January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings" to the asoc tree
by Mark Brown 07 Jan '19

07 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

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[alsa-devel] [PATCH v2 0/3] add Xilinx audio formatter driver
by Maruthi Srinivas Bayyavarapu 07 Jan '19

07 Jan '19

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 | 565 +++++++++++++++++++++ 4 files changed, 603 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 7

[alsa-devel] Applied "ASoC: xlnx: enable audio formatter driver build" to the asoc tree
by Mark Brown 07 Jan '19

07 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

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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 07 Jan '19

07 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

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[alsa-devel] [PATCH 00/13 v2] ASoC: audio/simple card tidyup/cleanup
by Kuninori Morimoto 07 Jan '19

07 Jan '19

Hi Mark As I mentioned on off-list mail, I noticed that previous patch-set included many exchanges in 1 patch. Because of this, the reviewing was very difficult. I had break downed it into few more small patches. I hope it becomes more review-able 1) : for "reg" property issue, again. 2) - 7) : missing 1CPU-1Codec DPCM case, and cleanup for audio-graph 8) - 13) : missing 1CPU-1Codec DPCM case, and cleanup for simple-card Kuninori Morimoto (13): 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: add link_info 5) ASoC: audio-graph-card: cleanup DAI link loop method - step1 6) ASoC: audio-graph-card: cleanup DAI link loop method - step2 7) ASoC: audio-graph-card: reduce naming prefix 8) ASoC: simple-card: add asoc_simple_card_get_conversion() 9) ASoC: simple-card: add 1 CPU : 1 Codec support again 10) ASoC: simple-card: add link_info 11) ASoC: simple-card: cleanup DAI link loop method - step1 12) ASoC: simple-card: cleanup DAI link loop method - step2 13) ASoC: simple-card: reduce naming prefix sound/soc/generic/audio-graph-card.c | 465 +++++++++++++++++++--------------- sound/soc/generic/simple-card-utils.c | 14 +- sound/soc/generic/simple-card.c | 454 ++++++++++++++++++++------------- 3 files changed, 553 insertions(+), 380 deletions(-) -- 2.7.4

2 45

[alsa-devel] Applied "ASoC: simple-card: add asoc_simple_card_get_conversion()" to the asoc tree
by Mark Brown 07 Jan '19

07 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

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[alsa-devel] Applied "ASoC: simple-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 07 Jan '19

07 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

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[alsa-devel] [PATCH 1/2] ASoC: rockchip: support ACODEC for rk3328
by Katsuhiro Suzuki 07 Jan '19

