November 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/4] ASoC: ti: davinci-mcasp: Handle incomplete DT node gracefully
by Peter Ujfalusi 09 Nov '20

09 Nov '20

Hi, The series is inspired by the effort to standardize TI's arm64 dtsi files to keep all nodes in 'okay' state and let the board dts files disable not needed peripherals (and not properly configured): https://lore.kernel.org/lkml/20201104224356.18040-1-nm@ti.com/ In the unlikely (or likely?) event when the dts misses to disable the McASP node which is not configured we currenly and luckily just manage to not crash as we had fixup code in place for legacy pdata boots. This however prints out a message which does not really help to identify the issue. This series will reduce some of the noise during boot (first patch) then adds the needed changes to handle the variations of 'okay': A - have all required DT properties for audio B - gpiochip is enabled A && !B -> everything is OK for audio, no gpiochip registered A && B -> everything is OK for audio, gpiochip is registered !A && B -> audio is not OK, gpiochip is registered. dev_dbg() for audio and do not register SOC DAI and PCM !A && !B -> audio is not OK, no gpiochip. dev_err() and fail the probe Regards, Peter --- Peter Ujfalusi (4): ASoC: ti: davinci-mcasp: Use platform_get_irq_byname_optional ASoC: ti: davinci-mcasp: Remove legacy dma_request parsing ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling ASoC: ti: davinci-mcasp: Handle missing required DT properties sound/soc/ti/davinci-mcasp.c | 294 ++++++++++++++--------------------- 1 file changed, 119 insertions(+), 175 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[PATCH] ASoC: rt1015: modification for calibration to get better performance
by jack.yu＠realtek.com 09 Nov '20

