January 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] SoundBlaterX G5
by Milan Oravec 17 Jan '17

17 Jan '17

Greetings to all, excuse me please if I'm posting to wrong mailing list. I've bought SoundBlaterX G5 USB soundcard/DAC and want to use it in linux. Sound-card works OOB in linux with snd_usb_audio driver but only at max. 96k/24 sample rate, but HW should be 192K/24 capable. Device volume button works too, but LED backlight doesn't change with volume set. Device needs to be plugged twice i to PC to get USB high speed and 96k support enabled. Here is alsa-info output after plugging in first time: http://www.alsa-project.org/db/?f=1805a3c80208a62da101a27e660a8f020ac6ea71 and after re-plugging: http://www.alsa-project.org/db/?f=74ca49f354e9a100ebf7ba9fe9f6eb82c019f3ef The device is 192K/24 capable under OSX but only when is DSP clock source set to direct stereo ( http://support.creative.com/kb/ShowArticle.aspx?sid=132116 ), this may be the problem in linux. Can anybody help/guide me to how to change this setting or tell me what to do please? If more info/debugging is needed I'm here to serve. How can I submit info about this sound-card to ALSA SoundCard Matrix? Thank you in advance! Kind regards Milan -- s pozdravom Milan Oravec CIKT UKF Nitra mobil: +421907670270 email:moravec@ukf.sk -------------------------------------------------------------------- Tuto spravu nepreveril ziadny antivirus - nebolo treba...

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Re: [alsa-devel] [PATCH v4 0/3] Generic HDMI codec: Add channel mapping control
by Arnaud Pouliquen 17 Jan '17

17 Jan '17

Hello, Any comments on this patch-set? what about the introduction of pcm new callback in DAI ops to use chmap helpers? Is it something reasonable, or should i come back on V1? Regards Arnaud > 2017-01-03 16:52 GMT+01:00 Arnaud Pouliquen <arnaud.pouliquen(a)st.com>: > Aim of this patch is to add 'Playback Channel Map' control to export > audio capabilities in term of HDMI sink speakers allocation. > > V4: > Abandon "Generic HDMI codec: Add channel mapping control" patch as it generates warnings during compilation. > > Workaround is to define 2 constant tables in hdmi-codec.c to declare channel mapping. > One for stereo and one for multichannel. > Consequence is that the behaviour is changed: > The chmap multichannel table export the HDMI CA configuration (tlv) and not only the one suuported by HDMI sink. > Furthermore the chmap control .get handler is overwritten to allow to export to user the selected configuration. > > - "ASoC: hdmi-codec: add channel mapping control": > - add hdmi_codec_stereo_chmaps and hdmi_codec_8ch_chmaps tables. > - implement chmap control get handler. > - "DRM: add help to get ELD speaker allocation" > => No delta vs V2. > - "ALSA pcm: allow non constant snd_pcm_chmap_elem" > => abandonned > - "ASoC: core: add optional pcm_new callback for DAI driver" > => No delta vs V2. > > V3: > - "ASoC: hdmi-codec: add channel mapping control": > => Minor fixes: > - select stereo speaker config by default if HDMI cable unplugged > - fix compilation warning. > - "DRM: add help to get ELD speaker allocation" > => No delta vs V2. > - "ALSA pcm: allow non constant snd_pcm_chmap_elem" > => No delta vs V2. > - "ASoC: core: add optional pcm_new callback for DAI driver" > => No delta vs V2. > > V2: > In this version I use chmap helper functions from pcm_lib.c. > It is quite tricky to use it from ASoC due to the relation chip of the controls > with the pcm runtime. > After several tries, my conclusion is that it must be handled in ASoC DAI driver. > Main reason is that the DAI driver can not provide snd_pcm_chmap struct > through kcontrol structure. But this induces that soc-core provides pcm runtime > structure to DAI driver during probe. > > Base on this conclusion. I reworked my patches by adding 2 > new patches in patch-set > 1) ALSA pcm: allow non constant snd_pcm_chmap_elem > This patch allows to handle non constant channel mapping. As ELD > information can change during runtime it is mandatory to properly > handle the feature. > In term of compatibility with legacy it should be straightforward, > as update just consists in suppressing the 'const' constraint. > > 2) ASoC: core: add optional pcm_new callback for DAI driver > This is the only way i found to be able to use > snd_pcm_add_chmap_ctls and associated controls helper functions. > From my point of view, this is the more simple way to add relationship > between DAI and associated pcm devices. > Notice that this patch, if accepted, makes the following one obsolete, > as it also answer to the associated topic: > [PATCH v5 0/2] ALSA controls management using index/device/sub-devices fields > (http://www.spinics.net/lists/alsa-devel/msg57639.html). > > If you estimate that this it not reasonable i will come back to my first version. > > V1: > This patch follows discussion initiate here: > [RFC] ASOC: HDMI audio info frame speaker allocation > http://www.spinics.net/lists/alsa-devel/msg57363.html > > The code is fully inspired from HDA driver. > On hw_param, HDMI sink speaker capabilities are exported via TLV ops > and a CEA allocation is choson, based on ELD information and the number of > channels requested by user. > > Mains differences with HDA implementation are: > - Control is read only > - Channel swap is not supported. Consequence is that unused channel in > the mid of CEA audio infoframe channel mapping are considered as > active. > example for channel allocation 0x02: FL, FR, 0, FC) > This configuration is only available for a 4 channels stream. > - Channel allocation table has been reordered and HDMI 2.0 is not > supported. > > Arnaud Pouliquen (4): > DRM: add help to get ELD speaker allocation > ASoC: core: add optional pcm_new callback for DAI driver > ASoC: hdmi-codec: add channel mapping control > > include/drm/drm_edid.h | 13 ++ > include/sound/soc-dai.h | 3 + > sound/soc/codecs/hdmi-codec.c | 380 +++++++++++++++++++++++++++++++++++++++++- > sound/soc/soc-core.c | 28 ++++ > 4 files changed, 423 insertions(+), 1 deletion(-) > > -- > 1.9.1 > >

