December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: audio-graph-scu-card: care multi DPCM codec_conf" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-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 7c2ae7981d73f9e4f10ed9d1dcb5bc6056189b30 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:32 +0000 Subject: [PATCH] ASoC: audio-graph-scu-card: care multi DPCM codec_conf Current audio-graph-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/audio-graph-scu-card.c | 40 +++++++++++++++++------- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index ed128d430060..e1b192ea147b 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -24,17 +24,18 @@ struct graph_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; 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) @@ -114,7 +115,8 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, 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 is_fe) + 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); @@ -159,6 +161,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, 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"; @@ -171,6 +175,9 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, 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; @@ -186,14 +193,12 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, 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"); /* check "prefix" from each node if top doesn't have */ - if (!priv->codec_conf.of_node) - snd_soc_of_parse_node_prefix(node, - &priv->codec_conf, + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(node, cconf, dai_link->codecs->of_node, PREFIX "prefix"); } @@ -232,7 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - int dai_idx, link_idx, ret; + int dai_idx, link_idx, conf_idx, ret; int rc, codec; if (!node) @@ -256,6 +261,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) link_idx = 0; dai_idx = 0; + conf_idx = 0; codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* @@ -284,7 +290,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, priv, &dai_idx, - link_idx++, !codec); + link_idx++, &conf_idx, + !codec); if (ret < 0) goto parse_of_err; } @@ -294,6 +301,14 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) 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: @@ -390,6 +405,7 @@ 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; + struct snd_soc_codec_conf *cconf; int lnum = 0, dnum = 0, cnum = 0; int ret, i; @@ -405,6 +421,7 @@ static int asoc_graph_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; @@ -423,6 +440,7 @@ 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); @@ -430,8 +448,8 @@ static int asoc_graph_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_graph_card_parse_of(priv); if (ret < 0) { -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-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 0e3460bceedc98176684602a34659d51cd8bd4ce Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:48 +0000 Subject: [PATCH] ASoC: audio-graph-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 audio-graph-card and audio-graph-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 audio-graph-card. It is same logic with audio-graph-scu-card, thus easy merging. This is prepare for merging audio card 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 | 57 +++++++++++++++++----------- 1 file changed, 34 insertions(+), 23 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1b158cc2df88..1da9532ef897 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -23,8 +23,8 @@ 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 asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ struct snd_soc_dai_link_component platform; unsigned int mclk_fs; @@ -33,6 +33,7 @@ struct graph_card_data { struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct gpio_desc *pa_gpio; }; @@ -75,13 +76,13 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai); + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); if (ret) return ret; - ret = asoc_simple_card_clk_enable(&dai_props->codec_dai); + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); if (ret) - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); return ret; } @@ -92,9 +93,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) 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->cpu_dai); - asoc_simple_card_clk_disable(&dai_props->codec_dai); + asoc_simple_card_clk_disable(dai_props->codec_dai); } static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, @@ -139,17 +140,16 @@ static const struct snd_soc_ops asoc_graph_card_ops = { 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 snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - struct graph_dai_props *dai_props = - graph_priv_to_props(priv, rtd->num); - int ret; + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - ret = asoc_simple_card_init_dai(codec, &dai_props->codec_dai); + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_init_dai(cpu, &dai_props->cpu_dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); if (ret < 0) return ret; @@ -158,17 +158,22 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, struct graph_card_data *priv, - int idx) + int *dai_idx, int link_idx) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); - struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; - struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; + 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 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; + cpu_dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + codec_dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) @@ -231,8 +236,8 @@ 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; - int rc, idx = 0; - int ret; + int rc, ret; + int link_idx, dai_idx; ret = asoc_simple_card_of_parse_widgets(card, NULL); if (ret < 0) @@ -245,8 +250,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* 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, idx++); + ret = asoc_graph_card_dai_link_of(it.node, priv, + &dai_idx, link_idx++); if (ret < 0) { of_node_put(it.node); @@ -291,6 +299,7 @@ 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; int num, ret, i; @@ -306,7 +315,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, num * 2, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -330,6 +340,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = graph_priv_to_card(priv); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max98373: Added max98373_reset for stable amp reset" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: max98373: Added max98373_reset for stable amp reset 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 20f2ab247d3b787af91c1aa5eb27c5061744c154 Mon Sep 17 00:00:00 2001 From: Ryan Lee <RyanS.Lee(a)maximintegrated.com> Date: Fri, 30 Nov 2018 03:21:09 +0000 Subject: [PATCH] ASoC: max98373: Added max98373_reset for stable amp reset This patch added max98373_reset function to avoid amp software reset failure and code duplication. Reset verification step has been added for stable amp reset and it repeats verification maximum 3 times when it is failed. Chip revision ID is available when the amp is in the idle state which means software reset is completed well. Additional 10ms delay was added for every retrial and maximum 30ms delay can be applied. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d01318f79..9c8616a7b61c 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v9 0/3] Huawei laptops
by Ayman Bagabas 04 Dec '18

