branch of my sound git tree. This will be an immutable branch once when merged to for-next, so that it can be merged to ASoC tree if any conflicting change needs to be applied. The major change in v2 series is the complete removal of AZX_DCAPS_I915_POWERWELL flag in patch#3, and other minor fixes suggested by Pierre. Takashi === Takashi Iwai (7): ALSA: hda/intel: Refactoring PM code ALSA: hda: Refactor display power management ALSA: hda/intel: Drop superfluous AZX_DCAPS_I915_POWERWELL checks ALSA: hda/intel: Properly free the display power at error path ALSA: hda: Make snd_hdac_display_power() void function ASoC: hdac_hdmi: Add missing display power-off at driver removal ALSA: hda/hdmi: Always set display_power_control for Intel HSW+ codecs include/sound/hda_codec.h | 1 + include/sound/hda_component.h | 11 +- include/sound/hdaudio.h | 7 +- sound/hda/hdac_component.c | 39 +++--- sound/hda/hdac_device.c | 17 --- sound/pci/hda/hda_codec.c | 16 ++- sound/pci/hda/hda_controller.c | 11 -- sound/pci/hda/hda_controller.h | 6 +- sound/pci/hda/hda_intel.c | 220 +++++++++++++-------------------- sound/pci/hda/patch_hdmi.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 24 +--- sound/soc/intel/skylake/skl.c | 40 ++---- 12 files changed, 148 insertions(+), 252 deletions(-) -- 2.19.2

2 9

[alsa-devel] [PATCH] ASoC: rsnd: remove RSND_REG_ from rsnd_reg
by Kuninori Morimoto 13 Dec '18

