January 2019 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] hp_pin was NULL value
by Takashi Iwai 30 Jan '19

30 Jan '19

On Wed, 09 Jan 2019 13:32:51 +0100, Kailang wrote: > > > Hi Takashi, > > > OK! I will test it. > So, if ALC225 need to do this. I will assigned it to alc225_init(). Right? > > >>But ALC294 (and ALC700) have no init_hook, so far? > > spec->init_hook = alc_default_init; > It has default init_hook. I use this to do standard headphone depop function. Ah, OK. Then we need a call chain from alc294_init(), static void alc294_init() { coef(); alc294_hp_init(); alc_default_init(); } thanks, Takashi > > BR, > Kailang > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2019年1月9日下午 07:29 > 至: Kailang > 副本: (alsa-devel(a)alsa-project.org) > 主旨: Re: hp_pin was NULL value > > On Wed, 09 Jan 2019 12:19:14 +0100, > Kailang wrote: > > > > Hi Takashi, > > > > But spec->init_hook will call at resume back. > > alc294_hp_init() function just need to call at one time. > > As already mentioned, even if it has to be called only once and does > something for the hardware, it'll be missing if the machine goes > hibernate (S4) and resume. In S4, the machine boots from scratch and > switches immediately to the saved image that calls resume callback. > What's worse for us is that this switch happens usually in initrd, > i.e. before the sound driver gets loaded. So, the machine goes to > resume without processing alc294_hp_init(). > > IOW, even if it's needed only once, you'll have to call it at resume > unless it has a significant drawback. > > (Strictly speaking, we may use thaw PM callback, but the HD-audio > codec driver doesn't use it, so there is only resume callback for > now.) > > > And it need to call it before call spec->init_hook. > > But ALC294 (and ALC700) have no init_hook, so far? > > > ALC225 also need to call similar procedure. I will post it later. > > OK, thanks. > > > Takashi > > > > > BR, > > Kailang > > ________________________________________ > > 從: Takashi Iwai [tiwai(a)suse.de] > > 寄件日期: 2019年1月9日下午 05:56 > > 至: Kailang > > 副本: (alsa-devel(a)alsa-project.org) > > 主旨: Re: hp_pin was NULL value > > > > On Wed, 09 Jan 2019 10:45:25 +0100, > > Kailang wrote: > > > > > > >>But this made me wonder whether we don't need to call this function at resume as well? Currently it's called only at probing. > > > Yes, it only call at probing. Just need to call it at boot time. > > > > > > -----Original Message----- > > > From: Takashi Iwai <tiwai(a)suse.de> > > > Sent: Wednesday, January 9, 2019 5:43 PM > > > To: Kailang <kailang(a)realtek.com> > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > Subject: Re: hp_pin was NULL value > > > > > > On Wed, 09 Jan 2019 10:31:33 +0100, > > > Kailang wrote: > > > > > > > > Hi Takashi, > > > > > > > > Could I move the alc294_hp_init(codec) to below line. > > > > Because hp_pin = spec->gen.autocfg.hp_pins[0] was null value when alc294_hp_init(codec) at original line. > > > > Or move alc269_parse_auto_config() upward. > > > > > > It looks OK to me. But this made me wonder whether we don't need to call this function at resume as well? Currently it's called only at probing. > > > > Hrm, but it modifies many COEFs, and are these all preserved with > > suspend? For example, there is also hibernation (S4), which is a > > switch from the boot without the audio driver initialization. Then > > we'll resume from the uninitialized state. I guess other COEF changes > > in that area should be moved as the additional init hook as well. > > > > Actually, if we do call this at resume, the change would be easier, a > > patch like below. > > > > > > thanks, > > > > Takashi > > > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -7404,6 +7404,20 @@ static void alc294_hp_init(struct hda_codec *codec) > > msleep(50); > > } > > > > +static void alc294_init(struct hda_codec *codec) > > +{ > > + /* UAJ MIC Vref control by verb */ > > + alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); > > + alc294_hp_init(codec); > > +} > > + > > +static void alc700_init(struct hda_codec *codec) > > +{ > > + /* Combo jack auto trigger control */ > > + alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); > > + alc294_hp_init(codec); > > +} > > + > > /* > > */ > > static int patch_alc269(struct hda_codec *codec) > > @@ -7528,8 +7542,7 @@ static int patch_alc269(struct hda_codec *codec) > > case 0x10ec0294: > > spec->codec_variant = ALC269_TYPE_ALC294; > > spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */ > > - alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */ > > - alc294_hp_init(codec); > > + spec->init_hook = alc294_init; > > break; > > case 0x10ec0300: > > spec->codec_variant = ALC269_TYPE_ALC300; > > @@ -7540,8 +7553,7 @@ static int patch_alc269(struct hda_codec *codec) > > case 0x10ec0703: > > spec->codec_variant = ALC269_TYPE_ALC700; > > spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ > > - alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ > > - alc294_hp_init(codec); > > + spec->init_hook = alc700_init; > > break; > > > > } > > > > > > ------Please consider the environment before printing this e-mail.

