March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: snappercl15: Convert to table based DAPM setup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/cirrus/snappercl15.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 29238a7..bdc88e5 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -65,18 +65,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; -} - static struct snd_soc_dai_link snappercl15_dai = { .name = "tlv320aic23", .stream_name = "AIC23", @@ -84,7 +72,6 @@ static struct snd_soc_dai_link snappercl15_dai = { .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", .platform_name = "ep93xx-i2s", - .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, .ops = &snappercl15_ops, @@ -95,6 +82,11 @@ static struct snd_soc_card snd_soc_snappercl15 = { .owner = THIS_MODULE, .dai_link = &snappercl15_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int snappercl15_probe(struct platform_device *pdev) -- 1.8.0

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[alsa-devel] [PATCH] sound: omap: n810: fix init with DT boot
by Aaro Koskinen 03 Mar '14

03 Mar '14

Since 3.14-rc1 only DT boot has been supported on N810. Make a minimal fix to retain functionality. This file should be properly converted to DT in longer term. Signed-off-by: Aaro Koskinen <aaro.koskinen(a)iki.fi> --- sound/soc/omap/n810.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 3fde9e4..d163e18 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -305,7 +305,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); -- 1.8.3.3

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[alsa-devel] Enable headset Mic for Dell DT
by Kailang 03 Mar '14

03 Mar '14

Hi Takashi, I created patch below for Dell DT for support headset MIC. Many Thanks. Kailang. >From 2f448d6f9ad7023befa91b80f86cb267311d3f8a Mon Sep 17 00:00:00 2001 From: Kailang Yang <kailang(a)realtek.com> Date: Mon, 3 Mar 2014 15:27:22 +0800 Subject: [PATCH] ALSA: hda/realtek - Add headset quirk for Dell DT This quirk is needed for the headset microphone to work. Signed-off-by: Kailang Yang <kailang(a)realtek.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 534796e..74456bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3594,21 +3594,38 @@ static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc255_set_default_jack_type(struct hda_codec *codec) +{ + /* Set to iphone type */ + alc_write_coef_idx(codec, 0x1b, 0x880b); + alc_write_coef_idx(codec, 0x45, 0xd089); + alc_write_coef_idx(codec, 0x1b, 0x080b); + alc_write_coef_idx(codec, 0x46, 0x0004); + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + msleep(30); +} + static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, const struct hda_fixup *fix, int action) { if (action == HDA_FIXUP_ACT_PRE_PROBE) { - /* Set to iphone type */ - alc_write_coef_idx(codec, 0x1b, 0x880b); - alc_write_coef_idx(codec, 0x45, 0xd089); - alc_write_coef_idx(codec, 0x1b, 0x080b); - alc_write_coef_idx(codec, 0x46, 0x0004); - alc_write_coef_idx(codec, 0x1b, 0x0c0b); - msleep(30); + alc255_set_default_jack_type(codec); } alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + alc255_set_default_jack_type(codec); + } + else + alc_fixup_headset_mode(codec, fix, action); +} + static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3875,7 +3892,9 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, + ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -4248,10 +4267,23 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_DELL2_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, }, + [ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc255_no_hp_mic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4314,6 +4346,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),

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[alsa-devel] [PATCH] ALSA: hda - Added inverted digital-mic handling for Acer TravelMate 8371
by Marius Knaust 03 Mar '14

03 Mar '14

Signed-off-by: Marius Knaust <marius.knaust(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 534796e..9fc7bb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4255,6 +4255,7 @@ static const struct hda_fixup alc269_fixups[] = { }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0283, "Acer TravelMate 8371", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),

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[alsa-devel] [PATCH 1/3] ALSA: hda - Make azx_attach_pcm_stream static
by Dylan Reid 03 Mar '14

03 Mar '14

It is only used in hda_controller.c now. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_controller.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6156d0a..97993e1 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -971,8 +971,8 @@ static void azx_pcm_free(struct snd_pcm *pcm) #define MAX_PREALLOC_SIZE (32 * 1024 * 1024) -int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, - struct hda_pcm *cpcm) +static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, + struct hda_pcm *cpcm) { struct azx *chip = bus->private_data; struct snd_pcm *pcm; -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH] ASoC: wm1133-ev1: Convert to table based DAPM setup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/wm1133-ev1.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index fce6325..804749a 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -214,12 +214,6 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets, - ARRAY_SIZE(wm1133_ev1_widgets)); - - snd_soc_dapm_add_routes(dapm, wm1133_ev1_map, - ARRAY_SIZE(wm1133_ev1_map)); - /* Headphone jack detection */ snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), @@ -257,6 +251,11 @@ static struct snd_soc_card wm1133_ev1 = { .owner = THIS_MODULE, .dai_link = &wm1133_ev1_dai, .num_links = 1, + + .dapm_widgets = wm1133_ev1_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets), + .dapm_routes = wm1133_ev1_map, + .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map), }; static struct platform_device *wm1133_ev1_snd_device; -- 1.8.0

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Re: [alsa-devel] 7.1 surround not working on ALC892/ Ubuntu 13.04
by Raymond Yau 03 Mar '14

