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

[alsa-devel] [PATCH v2] sound: omap-twl4030: do not complain on -EPROBE_DEFER when initializing
by Ladislav Michl 14 Mar '18

14 Mar '18

Don't complain on -EPROBE_DEFER when initializing, the driver probe will be retried later. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Changes: -v2: fixed typo in commit log, added Jarkko Nikula's ack sound/soc/omap/omap-twl4030.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index cccc316743fa..7c1f71884768 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -322,8 +322,10 @@ static int omap_twl4030_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", - ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "devm_snd_soc_register_card() failed: %d\n", + ret); return ret; } -- 2.16.1

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[alsa-devel] linux-next: merge conflict between drm-misc and sound trees
by Lukas Wunner 14 Mar '18

14 Mar '18

Dear Stephen, when rebuilding linux-next you're going to see a merge conflict in: sound/pci/hda/hda_intel.c betwen commit: 1ba8f9d30817 ("ALSA: hda: Add a power_save blacklist") from the sound tree and commit: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") from the drm-misc tree. I recommend fixing it up like below. -- >8 -- @@@ -2284,16 -2298,22 +2303,30 @@@ chip->running = 1; azx_add_card_list(chip); + /* + * The discrete GPU cannot power down unless the HDA controller runtime + * suspends, so activate runtime PM on codecs even if power_save == 0. + */ + if (use_vga_switcheroo(hda)) + list_for_each_codec(codec, &chip->bus) + codec->auto_runtime_pm = 1; + - snd_hda_set_power_save(&chip->bus, power_save * 1000); + val = power_save; + #ifdef CONFIG_PM + if (val == -1) { + const struct snd_pci_quirk *q; + + val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } + #endif /* CONFIG_PM */ + snd_hda_set_power_save(&chip->bus, val * 1000); - if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) + if (azx_has_pm_runtime(chip)) pm_runtime_put_autosuspend(&pci->dev); out_free:

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[alsa-devel] [Xen-devel][PATCH 0/2] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 14 Mar '18

14 Mar '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hi, all! Foreword ======== This change is aimed to add support for explicit back and front synchronization during playback and capture in response to comments raised during upstream attempt of the para-virtualized sound frontend driver for Xen [1], [2] and gather opinions from the relevant communities (ALSA, Xen) on the change. The relevant backend is implemented as a user-space application [3] and uses accompanying helper library [4]. Both frontend driver and backend were tested on real HW running Xen hypervisor (Renesas R-Car ARM based H3/M3 boards, x86) to make sure the proposed solution does work. Rationale ========= During the first attempt to upstream the Linux front driver [5] number of comments and concerns were raised, one of the biggest flaws in the design were questioned by both Clemens Ladisch [6] and Takashi Sakamoto [7]: the absence of synchronization between frontend and backend during capture/playback. Two options were discussed: “In design of ALSA PCM core, drivers are expected to synchronize to actual hardwares for semi-realtime data transmission. The synchronization is done by two points: 1) Interrupts to respond events from actual hardwares. 2) Positions of actual data transmission in any serial sound interfaces of actual hardwares. “ and finally a change to the existing protocol was suggested: “In 'include/xen/interface/io/sndif.h', there's no functionalities I described the above: 1. notifications from DomU to Dom0 about the size of period for interrupts from actual hardwares. Or no way from Dom0 to DomU about the configured size of the period. 2. notifications of the interrupts from actual hardwares to DomU.” This is implemented as a change to the sndif protocol and allows removing period emulation: 1. Introduced a new event channel from back to front 2. New event with number of bytes played/captured (XENSND_EVT_CUR_POS, to be used for sending snd_pcm_period_elapsed at frontend (in Linux implementation). Sent in bytes, not frames to make the protocol generic and consistent) 3. New request for playback/capture control (XENSND_OP_TRIGGER) with start/pause/stop/resume sub-ops 4. Playback/capture buffer size is set on the backend side via XENSND_FIELD_BUFFER_SIZE XenStore entry Waiting for your valuable comments, Thank you, Oleksandr [1] https://github.com/andr2000/linux/commits/snd_upstream_v1 [2] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/inc… [3] https://github.com/xen-troops/snd_be [4] https://github.com/xen-troops/libxenbe [5] https://lkml.org/lkml/2017/8/7/363 [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/123617.html [7] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/123744.html Oleksandr Andrushchenko (2): sndif: introduce protocol version sndif: add explicit back and front synchronization xen/include/public/io/sndif.h | 173 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 170 insertions(+), 3 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH v2 0/7] Modernize vga_switcheroo by using device link for HDA
by Lukas Wunner 14 Mar '18