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[alsa-devel] [PATCH] ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check
by Kuninori Morimoto 30 Jan '19

30 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") added new rsnd_ssi_prepare() and moved rsnd_ssi_master_clk_start() to .prepare. But, ssi user count (= ssi->usrcnt) is incremented at .init (= rsnd_ssi_init()). Because of these timing exchange, ssi->usrcnt check at rsnd_ssi_master_clk_start() should be adjusted. Otherwise, 2nd master clock setup will be no check. This patch fixup this issue. Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Reported-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Reported-by: Valentine Barshak <valentine.barshak(a)cogentembedded.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 45ef295..f5afab6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->usrcnt > 1) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix suspend/resume calls
by Rodrigo Vivi 29 Jan '19

29 Jan '19

Besides the compilation issue the resume function was calling the force_suspend. Fixes: d81f8c6b6ced ("ALSA: hda - Record the current power state before suspend/resume calls") Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Rodrigo Vivi <rodrigo.vivi(a)intel.com> --- sound/pci/hda/hda_codec.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 43bcc6f1ea8c..0b86aa2b649c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2894,6 +2894,8 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { + struct device *dev = hda_codec_dev(codec); + snd_hdac_enter_pm(&codec->core); if (codec->core.regmap) regcache_mark_dirty(codec->core.regmap); @@ -2955,10 +2957,16 @@ static int hda_codec_pm_suspend(struct device *dev) return pm_runtime_force_suspend(dev); } +static int hda_codec_pm_freeze(struct device *dev) +{ + dev->power.power_state = PMSG_FREEZE; + return pm_runtime_force_suspend(dev); +} + static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return pm_runtime_force_suspend(dev); + return pm_runtime_force_resume(dev); } static int hda_codec_pm_thaw(struct device *dev) -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda - Use standard device registration for beep
by Takashi Iwai 29 Jan '19

