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

[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

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[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

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[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

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[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