13 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd is using RSND_REG_xxx for register naming, and using RSND_REG_##f style macro for read/write. The biggest reason why it uses this style is that we can avoid non-existing register access. But, its demerit is sequential register access code will be very ugly. Current rsnd driver is well tested, so, let's remove RSND_REG_ from rsnd_reg, and cleanup sequential register access code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 36 +----- sound/soc/sh/rcar/ctu.c | 69 +++------- sound/soc/sh/rcar/dvc.c | 10 +- sound/soc/sh/rcar/gen.c | 23 ++-- sound/soc/sh/rcar/rsnd.h | 329 +++++++++++++++++++++++------------------------ sound/soc/sh/rcar/ssiu.c | 59 ++------- 6 files changed, 212 insertions(+), 314 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66..e821ccc7 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -249,28 +249,8 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, out = out << shift; mask = 0x0f1f << shift; - switch (id / 2) { - case 0: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out); - break; - case 1: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out); - break; - case 2: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out); - break; - case 3: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out); - break; - case 4: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out); - break; - } + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL(id / 2), mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL(id / 2), mask, out); if (en) rsnd_mod_bset(adg_mod, DIV_EN, en, en); @@ -299,17 +279,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) if (id == 8) return; - switch (id / 4) { - case 0: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val); - break; - case 1: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val); - break; - case 2: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val); - break; - } + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL(id / 4), mask, val); dev_dbg(dev, "AUDIO_CLK_SEL is 0x%x\n", val); } diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index ff41a5e..8cb06da 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -72,10 +72,7 @@ struct rsnd_ctu { struct rsnd_mod mod; struct rsnd_kctrl_cfg_m pass; - struct rsnd_kctrl_cfg_m sv0; - struct rsnd_kctrl_cfg_m sv1; - struct rsnd_kctrl_cfg_m sv2; - struct rsnd_kctrl_cfg_m sv3; + struct rsnd_kctrl_cfg_m sv[4]; struct rsnd_kctrl_cfg_s reset; int channels; u32 flags; @@ -120,7 +117,7 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); u32 cpmdr = 0; u32 scmdr = 0; - int i; + int i, j; for (i = 0; i < RSND_MAX_CHANNELS; i++) { u32 val = rsnd_kctrl_valm(ctu->pass, i); @@ -139,45 +136,13 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, rsnd_mod_write(mod, CTU_SCMDR, scmdr); - if (scmdr > 0) { - rsnd_mod_write(mod, CTU_SV00R, rsnd_kctrl_valm(ctu->sv0, 0)); - rsnd_mod_write(mod, CTU_SV01R, rsnd_kctrl_valm(ctu->sv0, 1)); - rsnd_mod_write(mod, CTU_SV02R, rsnd_kctrl_valm(ctu->sv0, 2)); - rsnd_mod_write(mod, CTU_SV03R, rsnd_kctrl_valm(ctu->sv0, 3)); - rsnd_mod_write(mod, CTU_SV04R, rsnd_kctrl_valm(ctu->sv0, 4)); - rsnd_mod_write(mod, CTU_SV05R, rsnd_kctrl_valm(ctu->sv0, 5)); - rsnd_mod_write(mod, CTU_SV06R, rsnd_kctrl_valm(ctu->sv0, 6)); - rsnd_mod_write(mod, CTU_SV07R, rsnd_kctrl_valm(ctu->sv0, 7)); - } - if (scmdr > 1) { - rsnd_mod_write(mod, CTU_SV10R, rsnd_kctrl_valm(ctu->sv1, 0)); - rsnd_mod_write(mod, CTU_SV11R, rsnd_kctrl_valm(ctu->sv1, 1)); - rsnd_mod_write(mod, CTU_SV12R, rsnd_kctrl_valm(ctu->sv1, 2)); - rsnd_mod_write(mod, CTU_SV13R, rsnd_kctrl_valm(ctu->sv1, 3)); - rsnd_mod_write(mod, CTU_SV14R, rsnd_kctrl_valm(ctu->sv1, 4)); - rsnd_mod_write(mod, CTU_SV15R, rsnd_kctrl_valm(ctu->sv1, 5)); - rsnd_mod_write(mod, CTU_SV16R, rsnd_kctrl_valm(ctu->sv1, 6)); - rsnd_mod_write(mod, CTU_SV17R, rsnd_kctrl_valm(ctu->sv1, 7)); - } - if (scmdr > 2) { - rsnd_mod_write(mod, CTU_SV20R, rsnd_kctrl_valm(ctu->sv2, 0)); - rsnd_mod_write(mod, CTU_SV21R, rsnd_kctrl_valm(ctu->sv2, 1)); - rsnd_mod_write(mod, CTU_SV22R, rsnd_kctrl_valm(ctu->sv2, 2)); - rsnd_mod_write(mod, CTU_SV23R, rsnd_kctrl_valm(ctu->sv2, 3)); - rsnd_mod_write(mod, CTU_SV24R, rsnd_kctrl_valm(ctu->sv2, 4)); - rsnd_mod_write(mod, CTU_SV25R, rsnd_kctrl_valm(ctu->sv2, 5)); - rsnd_mod_write(mod, CTU_SV26R, rsnd_kctrl_valm(ctu->sv2, 6)); - rsnd_mod_write(mod, CTU_SV27R, rsnd_kctrl_valm(ctu->sv2, 7)); - } - if (scmdr > 3) { - rsnd_mod_write(mod, CTU_SV30R, rsnd_kctrl_valm(ctu->sv3, 0)); - rsnd_mod_write(mod, CTU_SV31R, rsnd_kctrl_valm(ctu->sv3, 1)); - rsnd_mod_write(mod, CTU_SV32R, rsnd_kctrl_valm(ctu->sv3, 2)); - rsnd_mod_write(mod, CTU_SV33R, rsnd_kctrl_valm(ctu->sv3, 3)); - rsnd_mod_write(mod, CTU_SV34R, rsnd_kctrl_valm(ctu->sv3, 4)); - rsnd_mod_write(mod, CTU_SV35R, rsnd_kctrl_valm(ctu->sv3, 5)); - rsnd_mod_write(mod, CTU_SV36R, rsnd_kctrl_valm(ctu->sv3, 6)); - rsnd_mod_write(mod, CTU_SV37R, rsnd_kctrl_valm(ctu->sv3, 7)); + for (i = 0; i < 4; i++) { + + if (i >= scmdr) + break; + + for (j = 0; j < RSND_MAX_CHANNELS; j++) + rsnd_mod_write(mod, CTU_SVxxR(i, j), rsnd_kctrl_valm(ctu->sv[i], j)); } rsnd_mod_write(mod, CTU_CTUIR, 0); @@ -194,10 +159,10 @@ static void rsnd_ctu_value_reset(struct rsnd_dai_stream *io, for (i = 0; i < RSND_MAX_CHANNELS; i++) { rsnd_kctrl_valm(ctu->pass, i) = 0; - rsnd_kctrl_valm(ctu->sv0, i) = 0; - rsnd_kctrl_valm(ctu->sv1, i) = 0; - rsnd_kctrl_valm(ctu->sv2, i) = 0; - rsnd_kctrl_valm(ctu->sv3, i) = 0; + rsnd_kctrl_valm(ctu->sv[0], i) = 0; + rsnd_kctrl_valm(ctu->sv[1], i) = 0; + rsnd_kctrl_valm(ctu->sv[2], i) = 0; + rsnd_kctrl_valm(ctu->sv[3], i) = 0; } rsnd_kctrl_vals(ctu->reset) = 0; } @@ -247,7 +212,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv0, RSND_MAX_CHANNELS, + &ctu->sv[0], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -256,7 +221,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv1, RSND_MAX_CHANNELS, + &ctu->sv[1], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -265,7 +230,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv2, RSND_MAX_CHANNELS, + &ctu->sv[2], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -274,7 +239,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv3, RSND_MAX_CHANNELS, + &ctu->sv[3], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 27c3761..8d91c0e 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -86,14 +86,8 @@ static void rsnd_dvc_volume_parameter(struct rsnd_dai_stream *io, val[i] = rsnd_kctrl_valm(dvc->volume, i); /* Enable Digital Volume */ - rsnd_mod_write(mod, DVC_VOL0R, val[0]); - rsnd_mod_write(mod, DVC_VOL1R, val[1]); - rsnd_mod_write(mod, DVC_VOL2R, val[2]); - rsnd_mod_write(mod, DVC_VOL3R, val[3]); - rsnd_mod_write(mod, DVC_VOL4R, val[4]); - rsnd_mod_write(mod, DVC_VOL5R, val[5]); - rsnd_mod_write(mod, DVC_VOL6R, val[6]); - rsnd_mod_write(mod, DVC_VOL7R, val[7]); + for (i = 0; i < RSND_MAX_CHANNELS; i++) + rsnd_mod_write(mod, DVC_VOLxR(i), val[i]); } static void rsnd_dvc_volume_init(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3283d67..7cda601 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -26,8 +26,8 @@ struct rsnd_gen { struct regmap *regmap[RSND_BASE_MAX]; /* RSND_REG_MAX base */ - struct regmap_field *regs[RSND_REG_MAX]; - const char *reg_name[RSND_REG_MAX]; + struct regmap_field *regs[REG_MAX]; + const char *reg_name[REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) @@ -49,11 +49,11 @@ struct rsnd_regmap_field_conf { } /* single address mapping */ #define RSND_GEN_S_REG(id, offset) \ - RSND_REG_SET(RSND_REG_##id, offset, 0, #id) + RSND_REG_SET(id, offset, 0, #id) /* multi address mapping */ #define RSND_GEN_M_REG(id, offset, _id_offset) \ - RSND_REG_SET(RSND_REG_##id, offset, _id_offset, #id) + RSND_REG_SET(id, offset, _id_offset, #id) /* * basic function @@ -79,9 +79,9 @@ static int rsnd_mod_id_cmd(struct rsnd_mod *mod) return rsnd_mod_id(mod); } -u32 rsnd_read(struct rsnd_priv *priv, - struct rsnd_mod *mod, enum rsnd_reg reg) +u32 rsnd_mod_read(struct rsnd_mod *mod, enum rsnd_reg reg) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); u32 val; @@ -98,10 +98,10 @@ u32 rsnd_read(struct rsnd_priv *priv, return val; } -void rsnd_write(struct rsnd_priv *priv, - struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data) +void rsnd_mod_write(struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); @@ -115,9 +115,10 @@ void rsnd_write(struct rsnd_priv *priv, rsnd_reg_name(gen, reg), reg, data); } -void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 mask, u32 data) +void rsnd_mod_bset(struct rsnd_mod *mod, + enum rsnd_reg reg, u32 mask, u32 data) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 94d51fe..605e4b9 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -42,165 +42,175 @@ */ enum rsnd_reg { /* SCU (MIX/CTU/DVC) */ - RSND_REG_SRC_I_BUSIF_MODE, - RSND_REG_SRC_O_BUSIF_MODE, - RSND_REG_SRC_ROUTE_MODE0, - RSND_REG_SRC_SWRSR, - RSND_REG_SRC_SRCIR, - RSND_REG_SRC_ADINR, - RSND_REG_SRC_IFSCR, - RSND_REG_SRC_IFSVR, - RSND_REG_SRC_SRCCR, - RSND_REG_SRC_CTRL, - RSND_REG_SRC_BSDSR, - RSND_REG_SRC_BSISR, - RSND_REG_SRC_INT_ENABLE0, - RSND_REG_SRC_BUSIF_DALIGN, - RSND_REG_SRCIN_TIMSEL0, - RSND_REG_SRCIN_TIMSEL1, - RSND_REG_SRCIN_TIMSEL2, - RSND_REG_SRCIN_TIMSEL3, - RSND_REG_SRCIN_TIMSEL4, - RSND_REG_SRCOUT_TIMSEL0, - RSND_REG_SRCOUT_TIMSEL1, - RSND_REG_SRCOUT_TIMSEL2, - RSND_REG_SRCOUT_TIMSEL3, - RSND_REG_SRCOUT_TIMSEL4, - RSND_REG_SCU_SYS_STATUS0, - RSND_REG_SCU_SYS_STATUS1, - RSND_REG_SCU_SYS_INT_EN0, - RSND_REG_SCU_SYS_INT_EN1, - RSND_REG_CMD_CTRL, - RSND_REG_CMD_BUSIF_MODE, - RSND_REG_CMD_BUSIF_DALIGN, - RSND_REG_CMD_ROUTE_SLCT, - RSND_REG_CMDOUT_TIMSEL, - RSND_REG_CTU_SWRSR, - RSND_REG_CTU_CTUIR, - RSND_REG_CTU_ADINR, - RSND_REG_CTU_CPMDR, - RSND_REG_CTU_SCMDR, - RSND_REG_CTU_SV00R, - RSND_REG_CTU_SV01R, - RSND_REG_CTU_SV02R, - RSND_REG_CTU_SV03R, - RSND_REG_CTU_SV04R, - RSND_REG_CTU_SV05R, - RSND_REG_CTU_SV06R, - RSND_REG_CTU_SV07R, - RSND_REG_CTU_SV10R, - RSND_REG_CTU_SV11R, - RSND_REG_CTU_SV12R, - RSND_REG_CTU_SV13R, - RSND_REG_CTU_SV14R, - RSND_REG_CTU_SV15R, - RSND_REG_CTU_SV16R, - RSND_REG_CTU_SV17R, - RSND_REG_CTU_SV20R, - RSND_REG_CTU_SV21R, - RSND_REG_CTU_SV22R, - RSND_REG_CTU_SV23R, - RSND_REG_CTU_SV24R, - RSND_REG_CTU_SV25R, - RSND_REG_CTU_SV26R, - RSND_REG_CTU_SV27R, - RSND_REG_CTU_SV30R, - RSND_REG_CTU_SV31R, - RSND_REG_CTU_SV32R, - RSND_REG_CTU_SV33R, - RSND_REG_CTU_SV34R, - RSND_REG_CTU_SV35R, - RSND_REG_CTU_SV36R, - RSND_REG_CTU_SV37R, - RSND_REG_MIX_SWRSR, - RSND_REG_MIX_MIXIR, - RSND_REG_MIX_ADINR, - RSND_REG_MIX_MIXMR, - RSND_REG_MIX_MVPDR, - RSND_REG_MIX_MDBAR, - RSND_REG_MIX_MDBBR, - RSND_REG_MIX_MDBCR, - RSND_REG_MIX_MDBDR, - RSND_REG_MIX_MDBER, - RSND_REG_DVC_SWRSR, - RSND_REG_DVC_DVUIR, - RSND_REG_DVC_ADINR, - RSND_REG_DVC_DVUCR, - RSND_REG_DVC_ZCMCR, - RSND_REG_DVC_VOL0R, - RSND_REG_DVC_VOL1R, - RSND_REG_DVC_VOL2R, - RSND_REG_DVC_VOL3R, - RSND_REG_DVC_VOL4R, - RSND_REG_DVC_VOL5R, - RSND_REG_DVC_VOL6R, - RSND_REG_DVC_VOL7R, - RSND_REG_DVC_DVUER, - RSND_REG_DVC_VRCTR, - RSND_REG_DVC_VRPDR, - RSND_REG_DVC_VRDBR, + SRC_I_BUSIF_MODE, + SRC_O_BUSIF_MODE, + SRC_ROUTE_MODE0, + SRC_SWRSR, + SRC_SRCIR, + SRC_ADINR, + SRC_IFSCR, + SRC_IFSVR, + SRC_SRCCR, + SRC_CTRL, + SRC_BSDSR, + SRC_BSISR, + SRC_INT_ENABLE0, + SRC_BUSIF_DALIGN, + SRCIN_TIMSEL0, + SRCIN_TIMSEL1, + SRCIN_TIMSEL2, + SRCIN_TIMSEL3, + SRCIN_TIMSEL4, + SRCOUT_TIMSEL0, + SRCOUT_TIMSEL1, + SRCOUT_TIMSEL2, + SRCOUT_TIMSEL3, + SRCOUT_TIMSEL4, + SCU_SYS_STATUS0, + SCU_SYS_STATUS1, + SCU_SYS_INT_EN0, + SCU_SYS_INT_EN1, + CMD_CTRL, + CMD_BUSIF_MODE, + CMD_BUSIF_DALIGN, + CMD_ROUTE_SLCT, + CMDOUT_TIMSEL, + CTU_SWRSR, + CTU_CTUIR, + CTU_ADINR, + CTU_CPMDR, + CTU_SCMDR, + CTU_SV00R, + CTU_SV01R, + CTU_SV02R, + CTU_SV03R, + CTU_SV04R, + CTU_SV05R, + CTU_SV06R, + CTU_SV07R, + CTU_SV10R, + CTU_SV11R, + CTU_SV12R, + CTU_SV13R, + CTU_SV14R, + CTU_SV15R, + CTU_SV16R, + CTU_SV17R, + CTU_SV20R, + CTU_SV21R, + CTU_SV22R, + CTU_SV23R, + CTU_SV24R, + CTU_SV25R, + CTU_SV26R, + CTU_SV27R, + CTU_SV30R, + CTU_SV31R, + CTU_SV32R, + CTU_SV33R, + CTU_SV34R, + CTU_SV35R, + CTU_SV36R, + CTU_SV37R, + MIX_SWRSR, + MIX_MIXIR, + MIX_ADINR, + MIX_MIXMR, + MIX_MVPDR, + MIX_MDBAR, + MIX_MDBBR, + MIX_MDBCR, + MIX_MDBDR, + MIX_MDBER, + DVC_SWRSR, + DVC_DVUIR, + DVC_ADINR, + DVC_DVUCR, + DVC_ZCMCR, + DVC_VOL0R, + DVC_VOL1R, + DVC_VOL2R, + DVC_VOL3R, + DVC_VOL4R, + DVC_VOL5R, + DVC_VOL6R, + DVC_VOL7R, + DVC_DVUER, + DVC_VRCTR, + DVC_VRPDR, + DVC_VRDBR, /* ADG */ - RSND_REG_BRRA, - RSND_REG_BRRB, - RSND_REG_BRGCKR, - RSND_REG_DIV_EN, - RSND_REG_AUDIO_CLK_SEL0, - RSND_REG_AUDIO_CLK_SEL1, - RSND_REG_AUDIO_CLK_SEL2, + BRRA, + BRRB, + BRGCKR, + DIV_EN, + AUDIO_CLK_SEL0, + AUDIO_CLK_SEL1, + AUDIO_CLK_SEL2, /* SSIU */ - RSND_REG_SSI_MODE, - RSND_REG_SSI_MODE0, - RSND_REG_SSI_MODE1, - RSND_REG_SSI_MODE2, - RSND_REG_SSI_CONTROL, - RSND_REG_SSI_CTRL, - RSND_REG_SSI_BUSIF0_MODE, - RSND_REG_SSI_BUSIF0_ADINR, - RSND_REG_SSI_BUSIF0_DALIGN, - RSND_REG_SSI_BUSIF1_MODE, - RSND_REG_SSI_BUSIF1_ADINR, - RSND_REG_SSI_BUSIF1_DALIGN, - RSND_REG_SSI_BUSIF2_MODE, - RSND_REG_SSI_BUSIF2_ADINR, - RSND_REG_SSI_BUSIF2_DALIGN, - RSND_REG_SSI_BUSIF3_MODE, - RSND_REG_SSI_BUSIF3_ADINR, - RSND_REG_SSI_BUSIF3_DALIGN, - RSND_REG_SSI_BUSIF4_MODE, - RSND_REG_SSI_BUSIF4_ADINR, - RSND_REG_SSI_BUSIF4_DALIGN, - RSND_REG_SSI_BUSIF5_MODE, - RSND_REG_SSI_BUSIF5_ADINR, - RSND_REG_SSI_BUSIF5_DALIGN, - RSND_REG_SSI_BUSIF6_MODE, - RSND_REG_SSI_BUSIF6_ADINR, - RSND_REG_SSI_BUSIF6_DALIGN, - RSND_REG_SSI_BUSIF7_MODE, - RSND_REG_SSI_BUSIF7_ADINR, - RSND_REG_SSI_BUSIF7_DALIGN, - RSND_REG_SSI_INT_ENABLE, - RSND_REG_SSI_SYS_STATUS0, - RSND_REG_SSI_SYS_STATUS1, - RSND_REG_SSI_SYS_STATUS2, - RSND_REG_SSI_SYS_STATUS3, - RSND_REG_SSI_SYS_STATUS4, - RSND_REG_SSI_SYS_STATUS5, - RSND_REG_SSI_SYS_STATUS6, - RSND_REG_SSI_SYS_STATUS7, - RSND_REG_HDMI0_SEL, - RSND_REG_HDMI1_SEL, + SSI_MODE, + SSI_MODE0, + SSI_MODE1, + SSI_MODE2, + SSI_CONTROL, + SSI_CTRL, + SSI_BUSIF0_MODE, + SSI_BUSIF1_MODE, + SSI_BUSIF2_MODE, + SSI_BUSIF3_MODE, + SSI_BUSIF4_MODE, + SSI_BUSIF5_MODE, + SSI_BUSIF6_MODE, + SSI_BUSIF7_MODE, + SSI_BUSIF0_ADINR, + SSI_BUSIF1_ADINR, + SSI_BUSIF2_ADINR, + SSI_BUSIF3_ADINR, + SSI_BUSIF4_ADINR, + SSI_BUSIF5_ADINR, + SSI_BUSIF6_ADINR, + SSI_BUSIF7_ADINR, + SSI_BUSIF0_DALIGN, + SSI_BUSIF1_DALIGN, + SSI_BUSIF2_DALIGN, + SSI_BUSIF3_DALIGN, + SSI_BUSIF4_DALIGN, + SSI_BUSIF5_DALIGN, + SSI_BUSIF6_DALIGN, + SSI_BUSIF7_DALIGN, + SSI_INT_ENABLE, + SSI_SYS_STATUS0, + SSI_SYS_STATUS1, + SSI_SYS_STATUS2, + SSI_SYS_STATUS3, + SSI_SYS_STATUS4, + SSI_SYS_STATUS5, + SSI_SYS_STATUS6, + SSI_SYS_STATUS7, + HDMI0_SEL, + HDMI1_SEL, /* SSI */ - RSND_REG_SSICR, - RSND_REG_SSISR, - RSND_REG_SSITDR, - RSND_REG_SSIRDR, - RSND_REG_SSIWSR, + SSICR, + SSISR, + SSITDR, + SSIRDR, + SSIWSR, - RSND_REG_MAX, + REG_MAX, }; +#define SRCIN_TIMSEL(i) (SRCIN_TIMSEL0 + (i)) +#define SRCOUT_TIMSEL(i) (SRCOUT_TIMSEL0 + (i)) +#define CTU_SVxxR(i, j) (CTU_SV00R + (i * 8) + (j)) +#define DVC_VOLxR(i) (DVC_VOL0R + (i)) +#define AUDIO_CLK_SEL(i) (AUDIO_CLK_SEL0 + (i)) +#define SSI_BUSIF_MODE(i) (SSI_BUSIF0_MODE + (i)) +#define SSI_BUSIF_ADINR(i) (SSI_BUSIF0_ADINR + (i)) +#define SSI_BUSIF_DALIGN(i) (SSI_BUSIF0_DALIGN + (i)) +#define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) + struct rsnd_priv; struct rsnd_mod; @@ -210,20 +220,9 @@ struct rsnd_dai_stream; /* * R-Car basic functions */ -#define rsnd_mod_read(m, r) \ - rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r) -#define rsnd_mod_write(m, r, d) \ - rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) -#define rsnd_mod_bset(m, r, s, d) \ - rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) - -u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data); -void rsnd_force_write(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data); -void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, - u32 mask, u32 data); +u32 rsnd_mod_read(struct rsnd_mod *mod, enum rsnd_reg reg); +void rsnd_mod_write(struct rsnd_mod *mod, enum rsnd_reg reg, u32 data); +void rsnd_mod_bset(struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data); u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 04b1a48..c5934ad 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -65,6 +65,7 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, int id = rsnd_mod_id(mod); u32 mask1, val1; u32 mask2, val2; + int i; /* clear status */ switch (id) { @@ -73,16 +74,12 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, case 2: case 3: case 4: - rsnd_mod_write(mod, SSI_SYS_STATUS0, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS2, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS4, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS6, 0xf << (id * 4)); + for (i = 0; i < 4; i++) + rsnd_mod_write(mod, SSI_SYS_STATUS(i * 2), 0xf << (id * 4)); break; case 9: - rsnd_mod_write(mod, SSI_SYS_STATUS1, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS3, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS5, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS7, 0xf << 4); + for (i = 0; i < 4; i++) + rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << (id * 4)); break; } @@ -198,43 +195,15 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, id, busif); } -#define RSND_WRITE_BUSIF(i) \ - rsnd_mod_write(mod, SSI_BUSIF##i##_ADINR, \ - rsnd_get_adinr_bit(mod, io) | \ - (rsnd_io_is_play(io) ? \ - rsnd_runtime_channel_after_ctu(io) : \ - rsnd_runtime_channel_original(io))); \ - rsnd_mod_write(mod, SSI_BUSIF##i##_MODE, \ - rsnd_get_busif_shift(io, mod) | 1); \ - rsnd_mod_write(mod, SSI_BUSIF##i##_DALIGN, \ - rsnd_get_dalign(mod, io)) - - switch (busif) { - case 0: - RSND_WRITE_BUSIF(0); - break; - case 1: - RSND_WRITE_BUSIF(1); - break; - case 2: - RSND_WRITE_BUSIF(2); - break; - case 3: - RSND_WRITE_BUSIF(3); - break; - case 4: - RSND_WRITE_BUSIF(4); - break; - case 5: - RSND_WRITE_BUSIF(5); - break; - case 6: - RSND_WRITE_BUSIF(6); - break; - case 7: - RSND_WRITE_BUSIF(7); - break; - } + rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), + rsnd_get_adinr_bit(mod, io) | + (rsnd_io_is_play(io) ? + rsnd_runtime_channel_after_ctu(io) : + rsnd_runtime_channel_original(io))); + rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), + rsnd_get_busif_shift(io, mod) | 1); + rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), + rsnd_get_dalign(mod, io)); } if (has_hdmi0 || has_hdmi1) { -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M" to the asoc tree
by Mark Brown 13 Dec '18