29 Jan '19

Currently the registration and free of beep input device was done manually from the register and the disconnect callbacks of the assigned codec object. This seems working in most cases, but this may be a cause of some races at probe. Moreover, due to these manual calls, the total code became unnecessarily lengthy. This patch rewrites the beep registration code to follow the standard sound device object style. This allows us reducing the code, in addition to avoiding the nested device registration calls. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 151 ++++++++++++++++++++++------------------------ sound/pci/hda/hda_beep.h | 5 -- sound/pci/hda/hda_codec.c | 10 --- 3 files changed, 72 insertions(+), 94 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 066b5b59c4d7..b7d9160ed868 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -127,44 +127,6 @@ static void turn_off_beep(struct hda_beep *beep) } } -static void snd_hda_do_detach(struct hda_beep *beep) -{ - if (beep->registered) - input_unregister_device(beep->dev); - else - input_free_device(beep->dev); - beep->dev = NULL; - turn_off_beep(beep); -} - -static int snd_hda_do_attach(struct hda_beep *beep) -{ - struct input_dev *input_dev; - struct hda_codec *codec = beep->codec; - - input_dev = input_allocate_device(); - if (!input_dev) - return -ENOMEM; - - /* setup digital beep device */ - input_dev->name = "HDA Digital PCBeep"; - input_dev->phys = beep->phys; - input_dev->id.bustype = BUS_PCI; - input_dev->dev.parent = &codec->card->card_dev; - - input_dev->id.vendor = codec->core.vendor_id >> 16; - input_dev->id.product = codec->core.vendor_id & 0xffff; - input_dev->id.version = 0x01; - - input_dev->evbit[0] = BIT_MASK(EV_SND); - input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); - input_dev->event = snd_hda_beep_event; - input_set_drvdata(input_dev, beep); - - beep->dev = input_dev; - return 0; -} - /** * snd_hda_enable_beep_device - Turn on/off beep sound * @codec: the HDA codec @@ -186,6 +148,38 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) } EXPORT_SYMBOL_GPL(snd_hda_enable_beep_device); +static int beep_dev_register(struct snd_device *device) +{ + struct hda_beep *beep = device->device_data; + int err; + + err = input_register_device(beep->dev); + if (!err) + beep->registered = true; + return err; +} + +static int beep_dev_disconnect(struct snd_device *device) +{ + struct hda_beep *beep = device->device_data; + + if (beep->registered) + input_unregister_device(beep->dev); + else + input_free_device(beep->dev); + turn_off_beep(beep); + return 0; +} + +static int beep_dev_free(struct snd_device *device) +{ + struct hda_beep *beep = device->device_data; + + beep->codec->beep = NULL; + kfree(beep); + return 0; +} + /** * snd_hda_attach_beep_device - Attach a beep input device * @codec: the HDA codec @@ -194,14 +188,16 @@ EXPORT_SYMBOL_GPL(snd_hda_enable_beep_device); * Attach a beep object to the given widget. If beep hint is turned off * explicitly or beep_mode of the codec is turned off, this doesn't nothing. * - * The attached beep device has to be registered via - * snd_hda_register_beep_device() and released via snd_hda_detach_beep_device() - * appropriately. - * * Currently, only one beep device is allowed to each codec. */ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { + static struct snd_device_ops ops = { + .dev_register = beep_dev_register, + .dev_disconnect = beep_dev_disconnect, + .dev_free = beep_dev_free, + }; + struct input_dev *input_dev; struct hda_beep *beep; int err; @@ -226,14 +222,41 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); mutex_init(&beep->mutex); - err = snd_hda_do_attach(beep); - if (err < 0) { - kfree(beep); - codec->beep = NULL; - return err; + input_dev = input_allocate_device(); + if (!input_dev) { + err = -ENOMEM; + goto err_free; } + /* setup digital beep device */ + input_dev->name = "HDA Digital PCBeep"; + input_dev->phys = beep->phys; + input_dev->id.bustype = BUS_PCI; + input_dev->dev.parent = &codec->card->card_dev; + + input_dev->id.vendor = codec->core.vendor_id >> 16; + input_dev->id.product = codec->core.vendor_id & 0xffff; + input_dev->id.version = 0x01; + + input_dev->evbit[0] = BIT_MASK(EV_SND); + input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + input_dev->event = snd_hda_beep_event; + input_set_drvdata(input_dev, beep); + + beep->dev = input_dev; + + err = snd_device_new(codec->card, SNDRV_DEV_JACK, beep, &ops); + if (err < 0) + goto err_input; + return 0; + + err_input: + input_free_device(beep->dev); + err_free: + kfree(beep); + codec->beep = NULL; + return err; } EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device); @@ -243,41 +266,11 @@ EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device); */ void snd_hda_detach_beep_device(struct hda_codec *codec) { - struct hda_beep *beep = codec->beep; - if (beep) { - if (beep->dev) - snd_hda_do_detach(beep); - codec->beep = NULL; - kfree(beep); - } + if (!codec->bus->shutdown && codec->beep) + snd_device_free(codec->card, codec->beep); } EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device); -/** - * snd_hda_register_beep_device - Register the beep device - * @codec: the HDA codec - */ -int snd_hda_register_beep_device(struct hda_codec *codec) -{ - struct hda_beep *beep = codec->beep; - int err; - - if (!beep || !beep->dev) - return 0; - - err = input_register_device(beep->dev); - if (err < 0) { - codec_err(codec, "hda_beep: unable to register input device\n"); - input_free_device(beep->dev); - codec->beep = NULL; - kfree(beep); - return err; - } - beep->registered = true; - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_register_beep_device); - static bool ctl_has_mute(struct snd_kcontrol *kcontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index f1457c6b3969..a25358a4807a 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -34,7 +34,6 @@ struct hda_beep { int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); -int snd_hda_register_beep_device(struct hda_codec *codec); #else static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { @@ -43,9 +42,5 @@ static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) static inline void snd_hda_detach_beep_device(struct hda_codec *codec) { } -static inline int snd_hda_register_beep_device(struct hda_codec *codec) -{ - return 0; -} #endif #endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index dc7b342f00ef..5f2005098a60 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -813,7 +813,6 @@ void snd_hda_codec_register(struct hda_codec *codec) if (codec->registered) return; if (device_is_registered(hda_codec_dev(codec))) { - snd_hda_register_beep_device(codec); codec_display_power(codec, true); pm_runtime_enable(hda_codec_dev(codec)); /* it was powered up in snd_hda_codec_new(), now all done */ @@ -828,14 +827,6 @@ static int snd_hda_codec_dev_register(struct snd_device *device) return 0; } -static int snd_hda_codec_dev_disconnect(struct snd_device *device) -{ - struct hda_codec *codec = device->device_data; - - snd_hda_detach_beep_device(codec); - return 0; -} - static int snd_hda_codec_dev_free(struct snd_device *device) { struct hda_codec *codec = device->device_data; @@ -921,7 +912,6 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, int err; static struct snd_device_ops dev_ops = { .dev_register = snd_hda_codec_dev_register, - .dev_disconnect = snd_hda_codec_dev_disconnect, .dev_free = snd_hda_codec_dev_free, }; -- 2.16.4