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[alsa-devel] [PATCH v3 0/4] ASoC: Intel: clean up Kconfig
by Andy Shevchenko 17 Jan '17

17 Jan '17

Clean up Kconfig here: - select DW_DMAC_CORE because it's mandatory for some modules - remove redundant selects and dependencies - rename MFLD_PLATFORM to MID_PLATFORM to show that it covers more than Medfield Since v2: - add three more patches - rebase on top of latest linux-next Andy Shevchenko (4): ASoC: Intel: select DW_DMAC_CORE since it's mandatory ASoC: Intel: remove redundant select SND_SOC_INTEL_SST ASoC: Intel: remove ignored dependencies ASoC: Intel: rename SND_SST_MFLD_PLATFORM to SND_SST_MID_PLATFORM sound/soc/intel/Kconfig | 46 ++++++++++++++----------------------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/Makefile | 7 ++++--- 3 files changed, 20 insertions(+), 35 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v4 0/4] ASoC: Intel: clean up Kconfig
by Andy Shevchenko 17 Jan '17

17 Jan '17

Clean up Kconfig here: - select DW_DMAC_CORE because it's mandatory for some modules - remove redundant selects and dependencies - rename MFLD_PLATFORM to ATOM_HIFI2_PLATFORM to show that it covers more than *Field Since v3: - use proposed name as SND_SST_ATOM_HIFI2_PLATFORM (Pierre-Louis) - append Keyon's tag Since v2: - add three more patches - rebase on top of latest linux-next Andy Shevchenko (4): ASoC: Intel: select DW_DMAC_CORE since it's mandatory ASoC: Intel: remove redundant select SND_SOC_INTEL_SST ASoC: Intel: remove ignored dependencies ASoC: Intel: rename SND_SST_MFLD_PLATFORM to SND_SST_ATOM_HIFI2_PLATFORM sound/soc/intel/Kconfig | 46 +++++++++------------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/Makefile | 7 +++-- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + 4 files changed, 21 insertions(+), 35 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v6 0/4] ASoC: Intel: clean up Kconfig
by Andy Shevchenko 17 Jan '17

17 Jan '17

Clean up Kconfig here: - select DW_DMAC_CORE because it's mandatory for some modules - remove redundant selects and dependencies - rename MFLD_PLATFORM to ATOM_HIFI2_PLATFORM to show that it covers more than *Field Since v5: - DMA engine drivers have to be dependent to DMADEVICES (kbuild test bot) - due to above drop tags for patch 1 Since v4: - append Vinod's and Liam's ACKs Since v3: - use proposed name as SND_SST_ATOM_HIFI2_PLATFORM (Pierre-Louis) - append Keyon's tag Since v2: - add three more patches - rebase on top of latest linux-next Andy Shevchenko (4): ASoC: Intel: select DW_DMAC_CORE since it's mandatory ASoC: Intel: remove redundant select SND_SOC_INTEL_SST ASoC: Intel: remove ignored dependencies ASoC: Intel: rename SND_SST_MFLD_PLATFORM to SND_SST_ATOM_HIFI2_PLATFORM sound/soc/intel/Kconfig | 51 +++++++++++----------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/Makefile | 7 ++-- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + 4 files changed, 26 insertions(+), 35 deletions(-) -- 2.11.0

