April 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH v2 0/5] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 21 Apr '16

21 Apr '16

Refactor and go v2 because of the 60KB limitation This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. The interface to connect with BT chip is not finished yet, but as an upstream rookie I really need some comments. Any comments will be greatly appreciated. Garlic Tseng (5): ASoC: mediatek: Refine mt8173 driver ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add clock and irq control for 2701 platform driver ASoC: mediatek: add mt2701 platform driver implementation. ASoC: mediatek: Add mt2701-cs42448 driver and config option. .../devicetree/bindings/sound/mt2701-cs42448.txt | 39 + .../devicetree/bindings/sound/mtk2701-afe-pcm.txt | 150 ++ sound/soc/mediatek/Kconfig | 21 + sound/soc/mediatek/Makefile | 15 +- sound/soc/mediatek/mt2701/Makefile | 20 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 376 +++++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c | 259 +++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h | 28 + sound/soc/mediatek/mt2701/mtk2701-afe-common.h | 217 +++ sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c | 1752 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mtk2701-irq.c | 109 ++ sound/soc/mediatek/mt2701/mtk2701-irq.h | 30 + sound/soc/mediatek/mt2701/mtk2701-reg.h | 165 ++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/{ => mt8173}/mtk-afe-common.h | 0 sound/soc/mediatek/{ => mt8173}/mtk-afe-pcm.c | 0 20 files changed, 3186 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 Documentation/devicetree/bindings/sound/mtk2701-afe-pcm.txt create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) rename sound/soc/mediatek/{ => mt8173}/mtk-afe-common.h (100%) rename sound/soc/mediatek/{ => mt8173}/mtk-afe-pcm.c (100%) -- 1.9.1

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[alsa-devel] [PATCH v7] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 20 Apr '16

20 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v6: Use subnode to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 ++++ sound/soc/mediatek/mt8173-rt5650.c | 49 ++++++++++++++++++-- 2 files changed, 56 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..5bfa6b6 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,21 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + codec-capture { + sound-dai = <&rt5650 1>; + }; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..b8ed965 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,28 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_err(card->dev, "Can't get the right codec dai\n"); + return -EINVAL; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +126,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +170,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +198,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +222,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: ak4642: Remove CLK_IS_ROOT
by Stephen Boyd 20 Apr '16

20 Apr '16

This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> --- sound/soc/codecs/ak4642.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index cda27c22812a..1ee8506c06c7 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -608,9 +608,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev) of_property_read_string(np, "clock-output-names", &clk_name); - clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, - (parent_clk_name) ? 0 : CLK_IS_ROOT, - rate); + clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate); if (!IS_ERR(clk)) of_clk_add_provider(np, of_clk_src_simple_get, clk); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: rsnd: Remove CLK_IS_ROOT
by Stephen Boyd 20 Apr '16

20 Apr '16

This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> --- sound/soc/sh/rcar/adg.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 606399de684d..49354d17ea55 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, req_rate); + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, else { for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->onecell.clks = adg->clkout; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH 0/4] ASoC: add graph base connection on simple-card
by Kuninori Morimoto 20 Apr '16

20 Apr '16

Hi Mark Current simple-card is using "sound-dai" base connection on DT, but V4L2 is using graph base connection. For example HDMI case, we would like to use both connection. To above confusable connection method, and to reuse current resource, this patch adds new function, and detect both "sound-dai" and "remote-endpoint" on simple-card. like this sound { compatible = "simple-audio-card"; ... sndcpu: simple-audio-card,cpu { sound-dai = <&xxx>; }; sndcodec: simple-audio-card,codec { => remote-endpoint = <&out_hdmi>; }; }; xxx { ... ports { ... port@0 { reg = <0>; out_rgb: endpoint { }; }; port@1 { reg = <1>; => out_hdmi: endpoint { }; }; ... }; }; Kuninori Morimoto (4): of_graph: add of_graph_get_endpoint_count() ASoC: add of_parse_snd_connection_with_args() for sound-dai/graph connection ASoC: snd_soc_of_get_dai_name() uses of_parse_snd_soc_connection_with_args() ASoC: simple-card: probe both sound-dai and remote-endpoint Documentation/devicetree/bindings/sound/simple-card.txt | 19 +++++++++++++++++++ include/linux/of_graph.h | 11 +++++++++++ include/sound/soc.h | 2 ++ sound/soc/generic/simple-card.c | 9 ++++++--- sound/soc/soc-core.c | 45 +++++++++++++++++++++++++++++++++++++++++++-- 5 files changed, 81 insertions(+), 5 deletions(-)

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Re: [alsa-devel] Issues with ALSA:ice1724 sound cards on powerpc machines
by Julian Margetson 20 Apr '16