04 Dec '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v8: * Explicitly use NULL in acpi_get_handle * Drop __initconst from huawei_wmi_keymap Changes from v7: * Switch to wmi_driver * Use devm to allocate and manage devices * Skip registering LED subsystem if a ACPI controller method was not found Changes from v6: * 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 from v5: * Review tags Changes from v4: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes from v3: * Code formatting Changes from v2: * 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 | 16 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 226 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 265 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.2

5 8

[alsa-devel] [PATCH linux-next v2 6/6] ASoC: rsnd: add avb clocks
by jiada_wang＠mentor.com 04 Dec '18

04 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> There are AVB Counter Clocks in ADG, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch registers 8 AVB Counter Clocks when clock-cells of rcar_sound node is 2, Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/adg.c | 306 +++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/gen.c | 9 ++ sound/soc/sh/rcar/rsnd.h | 9 ++ 3 files changed, 315 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66588eb..2c03d420ae76 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -5,6 +5,8 @@ // Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/clk-provider.h> +#include <linux/of_address.h> +#include <dt-bindings/clock/renesas-adg.h> #include "rsnd.h" #define CLKA 0 @@ -21,13 +23,33 @@ #define BRRx_MASK(x) (0x3FF & x) +#define ADG_CLK_NAME "adg" +#define AVB_CLK_NAME "avb" +#define AVB_CLK_NUM 8 +#define AVB_CLK_NAME_SIZE 10 +#define AVB_MAX_RATE 25000000 +#define AVB_DIV_EN_COM BIT(31) +#define AVB_DIV_MASK 0x3ffff +#define AVB_MAX_DIV 0x3ffc0 + static struct rsnd_mod_ops adg_ops = { .name = "adg", }; +struct clk_avb { + struct clk_hw hw; + unsigned int idx; + struct rsnd_mod *mod; + /* lock reg access */ + spinlock_t *lock; +}; + +#define to_clk_avb(_hw) container_of(_hw, struct clk_avb, hw) + struct rsnd_adg { struct clk *clk[CLKMAX]; struct clk *clkout[CLKOUTMAX]; + struct clk *clkavb[AVB_CLK_NUM]; struct clk_onecell_data onecell; struct rsnd_mod mod; u32 flags; @@ -37,6 +59,7 @@ struct rsnd_adg { int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ + spinlock_t avb_lock; }; #define LRCLK_ASYNC (1 << 0) @@ -408,6 +431,239 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) } } +static struct clk *rsnd_adg_clk_src_twocell_get(struct of_phandle_args *clkspec, + void *data) +{ + unsigned int clkidx = clkspec->args[1]; + struct rsnd_adg *adg = data; + const char *type; + struct clk *clk; + + switch (clkspec->args[0]) { + case ADG_FIX: + type = "fixed"; + if (clkidx >= CLKOUTMAX) { + pr_err("Invalid %s clock index %u\n", type, + clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkout[clkidx]; + break; + case ADG_AVB: + type = "avb"; + if (clkidx >= AVB_CLK_NUM) { + pr_err("Invalid %s clock index %u\n", type, + clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkavb[clkidx]; + break; + default: + pr_err("Invalid ADG clock type %u\n", clkspec->args[0]); + return ERR_PTR(-EINVAL); + } + + return clk; +} + +static void clk_avb_div_write(struct rsnd_mod *mod, u32 data, int idx) +{ + switch (idx) { + case 0: + rsnd_mod_write(mod, AVB_CLK_DIV0, data); + break; + case 1: + rsnd_mod_write(mod, AVB_CLK_DIV1, data); + break; + case 2: + rsnd_mod_write(mod, AVB_CLK_DIV2, data); + break; + case 3: + rsnd_mod_write(mod, AVB_CLK_DIV3, data); + break; + case 4: + rsnd_mod_write(mod, AVB_CLK_DIV4, data); + break; + case 5: + rsnd_mod_write(mod, AVB_CLK_DIV5, data); + break; + case 6: + rsnd_mod_write(mod, AVB_CLK_DIV6, data); + break; + case 7: + rsnd_mod_write(mod, AVB_CLK_DIV7, data); + break; + } +} + +static u32 clk_avb_div_read(struct rsnd_mod *mod, int idx) +{ + u32 val = 0; + + switch (idx) { + case 0: + val = rsnd_mod_read(mod, AVB_CLK_DIV0); + break; + case 1: + val = rsnd_mod_read(mod, AVB_CLK_DIV1); + break; + case 2: + val = rsnd_mod_read(mod, AVB_CLK_DIV2); + break; + case 3: + val = rsnd_mod_read(mod, AVB_CLK_DIV3); + break; + case 4: + val = rsnd_mod_read(mod, AVB_CLK_DIV4); + break; + case 5: + val = rsnd_mod_read(mod, AVB_CLK_DIV5); + break; + case 6: + val = rsnd_mod_read(mod, AVB_CLK_DIV6); + break; + case 7: + val = rsnd_mod_read(mod, AVB_CLK_DIV7); + break; + } + + return val; +} + +static int clk_avb_is_enabled(struct clk_hw *hw) +{ + struct clk_avb *avb = to_clk_avb(hw); + + return rsnd_mod_read(avb->mod, AVB_CLK_CONFIG) & BIT(avb->idx); +} + +static int clk_avb_enabledisable(struct clk_hw *hw, int enable) +{ + struct clk_avb *avb = to_clk_avb(hw); + u32 val; + + spin_lock(avb->lock); + + val = rsnd_mod_read(avb->mod, AVB_CLK_CONFIG); + if (enable) + val |= BIT(avb->idx); + else + val &= ~BIT(avb->idx); + rsnd_mod_write(avb->mod, AVB_CLK_CONFIG, val); + + spin_unlock(avb->lock); + + return 0; +} + +static int clk_avb_enable(struct clk_hw *hw) +{ + return clk_avb_enabledisable(hw, 