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[alsa-devel] [RFC PATCH 0/2] ASoC: Use DMI name in sound card long name
by mengdong.lin＠linux.intel.com 17 Jan '17

17 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series is based on Liam's idea to use DMI name as the sound card short name in the audio workshop. After testing on serveral machines, we changed the design a little: - Not change the card name (short name), keep it simple. So the card ID won't change as well. - Use short name and DMI name together to make a readable and unqique long name. Machine drivers can use a new API to get this feature. We hope to get aligned on the kernel side at first, then we'll update UCM code in alsa-lib: when the sound server ask UCM to open a sound card, UCM will try to find the best configuration file that matches the card long name, and only if not avaiable, fallback to the default configuration file that matches the card name (short name). Liam Girdwood (1): ASoC: core: Add API to use DMI name in sound card long name Mengdong Lin (1): ASoC: Intel: Use DMI name for sound card long name in Broadwell machine driver include/sound/soc.h | 5 ++ sound/soc/intel/boards/broadwell.c | 2 + sound/soc/soc-core.c | 134 +++++++++++++++++++++++++++++++++++++ 3 files changed, 141 insertions(+) -- 2.5.0

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[alsa-devel] [PATCH] RFC: ASoC: dapm: handle probe deferrals
by Linus Walleij 17 Jan '17

17 Jan '17

This starts to handle probe deferrals on regulators and clocks on the ASoC DAPM. I came to this patch after audio stopped working on Ux500 ages ago and I finally looked into it to see what is wrong. I had messages like this in the console since a while back: ab8500-codec.0: ASoC: Failed to request audioclk: -517 ab8500-codec.0: ASoC: Failed to create DAPM control audioclk ab8500-codec.0: Failed to create new controls -12 snd-soc-mop500.0: ASoC: failed to instantiate card -12 snd-soc-mop500.0: Error: snd_soc_register_card failed (-12)! snd-soc-mop500: probe of snd-soc-mop500.0 failed with error -12 Apparently because the widget table for the codec looks like this (sound/soc/codecs/ab8500-codec.c): static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { /* Clocks */ SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), /* Regulators */ SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0, 0), So when we call snd_soc_register_codec() and any of these widgets get a deferred probe we do not get an -EPROBE_DEFER (-517) back as we should and instead we just fail. Apparently the code assumes that clocks and regulators must be available at this point and not defer. After this patch it rather looks like this: ab8500-codec.0: Failed to create new controls -517 snd-soc-mop500.0: ASoC: failed to instantiate card -517 snd-soc-mop500.0: Error: snd_soc_register_card failed (-517)! (...) abx500-clk.0: registered clocks for ab850x snd-soc-mop500.0: ab8500-codec-dai.0 <-> ux500-msp-i2s.1 mapping ok snd-soc-mop500.0: ab8500-codec-dai.1 <-> ux500-msp-i2s.3 mapping ok I'm pretty happy about the patch as it it, but I'm a bit uncertain on how to proceed: there are a lot of users of the external functions snd_soc_dapm_new_control() (111 sites) and that will now return an occassional error pointer, which is not handled in the calling sites. I want an indication from the maintainers whether I should just go in and augment all these call sites, or if deferred probe is frowned upon when it leads to this much overhead. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/soc-dapm.c | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-topology.c | 9 +++++++++ 2 files changed, 51 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 27dd02e57b31..b218cc7bd994 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -363,6 +363,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, snd_soc_dapm_new_control_unlocked(widget->dapm, &template); kfree(name); + if (IS_ERR(data->widget)) { + ret = PTR_ERR(data->widget); + goto err_data; + } if (!data->widget) { ret = -ENOMEM; goto err_data; @@ -397,6 +401,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked( widget->dapm, &template); kfree(name); + if (IS_ERR(data->widget)) { + ret = PTR_ERR(data->widget); + goto err_data; + } if (!data->widget) { ret = -ENOMEM; goto err_data; @@ -3403,11 +3411,22 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); w = snd_soc_dapm_new_control_unlocked(dapm, widget); + /* Do not nag about probe deferrals */ + if (IS_ERR(w)) { + int ret = PTR_ERR(w); + + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, + "ASoC: Failed to create DAPM control %s (%d)\n", + widget->name, ret); + goto out_unlock; + } if (!w) dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", widget->name); +out_unlock: mutex_unlock(&dapm->card->dapm_mutex); return w; } @@ -3430,6 +3449,8 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); + if (ret == -EPROBE_DEFER) + return ERR_PTR(ret); dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", w->name, ret); return NULL; @@ -3448,6 +3469,8 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); + if (ret == -EPROBE_DEFER) + return ERR_PTR(ret); dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", w->name, ret); return NULL; @@ -3566,6 +3589,16 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { w = snd_soc_dapm_new_control_unlocked(dapm, widget); + if (IS_ERR(w)) { + ret = PTR_ERR(w); + /* Do not nag about probe deferrals */ + if (ret == -EPROBE_DEFER) + break; + dev_err(dapm->dev, + "ASoC: Failed to create DAPM control %s (%d)\n", + widget->name, ret); + break; + } if (!w) { dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", @@ -3842,6 +3875,15 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); + if (IS_ERR(w)) { + ret = PTR_ERR(w); + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(card->dev, + "ASoC: Failed to create %s widget (%d)\n", + link_name, ret); + goto outfree_kcontrol_news; + } if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 65670b2b408c..37006c63891a 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1556,6 +1556,15 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, widget = snd_soc_dapm_new_control(dapm, &template); else widget = snd_soc_dapm_new_control_unlocked(dapm, &template); + if (IS_ERR(widget)) { + ret = PTR_ERR(widget); + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(tplg->dev, + "ASoC: failed to create widget %s controls (%d)\n", + w->name, ret); + goto hdr_err; + } if (widget == NULL) { dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", w->name); -- 2.9.3