13 Dec '18

The patch ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From add6883619a9e3bf9658eaff1a547354131bbcd9 Mon Sep 17 00:00:00 2001 From: Fabio Estevam <festevam(a)gmail.com> Date: Thu, 13 Dec 2018 00:08:38 -0200 Subject: [PATCH] ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M eukrea-tlv320.c machine driver runs on non-DT platforms and include <asm/mach-types.h> header file in order to be able to use some machine_is_eukrea_xxx() macros. Building it for ARM64 causes the following build error: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320 driver when ARM64 is selected. This is needed in preparation for the i.MX8M support. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> Acked-by: Shawn Guo <shawnguo(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb4a934..2e75b5bc5f1d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: fsl-sai: Fix typo in "transmitter"
by Fabio Estevam 13 Dec '18

13 Dec '18

Fix the spelling of "transmitter". Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index dd9e597..2e726b9 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -35,13 +35,13 @@ Required properties: - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transimitter and the + with Rx) which means both the transmitter and the receiver will send and receive data by following receiver's bit clocks and frame sync clocks. - fsl,sai-asynchronous: This is a boolean property. If present, indicating that SAI will work in the asynchronous mode, which - means both transimitter and receiver will send and + means both transmitter and receiver will send and receive data by following their own bit clocks and frame sync clocks separately. @@ -58,8 +58,8 @@ Optional properties (for mx6ul): Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both - transimitter and receiver will send and receive data by following clocks - of transimitter. + transmitter and receiver will send and receive data by following clocks + of transmitter. - fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. Example: -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: simple-card: Dereference pointer for memcpy sizeof in asoc_simple_card_probe
by Kuninori Morimoto 13 Dec '18

13 Dec '18

From: Nathan Chancellor <natechancellor(a)gmail.com> Commit 4fb7f4df49d3 ("ASoC: simple-card: use cpu/codec pointer on simple_dai_props") updated {cpu,codec}_dai to be pointers in struct simple_dai_props but didn't update these locations to dereference the pointers. This patch fixup it for non DT simple-card use case. Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 45d3e8c..fb18e51 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -665,6 +665,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct asoc_simple_card_info *cinfo; struct snd_soc_dai_link_component *codecs; struct snd_soc_dai_link_component *platform; + int dai_idx = 0; cinfo = dev->platform_data; if (!cinfo) { @@ -681,6 +682,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } + dai_props->cpu_dai = &priv->dais[dai_idx++]; + dai_props->codec_dai = &priv->dais[dai_idx++]; + codecs = dai_link->codecs; codecs->name = cinfo->codec; codecs->dai_name = cinfo->codec_dai.name; @@ -694,10 +698,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; - memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, - sizeof(priv->dai_props->cpu_dai)); - memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, - sizeof(priv->dai_props->codec_dai)); + memcpy(priv->dai_props->cpu_dai, &cinfo->cpu_dai, + sizeof(*priv->dai_props->cpu_dai)); + memcpy(priv->dai_props->codec_dai, &cinfo->codec_dai, + sizeof(*priv->dai_props->codec_dai)); } snd_soc_card_set_drvdata(card, priv); -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: meson: add axg spdif input support
by Jerome Brunet 13 Dec '18

13 Dec '18

This patchset adds the initial support the spdif input found on the axg SoC family. The capture itself works well but the rate detection could be enhanced in the future. Like several other drivers of this type, no check is done to verify if the rate of the input stream is consistent with the rate requested in hw_params() The automatic rate detection mechanism of this device and the related IRQ proved difficult to work with, as documented in the driver. Several approaches to work around this minor issue have been tried but none provided a complete and reliable solution so far. Hopefully more eyes on this will help figure this out. Since I authored all Amlogic ASoC drivers (and the related bugs) merged so far, I have added myself as maintainer of them in the last patch of this series. Jerome Brunet (4): ASoC: meson: axg-toddr: add support for spdifin backend ASoC: meson: add axg spdif input DT binding documentation ASoC: meson: add axg spdif input MAINTAINERS: Add Amlogic sound drivers entry .../bindings/sound/amlogic,axg-spdifin.txt | 22 + MAINTAINERS | 7 + sound/soc/meson/Kconfig | 9 +- sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-fifo.h | 3 +- sound/soc/meson/axg-spdifin.c | 521 ++++++++++++++++++ sound/soc/meson/axg-toddr.c | 15 +- 7 files changed, 567 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 sound/soc/meson/axg-spdifin.c -- 2.19.2