07 Jan '19

This patch adds support for audio CODEC core of rk3328. Rockchip does not publish detail specification of this core but driver source code is opened on their GitHub repository. https://github.com/rockchip-linux/kernel So I ported this code to linux-next and added some trivial fixes. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 23 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk3328_codec.c | 517 ++++++++++++++++++ sound/soc/codecs/rk3328_codec.h | 210 +++++++ 5 files changed, 757 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 sound/soc/codecs/rk3328_codec.c create mode 100644 sound/soc/codecs/rk3328_codec.h diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt new file mode 100644 index 000000000000..2469588c7ccb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt @@ -0,0 +1,23 @@ +* Rockchip Rk3328 internal codec + +Required properties: + +- compatible: "rockchip,rk3328-codec" +- reg: physical base address of the controller and length of memory mapped + region. +- rockchip,grf: the phandle of the syscon node for GRF register. +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should be "pclk". +- spk-depop-time-ms: speak depop time msec. + +Example for rk3328 internal codec: + +codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0x0 0xff410000 0x0 0x1000>; + rockchip,grf = <&grf>; + clocks = <&cru PCLK_ACODEC>; + clock-names = "pclk"; + spk-depop-time-ms = 100; + status = "disabled"; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62bdb7e333b8..19a5168e0b89 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,6 +129,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RK3328 select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C @@ -799,6 +800,10 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RK3328 + tristate "Rockchip RK3328 audio CODEC" + select REGMAP_MMIO + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 66f55d185620..0d924efbc8ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -132,6 +132,7 @@ snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rk3328-objs := rk3328_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1305-objs := rt1305.o @@ -398,6 +399,7 @@ obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c new file mode 100644 index 000000000000..71f3fc2d970c --- /dev/null +++ b/sound/soc/codecs/rk3328_codec.c @@ -0,0 +1,517 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk3328 ALSA SoC Audio driver +// +// Copyright (c) 2017, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/mfd/syscon.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include "rk3328_codec.h" + +/* + * volume setting + * 0: -39dB + * 26: 0dB + * 31: 6dB + * Step: 1.5dB + */ +#define OUT_VOLUME (0x18) +#define RK3328_GRF_SOC_CON2 (0x0408) +#define RK3328_GRF_SOC_CON10 (0x0428) +#define INITIAL_FREQ (11289600) + +struct rk3328_codec_priv { + struct regmap *regmap; + struct regmap *grf; + struct clk *mclk; + struct clk *pclk; + unsigned int sclk; + int spk_depop_time; /* msec */ +}; + +static const struct reg_default rk3328_codec_reg_defaults[] = { + { CODEC_RESET, 0x03 }, + { DAC_INIT_CTRL1, 0x00 }, + { DAC_INIT_CTRL2, 0x50 }, + { DAC_INIT_CTRL3, 0x0e }, + { DAC_PRECHARGE_CTRL, 0x01 }, + { DAC_PWR_CTRL, 0x00 }, + { DAC_CLK_CTRL, 0x00 }, + { HPMIX_CTRL, 0x00 }, + { HPOUT_CTRL, 0x00 }, + { HPOUTL_GAIN_CTRL, 0x00 }, + { HPOUTR_GAIN_CTRL, 0x00 }, + { HPOUT_POP_CTRL, 0x11 }, +}; + +static int rk3328_codec_reset(struct rk3328_codec_priv *rk3328) +{ + regmap_write(rk3328->regmap, CODEC_RESET, 