1); +} + +static void clk_avb_disable(struct clk_hw *hw) +{ + clk_avb_enabledisable(hw, 0); +} + +static unsigned long clk_avb_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_avb *avb = to_clk_avb(hw); + u32 div; + + div = clk_avb_div_read(avb->mod, avb->idx) & AVB_DIV_MASK; + if (!div) + return parent_rate; + + return parent_rate * 32 / div; +} + +static unsigned int clk_avb_calc_div(unsigned long rate, + unsigned long parent_rate) +{ + unsigned int div; + + if (!rate) + rate = 1; + + if (rate > AVB_MAX_RATE) + rate = AVB_MAX_RATE; + + div = DIV_ROUND_CLOSEST(parent_rate * 32, rate); + + if (div > AVB_MAX_DIV) + div = AVB_MAX_DIV; + + return div; +} + +static long clk_avb_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + unsigned int div = clk_avb_calc_div(rate, *parent_rate); + + return (*parent_rate * 32) / div; +} + +static int clk_avb_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct clk_avb *avb = to_clk_avb(hw); + unsigned int div = clk_avb_calc_div(rate, parent_rate); + u32 val; + + val = clk_avb_div_read(avb->mod, avb->idx) & ~AVB_DIV_MASK; + clk_avb_div_write(avb->mod, val | div, avb->idx); + + return 0; +} + +static const struct clk_ops clk_avb_ops = { + .enable = clk_avb_enable, + .disable = clk_avb_disable, + .is_enabled = clk_avb_is_enabled, + .recalc_rate = clk_avb_recalc_rate, + .round_rate = clk_avb_round_rate, + .set_rate = clk_avb_set_rate, +}; + +static struct clk *clk_register_avb(struct device *dev, struct rsnd_mod *mod, + unsigned int id, spinlock_t *lock) +{ + struct clk_init_data init; + struct clk_avb *avb; + struct clk *clk; + char name[AVB_CLK_NAME_SIZE]; + const char *parent_name = ADG_CLK_NAME; + + avb = devm_kzalloc(dev, sizeof(*avb), GFP_KERNEL); + if (!avb) + return ERR_PTR(-ENOMEM); + + snprintf(name, AVB_CLK_NAME_SIZE, "%s%u", AVB_CLK_NAME, id); + + avb->idx = id; + avb->lock = lock; + avb->mod = mod; + + /* Register the clock. */ + init.name = name; + init.ops = &clk_avb_ops; + init.flags = CLK_IS_BASIC; + init.parent_names = &parent_name; + init.num_parents = 1; + + avb->hw.init = &init; + + /* init DIV to a valid state */ + clk_avb_div_write(avb->mod, avb->idx, AVB_MAX_DIV); + + clk = devm_clk_register(dev, &avb->hw); + + return clk; +} + static void rsnd_adg_get_clkin(struct rsnd_priv *priv, struct rsnd_adg *adg) { @@ -436,6 +692,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, unsigned long req_48kHz_rate, req_441kHz_rate; int i, req_size; const char *parent_clk_name = NULL; + struct rsnd_mod *mod = rsnd_mod_get(adg); static const char * const clkout_name[] = { [CLKOUT] = "audio_clkout", [CLKOUT1] = "audio_clkout1", @@ -540,21 +797,23 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ of_property_read_u32(np, "#clock-cells", &count); - /* - * for clkout - */ - if (!count) { + + switch (count) { + case 0: + /* + * for clkout + */ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, 0, req_rate[0]); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); } - } - /* - * for clkout0/1/2/3 - */ - else { + break; + case 1: + /* + * for clkout0/1/2/3 + */ for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, @@ -566,6 +825,33 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->onecell.clk_num = CLKOUTMAX; of_clk_add_provider(np, of_clk_src_onecell_get, &adg->onecell); + break; + case 2: + /* + * for clkout0/1/2/3 and avb clocks + */ + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, 0, + req_rate[0]); + if (!IS_ERR(clk)) + adg->clkout[i] = clk; + } + + for (i = 0; i < AVB_CLK_NUM; i++) { + clk = clk_register_avb(dev, mod, i, &adg->avb_lock); + if (!IS_ERR(clk)) + adg->clkavb[i] = clk; + } + + of_clk_add_provider(np, rsnd_adg_clk_src_twocell_get, adg); + + rsnd_mod_write(mod, AVB_CLK_CONFIG, AVB_DIV_EN_COM); + + break; + default: + dev_err(dev, "Invalid clock-cell %d\n", count); + break; } rsnd_adg_get_clkout_end: @@ -612,6 +898,8 @@ int rsnd_adg_probe(struct rsnd_priv *priv) if (!adg) return -ENOMEM; + spin_lock_init(&adg->avb_lock); + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, 0, 0); if (ret) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca639404f2cd..1b000d03b76e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -355,6 +355,15 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + RSND_GEN_S_REG(AVB_CLK_DIV0, 0x11c), + RSND_GEN_S_REG(AVB_CLK_DIV1, 0x120), + RSND_GEN_S_REG(AVB_CLK_DIV2, 0x124), + RSND_GEN_S_REG(AVB_CLK_DIV3, 0x128), + RSND_GEN_S_REG(AVB_CLK_DIV4, 0x12c), + RSND_GEN_S_REG(AVB_CLK_DIV5, 0x130), + RSND_GEN_S_REG(AVB_CLK_DIV6, 0x134), + RSND_GEN_S_REG(AVB_CLK_DIV7, 0x138), + RSND_GEN_S_REG(AVB_CLK_CONFIG, 0x13c), }; static const struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129af6d1..d31b8a65985f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -148,6 +148,15 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AVB_CLK_DIV0, + RSND_REG_AVB_CLK_DIV1, + RSND_REG_AVB_CLK_DIV2, + RSND_REG_AVB_CLK_DIV3, + RSND_REG_AVB_CLK_DIV4, + RSND_REG_AVB_CLK_DIV5, + RSND_REG_AVB_CLK_DIV6, + RSND_REG_AVB_CLK_DIV7, + RSND_REG_AVB_CLK_CONFIG, /* SSIU */ RSND_REG_SSI_MODE, -- 2.17.0