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[alsa-devel] [asoc:topic/rt5659 25/26] sound/soc/codecs/rt5659.c:4236:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 17 Jan '17

17 Jan '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5659 head: eae39b5f4269260d5d8b35133ba0f4c5e2895b71 commit: 0230f088adcf537a60df9d80dce5b71059946dfe [25/26] ASoC: rt5659: declare rt5659_i2c_driver static coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/rt5659.c:4236:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 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 1/2] ASoC: mxs-saif: fix setting master base rate
by Jörg Krause 17 Jan '17

17 Jan '17

From: Jörg Krause <jk(a)lintech.de> The SAIF base oversample rates are either 512*fs or 384*fs. An additional divider exists within the SAIF to generate sub-multiples of these two base rates if MCLK is required by the codec. * The sub-rates for the 512x base rate are: 256x, 128x, 64x, and 32x. * The sub-rates for the 384x base rate are: 192x, 96x, and 48x. Setting the base rate depending on the modulo operation with 32 and 48 give wrong results for some mclk. If mclk=18.432MHz both modulo operations results in 0. As testing the result with 32 is done first, a wrong base rate of 512*fs is set instead of the correct 384*fs. Fix this by setting the base rate depending on the calculated sub-rate. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- sound/soc/mxs/mxs-saif.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 13631003cb7c..1e724533dc6e 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -119,23 +119,33 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * Set SAIF clock * * The SAIF clock should be either 384*fs or 512*fs. - * If MCLK is used, the SAIF clk ratio need to match mclk ratio. - * For 32x mclk, set saif clk as 512*fs. - * For 48x mclk, set saif clk as 384*fs. + * If MCLK is used, the SAIF clk ratio needs to match mclk ratio. + * For 256x, 128x, 64x, and 32x sub-rates, set saif clk as 512*fs. + * For 192x, 96x, and 48x sub-rates, set saif clk as 384*fs. * * If MCLK is not used, we just set saif clk to 512*fs. */ clk_prepare_enable(master_saif->clk); if (master_saif->mclk_in_use) { - if (mclk % 32 == 0) { + switch (mclk / rate) { + case 32: + case 64: + case 128: + case 256: + case 512: scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; ret = clk_set_rate(master_saif->clk, 512 * rate); - } else if (mclk % 48 == 0) { + break; + case 48: + case 96: + case 192: + case 384: scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; ret = clk_set_rate(master_saif->clk, 384 * rate); - } else { - /* SAIF MCLK should be either 32x or 48x */ + break; + default: + /* SAIF MCLK should be a sub-rate of 512x or 384x */ clk_disable_unprepare(master_saif->clk); return -EINVAL; } -- 2.11.0

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Re: [alsa-devel] [PATCH v2 07/10] dt-bindings: sound: Add sun8i audio documentation
by Icenowy Zheng 17 Jan '17

