December 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Intermittent cut out on the video
by Dolevo Jay 06 Dec '13

06 Dec '13

From: cmst(a)live.com To: clemens(a)ladisch.de Subject: RE: [alsa-devel] Intermittent cut out on the video Date: Mon, 2 Dec 2013 09:44:16 +0000 > Date: Thu, 28 Nov 2013 09:29:45 +0100 > From: clemens(a)ladisch.de > To: cmst(a)live.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Intermittent cut out on the video > > Dolevo Jay wrote: > > I have two sound cards on my system and I use the [multi plugin] to > > have audio from two jacks simultaneously. > > > > The audio coming from the jack on the pc is cutting out intermittently. > > It just cuts out for a fraction of a second and continues as normal > > and after some time it happens again. > > The two cards use different sample clocks, so it is not possible to feed > them reliably from the same stream. > > You would need software that measures that actual sample rate of both > devices and resamples its output appropriately. Try using PulseAudio. > > > Regards, > Clemens > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Hi, I have tried to apply pulseaudio but couldn't really manage it. Could you please tell me if I can fix it in the configuration file? I mean, for example, would it be solved if I resample the audio for the appropriate sound card in the asound.conf? What other option do I have other than pulseaudio? Thanks a lot.

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[alsa-devel] [PATCH] ALSA: hda - Allow capture-only configuration
by Takashi Iwai 06 Dec '13

06 Dec '13

We have blindly assumed that all valid configurations should have either analog or digital playback, but there can be capture-only configurations. The parser shouldn't escape in such a case. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2729e881541d..5ba3106b9712 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4334,7 +4334,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, spec->no_analog = 1; goto dig_only; } - return 0; /* can't find valid BIOS pin config */ + if (!cfg->num_inputs && !cfg->dig_in_pin) + return 0; /* can't find valid BIOS pin config */ } if (!spec->no_primary_hp && -- 1.8.5

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[alsa-devel] [PATCH] ALSA: hda - skip depop delay before D3 for Haswell and Valleyview2 display codec
by mengdong.lin＠intel.com 06 Dec '13

06 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch skips the default depop delay before D3 for Haswell (10 ms) and Valleyview2 (100 ms) display codec, to reduce codec suspend time. The analog part of display audio is implemented in the external display. Some displays have weak pop noise while others not when suspending, no matter there is the default delay or not. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2306645..e14359a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2022,6 +2022,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } + if (is_haswell(codec) || is_valleyview(codec)) { + codec->depop_delay = 0; + } + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.8.1.2

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[alsa-devel] [PATCH] ALSA: hda - Remove quirk for Dell Vostro 131
by David Henningsson 06 Dec '13

06 Dec '13

I've tested the old Dell Vostro 131 with the latest generic parser and it works just fine, and as a bonus we get better jack detection features in userspace. Therefore this quirk can be removed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1f2717f..3fbf288 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - do delayed suspend for HD-audio devices
by mengdong.lin＠intel.com 06 Dec '13

06 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> HD-Audio codecs tends to take a long time to supend and is time-consuming part in driver suspend. In this patch, codec suspending and the left operations are delayed to a worker thead, so that they can be done asynchronously in parallel with suspending of other devices. And azx_suspend_late() is defined as PM .suspend_late ops to wait for completion of the delayed suspending. Credit goes to Liam Girdwood, who suggested this workaround. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14..aed4cdf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -545,6 +545,11 @@ struct azx { struct work_struct probe_work; +#ifdef CONFIG_PM + struct workqueue_struct *pm_wq; /* workqueue to delay controller PM */ + struct work_struct suspend_work; /* task to delay controller suspend */ +#endif + /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; @@ -2908,20 +2913,11 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) /* * power management */ -static int azx_suspend(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct azx_pcm *p; - if (chip->disabled) - return 0; +static int azx_suspend_continue(struct azx *chip) +{ + struct pci_dev *pci = chip->pci; - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - azx_clear_irq_pending(chip); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); @@ -2937,6 +2933,41 @@ static int azx_suspend(struct device *dev) pci_set_power_state(pci, PCI_D3hot); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); + + return 0; +} + +static void azx_suspend_work(struct work_struct *work) +{ + azx_suspend_continue(container_of(work, struct azx, suspend_work)); +} + + +static int azx_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + if (chip->disabled) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + + queue_work(chip->pm_wq, &chip->suspend_work); + return 0; +} + +static int azx_suspend_late(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + flush_workqueue(chip->pm_wq); + return 0; } @@ -3057,6 +3088,7 @@ static int azx_runtime_idle(struct device *dev) #ifdef CONFIG_PM static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + .suspend_late = azx_suspend_late, SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; @@ -3272,6 +3304,12 @@ static int azx_free(struct azx *chip) hda_display_power(false); hda_i915_exit(); } + +#ifdef CONFIG_PM + if (chip->pm_wq) + destroy_workqueue(chip->pm_wq); +#endif + kfree(chip); return 0; @@ -3515,6 +3553,9 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, }; struct azx *chip; int err; +#ifdef CONFIG_PM + char wqname[24]; +#endif *rchip = NULL; @@ -3581,6 +3622,18 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, /* continue probing in work context as may trigger request module */ INIT_WORK(&chip->probe_work, azx_probe_work); +#ifdef CONFIG_PM + sprintf(wqname, "hda-azx-pm-wq%d", card->number); + chip->pm_wq = create_workqueue("wqname"); + if (!chip->pm_wq) { + snd_printk(KERN_ERR "cannot create PM workqueue\n"); + azx_free(chip); + return -ENOMEM; + } + + INIT_WORK(&chip->suspend_work, azx_suspend_work); +#endif + *rchip = chip; return 0; -- 1.8.1.2