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[alsa-devel] [PATCH linux-next v2 0/6] clk: renesas: adg: add AVB Clock
by jiada_wang＠mentor.com 04 Dec '18

04 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> on R-Car SoCs there are AVB Counter Clocks, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch-set adds 'adg' clock to R-Car Soc, and changes adg driver to register avb clocks when clock-cells of rcar_sound node is 2. --- v2: - expends adg register size and register avb clocks instead of add new clk-avb driver - Add adg clock v1: initial version Jiada Wang (2): dt-bindings: clock: add clock id for renesas adg clocks ASoC: rsnd: add avb clocks Takeshi Kihara (4): clk: renesas: r8a7795: Add ADG clock clk: renesas: r8a7796: Add ADG clock clk: renesas: r8a77990: Add ADG clocks clk: renesas: r8a77995: Add ADG clock drivers/clk/renesas/r8a7795-cpg-mssr.c | 1 + drivers/clk/renesas/r8a7796-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77990-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77995-cpg-mssr.c | 1 + include/dt-bindings/clock/renesas-adg.h | 11 + sound/soc/sh/rcar/adg.c | 306 +++++++++++++++++++++++- sound/soc/sh/rcar/gen.c | 9 + sound/soc/sh/rcar/rsnd.h | 9 + 8 files changed, 330 insertions(+), 9 deletions(-) create mode 100644 include/dt-bindings/clock/renesas-adg.h -- 2.17.0