0x00); + mdelay(10); + regmap_write(rk3328->regmap, CODEC_RESET, 0x03); + + return 0; +} + +static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = PIN_DIRECTION_IN | DAC_I2S_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val = PIN_DIRECTION_OUT | DAC_I2S_MODE_MASTER; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL1, + PIN_DIRECTION_MASK | DAC_I2S_MODE_MASK, val); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + val = DAC_MODE_PCM; + break; + case SND_SOC_DAIFMT_I2S: + val = DAC_MODE_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = DAC_MODE_RJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = DAC_MODE_LJM; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL2, + DAC_MODE_MASK, val); + + return 0; +} + +static void rk3328_analog_output(struct rk3328_codec_priv *rk3328, int mute) +{ + unsigned int val = BIT(17); + + if (mute) + val |= BIT(1); + + regmap_write(rk3328->grf, RK3328_GRF_SOC_CON10, val); +} + +static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val; + + if (mute) + val = HPOUTL_MUTE | HPOUTR_MUTE; + else + val = HPOUTL_UNMUTE | HPOUTR_UNMUTE; + + regmap_update_bits(rk3328->regmap, HPOUT_CTRL, + HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, val); + + return 0; +} + +static int rk3328_codec_power_on(struct rk3328_codec_priv *rk3328, int wait_ms) +{ + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_XCHARGE_MASK, DAC_CHARGE_PRECHARGE); + mdelay(10); + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_ALL_ON); + mdelay(wait_ms); + + return 0; +} + +static int rk3328_codec_power_off(struct rk3328_codec_priv *rk3328, int wait_ms) +{ + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_XCHARGE_MASK, DAC_CHARGE_DISCHARGE); + mdelay(10); + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_ALL_ON); + mdelay(wait_ms); + + return 0; +} + +static const struct rk3328_reg_msk_val playback_open_list[] = { + { DAC_PWR_CTRL, DAC_PWR_MASK, DAC_PWR_ON }, + { DAC_PWR_CTRL, DACL_PATH_REFV_MASK | DACR_PATH_REFV_MASK, + DACL_PATH_REFV_ON | DACR_PATH_REFV_ON }, + { DAC_PWR_CTRL, HPOUTL_ZERO_CROSSING_MASK | HPOUTR_ZERO_CROSSING_MASK, + HPOUTL_ZERO_CROSSING_ON | HPOUTR_ZERO_CROSSING_ON }, + { HPOUT_POP_CTRL, HPOUTR_POP_MASK | HPOUTL_POP_MASK, + HPOUTR_POP_WORK | HPOUTL_POP_WORK }, + { HPMIX_CTRL, HPMIXL_MASK | HPMIXR_MASK, HPMIXL_EN | HPMIXR_EN }, + { HPMIX_CTRL, HPMIXL_INIT_MASK | HPMIXR_INIT_MASK, + HPMIXL_INIT_EN | HPMIXR_INIT_EN }, + { HPOUT_CTRL, HPOUTL_MASK | HPOUTR_MASK, HPOUTL_EN | HPOUTR_EN }, + { HPOUT_CTRL, HPOUTL_INIT_MASK | HPOUTR_INIT_MASK, + HPOUTL_INIT_EN | HPOUTR_INIT_EN }, + { DAC_CLK_CTRL, DACL_REFV_MASK | DACR_REFV_MASK, + DACL_REFV_ON | DACR_REFV_ON }, + { DAC_CLK_CTRL, DACL_CLK_MASK | DACR_CLK_MASK, + DACL_CLK_ON | DACR_CLK_ON }, + { DAC_CLK_CTRL, DACL_MASK | DACR_MASK, DACL_ON | DACR_ON }, + { DAC_CLK_CTRL, DACL_INIT_MASK | DACR_INIT_MASK, + DACL_INIT_ON | DACR_INIT_ON }, + { DAC_SELECT, DACL_SELECT_MASK | DACR_SELECT_MASK, + DACL_SELECT | DACR_SELECT }, + { HPMIX_CTRL, HPMIXL_INIT2_MASK | HPMIXR_INIT2_MASK, + HPMIXL_INIT2_EN | HPMIXR_INIT2_EN }, + { HPOUT_CTRL, HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, + HPOUTL_UNMUTE | HPOUTR_UNMUTE }, +}; + +static int rk3328_codec_open_playback(struct rk3328_codec_priv *rk3328) +{ + int i; + + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_I); + + for (i = 0; i < ARRAY_SIZE(playback_open_list); i++) { + regmap_update_bits(rk3328->regmap, + playback_open_list[i].reg, + playback_open_list[i].msk, + playback_open_list[i].val); + mdelay(1); + } + + msleep(rk3328->spk_depop_time); + rk3328_analog_output(rk3328, 1); + + regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, + HPOUTL_GAIN_MASK, OUT_VOLUME); + regmap_update_bits(rk3328->regmap, HPOUTR_GAIN_CTRL, + HPOUTR_GAIN_MASK, OUT_VOLUME); + + return 0; +} + +static const struct rk3328_reg_msk_val playback_close_list[] = { + { HPMIX_CTRL, HPMIXL_INIT2_MASK | HPMIXR_INIT2_MASK, + HPMIXL_INIT2_DIS | HPMIXR_INIT2_DIS }, + { DAC_SELECT, DACL_SELECT_MASK | DACR_SELECT_MASK, + DACL_UNSELECT | DACR_UNSELECT }, + { HPOUT_CTRL, HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, + HPOUTL_MUTE | HPOUTR_MUTE }, + { HPOUT_CTRL, HPOUTL_INIT_MASK | HPOUTR_INIT_MASK, + HPOUTL_INIT_DIS | HPOUTR_INIT_DIS }, + { HPOUT_CTRL, HPOUTL_MASK | HPOUTR_MASK, HPOUTL_DIS | HPOUTR_DIS }, + { HPMIX_CTRL, HPMIXL_MASK | HPMIXR_MASK, HPMIXL_DIS | HPMIXR_DIS }, + { DAC_CLK_CTRL, DACL_MASK | DACR_MASK, DACL_OFF | DACR_OFF }, + { DAC_CLK_CTRL, DACL_CLK_MASK | DACR_CLK_MASK, + DACL_CLK_OFF | DACR_CLK_OFF }, + { DAC_CLK_CTRL, DACL_REFV_MASK | DACR_REFV_MASK, + DACL_REFV_OFF | DACR_REFV_OFF }, + { HPOUT_POP_CTRL, HPOUTR_POP_MASK | HPOUTL_POP_MASK, + HPOUTR_POP_XCHARGE | HPOUTL_POP_XCHARGE }, + { DAC_PWR_CTRL, DACL_PATH_REFV_MASK | DACR_PATH_REFV_MASK, + DACL_PATH_REFV_OFF | DACR_PATH_REFV_OFF }, + { DAC_PWR_CTRL, DAC_PWR_MASK, DAC_PWR_OFF }, + { HPMIX_CTRL, HPMIXL_INIT_MASK | HPMIXR_INIT_MASK, + HPMIXL_INIT_DIS | HPMIXR_INIT_DIS }, + { DAC_CLK_CTRL, DACL_INIT_MASK | DACR_INIT_MASK, + DACL_INIT_OFF | DACR_INIT_OFF }, +}; + +static int rk3328_codec_close_playback(struct rk3328_codec_priv *rk3328) +{ + size_t i; + + rk3328_analog_output(rk3328, 0); + + regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, + HPOUTL_GAIN_MASK, 0); + regmap_update_bits(rk3328->regmap, HPOUTR_GAIN_CTRL, + HPOUTR_GAIN_MASK, 0); + + for (i = 0; i < ARRAY_SIZE(playback_close_list); i++) { + regmap_update_bits(rk3328->regmap, + playback_close_list[i].reg, + playback_close_list[i].msk, + playback_close_list[i].val); + mdelay(1); + } + + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_I); + + return 0; +} + +static int rk3328_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = DAC_VDL_16BITS; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = DAC_VDL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = DAC_VDL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = DAC_VDL_32BITS; + break; + default: + return -EINVAL; + } + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL2, DAC_VDL_MASK, val); + + val = DAC_WL_32BITS | DAC_RST_DIS; + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL3, + DAC_WL_MASK | DAC_RST_MASK, val); + + return 0; +} + +static int rk3328_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + + return rk3328_codec_open_playback(rk3328); +} + +static void rk3328_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + + rk3328_codec_close_playback(rk3328); +} + +static const struct snd_soc_dai_ops rk3328_dai_ops = { + .hw_params = rk3328_hw_params, + .set_fmt = rk3328_set_dai_fmt, + .digital_mute = rk3328_digital_mute, + .startup = rk3328_pcm_startup, + .shutdown = rk3328_pcm_shutdown, +}; + +static struct snd_soc_dai_driver rk3328_dai[] = { + { + .name = "rk3328-hifi", + .id = RK3328_HIFI, + .playback = { + .stream_name = "HIFI Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .capture = { + .stream_name = "HIFI Capture", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .ops = &rk3328_dai_ops, + }, +}; + +static int rk3328_codec_probe(struct snd_soc_component *component) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(component); + + rk3328_codec_reset(rk3328); + rk3328_codec_power_on(rk3328, 0); + + return 0; +} + +static void rk3328_codec_remove(struct snd_soc_component *component) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(component); + + rk3328_codec_close_playback(rk3328); + rk3328_codec_power_off(rk3328, 0); +} + +static const struct snd_soc_component_driver soc_codec_rk3328 = { + .probe = rk3328_codec_probe, + .remove = rk3328_codec_remove, +}; + +static bool rk3328_codec_write_read_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CODEC_RESET: + case DAC_INIT_CTRL1: + case DAC_INIT_CTRL2: + case DAC_INIT_CTRL3: + case DAC_PRECHARGE_CTRL: + case DAC_PWR_CTRL: + case DAC_CLK_CTRL: + case HPMIX_CTRL: + case DAC_SELECT: + case HPOUT_CTRL: + case HPOUTL_GAIN_CTRL: + case HPOUTR_GAIN_CTRL: + case HPOUT_POP_CTRL: + return true; + default: + return false; + } +} + +static bool rk3328_codec_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CODEC_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_config rk3328_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = HPOUT_POP_CTRL, + .writeable_reg = rk3328_codec_write_read_reg, + .readable_reg = rk3328_codec_write_read_reg, + .volatile_reg = rk3328_codec_volatile_reg, + .reg_defaults = rk3328_codec_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk3328_codec_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +static int rk3328_platform_probe(struct platform_device *pdev) +{ + struct device_node *rk3328_np = pdev->dev.of_node; + struct rk3328_codec_priv *rk3328; + struct resource *res; + struct regmap *grf; + void __iomem *base; + int ret = 0; + + rk3328 = devm_kzalloc(&pdev->dev, sizeof(*rk3328), GFP_KERNEL); + if (!rk3328) + return -ENOMEM; + + grf = syscon_regmap_lookup_by_phandle(rk3328_np, + "rockchip,grf"); + if (IS_ERR(grf)) { + dev_err(&pdev->dev, "missing 'rockchip,grf'\n"); + return PTR_ERR(grf); + } + rk3328->grf = grf; + /* enable i2s_acodec_en */ + regmap_write(grf, RK3328_GRF_SOC_CON2, + (BIT(14) << 16 | BIT(14))); + + ret = of_property_read_u32(rk3328_np, "spk-depop-time-ms", + &rk3328->spk_depop_time); + if (ret < 0) { + dev_info(&pdev->dev, "spk_depop_time use default value.\n"); + rk3328->spk_depop_time = 200; + } + + rk3328_analog_output(rk3328, 0); + + rk3328->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk3328->mclk)) + return PTR_ERR(rk3328->mclk); + + ret = clk_prepare_enable(rk3328->mclk); + if (ret) + return ret; + clk_set_rate(rk3328->mclk, INITIAL_FREQ); + + rk3328->pclk = devm_clk_get(&pdev->dev, "pclk"); + if (IS_ERR(rk3328->pclk)) { + dev_err(&pdev->dev, "can't get acodec pclk\n"); + return PTR_ERR(rk3328->pclk); + } + + ret = clk_prepare_enable(rk3328->pclk); + if (ret < 0) { + dev_err(&pdev->dev, "failed to enable acodec pclk\n"); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &rk3328_codec_regmap_config); + if (IS_ERR(rk3328->regmap)) + return PTR_ERR(rk3328->regmap); + + platform_set_drvdata(pdev, rk3328); + + return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, + rk3328_dai, + ARRAY_SIZE(rk3328_dai)); +} + +static const struct of_device_id rk3328_codec_of_match[] = { + { .compatible = "rockchip,rk3328-codec", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk3328_codec_of_match); + +static struct platform_driver rk3328_codec_driver = { + .driver = { + .name = "rk3328-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rk3328_codec_of_match), + }, + .probe = rk3328_platform_probe, +}; +module_platform_driver(rk3328_codec_driver); + +MODULE_AUTHOR("Sugar Zhang <sugar.zhang(a)rock-chips.com>"); +MODULE_DESCRIPTION("ASoC rk3328 codec driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk3328_codec.h b/sound/soc/codecs/rk3328_codec.h new file mode 100644 index 000000000000..655103586241 --- /dev/null +++ b/sound/soc/codecs/rk3328_codec.h @@ -0,0 +1,210 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rk3328 ALSA SoC Audio driver + * + * Copyright (c) 2017, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + */ + +#ifndef _RK3328_CODEC_H +#define _RK3328_CODEC_H + +#include <linux/bitfield.h> + +/* codec register */ +#define CODEC_RESET (0x00 << 2) +#define DAC_INIT_CTRL1 (0x03 << 2) +#define DAC_INIT_CTRL2 (0x04 << 2) +#define DAC_INIT_CTRL3 (0x05 << 2) +#define DAC_PRECHARGE_CTRL (0x22 << 2) +#define DAC_PWR_CTRL (0x23 << 2) +#define DAC_CLK_CTRL (0x24 << 2) +#define HPMIX_CTRL (0x25 << 2) +#define DAC_SELECT (0x26 << 2) +#define HPOUT_CTRL (0x27 << 2) +#define HPOUTL_GAIN_CTRL (0x28 << 2) +#define HPOUTR_GAIN_CTRL (0x29 << 2) +#define HPOUT_POP_CTRL (0x2a << 2) + +/* REG00: CODEC_RESET */ +#define PWR_RST_BYPASS_DIS (0x0 << 6) +#define PWR_RST_BYPASS_EN (0x1 << 6) +#define DIG_CORE_RST (0x0 << 1) +#define DIG_CORE_WORK (0x1 << 1) +#define SYS_RST (0x0 << 0) +#define SYS_WORK (0x1 << 0) + +/* REG03: DAC_INIT_CTRL1 */ +#define PIN_DIRECTION_MASK BIT(5) +#define PIN_DIRECTION_IN (0x0 << 5) +#define PIN_DIRECTION_OUT (0x1 << 5) +#define DAC_I2S_MODE_MASK BIT(4) +#define DAC_I2S_MODE_SLAVE (0x0 << 4) +#define DAC_I2S_MODE_MASTER (0x1 << 4) + +/* REG04: DAC_INIT_CTRL2 */ +#define DAC_I2S_LRP_MASK BIT(7) +#define DAC_I2S_LRP_NORMAL (0x0 << 7) +#define DAC_I2S_LRP_REVERSAL (0x1 << 7) +#define DAC_VDL_MASK GENMASK(6, 5) +#define DAC_VDL_16BITS (0x0 << 5) +#define DAC_VDL_20BITS (0x1 << 5) +#define DAC_VDL_24BITS (0x2 << 5) +#define DAC_VDL_32BITS (0x3 << 5) +#define DAC_MODE_MASK GENMASK(4, 3) +#define DAC_MODE_RJM (0x0 << 3) +#define DAC_MODE_LJM (0x1 << 3) +#define DAC_MODE_I2S (0x2 << 3) +#define DAC_MODE_PCM (0x3 << 3) +#define DAC_LR_SWAP_MASK BIT(2) +#define DAC_LR_SWAP_DIS (0x0 << 2) +#define DAC_LR_SWAP_EN (0x1 << 2) + +/* REG05: DAC_INIT_CTRL3 */ +#define DAC_WL_MASK GENMASK(3, 2) +#define DAC_WL_16BITS (0x0 << 2) +#define DAC_WL_20BITS (0x1 << 2) +#define DAC_WL_24BITS (0x2 << 2) +#define DAC_WL_32BITS (0x3 << 2) +#define DAC_RST_MASK BIT(1) +#define DAC_RST_EN (0x0 << 1) +#define DAC_RST_DIS (0x1 << 1) +#define DAC_BCP_MASK BIT(0) +#define DAC_BCP_NORMAL (0x0 << 0) +#define DAC_BCP_REVERSAL (0x1 << 0) + +/* REG22: DAC_PRECHARGE_CTRL */ +#define DAC_CHARGE_XCHARGE_MASK BIT(7) +#define DAC_CHARGE_DISCHARGE (0x0 << 7) +#define DAC_CHARGE_PRECHARGE (0x1 << 7) +#define DAC_CHARGE_CURRENT_64I_MASK BIT(6) +#define DAC_CHARGE_CURRENT_64I (0x1 << 6) +#define DAC_CHARGE_CURRENT_32I_MASK BIT(5) +#define DAC_CHARGE_CURRENT_32I (0x1 << 5) +#define DAC_CHARGE_CURRENT_16I_MASK BIT(4) +#define DAC_CHARGE_CURRENT_16I (0x1 << 4) +#define DAC_CHARGE_CURRENT_08I_MASK BIT(3) +#define DAC_CHARGE_CURRENT_08I (0x1 << 3) +#define DAC_CHARGE_CURRENT_04I_MASK BIT(2) +#define DAC_CHARGE_CURRENT_04I (0x1 << 2) +#define DAC_CHARGE_CURRENT_02I_MASK BIT(1) +#define DAC_CHARGE_CURRENT_02I (0x1 << 1) +#define DAC_CHARGE_CURRENT_I_MASK BIT(0) +#define DAC_CHARGE_CURRENT_I (0x1 << 0) +#define DAC_CHARGE_CURRENT_ALL_MASK