14 Mar '18

On Tue, Mar 06, 2018 at 11:29:40AM +0100, Daniel Vetter wrote: > On Sat, Mar 03, 2018 at 10:53:24AM +0100, Lukas Wunner wrote: > > Modernize vga_switcheroo by using a device link to enforce a runtime PM > > dependency from an HDA controller to the GPU it's integrated into, v2. > > > > https://github.com/l1k/linux/commits/switcheroo_devlink_v2 > > This all looks really reasonable and like a good cleanup, but it's a bit > too much detail so I'll punt review to someone else with more clue. Patches [3/7] to [7/7] were reviewed by Peter Wu, the HDA bits in patch [5/7] additionally by Takashi. Patch [2/7] was acked by Bjorn. There was no ack for patch [1/7] (authored by Rafael), but it adressed the objection Bjorn raised against my original patch, so I'm assuming Bjorn is okay with it. (Bjorn, please let me know if that isn't the case.) The series has been tested on 5 systems, which raises the confidence: 2x AMD PowerXpress (Mike Lothian, Kai Heng Feng) 2x Nvidia Optimus (Denis Lisov, Peter Wu) 1x MacBook Pro The issues found during Peter Wu's thorough testing appear to all be unrelated to this series, as per my e-mail yesterday. If there are no objections, I plan to push the series to drm-misc-next by the middle of the coming week so that it would still catch the last train to 4.17. Thanks, Lukas

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[alsa-devel] [PATCH - alsa-utils 1/1] aplay: Fix wav file not being split on 32 bit platforms
by erwin 13 Mar '18

13 Mar '18

On my 32 bit armhf board arecord exits because of write() returning EFBIG when the output file size reaches 2147483647 bytes. To fix this, include generated header file before system header files so that _FILE_OFFSET_BITS=64 is used properly, as required in documentation "man feature_test_macros". Signed-off-by: erwin <nebelbank(a)posteo.de> diff --git a/aplay/aplay.c b/aplay/aplay.c index 6b740c2..bbd7fff 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -27,6 +27,7 @@ */ #define _GNU_SOURCE +#include "aconfig.h" #include <stdio.h> #include <malloc.h> #include <unistd.h> @@ -49,7 +50,6 @@ #include <sys/stat.h> #include <sys/types.h> #include <endian.h> -#include "aconfig.h" #include "gettext.h" #include "formats.h" #include "version.h" -- 2.14.1

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[alsa-devel] [asoc:topic/intel 22/38] sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type
by kbuild test robot 13 Mar '18