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[alsa-devel] [PATCH 1/2] dt-bindings: ASoC: Add Spreadtrum DMA platform documentation
by Baolin Wang 29 Jan '19

29 Jan '19

Add documentation for Spreadtrum DMA platform driver. Signed-off-by: Baolin Wang <baolin.wang(a)linaro.org> --- .../devicetree/bindings/sound/sprd-pcm.txt | 23 ++++++++++++++++++++ 1 file changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sprd-pcm.txt diff --git a/Documentation/devicetree/bindings/sound/sprd-pcm.txt b/Documentation/devicetree/bindings/sound/sprd-pcm.txt new file mode 100644 index 0000000..4b23e84 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd-pcm.txt @@ -0,0 +1,23 @@ +* Spreadtrum DMA platfrom bindings + +Required properties: +- compatible: Should be "sprd,pcm-platform". +- dmas: Specify the list of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + +Example: + + audio_platform:platform@0 { + compatible = "sprd,pcm-platform"; + dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, + <&agcp_dma 3 3>, <&agcp_dma 4 4>, + <&agcp_dma 5 5>, <&agcp_dma 6 6>, + <&agcp_dma 7 7>, <&agcp_dma 8 8>, + <&agcp_dma 9 9>, <&agcp_dma 10 10>; + dma-names = "normal_p_l", "normal_p_r", + "normal_c_l", "normal_c_r", + "voice_c", "fast_p", + "loop_c", "loop_p", + "voip_c", "voip_p"; + }; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Intel: cht_bsw_rt5672: remove useless test
by Pierre-Louis Bossart 29 Jan '19

29 Jan '19

For some reason we test if the machine is passed as a parameter before fixing up the codec name. This is unnecessary, generates false positives in static analysis tools and done only in this machine driver, remove and adjust indentation. Reported-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index f1c1f9dd5353..3d5a2b3a06f0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -411,18 +411,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) strcpy(drv->codec_name, RT5672_I2C_DEFAULT); /* fixup codec name based on HID */ - if (mach) { - i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); - if (i2c_name) { - snprintf(drv->codec_name, sizeof(drv->codec_name), - "i2c-%s", i2c_name); - for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) { - if (!strcmp(cht_dailink[i].codec_name, - RT5672_I2C_DEFAULT)) { - cht_dailink[i].codec_name = - drv->codec_name; - break; - } + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); + if (i2c_name) { + snprintf(drv->codec_name, sizeof(drv->codec_name), + "i2c-%s", i2c_name); + for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) { + if (!strcmp(cht_dailink[i].codec_name, + RT5672_I2C_DEFAULT)) { + cht_dailink[i].codec_name = drv->codec_name; + break; } } } -- 2.17.1

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Re: [alsa-devel] linux-next: Tree for Jan 29 (SLIMbus & SND_SOC_WCD9335)
by Randy Dunlap 29 Jan '19

29 Jan '19

On 1/28/19 10:11 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20190125: > on x86_64: CONFIG_SND_SOC_WCD9335=m which selects REGMAP_SLIMBUS, but SLIMBUS is not set/enabled. ERROR: "slim_driver_unregister" [sound/soc/codecs/snd-soc-wcd9335.ko] undefined! ERROR: "__slim_driver_register" [sound/soc/codecs/snd-soc-wcd9335.ko] undefined! ERROR: "slim_read" [drivers/base/regmap/regmap-slimbus.ko] undefined! ERROR: "slim_write" [drivers/base/regmap/regmap-slimbus.ko] undefined! Full randconfig file is attached. -- ~Randy