17 Jan '17

2017年1月18日 00:51于 Maxime Ripard <maxime.ripard(a)free-electrons.com>写道： > > Hi, > > On Tue, Jan 17, 2017 at 03:02:27PM +0100, Mylène Josserand wrote: > > Add the documentation for dt-binding of the digital audio codec driver > > and the audio card driver for Sun8i SoCs. > > > > Signed-off-by: Mylène Josserand <mylene.josserand(a)free-electrons.com> > > One small comment below, > > > --- > > .../devicetree/bindings/sound/sun8i-codec.txt | 76 ++++++++++++++++++++++ > > 1 file changed, 76 insertions(+) > > create mode 100644 Documentation/devicetree/bindings/sound/sun8i-codec.txt > > > > diff --git a/Documentation/devicetree/bindings/sound/sun8i-codec.txt b/Documentation/devicetree/bindings/sound/sun8i-codec.txt > > new file mode 100644 > > index 000000000000..ce3c05219e33 > > --- /dev/null > > +++ b/Documentation/devicetree/bindings/sound/sun8i-codec.txt > > @@ -0,0 +1,76 @@ > > +Allwinner SUN8I audio codec > > +------------------------------------ > > + > > +On Sun8i SoCs, and particularly on A33, the audio is separated in > > Technically that's not true on all the SoCs of the sun8i family, but > only a few of them (A33 and H3 iirc). H3 is still the old-style sunxi codec. I think A64 has the same digital codec part (but A64's analog part is different) > > This driver is only made for the A33 at the moment, so you should only > mention it (and you should rename that file as well). > > Once done, > Acked-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> > > Thanks, > Maxime > > > > +different parts: > > + - A DAI driver. It uses the "sun4i-i2s" driver which is > > + documented here: > > + Documentation/devicetree/bindings/sound/sun4i-i2s.txt > > + - An analog part of the codec which is handled as PRCM registers. > > + See Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt > > + - An digital part of the codec which is documented in this current > > + binding documentation. > > + - And finally, an audio card which links all the above components. > > + The simple-audio card will be used. > > + See Documentation/devicetree/bindings/sound/simple-card.txt > > + > > +This bindings documentation exposes Sun8i codec (digital part). > > + > > +Required properties: > > +- compatible: must be "allwinner,sun8i-a33-codec" > > +- reg: must contain the registers location and length > > +- interrupts: must contain the codec interrupt > > +- clocks: a list of phandle + clock-specifer pairs, one for each entry > > + in clock-names. > > +- clock-names: should contain followings: > > + - "bus": the parent APB clock for this controller > > + - "mod": the parent module clock > > + > > +Example: > > +codec: codec@01c22e00 { > > + #sound-dai-cells = <0>; > > + compatible = "allwinner,sun8i-a33-codec"; > > + reg = <0x01c22e00 0x400>; > > + interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; > > + clocks = <&ccu CLK_BUS_CODEC>, <&ccu CLK_AC_DIG>; > > + clock-names = "bus", "mod"; > > + status = "disabled"; > > +}; > > + > > +Here is an example to add a sound card and the codec binding on sun8i SoCs that > > +are similar to A33 using simple-card: > > + > > + sound { > > + compatible = "simple-audio-card"; > > + simple-audio-card,name = "Sun8i Audio Card"; > > + simple-audio-card,format = "i2s"; > > + simple-audio-card,frame-master = <&link_codec>; > > + simple-audio-card,bitclock-master = <&link_codec>; > > + simple-audio-card,mclk-fs = <512>; > > + simple-audio-card,aux-devs = <&codec_analog>; > > + simple-audio-card,routing = > > + "Left DAC", "Digital Left DAC", > > + "Right DAC", "Digital Right DAC"; > > + > > + simple-audio-card,cpu { > > + sound-dai = <&dai>; > > + }; > > + > > + link_codec: simple-audio-card,codec { > > + sound-dai = <&codec>; > > + }; > > + > > + soc@01c00000 { > > + [...] > > + > > + codec: codec@01c22e00 { > > + #sound-dai-cells = <0>; > > + compatible = "allwinner,sun8i-a33-codec"; > > + reg = <0x01c22e00 0x400>; > > + interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; > > + clocks = <&ccu CLK_BUS_CODEC>, <&ccu CLK_AC_DIG>; > > + clock-names = "bus", "mod"; > > + status = "disabled"; > > + }; > > + }; > > + > > -- > > 2.11.0 > > > > -- > Maxime Ripard, Free Electrons > Embedded Linux and Kernel engineering > http://free-electrons.com > > _______________________________________________ > linux-arm-kernel mailing list > linux-arm-kernel(a)lists.infradead.org > http://lists.infradead.org/mailman/listinfo/linux-arm-kernel

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