GENMASK(6, 0) +#define DAC_CHARGE_CURRENT_ALL_OFF 0x00 +#define DAC_CHARGE_CURRENT_ALL_ON 0x7f + +/* REG23: DAC_PWR_CTRL */ +#define DAC_PWR_MASK BIT(6) +#define DAC_PWR_OFF (0x0 << 6) +#define DAC_PWR_ON (0x1 << 6) +#define DACL_PATH_REFV_MASK BIT(5) +#define DACL_PATH_REFV_OFF (0x0 << 5) +#define DACL_PATH_REFV_ON (0x1 << 5) +#define HPOUTL_ZERO_CROSSING_MASK BIT(4) +#define HPOUTL_ZERO_CROSSING_OFF (0x0 << 4) +#define HPOUTL_ZERO_CROSSING_ON (0x1 << 4) +#define DACR_PATH_REFV_MASK BIT(1) +#define DACR_PATH_REFV_OFF (0x0 << 1) +#define DACR_PATH_REFV_ON (0x1 << 1) +#define HPOUTR_ZERO_CROSSING_MASK BIT(0) +#define HPOUTR_ZERO_CROSSING_OFF (0x0 << 0) +#define HPOUTR_ZERO_CROSSING_ON (0x1 << 0) + +/* REG24: DAC_CLK_CTRL */ +#define DACL_REFV_MASK BIT(7) +#define DACL_REFV_OFF (0x0 << 7) +#define DACL_REFV_ON (0x1 << 7) +#define DACL_CLK_MASK BIT(6) +#define DACL_CLK_OFF (0x0 << 6) +#define DACL_CLK_ON (0x1 << 6) +#define DACL_MASK BIT(5) +#define DACL_OFF (0x0 << 5) +#define DACL_ON (0x1 << 5) +#define DACL_INIT_MASK BIT(4) +#define DACL_INIT_OFF (0x0 << 4) +#define DACL_INIT_ON (0x1 << 4) +#define DACR_REFV_MASK BIT(3) +#define DACR_REFV_OFF (0x0 << 3) +#define DACR_REFV_ON (0x1 << 3) +#define DACR_CLK_MASK BIT(2) +#define DACR_CLK_OFF (0x0 << 2) +#define DACR_CLK_ON (0x1 << 2) +#define DACR_MASK BIT(1) +#define DACR_OFF (0x0 << 1) +#define DACR_ON (0x1 << 1) +#define DACR_INIT_MASK BIT(0) +#define DACR_INIT_OFF (0x0 << 0) +#define DACR_INIT_ON (0x1 << 0) + +/* REG25: HPMIX_CTRL*/ +#define HPMIXL_MASK BIT(6) +#define HPMIXL_DIS (0x0 << 6) +#define HPMIXL_EN (0x1 << 6) +#define HPMIXL_INIT_MASK BIT(5) +#define HPMIXL_INIT_DIS (0x0 << 5) +#define HPMIXL_INIT_EN (0x1 << 5) +#define HPMIXL_INIT2_MASK BIT(4) +#define HPMIXL_INIT2_DIS (0x0 << 4) +#define HPMIXL_INIT2_EN (0x1 << 4) +#define HPMIXR_MASK BIT(2) +#define HPMIXR_DIS (0x0 << 2) +#define HPMIXR_EN (0x1 << 2) +#define HPMIXR_INIT_MASK BIT(1) +#define HPMIXR_INIT_DIS (0x0 << 1) +#define HPMIXR_INIT_EN (0x1 << 1) +#define HPMIXR_INIT2_MASK BIT(0) +#define HPMIXR_INIT2_DIS (0x0 << 0) +#define HPMIXR_INIT2_EN (0x1 << 0) + +/* REG26: DAC_SELECT */ +#define DACL_SELECT_MASK BIT(4) +#define DACL_UNSELECT (0x0 << 4) +#define DACL_SELECT (0x1 << 4) +#define DACR_SELECT_MASK BIT(0) +#define DACR_UNSELECT (0x0 << 0) +#define DACR_SELECT (0x1 << 0) + +/* REG27: HPOUT_CTRL */ +#define HPOUTL_MASK BIT(7) +#define HPOUTL_DIS (0x0 << 7) +#define HPOUTL_EN (0x1 << 7) +#define HPOUTL_INIT_MASK BIT(6) +#define HPOUTL_INIT_DIS (0x0 << 6) +#define HPOUTL_INIT_EN (0x1 << 6) +#define HPOUTL_MUTE_MASK BIT(5) +#define HPOUTL_MUTE (0x0 << 5) +#define HPOUTL_UNMUTE (0x1 << 5) +#define HPOUTR_MASK BIT(4) +#define HPOUTR_DIS (0x0 << 4) +#define HPOUTR_EN (0x1 << 4) +#define HPOUTR_INIT_MASK BIT(3) +#define HPOUTR_INIT_DIS (0x0 << 3) +#define HPOUTR_INIT_EN (0x1 << 3) +#define HPOUTR_MUTE_MASK BIT(2) +#define HPOUTR_MUTE (0x0 << 2) +#define HPOUTR_UNMUTE (0x1 << 2) + +/* REG28: HPOUTL_GAIN_CTRL */ +#define HPOUTL_GAIN_MASK GENMASK(4, 0) + +/* REG29: HPOUTR_GAIN_CTRL */ +#define HPOUTR_GAIN_MASK GENMASK(4, 0) + +/* REG2a: HPOUT_POP_CTRL */ +#define HPOUTR_POP_MASK GENMASK(5, 4) +#define HPOUTR_POP_XCHARGE (0x1 << 4) +#define HPOUTR_POP_WORK (0x2 << 4) +#define HPOUTL_POP_MASK GENMASK(1, 0) +#define HPOUTL_POP_XCHARGE (0x1 << 0) +#define HPOUTL_POP_WORK (0x2 << 0) + +#define RK3328_HIFI 0 + +struct rk3328_reg_msk_val { + unsigned int reg; + unsigned int msk; + unsigned int val; +}; + +#endif -- 2.19.2