2 8

[alsa-devel] [PATCH] ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M
by Fabio Estevam 13 Dec '18

13 Dec '18

eukrea-tlv320.c machine driver runs on non-DT platforms and include <asm/mach-types.h> header file in order to be able to use some machine_is_eukrea_xxx() macros. Since commit 0e1532bfecf4 ("arm64: add basic Kconfig symbols for i.MX8") the ARCH_MXC symbol is visible on ARM64 too. Building it for ARM64 causes the following build error: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320 driver when ARM64 is selected. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Hi Mark, Commit commit 0e1532bfecf4 ("arm64: add basic Kconfig symbols for i.MX8") is present in Shawn's tree, so maybe this should be taken by Shawn? sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb..2e75b5b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.7.4

3 3

[alsa-devel] [PATCH] ASoC: fsl: Add a dependency on ARM for Eukrea
by Mark Brown 13 Dec '18

13 Dec '18

The MXC architecture became buildable on arm64 which doesn't have a mach-types.h so add an explicit 32 bit dependency. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb4a934..ff2d4a8783f9 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_ARM && ARCH_MXC && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.19.0.rc2

1 0

[alsa-devel] [bunk@debian.org] Bug#916052: alsa-oss FTBFS with glibc 2.28
by Elimar Riesebieter 13 Dec '18

13 Dec '18

Control: forwarded -1 alsa-devel(a)alsa-project.org Hi, could you please investigate? It seems as alsa-oss doesn't built with glibc 2.28. Thanks in advance Elimar -- Learned men are the cisterns of knowledge, not the fountainheads ;-)

2 2

[alsa-devel] [PATCH v11 0/3] Huawei laptops
by Ayman Bagabas 13 Dec '18

13 Dec '18

This patch set is based on the new audio LED triggers in topic/leds-trigger branch from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes in v11: * Minor code changes Changes in v10: * Use ec_get_handle instead of acpi_get_handle since we are using the ec device * Switch to WMI0_EXPENSIVE_GUID and wmi_query_block to fetch keycode when WMI0 is used Changes in v9: * Explicitly use NULL in acpi_get_handle * Drop __initconst from huawei_wmi_keymap Changes in v8: * Switch to wmi_driver * Use devm to allocate and manage devices * Skip registering LED subsystem if a ACPI controller method was not found Changes in v7: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes in v6: * Review tags Changes in v5: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes in v4: * Code formatting Changes in v3: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 17 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 208 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 ++++ 4 files changed, 248 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.2

2 4

[alsa-devel] [PATCH v2] ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M
by Fabio Estevam 13 Dec '18

13 Dec '18

eukrea-tlv320.c machine driver runs on non-DT platforms and include <asm/mach-types.h> header file in order to be able to use some machine_is_eukrea_xxx() macros. Building it for ARM64 causes the following build error: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320 driver when ARM64 is selected. This is needed in preparation for the i.MX8M support. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Changes since v1: - Drop the mention to the i.MX8M commit that is currently only in Shawn's tree. sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb..2e75b5b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: simple-card: Dereference pointer for memcpy sizeof in asoc_simple_card_probe
by Nathan Chancellor 13 Dec '18

13 Dec '18

Clang warns: sound/soc/generic/simple-card.c:462:6: warning: argument to 'sizeof' in 'memcpy' call is the same pointer type 'struct asoc_simple_dai *' as the source; expected 'struct asoc_simple_dai' or an explicit length [-Wsizeof-pointer-memaccess] sizeof(priv->dai_props->cpu_dai)); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/generic/simple-card.c:464:6: warning: argument to 'sizeof' in 'memcpy' call is the same pointer type 'struct asoc_simple_dai *' as the source; expected 'struct asoc_simple_dai' or an explicit length [-Wsizeof-pointer-memaccess] sizeof(priv->dai_props->codec_dai)); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 2 warnings generated. Commit 4fb7f4df49d3 ("ASoC: simple-card: use cpu/codec pointer on simple_dai_props") updated {cpu,codec}_dai to be pointers in struct simple_dai_props but didn't update these locations to dereference the pointers to get the proper size of their contents. Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/generic/simple-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 766123485d7c..d4738d3eb2f1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -459,9 +459,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, - sizeof(priv->dai_props->cpu_dai)); + sizeof(*priv->dai_props->cpu_dai)); memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, - sizeof(priv->dai_props->codec_dai)); + sizeof(*priv->dai_props->codec_dai)); } snd_soc_card_set_drvdata(card, priv); -- 2.20.0

2 2

[alsa-devel] [PATCH] ALSA: pcm: Fix potential Spectre v1 vulnerability
by Gustavo A. R. Silva 13 Dec '18

13 Dec '18

stream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/pcm.c:140 snd_pcm_control_ioctl() warn: potential spectre issue 'pcm->streams' [r] (local cap) Fix this by sanitizing stream before using it to index pcm->streams Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index fdb9b92fc8d6..23141003a90d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -32,6 +32,8 @@ #include <sound/control.h> #include <sound/info.h> +#include <linux/nospec.h> + #include "pcm_local.h" MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); @@ -129,6 +131,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -EFAULT; if (stream < 0 || stream > 1) return -EINVAL; + stream = array_index_nospec(stream, 2); if (get_user(subdevice, &info->subdevice)) return -EFAULT; mutex_lock(&register_mutex); -- 2.17.1

3 2

[alsa-devel] [PATCH] alsa-oss: fix FTBFS with glibc v2.28
by GitHub pull_request - opened 13 Dec '18

13 Dec '18

alsa-project/alsa-oss pull request #1 was opened from takaswie: In glibc upstream, libio/libio.h had been deprecated for applications and became internal header in its v2.28 release. We can see below message in its release note[1]. * The nonstandard header files <libio.h> and <_G_config.h> are no longer installed. Software that was using either header should be updated to use standard <stdio.h> interfaces instead. This brings FTBFS with glibc v2.28 due to missing <libio.h>. This commit fixes it. [1] https://www.sourceware.org/ml/libc-alpha/2018-08/msg00003.html Reported-by: Adrian Bunk <bunk(a)debian.org> Reference: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=916052 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> ``` --- alsa/stdioemu.c | 1 - 1 file changed, 1 deletion(-) ``` Request URL : https://github.com/alsa-project/alsa-oss/pull/1 Patch URL : https://github.com/alsa-project/alsa-oss/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-oss

1 0

[alsa-devel] [PATCH] ALSA: emux_hwdep: Fix potential Spectre v1 vulnerabilities
by Gustavo A. R. Silva 12 Dec '18

12 Dec '18

info.mode and info.port are indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. These issues were detected with the help of Smatch: sound/synth/emux/emux_hwdep.c:72 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[i]->ctrls' [w] (local cap) sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs' [w] (local cap) sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[info.port]->ctrls' [w] (local cap) Fix this by sanitizing both info.mode and info.port before using them to index emu->portptrs[i]->ctrls, emu->portptrs[info.port]->ctrls and emu->portptrs. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/synth/emux/emux_hwdep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index e557946718a9..a0d5cd99572f 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -24,6 +24,7 @@ #include <linux/uaccess.h> #include "emux_voice.h" +#include <linux/nospec.h> #define TMP_CLIENT_ID 0x1001 @@ -66,13 +67,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg) return -EFAULT; if (info.mode < 0 || info.mode >= EMUX_MD_END) return -EINVAL; + info.mode = array_index_nospec(info.mode, EMUX_MD_END); if (info.port < 0) { for (i = 0; i < emu->num_ports; i++) emu->portptrs[i]->ctrls[info.mode] = info.value; } else { - if (info.port < emu->num_ports) + if (info.port < emu->num_ports) { + info.port = array_index_nospec(info.port, emu->num_ports); emu->portptrs[info.port]->ctrls[info.mode] = info.value; + } } return 0; } -- 2.17.1

3 2

[alsa-devel] [PATCH v10 0/3] Huawei laptops
by Ayman Bagabas 12 Dec '18

12 Dec '18

This patch set is based on the new audio LED triggers in topic/leds-trigger branch from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes in v10: * Use ec_get_handle instead of acpi_get_handle since we are using the ec device * Switch to WMI0_EXPENSIVE_GUID and wmi_query_block to fetch keycode when WMI0 is used Changes in v9: * Explicitly use NULL in acpi_get_handle * Drop __initconst from huawei_wmi_keymap Changes in v8: * Switch to wmi_driver * Use devm to allocate and manage devices * Skip registering LED subsystem if a ACPI controller method was not found Changes in v7: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes in v6: * Review tags Changes in v5: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes in v4: * Code formatting Changes in v3: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 17 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 220 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 260 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.2

4 10

[alsa-devel] [PATCH 0/6] ASoC: merge audio-graph-scu into audio-graph - final step
by Kuninori Morimoto 12 Dec '18

12 Dec '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. This is final step patches. Kuninori Morimoto (6): ASoC: audio-graph-card: merge audio-graph-scu-card on Doc ASoC: audio-graph-card: merge audio-graph-scu-card ASoC: audio-graph-card: tidyup mclk-fs method ASoC: audio-graph-card: tidyup convert_rate/channel method ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc ASoC: audio-graph-scu-card: remove audio-graph-scu-card .../devicetree/bindings/sound/audio-graph-card.txt | 222 +++++++++ .../bindings/sound/audio-graph-scu-card.txt | 123 ----- sound/soc/generic/Kconfig | 10 +- sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-card.c | 397 +++++++++++++--- sound/soc/generic/audio-graph-scu-card.c | 501 --------------------- 6 files changed, 563 insertions(+), 692 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt delete mode 100644 sound/soc/generic/audio-graph-scu-card.c -- 2.7.4

1 7

[alsa-devel] [PATCH v2] ASoC: Intel: kbl_rt5660: Add a new machine driver for kbl with rt5660
by Hui Wang 12 Dec '18