13 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: fc9fdd61c4809b14faa9b84fe3d8f4167a836326 commit: 3bf045d15f52529e8abe26543dbe22af44efb41a [22/38] ASoC: Intel: fix argument error in nau8824 machine config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 3bf045d15f52529e8abe26543dbe22af44efb41a # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/intel/boards/cht_bsw_nau8824.c: In function 'cht_codec_init': >> sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type [-Werror=incompatible-pointer-types] nau8824_enable_jack_detect(component, jack); ^~~~~~~~~ In file included from sound/soc/intel/boards/cht_bsw_nau8824.c:31:0: sound/soc/intel/boards/../../codecs/nau8824.h:474:5: note: expected 'struct snd_soc_codec *' but argument is of type 'struct snd_soc_component *' int nau8824_enable_jack_detect(struct snd_soc_codec *codec, ^~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/nau8824_enable_jack_detect +137 sound/soc/intel/boards/cht_bsw_nau8824.c 101 102 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) 103 { 104 struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); 105 struct snd_soc_jack *jack = &ctx->jack; 106 struct snd_soc_dai *codec_dai = runtime->codec_dai; 107 struct snd_soc_component *component = codec_dai->component; 108 int ret, jack_type; 109 110 /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ 111 ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); 112 if (ret < 0) { 113 dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); 114 return ret; 115 } 116 117 /* NAU88L24 supports 4 butons headset detection 118 * KEY_MEDIA 119 * KEY_VOICECOMMAND 120 * KEY_VOLUMEUP 121 * KEY_VOLUMEDOWN 122 */ 123 jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | 124 SND_JACK_BTN_2 | SND_JACK_BTN_3; 125 ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, 126 cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); 127 if (ret) { 128 dev_err(runtime->dev, 129 "Headset Jack creation failed %d\n", ret); 130 return ret; 131 } 132 snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); 133 snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); 134 snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); 135 snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); 136 > 137 nau8824_enable_jack_detect(component, jack); 138 139 return ret; 140 } 141 --- 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 v2] ASoC: dapm: add support to pinctrl dapm
by srinivas.kandagatla＠linaro.org 13 Mar '18

13 Mar '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Purpose of having pinctrl dapm is to dynamically put the pins in low power state when they are not actively used by the audio and saving power. Without this each driver has to set the pinctrl states, either during probe or dynamically depending on the callbacks received from ASoC core. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - removed synchronize with the DAPM async queue include/sound/soc-dapm.h | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 58 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 344b96c206a3..a6ce2de4e20a 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -269,6 +269,13 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ .on_val = wflags} +#define SND_SOC_DAPM_PINCTRL(wname, active, sleep) \ +{ .id = snd_soc_dapm_pinctrl, .name = wname, \ + .priv = (&(struct snd_soc_dapm_pinctrl_priv) \ + { .active_state = active, .sleep_state = sleep,}), \ + .reg = SND_SOC_NOPM, .event = dapm_pinctrl_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ @@ -374,6 +381,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_pinctrl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -500,6 +509,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_pinctrl, /* pinctrl */ snd_soc_dapm_regulator_supply, /* external regulator */ snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ @@ -581,6 +591,7 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ + struct pinctrl *pinctrl; /* attached pinctrl */ const struct snd_soc_pcm_stream *params; /* params for dai links */ unsigned int num_params; /* number of params for dai links */ unsigned int params_select; /* currently selected param for dai link */ @@ -683,6 +694,11 @@ struct snd_soc_dapm_stats { int neighbour_checks; }; +struct snd_soc_dapm_pinctrl_priv { + const char *active_state; + const char *sleep_state; +}; + /** * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level * @dapm: The DAPM context to initialize diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 40085a4c8f92..ac4f13230faf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/pinctrl/consumer.h> #include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> @@ -72,6 +73,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_pinctrl] = 1, [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, @@ -119,6 +121,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_supply] = 12, [snd_soc_dapm_clock_supply] = 13, + [snd_soc_dapm_pinctrl] = 13, [snd_soc_dapm_regulator_supply] = 13, [snd_soc_dapm_post] = 14, }; @@ -1287,6 +1290,31 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for pinctrl widget. + */ +int dapm_pinctrl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_dapm_pinctrl_priv *priv = w->priv; + struct pinctrl *p = w->pinctrl; + struct pinctrl_state *s; + + if (!p || !priv) + return -EIO; + + if (SND_SOC_DAPM_EVENT_ON(event)) + s = pinctrl_lookup_state(p, priv->active_state); + else + s = pinctrl_lookup_state(p, priv->sleep_state); + + if (IS_ERR(s)) + return PTR_ERR(s); + + return pinctrl_select_state(p, s); +} +EXPORT_SYMBOL_GPL(dapm_pinctrl_event); + /* * Handler for clock supply widget. */ @@ -1900,6 +1928,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) @@ -2313,6 +2342,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", @@ -3462,6 +3492,17 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name, ret); } break; + case snd_soc_dapm_pinctrl: + w->pinctrl = devm_pinctrl_get(dapm->dev); + if (IS_ERR_OR_NULL(w->pinctrl)) { + ret = PTR_ERR(w->pinctrl); + if (ret == -EPROBE_DEFER) + return ERR_PTR(ret); + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + return NULL; + } + break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); @@ -3541,6 +3582,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: case snd_soc_dapm_kcontrol: w->is_supply = 1; -- 2.15.1

