From eefb8be4a4fb4aa9005fc092a88d66fe7cf1adc2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2013 16:26:15 +0200 Subject: ALSA: hda - Remove analog mic pin override from STAC9228 dell-bios quirk The current fixup for dell-bios model with STAC9228 codec contains the override of pin 0x0c for analog mic. But this is actually just adding a bogus pin and confuses the parser. Better to remove it for the auto-mic switching. Meanwhile, for a possible regression, keep the old configuration as model=dell-bios-amic, so that people can test it again quickly. Tested on Dell 1420n laptop. Reported-and-tested-by: Eric Shattow Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + 1 file changed, 1 insertion(+) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 809d72b8eff1..a46ddb85e83a 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -244,6 +244,7 @@ STAC9227/9228/9229/927x 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + dell-bios-amic Fixes with Dell BIOS setup including analog mic volknob Fixes with volume-knob widget 0x24 auto BIOS setup (default) -- cgit v1.2.3 From da96fb5b0185d27faab0746f872d22b0cee7b026 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2013 16:54:36 +0200 Subject: ALSA: hda - Fix invalid multi-io creation on VAIO-Z laptops VAIO-Z laptops need to use the specific DAC for the speaker output by some unknown reason although the codec itself supports the flexible connection. So we implemented a workaround by a new flag, no_primary_hp, for assigning the speaker pin first. This worked until 3.8 kernel, but it got broken because the driver learned for a better multi-io pin mapping, and not it can assign two mic pins for multi-io. Since the multi-io requires to be the primary output, the hp and two mic pins are assigned in prior to the speaker in the end. Although the machine has two mic pins, one of them is used as a noise- canceling headphone, thus it's no real retaskable mic jack. Thus, at best, we can disable the multi-io assignment and make the parser behavior back to the state before the multi-io. This patch adds again a new flag, no_multi_io, to indicate that the device has no multi-io capability, and set it in the fixup for VAIO-Z. The no_multi_io flag itself can be used generically, added via a helper line, too. Reported-by: Tormen Reported-by: Adam Williamson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 2 ++ sound/pci/hda/hda_generic.c | 14 ++++++++++---- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 4 +++- 4 files changed, 16 insertions(+), 5 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index c3c912d023cc..42a0a39b77e6 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -454,6 +454,8 @@ The generic parser supports the following hints: - need_dac_fix (bool): limits the DACs depending on the channel count - primary_hp (bool): probe headphone jacks as the primary outputs; default true +- multi_io (bool): try probing multi-I/O config (e.g. shared + line-in/surround, mic/clfe jacks) - multi_cap_vol (bool): provide multiple capture volumes - inv_dmic_split (bool): provide split internal mic volume/switch for phase-inverted digital mics diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f5c2d1ff1a09..f6c0344258ac 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "primary_hp"); if (val >= 0) spec->no_primary_hp = !val; + val = snd_hda_get_bool_hint(codec, "multi_io"); + if (val >= 0) + spec->no_multi_io = !val; val = snd_hda_get_bool_hint(codec, "multi_cap_vol"); if (val >= 0) spec->multi_cap_vol = !!val; @@ -1541,7 +1544,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, cfg->speaker_pins, spec->multiout.extra_out_nid, spec->speaker_paths); - if (fill_mio_first && cfg->line_outs == 1 && + if (!spec->no_multi_io && + fill_mio_first && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = fill_multi_ios(codec, cfg->line_out_pins[0], true); if (!err) @@ -1554,7 +1558,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, spec->private_dac_nids, spec->out_paths, spec->main_out_badness); - if (fill_mio_first && + if (!spec->no_multi_io && fill_mio_first && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { /* try to fill multi-io first */ err = fill_multi_ios(codec, cfg->line_out_pins[0], false); @@ -1582,7 +1586,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, return err; badness += err; } - if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + if (!spec->no_multi_io && + cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = fill_multi_ios(codec, cfg->line_out_pins[0], false); if (err < 0) return err; @@ -1600,7 +1605,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, check_aamix_out_path(codec, spec->speaker_paths[0]); } - if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + if (!spec->no_multi_io && + cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2) spec->multi_ios = 1; /* give badness */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index e199a852388b..48d44026705b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -220,6 +220,7 @@ struct hda_gen_spec { unsigned int hp_mic:1; /* Allow HP as a mic-in */ unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */ unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */ + unsigned int no_multi_io:1; /* Don't try multi I/O config */ unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */ unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */ unsigned int own_eapd_ctl:1; /* set EAPD by own function */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 04a69e3fca47..ad7a0985edfe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1845,8 +1845,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.no_primary_hp = 1; + spec->gen.no_multi_io = 1; + } } static const struct hda_fixup alc882_fixups[] = { -- cgit v1.2.3 From eb63231830360f5acfea5dd2b545d7a14476bc3a Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Wed, 14 Aug 2013 12:27:33 +0200 Subject: ASoc: kirkwood: add DT support to the mvebu audio subsystem This patch adds DT support to the audio subsystem of the mvebu family (Kirkwood, Dove, Armada 370). Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mvebu-audio.txt | 29 ++++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 26 ++++++++++++++----- 2 files changed, 49 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mvebu-audio.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 000000000000..7e5fd37c1b3f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) audio controller + +Required properties: + +- compatible: "marvell,mvebu-audio" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "extclk" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-audio"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index e5f3f7a9ea26..7fce340ab3ef 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -22,6 +22,8 @@ #include #include #include +#include + #include "kirkwood.h" #define DRV_NAME "mvebu-audio" @@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); return -EINVAL; } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); return PTR_ERR(priv->clk); @@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id mvebu_audio_of_match[] = { + { .compatible = "marvell,mvebu-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mvebu_audio_of_match), }, }; -- cgit v1.2.3 From 2a956ec04b3703809b6cf500dbee450e44f3a70c Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 28 Aug 2013 12:04:46 +0800 Subject: ASoC: fsl: Add S/PDIF machine driver This patch implements a device-tree-only machine driver for Freescale i.MX series Soc. It works with spdif_transmitter/spdif_receiver and fsl_spdif.c drivers. Signed-off-by: Nicolin Chen Acked-by: Stephen Warren Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 +++++ sound/soc/fsl/Kconfig | 11 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-spdif.c | 148 +++++++++++++++++++++ 4 files changed, 195 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/imx-spdif.c (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 000000000000..7d13479f9c3c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,34 @@ +Freescale i.MX audio complex with S/PDIF transceiver + +Required properties: + + - compatible : "fsl,imx-audio-spdif" + + - model : The user-visible name of this sound complex + + - spdif-controller : The phandle of the i.MX S/PDIF controller + + +Optional properties: + + - spdif-out : This is a boolean property. If present, the transmitting + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF out connector/jack on the board or it's connecting to some + other IP block, such as an HDMI encoder/display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF in connector/jack on the board. + +* Note: At least one of these two properties should be set in the DT binding. + + +Example: + +sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index cd088cc8c866..a70838034600 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_SPDIF + tristate "SoC Audio support for i.MX boards with S/PDIF" + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SPDIF + select SND_SOC_FSL_UTILS + select SND_SOC_SPDIF + help + SoC Audio support for i.MX boards with S/PDIF + Say Y if you want to add support for SoC audio on an i.MX board with + a S/DPDIF. + config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13783 && ARM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4b5970e014dd..e2aaff717f8a 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o +snd-soc-imx-spdif-objs :=imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 000000000000..816013b0ebba --- /dev/null +++ b/sound/soc/fsl/imx-spdif.c @@ -0,0 +1,148 @@ +/* + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include +#include +#include + +struct imx_spdif_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + struct platform_device *txdev; + struct platform_device *rxdev; +}; + +static int imx_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct imx_spdif_data *data; + int ret = 0, num_links = 0; + + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(&pdev->dev, "failed to allocate memory\n"); + ret = -ENOMEM; + goto end; + } + + if (of_property_read_bool(np, "spdif-out")) { + data->dai[num_links].name = "S/PDIF TX"; + data->dai[num_links].stream_name = "S/PDIF PCM Playback"; + data->dai[num_links].codec_dai_name = "dit-hifi"; + data->dai[num_links].codec_name = "spdif-dit"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(data->txdev)) { + ret = PTR_ERR(data->txdev); + dev_err(&pdev->dev, "register dit failed: %d\n", ret); + goto end; + } + } + + if (of_property_read_bool(np, "spdif-in")) { + data->dai[num_links].name = "S/PDIF RX"; + data->dai[num_links].stream_name = "S/PDIF PCM Capture"; + data->dai[num_links].codec_dai_name = "dir-hifi"; + data->dai[num_links].codec_name = "spdif-dir"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(data->rxdev)) { + ret = PTR_ERR(data->rxdev); + dev_err(&pdev->dev, "register dir failed: %d\n", ret); + goto error_dit; + } + } + + if (!num_links) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto error_dir; + } + + data->card.dev = &pdev->dev; + data->card.num_links = num_links; + data->card.dai_link = data->dai; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto error_dir; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto error_dir; + } + + platform_set_drvdata(pdev, data); + + goto end; + +error_dir: + if (data->rxdev) + platform_device_unregister(data->rxdev); +error_dit: + if (data->txdev) + platform_device_unregister(data->txdev); +end: + if (spdif_np) + of_node_put(spdif_np); + + return ret; +} + +static int imx_spdif_audio_remove(struct platform_device *pdev) +{ + struct imx_spdif_data *data = platform_get_drvdata(pdev); + + if (data->rxdev) + platform_device_unregister(data->rxdev); + if (data->txdev) + platform_device_unregister(data->txdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_spdif_dt_ids[] = { + { .compatible = "fsl,imx-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); + +static struct platform_driver imx_spdif_driver = { + .driver = { + .name = "imx-spdif", + .owner = THIS_MODULE, + .of_match_table = imx_spdif_dt_ids, + }, + .probe = imx_spdif_audio_probe, + .remove = imx_spdif_audio_remove, +}; + +module_platform_driver(imx_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-spdif"); -- cgit v1.2.3