12 Dec '18

The new Dell IoT platform uses kabylake + alc3277 codec, and alc3277 shares the driver with the codec rt5660, here we generate a new machine driver based on kbl_da7219_max98357a. The audio design on this IoT platform is as below: - Intel kabylake platform - connect the codec ALC3277 via SSP0 - line-out and line-in with Micbias jacks - line-out mute control and jack detection of line-out and line-in - two HDMI ports with audio capability Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/intel/boards/Kconfig | 10 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_rt5660.c | 536 ++++++++++++++++++ .../intel/common/soc-acpi-intel-kbl-match.c | 5 + 4 files changed, 553 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_rt5660.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index b177db2a0dbb..3839d6205fcf 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -293,6 +293,16 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_KBL_RT5660_MACH + tristate "KBL with RT5660 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5660 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5660 I2S audio codec. + Say Y if you have such a device. + config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 5381e27df9cc..bf072ea299b7 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -20,6 +20,7 @@ snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o +snd-soc-kbl_rt5660-objs := kbl_rt5660.o snd-soc-skl_rt286-objs := skl_rt286.o snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o @@ -46,6 +47,7 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max9 obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o +obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5660_MACH) += snd-soc-kbl_rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c new file mode 100644 index 000000000000..0482e6a7fc2a --- /dev/null +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -0,0 +1,536 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018-19 Intel Corporation. + +/* + * Intel Kabylake I2S Machine Driver with RT5660 Codec + * + * Modified from: + * Intel Kabylake I2S Machine driver supporting MAXIM98357a and + * DA7219 codecs + * Also referred to: + * Intel Broadwell I2S Machine driver supporting RT5677 codec + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../../codecs/hdac_hdmi.h" +#include "../../codecs/rt5660.h" + +#define KBL_RT5660_CODEC_DAI "rt5660-aif1" +#define DUAL_CHANNEL 2 + +static struct snd_soc_card *kabylake_audio_card; +static struct snd_soc_jack skylake_hdmi[3]; +static struct snd_soc_jack lineout_jack; +static struct snd_soc_jack mic_jack; + +struct kbl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct kbl_codec_private { + struct gpio_desc *gpio_lo_mute; + struct list_head hdmi_pcm_list; +}; + +enum { + KBL_DPCM_AUDIO_PB = 0, + KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_HDMI1_PB, + KBL_DPCM_AUDIO_HDMI2_PB, + KBL_DPCM_AUDIO_HDMI3_PB, +}; + +#define GPIO_LINEOUT_MUTE_INDEX 0 +#define GPIO_LINEOUT_DET_INDEX 3 +#define GPIO_LINEIN_DET_INDEX 4 + +static const struct acpi_gpio_params lineout_mute_gpio = { GPIO_LINEOUT_MUTE_INDEX, 0, true }; +static const struct acpi_gpio_params lineout_det_gpio = { GPIO_LINEOUT_DET_INDEX, 0, false }; +static const struct acpi_gpio_params mic_det_gpio = { GPIO_LINEIN_DET_INDEX, 0, false }; + + +static const struct acpi_gpio_mapping acpi_rt5660_gpios[] = { + { "lineout-mute-gpios", &lineout_mute_gpio , 1 }, + { "lineout-det-gpios", &lineout_det_gpio, 1 }, + { "mic-det-gpios", &mic_det_gpio, 1 }, + { NULL }, +}; + +static struct snd_soc_jack_pin lineout_jack_pin = { + .pin = "Line Out", + .mask = SND_JACK_LINEOUT, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Line In", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio lineout_jack_gpio = { + .name = "lineout-det", + .report = SND_JACK_LINEOUT, + .debounce_time = 200, +}; + +static struct snd_soc_jack_gpio mic_jack_gpio = { + .name = "mic-det", + .report = SND_JACK_MICROPHONE, + .debounce_time = 200, +}; + +static int kabylake_5660_event_lineout(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(dapm->card); + + gpiod_set_value_cansleep(priv->gpio_lo_mute, + !(SND_SOC_DAPM_EVENT_ON(event))); + + return 0; +} + +static const struct snd_kcontrol_new kabylake_rt5660_controls[] = { + SOC_DAPM_PIN_SWITCH("Line In"), + SOC_DAPM_PIN_SWITCH("Line Out"), +}; + +static const struct snd_soc_dapm_widget kabylake_rt5660_widgets[] = { + SND_SOC_DAPM_MIC("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", kabylake_5660_event_lineout), + SND_SOC_DAPM_SPK("DP", NULL), + SND_SOC_DAPM_SPK("HDMI", NULL), +}; + +static const struct snd_soc_dapm_route kabylake_rt5660_map[] = { + /* other jacks */ + {"IN1P", NULL, "Line In"}, + {"IN2P", NULL, "Line In"}, + {"Line Out", NULL, "LOUTR"}, + {"Line Out", NULL, "LOUTL"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, +}; + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + 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); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios); + if (ret) + dev_warn(component->dev, "Failed to add driver gpios\n"); + + /* Request rt5660 GPIO for lineout mute control */ + ctx->gpio_lo_mute = devm_gpiod_get(component->dev, "lineout-mute", + GPIOD_OUT_HIGH); + if (IS_ERR(ctx->gpio_lo_mute)) { + dev_err(component->dev, "Can't find GPIO_MUTE# gpio\n"); + return PTR_ERR(ctx->gpio_lo_mute); + } + + /* Create and initialize headphone jack */ + if (!snd_soc_card_jack_new(rtd->card, "Lineout Jack", + SND_JACK_LINEOUT, &lineout_jack, + &lineout_jack_pin, 1)) { + lineout_jack_gpio.gpiod_dev = component->dev; + if (snd_soc_jack_add_gpios(&lineout_jack, 1, + &lineout_jack_gpio)) + dev_err(component->dev, "Can't add Lineout jack gpio\n"); + } else + dev_err(component->dev, "Can't create Lineout jack\n"); + + /* Create and initialize mic jack */ + if (!snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, &mic_jack, + &mic_jack_pin, 1)) { + mic_jack_gpio.gpiod_dev = component->dev; + if (snd_soc_jack_add_gpios(&mic_jack, 1, &mic_jack_gpio)) + dev_err(component->dev, "Can't add mic jack gpio\n"); + } else + dev_err(component->dev, "Can't create mic jack\n"); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(dapm, "BST1"); + snd_soc_dapm_force_enable_pin(dapm, "BST2"); + + return 0; +} + +static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct kbl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = device; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); +} + +static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); +} + +static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); +} + +static int kabylake_rt5660_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5660_SCLK_S_PLL1, params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, + RT5660_PLL1_S_BCLK, + params_rate(params) * 50, + params_rate(params) * 512); + if (ret < 0) + dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops kabylake_rt5660_ops = { + .hw_params = kabylake_rt5660_hw_params, +}; + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int kbl_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 48Khz + * stereo + * 16 bit audio + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + + return 0; +} + +static const struct snd_soc_ops kabylake_rt5660_fe_ops = { + .startup = kbl_fe_startup, +}; + +/* kabylake digital audio interface glue - connects rt5660 codec <--> CPU */ +static struct snd_soc_dai_link kabylake_rt5660_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_rt5660_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_rt5660_fe_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-10EC3277:00", + .codec_dai_name = KBL_RT5660_CODEC_DAI, + .init = kabylake_rt5660_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp0_fixup, + .ops = &kabylake_rt5660_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "iDisp1", + .id = 1, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 2, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 3, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + + +#define NAME_SIZE 32 +static int kabylake_card_late_probe(struct snd_soc_card *card) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); + struct kbl_hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &skylake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &skylake_hdmi[i]); + if (err < 0) + return err; + + i++; + + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* kabylake audio machine driver for rt5660 */ +static struct snd_soc_card kabylake_audio_card_rt5660 = { + .name = "kblrt5660", + .owner = THIS_MODULE, + .dai_link = kabylake_rt5660_dais, + .num_links = ARRAY_SIZE(kabylake_rt5660_dais), + .controls = kabylake_rt5660_controls, + .num_controls = ARRAY_SIZE(kabylake_rt5660_controls), + .dapm_widgets = kabylake_rt5660_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_rt5660_widgets), + .dapm_routes = kabylake_rt5660_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static int kabylake_audio_probe(struct platform_device *pdev) +{ + struct kbl_codec_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + kabylake_audio_card = + (struct snd_soc_card *)pdev->id_entry->driver_data; + + kabylake_audio_card->dev = &pdev->dev; + snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); +} + +static const struct platform_device_id kbl_board_ids[] = { + { + .name = "kbl_rt5660", + .driver_data = + (kernel_ulong_t)&kabylake_audio_card_rt5660, + }, + { } +}; + +static struct platform_driver kabylake_audio = { + .probe = kabylake_audio_probe, + .driver = { + .name = "kbl_rt5660", + .pm = &snd_soc_pm_ops, + }, + .id_table = kbl_board_ids, +}; + +module_platform_driver(kabylake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Audio Machine driver-RT5660 in I2S mode"); +MODULE_AUTHOR("Hui Wang <hui.wang(a)canonical.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:kbl_rt5660"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index a317b7790fce..1e41c7ded9e9 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -96,6 +96,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .quirk_data = &kbl_7219_98927_codecs, .pdata = &skl_dmic_data }, + { + .id = "10EC3277", + .drv_name = "kbl_rt5660", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); -- 2.17.1

2 3

[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirks for Dell Optiplex 5060, 7060 for audio out while using headphones
by Girija Kumar Kasinadhuni 12 Dec '18

12 Dec '18

There are two audio ports on these machines - headset and mic. There' s no sound when hp is inserted into the headset port, though the hp is detected. Without pulseaudio, this issue is seen on both the Optiplex machines. The issue goes away with the "options snd-hda-intel model=dell-headset-dock" quirk in the /etc/modprobe.d/alsa-base.conf or /etc/modprobe.d/alsa.conf Getting this fix into the kernel instead of fixing in an application layer above, or in a .conf file. This is also helpful for audio clients using the ALSA api like cras from ChromiumOS. Signed-off-by: Girija Kumar Kasinadhuni <gkumar(a)neverware.com> --- alsa-info: Optiplex 5060 - - without fix: http://www.alsa-project.org/db/?f=2cf3c4354e82bde4aa480612a914535982fa3f91 - with fix: http://www.alsa-project.org/db/?f=7a928960400ec828b82632411284ad8a63727b92 Optiplex 7060 - without fix: http://www.alsa-project.org/db/?f=69fa554faa712e7cad1f188be4a5192353d64f2c - with fix: http://www.alsa-project.org/db/?f=80ce1fb178b377b27b0795ac1c55e8cbee1b5b96 sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8d75597028ee..581f6a288abf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6444,6 +6444,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x085b, "Dell Optiplex 5060", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x085a, "Dell Optiplex 7060", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.17.1

2 1

[alsa-devel] RME Fireface alsa driver
by thet 11 Dec '18

11 Dec '18

I posted this a couple of weeks ago in the alsa users list but there was no response. I hope someone here can help. I have a FF400 and I have been using ffado with it. It came to my attention that there exists a ALSA driver for the FF400 since the midi was working over alsa and midi is not supported in the ffado driver. Also my system can see the FF400 in the sound control panel without jack or ffado running. I am using Mint 19 cinnamon and the liquorix kernel. (though so far I have not noticed any difference between the liquorix kernel and the generic one except pre-empt performance). I would like to try using just the alsa driver and not ffado. Since I'm using the alsa midi support anyway it would be less total software to use the alsa driver for everything - assuming it supports everything I need for serious music recording. However I can't get any sound out of the FF400 using the ALSA driver - it works ok with FFADO though. Also the midi works - presumably using the alsa driver. I have tried using alsa directly in the sound control panel, using the alsa output from audacious, and the alsa driver from jack. I get an error about setting the frames/period/buffer size. The red light on the FF400 that shows no connection to a host stays on - even though the system can see it is present. This light goes off if I start ffado and jack, or if I start ffado mixer. Is there a guide somewhere on how to set up the ALSA drivers for the FF400? thanks! Thet