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[alsa-devel] [PATCH - JACK plugin v2 0/2] jack: Support to connect multiple JACK ports with same ALSA channel
by twischer＠de.adit-jv.com 29 Jan '19

29 Jan '19

On 1/29/19 14:45, Takashi Iwai wrote: > name is already dereferenced in strlen(), so it's too late here. > i.e. name_size has to be evaluated after this NULL check, if any. > Or just drop this NULL check. It's an internal function and the name > is never NULL, practically seen. I have removed it. > name_size is the exact size of string + 1, so this is equivalent with > strcpy(). I have added a comment and I am using strcpy(), now. Best regards Timo

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[alsa-devel] [PATCH - JACK plugin 0/2] jack: Support to connect multiple ports
by twischer＠de.adit-jv.com 29 Jan '19

29 Jan '19

Hi all, the following patches enable the ALSA JACK plugin to connect the same ALSA channel to multiple JACK ports. An example configuration is given in the commit message of the second commit. Best regards Timo

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[alsa-devel] [PATCH] ucm: Add ucm files for DB820c board
by Srinivas Kandagatla 29 Jan '19

29 Jan '19

DB820c board is based of MSM8996 Qualcomm SoC, which has support for both Digital and Analog audio. Digital audio is over HDMI and analog is over WCD9335 codec via SLIMbus. Board itself has HDMI port, a 3.5mm audio Jack and an Audio expansion connector. This patch adds support for HDMI port and 3.5mm jack. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- configure.ac | 1 + src/conf/ucm/DB820c/DB820c.conf | 9 +++ src/conf/ucm/DB820c/HDMI | 37 +++++++++++ src/conf/ucm/DB820c/HiFi | 110 ++++++++++++++++++++++++++++++++ src/conf/ucm/DB820c/Makefile.am | 4 ++ src/conf/ucm/Makefile.am | 1 + 6 files changed, 162 insertions(+) create mode 100644 src/conf/ucm/DB820c/DB820c.conf create mode 100644 src/conf/ucm/DB820c/HDMI create mode 100644 src/conf/ucm/DB820c/HiFi create mode 100644 src/conf/ucm/DB820c/Makefile.am diff --git a/configure.ac b/configure.ac index a0c346ef..e9e1a369 100644 --- a/configure.ac +++ b/configure.ac @@ -747,6 +747,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/Makefile \ src/conf/ucm/DAISY-I2S/Makefile \ src/conf/ucm/DB410c/Makefile \ + src/conf/ucm/DB820c/Makefile \ src/conf/ucm/Dell-WD15-Dock/Makefile \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/gpd-win-pocket-rt5645/Makefile \ diff --git a/src/conf/ucm/DB820c/DB820c.conf b/src/conf/ucm/DB820c/DB820c.conf new file mode 100644 index 00000000..58b7ff4e --- /dev/null +++ b/src/conf/ucm/DB820c/DB820c.conf @@ -0,0 +1,9 @@ +SectionUseCase."HiFi" { + File "HiFi" + Comment "HiFi quality Music." +} + +SectionUseCase."HDMI" { + File "HDMI" + Comment "HDMI output." +} diff --git a/src/conf/ucm/DB820c/HDMI b/src/conf/ucm/DB820c/HDMI new file mode 100644 index 00000000..39b28692 --- /dev/null +++ b/src/conf/ucm/DB820c/HDMI @@ -0,0 +1,37 @@ +# Use case configuration for DB820c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + EnableSequence [ + cdev "hw:0" + cset "name='HDMI Mixer MultiMedia1' 1" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='HDMI Mixer MultiMedia1' 0" + ] + Value { + TQ "HiFi" + PlaybackPCM "plughw:0,0" + } +} + +SectionDevice."HDMI-stereo" { + #Name "HDMI-stereo" + Comment "HDMI Digital Stereo Output" + + EnableSequence [ + cdev "hw:0" + cset "name='HDMI Mixer MultiMedia1' 1" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='HDMI Mixer MultiMedia1' 0" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/DB820c/HiFi b/src/conf/ucm/DB820c/HiFi new file mode 100644 index 00000000..4457329f --- /dev/null +++ b/src/conf/ucm/DB820c/HiFi @@ -0,0 +1,110 @@ +# Use case configuration for DB820c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + + EnableSequence [ + cdev "hw:0" + cset "name='SLIM RX0 MUX' ZERO" + cset "name='SLIM RX1 MUX' ZERO" + cset "name='SLIM RX2 MUX' ZERO" + cset "name='SLIM RX3 MUX' ZERO" + cset "name='SLIM RX4 MUX' ZERO" + cset "name='SLIM RX5 MUX' AIF4_PB" + cset "name='SLIM RX6 MUX' AIF4_PB" + cset "name='SLIM RX7 MUX' ZERO" + cset "name='RX INT1_2 MUX' RX5" + cset "name='RX INT2_2 MUX' RX6" + ## gain to 0dB + cset "name='RX5 Digital Volume' 68" + ## gain to 0dB + cset "name='RX6 Digital Volume' 68" + cset "name='SLIMBUS_6_RX Audio Mixer MultiMedia2' 1" + cset "name='MultiMedia3 Mixer SLIMBUS_0_TX' 1" + cset "name='RX INT1 DEM MUX' CLSH_DSM_OUT" + cset "name='RX INT2 DEM MUX' CLSH_DSM_OUT" + cset "name='AIF1_CAP Mixer SLIM TX0' 1" + cset "name='SLIM TX0 MUX' DEC0" + cset "name='ADC2 Volume' 12" + cset "name='ADC MUX0' AMIC" + cset "name='AMIC MUX0' ADC2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='SLIMBUS_6_RX Audio Mixer MultiMedia2' 0" + cset "name='MultiMedia3 Mixer SLIMBUS_0_TX' 0" + ] + + # ALSA PCM + Value { + # ALSA PCM device for HiFi + PlaybackPCM "plughw:0,1" + CapturePCM "plughw:0,2" + } +} + +SectionDevice."Headphones" { + Comment "Headphones playback" + + EnableSequence [ + cdev "hw:0" + cset "name='SLIM RX0 MUX' ZERO" + cset "name='SLIM RX1 MUX' ZERO" + cset "name='SLIM RX2 MUX' ZERO" + cset "name='SLIM RX3 MUX' ZERO" + cset "name='SLIM RX4 MUX' ZERO" + cset "name='SLIM RX5 MUX' AIF4_PB" + cset "name='SLIM RX6 MUX' AIF4_PB" + cset "name='SLIM RX7 MUX' ZERO" + cset "name='RX INT1_2 MUX' RX5" + cset "name='RX INT2_2 MUX' RX6" + ## gain to 0dB + cset "name='RX5 Digital Volume' 68" + ## gain to 0dB + cset "name='RX6 Digital Volume' 68" + cset "name='SLIMBUS_6_RX Audio Mixer MultiMedia2' 1" + cset "name='RX INT1 DEM MUX' CLSH_DSM_OUT" + cset "name='RX INT2 DEM MUX' CLSH_DSM_OUT" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='RX5 Digital Volume' 0" + cset "name='RX6 Digital Volume' 0" + cset "name='SLIM RX5 MUX' ZERO" + cset "name='SLIM RX6 MUX' ZERO" + cset "name='SLIMBUS_6_RX Audio Mixer MultiMedia2' 0" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Handset" { + Comment "Headset Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='MultiMedia3 Mixer SLIMBUS_0_TX' 1" + cset "name='AIF1_CAP Mixer SLIM TX0' 1" + cset "name='SLIM TX0 MUX' DEC0" + cset "name='ADC2 Volume' 12" + cset "name='ADC MUX0' AMIC" + cset "name='AMIC MUX0' ADC2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='MultiMedia3 Mixer SLIMBUS_0_TX' 0" + cset "name='AIF1_CAP Mixer SLIM TX0' 0" + cset "name='AMIC MUX0' ZERO" + cset "name='SLIM TX0 MUX' ZERO" + cset "name='ADC2 Volume' 0" + ] + + Value { + CaptureChannels "1" + } +} diff --git a/src/conf/ucm/DB820c/Makefile.am b/src/conf/ucm/DB820c/Makefile.am new file mode 100644 index 00000000..16e985e5 --- /dev/null +++ b/src/conf/ucm/DB820c/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/DB820c +ucm_DATA = DB820c.conf HDMI HiFi +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index ee850ee6..e9f88ed6 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -31,6 +31,7 @@ chtrt5650 \ cube-i1_TF-Defaultstring-CherryTrailCR \ DAISY-I2S \ DB410c \ +DB820c \ Dell-WD15-Dock \ GoogleNyan \ gpd-win-pocket-rt5645 \ -- 2.20.1

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