03 Mar '14

> I am unable to get 7.1 surround sound option on my system. > Alsamixer is not giving me any option to choose 8 channels. > Surround 5.1 is working correctly. > I am not sure how to configure the system to get the 8 channel setup working. > >Distributor ID: Ubuntu >Description: Ubuntu 13.04 >Release: 13.04 >Codename: raring >Attaching all the sound-card information I could gather from the system. >Please let me know in case any additional information is needed to debug the issue. > user1 <at> user1-MI970VF:~/ufouser$ cat /proc/asound/card0/codec#* | grep Codec > Codec: Realtek ALC892 For motherboard with 3 audio jacks at rear panel and 10 channels HDA codec you need to retask headphone at front panel as side line out the easy way is to use hdajackretask (advanced override) or early patching to retask if green line out at rear panel has sequence 1 in pin default. 1) blue line jack as Line Out with sequence 2 2) pink mic jack as Line Out with sequence 3 3) front panel headphone to Line out with sequence 4 the three retasked line out should have same default association as the green line out in theory, it is possible to add the retask of headphone to line out into the existing channel mode (8ch) when the codec has extra DAC and volume control using multi_Ios pin butt need to hack the jack detection control, disable auto mute and independent headphone when the headphone is retasked

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[alsa-devel] [PATCHv2 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 03 Mar '14

03 Mar '14

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[alsa-devel] [PATCH 1/2] ASoC: dapm: Power off all widgets in the snd_soc_dapm_shutdown
by Xiang Xiao 03 Mar '14

03 Mar '14

The widgets generated by the machine driver need to power off too. Signed-off-by: Xiang Xiao <xiaoxiang(a)xiaomi.com> --- sound/soc/soc-dapm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3651a37..00ed0e9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3899,7 +3899,7 @@ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) +static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) { struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; @@ -3939,12 +3939,12 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) */ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { - struct snd_soc_codec *codec; + struct snd_soc_dapm_context *dapm; - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - soc_dapm_shutdown_codec(&codec->dapm); - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) - snd_soc_dapm_set_bias_level(&codec->dapm, + list_for_each_entry(dapm, &card->dapm_list, list) { + soc_dapm_shutdown_dapm(dapm); + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_OFF); } } -- 1.8.1.2

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[alsa-devel] [PATCHv9][ 1/2] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 03 Mar '14

03 Mar '14

Cc: Eric Bénard <eric(a)eukrea.com> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v9->v10: - Better Cc list - Some non-arm architecture that have CONFIG_OF could select that driver, but they lacked asm/mach-types.h, so the compilation failed. That has been fixed by permitting, when CONFIG_OF is selected, to select the driver only if CONFIG_ARM is set. ChangeLog v7->v8: - fixed the vendor prefixes in the dts bindings. - updated the code and the documentation accordingly. - removed the need of an audio-codec property, since it's available in the ssi bindings. - the dai codec is now retrived trough the dt in order to support machines that have its codec on another i2c bus number. - driver compatible is now similar to the ones of the other drivers. ChangeLog v6->v7: - The cleanups went into another patch. - The support for fsl_ssi doesn't depend on dt anymore. - platform_name = "imx-ssi.0" is still set when in non-dt mode. --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 108 +++++++++++++++++--- 3 files changed, 121 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..0d7985c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,21 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,asoc-tlv320" +- eukrea,model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- fsl,mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,asoc-tlv320"; + eukrea,model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,mux-int-port = <2>; + fsl,mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f14..b8f8703 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,12 +168,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || (OF && ARM) depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 5983740..eb093d5 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -15,8 +15,11 @@ * */ +#include <linux/errno.h> #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,6 +29,7 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 @@ -41,7 +45,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) { + /* fsl_ssi lacks the set_fmt ops. */ + if (ret && ret != -ENOTSUPP) { dev_err(cpu_dai->dev, "Failed to set the cpu dai format.\n"); return ret; @@ -63,11 +68,13 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, "Failed to set the codec sysclk.\n"); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); - if (ret) { + /* fsl_ssi lacks the set_sysclk ops */ + if (ret && ret != -EINVAL) { dev_err(cpu_dai->dev, "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; @@ -84,14 +91,10 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", - .codec_name = "tlv320aic23-codec.0-001a", - .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, }; static struct snd_soc_card eukrea_tlv320 = { - .name = "cpuimx-audio", .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, @@ -101,8 +104,65 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; - if (machine_is_eukrea_cpuimx27()) { + eukrea_tlv320.dev = &pdev->dev; + if (np) { + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, + "eukrea,model"); + if (ret) { + dev_err(&pdev->dev, + "eukrea,model node missing or invalid.\n"); + goto err; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + if (!ssi_np) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); + if (codec_np) + eukrea_tlv320_dai.codec_of_node = codec_np; + else + dev_err(&pdev->dev, "codec-handle node missing or invalid.\n"); + + ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-int-port node missing or invalid.\n"); + return ret; + } + ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-ext-port node missing or invalid.\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_of_node = ssi_np; + eukrea_tlv320_dai.platform_of_node = ssi_np; + } else { + eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0"; + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a"; + eukrea_tlv320.name = "cpuimx-audio"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -119,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -134,14 +198,27 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,asoc-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing or invalid audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } - eukrea_tlv320.dev = &pdev->dev; ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -153,10 +230,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,asoc-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, .remove = eukrea_tlv320_remove, -- 1.7.9.5

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