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Re: [alsa-devel] [PATCH] usb-audio: fix warning
by Takashi Iwai 05 Dec '13

05 Dec '13

At Thu, 5 Dec 2013 14:32:43 -0500 (EST), Mikulas Patocka wrote: > > Fix the following warning when optimizing for size with gcc-4.6.4: > sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized] > > Signed-off-by: Mikulas Patocka <mpatocka(a)redhat.com> Thanks, applied. Takashi > > --- > sound/usb/mixer_quirks.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > Index: linux-3.12.3/sound/usb/mixer_quirks.c > =================================================================== > --- linux-3.12.3.orig/sound/usb/mixer_quirks.c 2013-12-05 20:21:18.000000000 +0100 > +++ linux-3.12.3/sound/usb/mixer_quirks.c 2013-12-05 20:21:23.000000000 +0100 > @@ -1520,7 +1520,7 @@ static int snd_microii_controls_create(s > return err; > } > > - return err; > + return 0; > } > > int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) >

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[alsa-devel] [PATCH] ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV
by Nicolin Chen 05 Dec '13

05 Dec '13

DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK, which would cause the calculation result from DSPCLK_DIV invalid since bit DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK while the driver won't calculate it again for the current instance. In this circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value. So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for calculation and then disables it afterward. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 07da601..97db3b4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_CLOCKING_4, WM8962_SYSCLK_RATE_MASK, clocking4); + /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK. + * So we here provisionally enable it and then disable it afterward + * if current bias_level hasn't reached SND_SOC_BIAS_ON. + */ + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + dspclk = snd_soc_read(codec, WM8962_CLOCKING1); + + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, 0); + if (dspclk < 0) { dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); return; -- 1.8.4

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[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Do FLL configuration in hw_params() and hw_free()
by Nicolin Chen 05 Dec '13

05 Dec '13

Previously, we couldn't use hw_params() and hw_free() to open and close FLL becuase there might be race between two simmultaneous substreams and cause FLL configuration being changed and accordingly mulfunction. So we adopted DAPM to control it since there won't be any race in DAPM. However, if we want to playback a different sample rate file, we need to wait for DAPM to change its bias_level to reconfigure FLL. But after we introduced full symmetry protection in the soc-pcm, we don't need to worry about the race any more. And the instance by using hw_params() and hw_free() to control FLL would allow us to support flexible use cases, 'aplay -Dhw:0 44k16bit.wav 48k24bit.wav 32k32bit.wav' for example. Thus this patch mainly moves FLL configuration from set_bias_level() to hw_params() and hw_free() so as to enchance the sound card's capability. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 147 +++++++++++++++++++++------------------------ 1 file changed, 70 insertions(+), 77 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 61e4885..df67a27 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -28,6 +28,9 @@ #define DAI_NAME_SIZE 32 +#define IMX_WM8962_DAI_FMT \ + (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) + struct imx_wm8962_data { struct snd_soc_dai_link dai; struct snd_soc_card card; @@ -49,92 +52,84 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; -static int sample_rate = 44100; -static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - sample_rate = params_rate(params); - sample_format = params_format(params); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + snd_pcm_format_t sample_format = params_format(params); + u32 pll_out, sample_rate = params_rate(params); + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, IMX_WM8962_DAI_FMT); + if (ret) { + dev_err(dev, "failed to set codec dai fmt: %d\n", ret); + return ret; + } + + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + int ret; + + /* Don't diable FLL if still having multiple substreams running */ + if (codec_dai->active != 1) + return 0; + + /* Switch to MCLK as sysclk once so as to disable FLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + /* Disable FLL so that we can reset its output freq later */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 0, 0); + if (ret < 0) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } return 0; } static struct snd_soc_ops imx_hifi_ops = { .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, }; -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; - else - pll_out = sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - dapm->bias_level = level; - - return 0; -} - static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; @@ -248,8 +243,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + data->dai.dai_fmt = IMX_WM8962_DAI_FMT; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); @@ -264,7 +258,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { -- 1.8.4

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[alsa-devel] [PATCH] ASoC: samsung: Ensure DMA data is initialised for secondary DAI
by Mark Brown 05 Dec '13

05 Dec '13

From: Mark Brown <broonie(a)linaro.org> While testing the upcoming dmaengine conversion for the driver Padma noticed that the DMA data for the secondard DAI is not being correctly initialised and provided a fix on top of the conversion. Also do this fix for mainline. Reported-by: Padmavathi Venna <padma.v(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> Cc: stable(a)vger.kernel.org --- Compile tested only, I don't have hardware to test. sound/soc/samsung/i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..f9f2227421ea 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + snd_soc_dai_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { -- 1.8.5.1

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[alsa-devel] [PATCH 1/2] ASoC: add DT binding document for MAX98090 CODEC
by Stephen Warren 05 Dec '13

05 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> This binding mainly serves to document the list of input and output pins that may be used in a sound card's audio routing table. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This series doesn't depend on anything; it could be applied anywhere in the ASoC tree. .../devicetree/bindings/sound/max98090.txt | 43 ++++++++++++++++++++++ 1 file changed, 43 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98090.txt diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt new file mode 100644 index 000000000000..e4c8b36dcf89 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -0,0 +1,43 @@ +MAX98090 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98090". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Pins on the device (for linking into audio routes): + + * MIC1 + * MIC2 + * DMICL + * DMICR + * IN1 + * IN2 + * IN3 + * IN4 + * IN5 + * IN6 + * IN12 + * IN34 + * IN56 + * HPL + * HPR + * SPKL + * SPKR + * RCVL + * RCVR + * MICBIAS + +Example: + +audio-codec@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; +}; -- 1.8.1.5

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