2 1

[alsa-devel] [PATCH 0/5 v2] ASoC: merge simple-scu-card into simple-card - step1
by Kuninori Morimoto 11 Dec '18

11 Dec '18

Hi Mark These are based on latest "for-4.21" branch. Kuninori Morimoto (5): ASoC: simple-scu-card: add dai-link support ASoC: simple-scu-card: care link / dai count ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props ASoC: simple-scu-card: care multi DPCM codec_conf sound/soc/generic/simple-scu-card.c | 240 +++++++++++++++++++++++++++--------- 1 file changed, 185 insertions(+), 55 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 6

[alsa-devel] Applied "ASoC: simple-scu-card: care link / dai count" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: care link / dai count has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b8d8f9a58fd971ec215fb8bfec5a4fb6e539a3ab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:08 +0000 Subject: [PATCH] ASoC: simple-scu-card: care link / dai count In DPCM case, it uses CPU-dummy / dummy-Codec dai links. If sound card is caring only DPCM, link count = dai count, but, if non DPCM case, link count != dai count. Now, we want to merge simple-card and simple-scu-card, then, we need to care both link / dai count more carefly This patch cares it, and prepare for merging simple card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 90 ++++++++++++++++++++++++++--- 1 file changed, 83 insertions(+), 7 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 666ace6175be..1c6382fa54a0 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -256,6 +256,80 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) return 0; } +static void asoc_simple_card_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) +{ + struct device_node *top = dev->of_node; + struct device_node *node; + int loop; + int num; + + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for "dummy-Codec" + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ + if (!top) { + (*link_num) = 1; + (*dais_num) = 2; + (*ccnf_num) = 0; + return; + } + + loop = 1; + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = top; + loop = 0; + } + + do { + num = of_get_child_count(node); + (*dais_num) += num; + if (num > 2) { + (*link_num) += num; + (*ccnf_num)++; + } else { + (*link_num)++; + } + node = of_get_next_child(top, node); + } while (loop && node); +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -263,18 +337,20 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_dai_props *dai_props; struct snd_soc_card *card; struct device *dev = &pdev->dev; - struct device_node *np = dev->of_node; - int num, ret, i; + int ret, i; + int lnum = 0, dnum = 0, cnum = 0; /* Allocate the private data */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - num = of_get_child_count(np); + asoc_simple_card_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) + return -EINVAL; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); if (!dai_props || !dai_link) return -ENOMEM; @@ -284,7 +360,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) * see * soc-core.c :: snd_soc_init_multicodec() */ - for (i = 0; i < num; i++) { + for (i = 0; i < lnum; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; dai_link[i].platform = &dai_props[i].platform; @@ -298,7 +374,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->dev = dev; card->dai_link = priv->dai_link; - card->num_links = num; + card->num_links = lnum; card->codec_conf = &priv->codec_conf; card->num_configs = 1; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 09ee833679761f4995daaa6fd6200c77b8fcdaef Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:18 +0000 Subject: [PATCH] ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing simple-scu-card.c is supporting "convert-rate/channels" which is used for DPCM. But, sound card might have multi codecs, and each codec might need each convert-rate/channels. This patch supports each codec's convert-rate/channles support. top node convert-rate/channels will overwrite settings if exist. It can't support each codec's convert-rate/channels if sound card had multi codecs without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 1c6382fa54a0..3c4e28995872 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -26,6 +26,7 @@ struct simple_card_data { struct asoc_simple_dai dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -86,7 +87,11 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + /* overwrite by top level adata if exist */ asoc_simple_card_convert_fixup(&priv->adata, params); return 0; @@ -177,6 +182,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, "prefix"); } + asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); + ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); if (ret) return ret; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e6a3ff2ed4c9c4df217c64e0fc0a280a24055790 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:27 +0000 Subject: [PATCH] ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and non DPCM case, it uses CPU-Codec dai links. Now, we want to merge simple-card and simple-scu-card. These sound cards are using silimar but not same logic on each functions. Then, of course we want to share same logic. To compromise, this patch uses cpu/codec pointer on simple-scu-card. It is same logic with simple-card, thus easy merging. This is prepare for merging simple card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 77 +++++++++++++++++++---------- 1 file changed, 52 insertions(+), 25 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 3c4e28995872..944a6f435f88 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -23,12 +23,14 @@ struct simple_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; struct simple_dai_props { - struct asoc_simple_dai dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; }; @@ -47,8 +49,17 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); - return asoc_simple_card_clk_enable(&dai_props->dai); + return ret; } static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) @@ -58,7 +69,9 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); } static const struct snd_soc_ops asoc_simple_card_ops = { @@ -68,19 +81,21 @@ static const struct snd_soc_ops asoc_simple_card_ops = { static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai; - struct snd_soc_dai_link *dai_link; - struct simple_dai_props *dai_props; - int num = rtd->num; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; - dai_link = simple_priv_to_link(priv, num); - dai_props = simple_priv_to_props(priv, num); - dai = dai_link->dynamic ? - rtd->cpu_dai : - rtd->codec_dai; + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; - return asoc_simple_card_init_dai(dai, &dai_props->dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; } static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -101,13 +116,14 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, struct device_node *np, struct device_node *codec, struct simple_card_data *priv, - int idx, bool is_fe, + int *dai_idx, int link_idx, int is_fe, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); struct snd_soc_card *card = simple_priv_to_card(priv); + struct asoc_simple_dai *dai; char *prefix = ""; int ret; @@ -129,12 +145,15 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); if (ret < 0) return ret; @@ -155,11 +174,14 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); if (ret < 0) return ret; @@ -184,7 +206,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); - ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); + ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) return ret; @@ -215,7 +237,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *codec; struct snd_soc_card *card = simple_priv_to_card(priv); bool is_fe; - int ret, i, loop; + int ret, loop; + int dai_idx, link_idx; if (!top) return -EINVAL; @@ -231,7 +254,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) asoc_simple_card_parse_convert(dev, top, PREFIX, &priv->adata); loop = 1; - i = 0; + link_idx = 0; + dai_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { node = dev->of_node; @@ -248,10 +272,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) is_fe = (np != codec); ret = asoc_simple_card_dai_link_of(node, np, codec, priv, - i, is_fe, !loop); + &dai_idx, link_idx++, + is_fe, !loop); if (ret < 0) return ret; - i++; } node = of_get_next_child(top, node); } while (loop && node); @@ -342,6 +366,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; + struct asoc_simple_dai *dais; struct snd_soc_card *card; struct device *dev = &pdev->dev; int ret, i; @@ -358,7 +383,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -375,6 +401,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = simple_priv_to_card(priv); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: simple-scu-card: care multi DPCM codec_conf" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: care multi DPCM codec_conf has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0a26d6df4fd8aa8f4fb444446d162e01c3e349c0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:37 +0000 Subject: [PATCH] ASoC: simple-scu-card: care multi DPCM codec_conf Current simple-scu-card didn't care about codec_conf for multi DPCM case. This patch cares it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 30 +++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 944a6f435f88..9d7299d536a8 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -21,17 +21,18 @@ struct simple_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; struct simple_dai_props { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; }; #define simple_priv_to_card(priv) (&(priv)->snd_card) @@ -116,7 +117,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, struct device_node *np, struct device_node *codec, struct simple_card_data *priv, - int *dai_idx, int link_idx, int is_fe, + int *dai_idx, int link_idx, + int *conf_idx, int is_fe, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); @@ -165,6 +167,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); } else { + struct snd_soc_codec_conf *cconf; + /* FE is dummy */ dai_link->cpu_of_node = NULL; dai_link->cpu_dai_name = "snd-soc-dummy-dai"; @@ -177,6 +181,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai = dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) return ret; @@ -192,14 +199,12 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, return ret; /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, - &priv->codec_conf, + snd_soc_of_parse_audio_prefix(card, cconf, dai_link->codecs->of_node, PREFIX "prefix"); /* check "prefix" from each node if top doesn't have */ - if (!priv->codec_conf.of_node) - snd_soc_of_parse_node_prefix(np, - &priv->codec_conf, + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(np, cconf, dai_link->codecs->of_node, "prefix"); } @@ -238,7 +243,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct snd_soc_card *card = simple_priv_to_card(priv); bool is_fe; int ret, loop; - int dai_idx, link_idx; + int dai_idx, link_idx, conf_idx; if (!top) return -EINVAL; @@ -256,6 +261,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) loop = 1; link_idx = 0; dai_idx = 0; + conf_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { node = dev->of_node; @@ -273,6 +279,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) ret = asoc_simple_card_dai_link_of(node, np, codec, priv, &dai_idx, link_idx++, + &conf_idx, is_fe, !loop); if (ret < 0) return ret; @@ -368,6 +375,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; struct snd_soc_card *card; + struct snd_soc_codec_conf *cconf; struct device *dev = &pdev->dev; int ret, i; int lnum = 0, dnum = 0, cnum = 0; @@ -384,6 +392,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -402,6 +411,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = simple_priv_to_card(priv); @@ -409,8 +419,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = lnum; - card->codec_conf = &priv->codec_conf; - card->num_configs = 1; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_simple_card_parse_of(priv); if (ret < 0) { -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 0/7] ALSA: HD-audio display power fixes
by Takashi Iwai 11 Dec '18

11 Dec '18

Hi, this patch is about fixing possible HD-audio display power unbalance. Basically the "fix" is done by refactoring the whole relevant code. It starts from the Intel HD-audio runtime PM refactoring, followed by the display PM API change, and lots of code cleanups. Since it changes the display power API function, it hits both legacy and ASoC drivers. The patches are kept in topic/hda-pm-refactor branch of my sound git tree. This will be an immutable branch once when merged to for-next, so that it can be merged to ASoC tree if any conflicting change needs to be applied. Takashi === Takashi Iwai (7): ALSA: hda/intel: Refactoring PM code ALSA: hda: Refactor display power management ALSA: hda/intel: Drop superfluous AZX_DCAPS_I915_POWERWELL checks ALSA: hda/intel: Properly free the display power at error path ALSA: hda: Make snd_hdac_display_power() void function ASoC: hdac_hdmi: Add missing display power-off at driver removal ALSA: hda/hdmi: Always set display_power_control for Intel HSW+ codecs include/sound/hda_codec.h | 1 + include/sound/hda_component.h | 9 +- include/sound/hdaudio.h | 7 +- sound/hda/hdac_component.c | 39 ++++--- sound/hda/hdac_device.c | 17 --- sound/pci/hda/hda_codec.c | 16 ++- sound/pci/hda/hda_controller.c | 11 -- sound/pci/hda/hda_intel.c | 192 +++++++++++++-------------------- sound/pci/hda/patch_hdmi.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 22 ++-- sound/soc/intel/skylake/skl.c | 40 ++----- 11 files changed, 134 insertions(+), 228 deletions(-) -- 2.19.2