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[alsa-devel] How to inform application about sample rate changed from device.
by Chakravarthi Pradeep 04 Dec '18

04 Dec '18

Hi All, I'm working in FPGA device, which can provide raw video and pcm data. I'm using v4l2 and alsa framework for video and audio respectively. Audio flow path is using ALSA framework and video flow path is using v4l2 framework. Device can generate different audio sample rate such as 44.1KHz, 48KHz and 96KHz. VLC can render video and audio smoothly with 48KHz of sample rate, 2 channel and video with FHD(1920x1080). When device is set with 44.1KHz and 96KHz then vlc audio is breaking.I'm using vlc version as 2.2.8 and there is no way to inform how to set the sample rate for audio device. Below are queries: 1) When device is generating audio sample rate of 44.1KHz then how to inform VLC that use 44.1KHz sample rate(same is for 96KHz) 2) what is command for vlc with video and audio device setting for width=1920 height= 1080 colorspace = RV24, channel =2,sample rate 44.1KHz. Assume video device is (/dev/video0 and audio device is hw:1.0) -- Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

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[alsa-devel] [PATCH 1/2] ALSA: hda/tegra - Probe up to 8 codecs
by Thierry Reding 03 Dec '18

03 Dec '18

From: Thierry Reding <treding(a)nvidia.com> Recent devices support more than the 4 codecs that the AZX core will probe by default. Probe up to 8 codecs to make sure all of them are enumerated. Suggested-by: Sameer Pujar <spujar(a)nvidia.com> Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 84ea6069d28b..83befd8d43e8 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -531,7 +531,7 @@ static void hda_tegra_probe_work(struct work_struct *work) goto out_free; /* create codec instances */ - err = azx_probe_codecs(chip, 0); + err = azx_probe_codecs(chip, 8); if (err < 0) goto out_free; -- 2.19.1

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[alsa-devel] [PATCH v8 0/3] Huawei laptops
by Ayman Bagabas 03 Dec '18

03 Dec '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v7: * Switch to wmi_driver * Use devm to allocate and register devices * Skip registering LED subsystem if a ACPI controller method was not found Changes from v6: * 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 from v5: * Review tags Changes from v4: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes from v3: * Code formatting Changes from v2: * 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 | 16 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 238 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 277 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.1

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[alsa-devel] UAF write in usb_audio_probe
by Mathias Payer 03 Dec '18

03 Dec '18

Hi there, We are submitting a patch for an UAF write in usb_audio_probe. A malicious USB device can send a broken device configuration that will trigger a free of the underlying object before it is decremented. I'm sending the patch and KASAN report after discussing with Takashi on the security mailing list so that he can merge. The attacker needs local access to plug in a malicious USB device that replays the trace (e.g., through FaceDancer) to get read/write primitives in the kernel. For, e.g., Android or locked Desktops this becomes security critical. This bug is present in 4.14.81 through to HEAD (i.e., all versions we tested). The place where UAF is triggered is in `usb_audio_probe` in `sound/usb/card.c`. In the error handling code in `usb_audio_probe`, if the number of interfaces read from the device side is zero, `chip->card` object is freed. Unfortunately, the `chip` object is part of the `card` object, thus, when `chip->card` is freed, `chip` is also freed. However, in the error handling code, the `active` field will be written to by an `atomic_dec` operation, resulting in an UAF write. ``` static int usb_audio_probe(struct usb_interface *intf, const struct usb_device_id *usb_id) ..... __error: if (chip) { if (!chip->num_interfaces) snd_card_free(chip->card); // UAF write atomic_dec(&chip->active); } mutex_unlock(&register_mutex); return err; ``` The attacker can race the snd_card_free and the atomic_dec by attaching new USB devices (the attacker can time when what parts of the first malicious device are read then play with attaching the second device; the race is arbitrarily repeatable). The new USB device descriptors will use the recently freed memory which is then modified by the atomic_dec, resulting in an attacker-controlled decrement operation to, e.g., a USB data structure under the attacker's control. We have also attached the KASAN report from when the bug was triggered. The patch for card.c is attached. It moves the atomic_dec to above the free of the chip memory object, ensuring that, if no cards are left, the decrement happens before the free. We've also added a comment to clarify that the two objects are dependent. Thanks, Hui Peng Mathias Payer

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