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[alsa-devel] [PATCH] ASoC: da7219: clkdev_drop usage depends on CONFIG_COMMON_CLK
by Adam Thomson 13 Mar '18

13 Mar '18

Fixes: ASoC: da7219: Add common clock usage for providing DAI clks clkdev_drop usage in the codec remove function should be dependent on if CONFIG_COMMON_CLK is defined for the platform, otherwise it can cause build failures for platforms that do not support this. The clkdev_* functions are still defined for those platforms, in headers and source but the functions are not linked in. This patch resolves this issue, so clkdev_drop is only used if CONFIG_COMMON_CLK is defined. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 4412159..08f55d7 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1966,8 +1966,10 @@ static void da7219_remove(struct snd_soc_component *component) da7219_aad_exit(component); +#ifdef CONFIG_COMMON_CLK if (da7219->dai_clks_lookup) clkdev_drop(da7219->dai_clks_lookup); +#endif /* Supplies */ regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); -- 1.9.1

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[alsa-devel] [PATCH 4/6] ASoC: Snow Chromebook DT bindings update
by Sylwester Nawrocki 13 Mar '18

13 Mar '18

The new DT bindings are introduced so sound on the HDMI interface is also supported. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Documentation/devicetree/bindings/sound/snow.txt | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 6df74f15687f..80fd9a87bb3f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -5,8 +5,17 @@ Required properties: "google,snow-audio-max98090" or "google,snow-audio-max98091" or "google,snow-audio-max98095" -- samsung,i2s-controller: The phandle of the Samsung I2S controller -- samsung,audio-codec: The phandle of the audio codec +- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller +- samsung,audio-codec (deprecated): The phandle of the audio codec + +Required sub-nodes: + + - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S + controller + - 'codec' subnode with a 'sound-dai' property containing list of phandles + to the CODEC nodes, first entry must be the phandle of the MAX98090, + MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible + string) and the second entry must be the phandle of the HDMI IP block node Optional: - samsung,model: The name of the sound-card -- 2.14.2

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[alsa-devel] [PATCH v3 1/2] ASoC: samsung: Snow DT bindings update
by Sylwester Nawrocki 13 Mar '18

13 Mar '18

The new DT bindings are introduced so sound on the HDMI interface can also be supported. The exisiting CPU and CODEC DAI phandle properties are marked as deprecated. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes since v2: - edited summary line. --- Documentation/devicetree/bindings/sound/snow.txt | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 6df74f15687f..80fd9a87bb3f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -5,8 +5,17 @@ Required properties: "google,snow-audio-max98090" or "google,snow-audio-max98091" or "google,snow-audio-max98095" -- samsung,i2s-controller: The phandle of the Samsung I2S controller -- samsung,audio-codec: The phandle of the audio codec +- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller +- samsung,audio-codec (deprecated): The phandle of the audio codec + +Required sub-nodes: + + - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S + controller + - 'codec' subnode with a 'sound-dai' property containing list of phandles + to the CODEC nodes, first entry must be the phandle of the MAX98090, + MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible + string) and the second entry must be the phandle of the HDMI IP block node Optional: - samsung,model: The name of the sound-card -- 2.14.2

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