3 15

[alsa-devel] Realtek ALC294 doesn't work in 4.20-rc6
by Anthony Ettinger 11 Dec '18

11 Dec '18

Does anyone know when a fix will be applied? This is affecting Asus Zenbook UX433F -- Anthony Ettinger http://linkedin.com/in/chovy

2 1

[alsa-devel] [PATCH 0/8] ALSA: fireface: add support for Fireface 800 with MIDI functionality only
by Takashi Sakamoto 11 Dec '18

11 Dec '18

Hi, In 2004, RME GmbH shipped Fireface 800 as a flagship model of its Fireface series. This model consists of some chips: - TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus - TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus - Xilinx Spartan-3 FPGA XC3S400 - Xilinx High-Performance CPLD XC9572XL A remarkable point of this model is that its physical/link layer support S800 (800Mbps) mode of IEEE 1394b. Within devices of audio and music unit on IEEE 1394 bus, this is an unique model to support it, as long as I know. This commit adds support for the model with its MIDI functionality only. The functionality is quite similar to the one of Fireface 400. In patch 01-06, some codes specific to Fireface 400 are arranged to be shared by both models. On the other hand, there're some differences. The largest difference is that the model has no register to switch enabling/disabling tx asynchronous transactions for MIDI messages. Therefore the functionality work well without any assist by userspace. For this work, I use v2.77 firmware and v3.1.24 driver shipped by RME GmbH. If finding some differences between older firmware, please report it with enough information for debug. To help users for test, I prepare a remote branch including these patches. This can be backport to Linux v4.17 or later. https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-midi Takashi Sakamoto (8): ALSA: fireface: share some registers for status of clock synchronization ALSA: fireface: share status and configuration dump ALSA: fireface: share helper function to get current sampling rate and clock source ALSA: fireface: add support for second optical interface for ADAT stream ALSA: fireface: share register for async transaction of MIDI messages ALSA: fireface: add driver data for register for MIDI high address ALSA: fireface: localize a handler for MIDI messages on tx transaction ALSA: fireface: add support for Fireface 800 with MIDI functionality only sound/firewire/Kconfig | 1 + sound/firewire/fireface/Makefile | 3 +- sound/firewire/fireface/ff-pcm.c | 2 +- sound/firewire/fireface/ff-proc.c | 193 +++++++++++++- sound/firewire/fireface/ff-protocol-ff400.c | 282 ++------------------ sound/firewire/fireface/ff-protocol-ff800.c | 27 ++ sound/firewire/fireface/ff-stream.c | 2 +- sound/firewire/fireface/ff-transaction.c | 157 ++++++----- sound/firewire/fireface/ff.c | 54 +++- sound/firewire/fireface/ff.h | 31 ++- 10 files changed, 406 insertions(+), 346 deletions(-) create mode 100644 sound/firewire/fireface/ff-protocol-ff800.c -- 2.19.1

2 9

[alsa-devel] [PATCH] ALSA: hda/ca0132 - make pci_iounmap() call conditional
by Arnd Bergmann 11 Dec '18

11 Dec '18

When building without CONFIG_PCI, we can (depending on the architecture) get a link failure: ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined! Adding a compile-time check for PCI gets it to work correctly on 32-bit ARM. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c40cb6336017..e5bdbc245682 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -8451,7 +8451,7 @@ static void ca0132_free(struct hda_codec *codec) ca0132_exit_chip(codec); snd_hda_power_down(codec); - if (spec->mem_base) + if (IS_ENABLED(CONFIG_PCI) && spec->mem_base) pci_iounmap(codec->bus->pci, spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); -- 2.20.0

2 1

[alsa-devel] [PATCH 0/7] ASoC: merge simple-scu-card into simple-card - step1
by Kuninori Morimoto 11 Dec '18

11 Dec '18

Hi Mark I'm posting "merge audio-graph-card" patch-set now. And, I decided to samethings to "simple-card". These are prepare for merging. Kuninori Morimoto (7): ASoC: simple-card: use cpu/codec pointer on simple_dai_props ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing ASoC: simple-scu-card: add dai-link support ASoC: simple-scu-card: care link / dai count ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props ASoC: simple-scu-card: care multi DPCM codec_conf sound/soc/generic/simple-card.c | 62 +++++---- sound/soc/generic/simple-scu-card.c | 261 +++++++++++++++++++++++++++--------- 2 files changed, 235 insertions(+), 88 deletions(-) -- 2.7.4

2 10

[alsa-devel] [PATCH 1/2] ASoC: Intel: bytcr_rt5640: Add quirk for the Prowise PT301 tablet
by Hans de Goede 11 Dec '18

11 Dec '18

Add a quirk for the Prowise PT301 tablet, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. Also it uses IN1 for its MIC and JD2 for jack-detect, rather then the default IN3 and JD1. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8587bd3d1cc1..6df902e3a380 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -674,6 +674,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { + /* Prowise PT301 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Prowise"), + DMI_MATCH(DMI_PRODUCT_NAME, "PT301"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), -- 2.19.2

3 5

[alsa-devel] Applied "ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f63196eaa83e033de523602324ff91288390a67 Mon Sep 17 00:00:00 2001 From: Zhuohao Lee <zhuohao(a)chromium.org> Date: Mon, 10 Dec 2018 12:32:13 +0800 Subject: [PATCH] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping >From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND. Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 58eb0fe69978..723a4935ed76 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: sdm845: add rt5663 codec select
by Arnd Bergmann 11 Dec '18

11 Dec '18

Calling into the codec driver adds a dependency on that being reachable from the module: ERROR: "rt5663_sel_asrc_clk_src" [sound/soc/qcom/snd-soc-sdm845.ko] undefined! Add the corresponding select statement, as it is done in the other user (Intel). Fixes: f7485875a687 ("ASoC: sdm845: Add configuration for headset codec") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 2a4c912d1e48..1b9fc0665792 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -100,6 +100,7 @@ config SND_SOC_SDM845 depends on QCOM_APR select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON + select SND_SOC_RT5663 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.20.0

2 1

[alsa-devel] [PATCH] ASoC: simple-card-utils: fix build warning without CONFIG_OF
by Arnd Bergmann 11 Dec '18

11 Dec '18

When CONFIG_OF is disabled, of_graph_parse_endpoint() does not initialize 'info', and gcc can see that: sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_card_parse_graph_dai': sound/soc/generic/simple-card-utils.c:284:13: error: 'info.port' may be used uninitialized in this function [-Werror=maybe-uninitialized] It's probably best to check the return code anyway, and that also takes care of the warning. Fixes: b6f3fc005a2c ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/generic/simple-card-utils.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 6a31d07976b9..17d8aee43835 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -280,7 +280,10 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) * Non HDMI sound case, counting port/endpoint on its DT * is enough. Let's count it. */ - of_graph_parse_endpoint(ep, &info); + ret = of_graph_parse_endpoint(ep, &info); + if (ret) + return -ENXIO; + return info.port; } -- 2.20.0

3 2

[alsa-devel] Applied "ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dbd08fe59cf8c2f4bad0a72e912913c20db4774e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 4 Dec 2018 08:20:13 +0000 Subject: [PATCH] ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing Current simple-scu-card driver is parsing codec position for DPCM and consider DAI format. But, current operation is doing totally pointless, because it should be called for each CPU/Codec pair. Let's tidyup asoc_simple_card_parse_daifmt() timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 37 +++++++++++++++++------------ 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 656abe2015e1..24099e61d1b1 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -92,17 +92,24 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int asoc_simple_card_dai_link_of(struct device_node *np, +static int asoc_simple_card_dai_link_of(struct device_node *link, + struct device_node *np, + struct device_node *codec, struct simple_card_data *priv, - unsigned int daifmt, - int idx, bool is_fe) + int idx, bool is_fe, + bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); struct snd_soc_card *card = simple_priv_to_card(priv); + char *prefix = ""; int ret; + /* For single DAI link & old style of DT node */ + if (is_top_level_node) + prefix = PREFIX; + if (is_fe) { int is_single_links = 0; struct snd_soc_dai_link_component *codecs; @@ -178,7 +185,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret < 0) return ret; - dai_link->dai_fmt = daifmt; + ret = asoc_simple_card_parse_daifmt(dev, link, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + return ret; + dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &asoc_simple_card_ops; @@ -191,10 +202,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + struct device_node *node = dev->of_node; struct device_node *np; + struct device_node *codec; struct snd_soc_card *card = simple_priv_to_card(priv); - struct device_node *node = dev->of_node; - unsigned int daifmt = 0; bool is_fe; int ret, i; @@ -211,22 +222,18 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) asoc_simple_card_parse_convert(dev, node, PREFIX, &priv->adata); - /* find 1st codec */ - np = of_get_child_by_name(node, PREFIX "codec"); - if (!np) + i = 0; + codec = of_get_child_by_name(node, PREFIX "codec"); + if (!codec) return -ENODEV; - ret = asoc_simple_card_parse_daifmt(dev, node, np, PREFIX, &daifmt); - if (ret < 0) - return ret; - - i = 0; for_each_child_of_node(node, np) { is_fe = false; if (of_node_name_eq(np, PREFIX "cpu")) is_fe = true; - ret = asoc_simple_card_dai_link_of(np, priv, daifmt, i, is_fe); + ret = asoc_simple_card_dai_link_of(node, np, codec, priv, + i, is_fe, true); if (ret < 0) return ret; i++; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH - AAF PCM plugin 0/7] Follow-up improvements
by Andre Guedes 10 Dec '18