20 Apr '16

On 2/17/2016 2:29 PM, Julian Margetson wrote: > Hi > > Having an issue issue with ice1724 sound cards on powerpc motherboards. > Motherboards in question Acube Sam460ex . > The card appears as a valid device but there is no sound output on the > majority of boots of the computer. > Logs attached. > On the first set boot audio playback produced sound ok. > On the second set boot no sound was produce with audio playback > > > Regards > Julian Margetson > > > Anyone have any ideas ? Resubmitting again with links to logs instead of included as attachments. The issue also seen on A-Eon Cyrus P5020 motherboard . https://www.dropbox.com/s/72labcje9z1r1b7/1alsa-info.txt.7CLHMG0Gap?dl=0 https://www.dropbox.com/s/v1pn5kjqrvwu16f/boot_1.log?dl=0 https://www.dropbox.com/s/uuaa14rkpmmz1o0/2alsa-info.txt.VKiEVuytiJ?dl=0 https://www.dropbox.com/s/wwu2f5t93l8g26d/boot_2.log?dl=0 Thanks

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[alsa-devel] GPIO control from ALSA
by Petr Kulhavy 20 Apr '16

20 Apr '16

Hello, I'm writing a driver for ASoC "sound card" hardware which uses CPU GPIO to mute/unmute an analogue amplifier and another GPIO to control an analogue switch for line/mic switching. Is there an abstraction in ALSA to model this kind of hardware? Thanks Petr

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[alsa-devel] [PATCH 1/1] ALSA: hda - add PCI ID for Intel Broxton-T
by han.lu＠intel.com 20 Apr '16

20 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Add HD Audio Device PCI ID for the Intel Broxton-T platform. It is an HDA Intel PCH controller. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b680be0..637b8a0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2232,6 +2232,9 @@ static const struct pci_device_id azx_ids[] = { /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, + /* Broxton-T */ + { PCI_DEVICE(0x8086, 0x1a98), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, -- 2.5.0

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[alsa-devel] [asoc:topic/mtk 1/1] sound/soc/mediatek/mt8173-rt5650.c:236:3: error: implicit declaration of function 'snd_soc_get_dai_name'
by kbuild test robot 20 Apr '16

20 Apr '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/mtk head: 0beeb8d0b434b8b007ac8058fb2cac5997fff6df commit: 0beeb8d0b434b8b007ac8058fb2cac5997fff6df [1/1] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver config: arm-allmodconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 0beeb8d0b434b8b007ac8058fb2cac5997fff6df # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc/mediatek/mt8173-rt5650.c: In function 'mt8173_rt5650_dev_probe': >> sound/soc/mediatek/mt8173-rt5650.c:236:3: error: implicit declaration of function 'snd_soc_get_dai_name' [-Werror=implicit-function-declaration] ret = snd_soc_get_dai_name(&args, &codec_dai_capture); ^ cc1: some warnings being treated as errors vim +/snd_soc_get_dai_name +236 sound/soc/mediatek/mt8173-rt5650.c 230 dev_err(&pdev->dev, 231 "%s capture-dai name fail %d\n", 232 __func__, ret); 233 return ret; 234 } 235 > 236 ret = snd_soc_get_dai_name(&args, &codec_dai_capture); 237 mt8173_rt5650_codecs[1].dai_name = codec_dai_capture; 238 } 239 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ALSA: hda - Keep powering up ADCs on Cirrus codecs
by Takashi Iwai 19 Apr '16

19 Apr '16

Although one weird behavior about the input path (inconsistent D0/D3 switch) on Cirrus CS420x codecs was fixed in the previous commit, there is still an issue on some Mac machines: the capture stream stalls when switching the ADCs on the fly. More badly, this keeps stuck until the next reboot. The dynamic ADC switching is already a bit fragile and assuming optimistically that the chip accepts the frequent power changes. On Cirrus codecs, this doesn't seem applicable. As a quick workaround, we pin down the ADCs to keep up in D0 when spec->dyn_adc_switch is set. In this way, the ADCs are kept up only for the system that were confirmed to be broken. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116171 Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a47e8ae0eb30..80bbadc83721 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -361,6 +361,7 @@ static int cs_parse_auto_config(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; int err; + int i; err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); if (err < 0) @@ -370,6 +371,19 @@ static int cs_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + /* keep the ADCs powered up when it's dynamically switchable */ + if (spec->gen.dyn_adc_switch) { + unsigned int done = 0; + for (i = 0; i < spec->gen.input_mux.num_items; i++) { + int idx = spec->gen.dyn_adc_idx[i]; + if (done & (1 << idx)) + continue; + snd_hda_gen_fix_pin_power(codec, + spec->gen.adc_nids[idx]); + done |= 1 << idx; + } + } + return 0; } -- 2.8.1

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