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[alsa-devel] [PATCH] ASoC: xlnx: fix error handling in xlnx_formatter_pcm_probe
by Gustavo A. R. Silva 07 Jan '19

07 Jan '19

Currently, if platform_get_irq_byname() fails, the returned error turns into a huge value, once it is being store into a variable of type unsigned int, hence never actually reporting any error and causing unexpected behavior when using the values stored in aud_drv_data->s2mm_irq and aud_drv_data->mm2s_irq. Fix this by changing the type of variables s2mm_irq and mm2s_irq in structure xlnx_pcm_drv_data from unsigned int to int. Addresses-Coverity-ID: 1476096 ("Unsigned compared against 0") Fixes: 796175a94a7f ("ASoC: xlnx: add pcm formatter platform driver") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index f7235f7664d7..d2194da928e7 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -76,8 +76,8 @@ struct xlnx_pcm_drv_data { void __iomem *mmio; bool s2mm_presence; bool mm2s_presence; - unsigned int s2mm_irq; - unsigned int mm2s_irq; + int s2mm_irq; + int mm2s_irq; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; struct clk *axi_clk; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rockchip: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 07 Jan '19

07 Jan '19

From: kbuild test robot <fengguang.wu(a)intel.com> sound/soc/codecs/rk3328_codec.c:508:6-11: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Fixes: dc1b33660692 ("ASoC: rockchip: support ACODEC for rk3328") CC: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> Signed-off-by: kbuild test robot <fengguang.wu(a)intel.com> --- tree: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master head: 6a1d293238c1774cef23c25b5188b0c134c29a34 commit: dc1b33660692441867a8ec5be147915dfa3bc8cb [11432/12209] ASoC: rockchip: support ACODEC for rk3328 rk3328_codec.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -505,7 +505,6 @@ MODULE_DEVICE_TABLE(of, rk3328_codec_of_ static struct platform_driver rk3328_codec_driver = { .driver = { .name = "rk3328-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(rk3328_codec_of_match), }, .probe = rk3328_platform_probe,

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