10 Dec '18

Hi all, This patch series provides some follow-up improvements to the AVTP Audio Format (AAF) plugin. The highlight of this series is the implementation of a transmission offload mechanism which improves the plugin performance considerably. Details are provided in the following paragraphs. Currently, the AVTPDU transmission interval is controlled by the AAF plugin, in software. At every timer expiration, the plugin transmits one AVTPDU. This has some implications in terms of task scheduling. For instance, in class A streams, AVTPDUs are transmitted at every 125 us which means the application task should be scheduled-in at every 125 us. In a general purpose Linux system, such scheduling interval can be hard to cope with in the long-run. To mitigate that issue, this series introduces a transmission offload mechanism that leverages the SO_TXTIME sockopt and ETF qdisc features recently introduced to kernel 4.19. Instead of sending one AVTPDU at every timer expiration, the plugin sends several AVTPDUs at once to the kernel, configuring their Tx time so the transmission interval is maintained. The kernel can then offload packet transmission to the hardware (if the network controller supports it), providing more transmission time accuracy. This offloading mechanism enables the application to sleep for longer times, easing on task scheduling. To illustrate the improvements provided by this series, I ran a before-after experiment. The experiment setup consisted in 2 PCs with Intel i210 card connected back-to-back running an up-to-date Archlinux with kernel 4.19.4. I ran 'aplay' for 5 minutes on one PC and captured the AVTPDUs on the other PC. The metric under evaluation is the transmission interval and it is measured by checking the 'time_delta' information from ethernet frames captured at the receiving side. If you're interested in reproducing the experiments, let me know and I can share my helper scripts. The table below shows the experiment outcome for a Class A, stereo, 16-bit sample, 48 kHz stream. The unit is nanoseconds. | Mean | Stdev | Min | Max | Range | -------+--------+---------+---------+---------+---------+ Before | 125000 | 1154 | 75311 | 172144 | 96833 | After | 125000 | 18 | 124960 | 125048 | 88 | Before this patchset, the transmission interval mean is equal to the optimal value (Class A stream -> 125 us interval), and it is kept the same after the patchset. However, the dispersion measurements had improved considerably, meaning the system is consistently transmitting AVTPDUs at the correct interval. Finally, to help the review process, here follows a quick summary of the patches within this series: * PATCH 1: fixes a type in the plugin documentation. * PATCH 2: enables the user to configure the presentation time tolerance. * PATCH 3-5: refactor the code in order to land the offload mechanism code smoothly. * PATCH 6-7: implement the transmission offload mechanism. This series can also be found in my alsa-plugins tree in github [1]. Regards, Andre [1] https://github.com/aguedes/alsa-plugins Andre Guedes (7): doc: Fix typo in AAF doc aaf: Add presentation time tolerance aaf: Refactor AVTPDU transmission routines aaf: Refactor AVTPDU reception routines aaf: Refactor timeout routines aaf: Tx multiple AVTPDUs per media clock tick aaf: AVTPDU transmission periodicity aaf/pcm_aaf.c | 600 +++++++++++++++++++++++++++++++++----------------- configure.ac | 2 +- doc/aaf.txt | 40 +++- 3 files changed, 429 insertions(+), 213 deletions(-) -- 2.19.1

3 10

[alsa-devel] [PATCH v1] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping
by Zhuohao Lee 10 Dec '18

10 Dec '18

>From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND Change-Id: I67f929f226883bd289cb7beb0a7b23f40e7121e2 Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..d3531f0eb357 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.20.0.rc2.403.gdbc3b29805-goog

3 4

[alsa-devel] [PATCH 0/2] ALSA: HDAudio: enable dynamic selection between legacy and Skylake drivers
by Pierre-Louis Bossart 10 Dec '18

10 Dec '18

This patchset depends on the "ASoC: Intel: Skylake: probe and Kconfig simplifications" series. Add the same logic based on PCI class/subclass/prog-if in the HDaudio legacy probe, and add a module parameter to bypass or force the automatic detection should it ever fail. This patchset was tested on a SKL device with the DSP disabled and a WHL device with the DSP enabled. All 6 combinations of module parameters provided the expected behavior. The SOF driver will use exactly the same mechanism, and add a mutual exclusion with the Skylake/SST driver to avoid having 3 drivers for the same PCI ID. Two is complicated enough. Pierre-Louis Bossart (2): ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection include/sound/hdaudio.h | 6 ++++ sound/pci/hda/Kconfig | 46 +++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 53 +++++++++++++++++++++++++++++----- sound/soc/intel/Kconfig | 6 ++++ sound/soc/intel/skylake/skl.c | 44 +++++++++++++++++++--------- 6 files changed, 135 insertions(+), 22 deletions(-) -- 2.17.1

2 12

[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping
by Zhuohao Lee 10 Dec '18

10 Dec '18

>From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND. Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..d3531f0eb357 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.20.0.rc2.403.gdbc3b29805-goog

2 1

[alsa-devel] [PATCH - snd_pcm_extplug_set_param_link() 1/1] pcm: extplug: Keep format and channels the same if requested
by twischer＠de.adit-jv.com 10 Dec '18

10 Dec '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this patch it is not possible to link the channel and format parameter if snd_pcm_extplug_set_param_*() or snd_pcm_extplug_set_slave_param_*() is called. Therefore the client and slave parameter can differ. So the extplug has to implement conversion. To avoid this the new snd_pcm_extplug_set_param_link() function can be called. As a reproduction sceanrio the following extplug source code can be used: static snd_pcm_sframes_t my_transfer(snd_pcm_extplug_t *e, const snd_pcm_channel_area_t *da, snd_pcm_uframes_t dof, const snd_pcm_channel_area_t *sa, snd_pcm_uframes_t sof, snd_pcm_uframes_t s) { return s; } static const snd_pcm_extplug_callback_t my_own_callback = { .transfer = my_transfer }; SND_PCM_PLUGIN_DEFINE_FUNC(my_plug) { snd_config_iterator_t i, next; snd_config_t *slave = NULL; snd_pcm_extplug_t *myplug; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0) continue; if (strcmp(id, "slave") == 0) { slave = n; continue; } return -EINVAL; } myplug = calloc(1, sizeof(*myplug)); myplug->version = SND_PCM_EXTPLUG_VERSION; myplug->callback = &my_own_callback; snd_pcm_extplug_create(myplug, name, root, slave, stream, mode); snd_pcm_extplug_set_param_minmax(myplug, SND_PCM_EXTPLUG_HW_CHANNELS, 1, 16); // snd_pcm_extplug_set_param_link(myplug, SND_PCM_EXTPLUG_HW_CHANNELS, 1); *pcmp = myplug->pcm; return 0; } SND_PCM_PLUGIN_SYMBOL(my_plug); To use this plugin the following ALSA configuration is required: pcm.myplug { type my_plug slave.pcm hw:Dummy } With this configuration without this patch snd_pcm_hw_params_get_channels_max() will always return 16 channels independent of the supported channels of the dummy device. Due to that for example the start up of JACK would fail: $ modprobe snd_dummy $ jackd -d alsa -P myplug ALSA: cannot set channel count to 16 for playback ALSA: cannot configure playback channel Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/include/pcm_extplug.h b/include/pcm_extplug.h index ced934f..e5c02d4 100644 --- a/include/pcm_extplug.h +++ b/include/pcm_extplug.h @@ -184,6 +184,8 @@ int snd_pcm_extplug_set_param_list(snd_pcm_extplug_t *extplug, int type, unsigne int snd_pcm_extplug_set_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max); int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, unsigned int num_list, const unsigned int *list); int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max); +int snd_pcm_extplug_set_param_link(snd_pcm_extplug_t *extplug, int type, + int keep_link); /** * set the parameter constraint with a single value diff --git a/src/pcm/pcm_ext_parm.h b/src/pcm/pcm_ext_parm.h index d25f2ab..7b99bef 100644 --- a/src/pcm/pcm_ext_parm.h +++ b/src/pcm/pcm_ext_parm.h @@ -5,6 +5,7 @@ struct snd_ext_parm { unsigned int *list; unsigned int active: 1; unsigned int integer: 1; + unsigned int keep_link: 1; }; static inline snd_mask_t *hw_param_mask(snd_pcm_hw_params_t *params, diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index 1f887c5..94002dc 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -249,7 +249,7 @@ static unsigned int get_links(struct snd_ext_parm *params) SND_PCM_HW_PARBIT_TICK_TIME); for (i = 0; i < SND_PCM_EXTPLUG_HW_PARAMS; i++) { - if (params[i].active) + if (params[i].active && !params[i].keep_link) links &= ~excl_parbits[i]; } return links; @@ -642,6 +642,17 @@ as former functions. To clear the parameter constraints, call #snd_pcm_extplug_params_reset() function. +When using snd_pcm_extplug_set_param_*() or snd_pcm_extplug_set_slave_param_*() +for any parameter. This parameter is no longer linked between the client and +slave PCM. Therefore it could differ and the extplug has to support conversion +between all valid parameter configurations. To keep the client and slave +parameter linked #snd_pcm_extplug_set_param_link() can be used for the +corresponding parameter. For example if the extplug does not support channel nor +format conversion the supported client parameters can be limited with +snd_pcm_extplug_set_param_*() and afterwards +#snd_pcm_extplug_set_param_link(ext, SND_PCM_EXTPLUG_HW_FORMAT, 1) and +#snd_pcm_extplug_set_param_link(ext, SND_PCM_EXTPLUG_HW_CHANNELS, 1) should be +called to keep the client and slave parameters the same. */ /** @@ -849,3 +860,26 @@ int snd_pcm_extplug_set_param_minmax(snd_pcm_extplug_t *extplug, int type, unsig return snd_ext_parm_set_minmax(&ext->params[type], min, max); } +/** + * @brief Keep the client and slave format/channels the same if requested. This + * is for example useful if this extplug does not support any channel + * conversion. + * @param extplug the extplug handle + * @param type parameter type + * @param keep_link if 1 the parameter identified by type will be kept the same + * for the client and slave PCM of this extplug + * @return 0 if successful, or a negative error code + */ +int snd_pcm_extplug_set_param_link(snd_pcm_extplug_t *extplug, int type, + int keep_link) +{ + extplug_priv_t *ext = extplug->pcm->private_data; + + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { + SNDERR("EXTPLUG: invalid parameter type %d", type); + return -EINVAL; + } + ext->params[type].keep_link = keep_link ? 1 : 0; + ext->sparams[type].keep_link = keep_link ? 1 : 0; + return 0; +} -- 2.7.4

2 1

Re: [alsa-devel] [PATCH 1/3] ALSA: hda/realtek: ALC294 mic and headset-mode fixups for ASUS X542UN
by Jian-Hong Pan 10 Dec '18

10 Dec '18

> Kailang <kailang(a)realtek.com> 於 2018年12月5日週三下午4:36寫道： > > > > Hi Jian-Hong, > > > > Could you test to change the model to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC? > > > > .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE ==> change to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC We do not have ASUS X542UN in hand right now, but we have ASUS X542UQ which goes with the same quirk as X542UN. I tested with ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, and it also works. So, I am going to send new patch set using ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC for ALC294. Thanks, Jian-Hong Pan

3 13

[alsa-devel] Patch for ALC294 Realtek soundcard in ASUS Zenbook UX433F?
by Anthony Ettinger 10 Dec '18

10 Dec '18

I saw something come across the mailing list earlier today, that supposedly fixes the no sound issues with this asus zenbook. Does anyone know when it will make it to the kernel master? I tried 4.20-rc6 but sound still doesn't work. -- Anthony Ettingers http://linkedin.com/in/chovy

1 0