09 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Modification for calibration to get better performance. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 101 ++++++++++++++++++-------------------- 1 file changed, 48 insertions(+), 53 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 52787fbd522f..32d748cc5ae3 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -497,18 +497,40 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015) snd_soc_dapm_mutex_lock(&component->dapm); regcache_cache_bypass(regmap, true); - regmap_write(regmap, RT1015_PWR1, 0xd7df); - regmap_write(regmap, RT1015_PWR4, 0x00b2); - regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2008); + regmap_write(regmap, RT1015_PWR9, 0xAA60); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0089); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008A); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008C); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008D); + regmap_write(regmap, RT1015_PWR4, 0x80B2); + regmap_write(regmap, RT1015_CLASSD_SEQ, 0x5797); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2100); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0100); + regmap_write(regmap, RT1015_PWR5, 0x2175); + regmap_write(regmap, RT1015_MIXER1, 0x005D); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); + regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12F7); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x1205); + msleep(200); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2000); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0180); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x0A05); + msleep(200); + regmap_write(regmap, RT1015_PWR4, 0x00B2); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2028); regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); - regmap_write(regmap, RT1015_GAT_BOOST, 0x0efe); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000d); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000e); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a00); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a01); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a05); - msleep(500); - regmap_write(regmap, RT1015_PWR1, 0x0); + regmap_write(regmap, RT1015_PWR5, 0x0175); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x1721); + regmap_write(regmap, RT1015_CLASSD_SEQ, 0x570E); + regmap_write(regmap, RT1015_MIXER1, 0x203D); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5A01); + regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12FF); + regmap_write(regmap, RT1015_GAT_BOOST, 0x0eFE); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008E); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0088); + regmap_write(regmap, RT1015_SYS_RST1, 0x05F5); + regmap_write(regmap, RT1015_SYS_RST2, 0x0b9a); regcache_cache_bypass(regmap, false); regcache_mark_dirty(regmap); @@ -603,6 +625,8 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b0a); snd_soc_component_write(component, RT1015_GAT_BOOST, 0xacfe); snd_soc_component_write(component, @@ -610,10 +634,14 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, snd_soc_component_write(component, RT1015_GAT_BOOST, 0xecfe); } else { + snd_soc_component_write(component, + 0x032d, 0xaa60); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x026e); + RT1015_SYS_RST2, 0x0b0a); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x008e); } break; @@ -627,11 +655,17 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, RT1015_PWR9, 0xa800); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f5); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b9a); } else { snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x0268); + 0x032d, 0xaa60); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x0088); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f5); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b9a); } rt1015->dac_is_used = 0; @@ -664,38 +698,12 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, } static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("LDO2", RT1015_PWR1, RT1015_PWR_LDO2_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("INT RC CLK", RT1015_PWR1, RT1015_PWR_INTCLK_BIT, - 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("ISENSE", RT1015_PWR1, RT1015_PWR_ISENSE_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("VSENSE", RT1015_PWR1, RT1015_PWR_VSENSE_BIT, 0, - NULL, 0), SND_SOC_DAPM_SUPPLY("PLL", RT1015_PWR1, RT1015_PWR_PLL_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("BG1 BG2", RT1015_PWR1, RT1015_PWR_BG_1_2_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("MBIAS BG", RT1015_PWR1, RT1015_PWR_MBIAS_BG_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("VBAT", RT1015_PWR1, RT1015_PWR_VBAT_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("MBIAS", RT1015_PWR1, RT1015_PWR_MBIAS_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("ADCV", RT1015_PWR1, RT1015_PWR_ADCV_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("MIXERV", RT1015_PWR1, RT1015_PWR_MIXERV_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("SUMV", RT1015_PWR1, RT1015_PWR_SUMV_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("VREFLV", RT1015_PWR1, RT1015_PWR_VREFLV_BIT, 0, - NULL, 0), - SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC_E("DAC", NULL, RT1015_PWR1, RT1015_PWR_DAC_BIT, 0, + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, rt1015_amp_drv_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPO"), @@ -703,19 +711,7 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { static const struct snd_soc_dapm_route rt1015_dapm_routes[] = { { "DAC", NULL, "AIFRX" }, - { "DAC", NULL, "LDO2" }, { "DAC", NULL, "PLL", rt1015_is_sys_clk_from_pll}, - { "DAC", NULL, "INT RC CLK" }, - { "DAC", NULL, "ISENSE" }, - { "DAC", NULL, "VSENSE" }, - { "DAC", NULL, "BG1 BG2" }, - { "DAC", NULL, "MBIAS BG" }, - { "DAC", NULL, "VBAT" }, - { "DAC", NULL, "MBIAS" }, - { "DAC", NULL, "ADCV" }, - { "DAC", NULL, "MIXERV" }, - { "DAC", NULL, "SUMV" }, - { "DAC", NULL, "VREFLV" }, { "Amp Drv", NULL, "DAC" }, { "SPO", NULL, "Amp Drv" }, }; @@ -1058,7 +1054,6 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; rt1015->cali_done = 0; - snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); -- 2.29.0

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[PATCH] ASoC: fsl_aud2htx: Remove dev_err() usage after platform_get_irq()
by Shengjiu Wang 09 Nov '20

09 Nov '20

platform_get_irq() would print error message internally, so dev_err() after platform_get_irq() is not needed Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_aud2htx.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 124aeb70f24e..4091ccc7c3e9 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -211,11 +211,8 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) } irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", - dev_name(&pdev->dev)); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, fsl_aud2htx_isr, 0, dev_name(&pdev->dev), aud2htx); -- 2.27.0

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[PATCH v1 0/2] Support SC7180 sound card variation without headset
by xuyuqing 09 Nov '20

09 Nov '20

This series is based on patch [v5,2/2] ASoC: qcom: sc7180: Modify machine driver for 2mic Use compatible string to let machine driver handle the case where ALC5682 headset codec is replaced with ADAU7002 dmic codec xuyuqing (2): ASoC: google: dt-bindings: add new compatible for sc7180-coachz ASoC: qcom: sc7180: Modify machine driver for adau7002 .../bindings/sound/google,sc7180-trogdor.yaml | 4 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7180.c | 119 ++++++++++++++++-- 3 files changed, 112 insertions(+), 12 deletions(-) -- 2.25.1

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[PATCH] ASoC: Intel: Skylake: Add alternative topology binary name
by Mateusz Gorski 09 Nov '20

09 Nov '20

[ Upstream commit 1b290ef023b3eeb4f4688b582fecb773915ef937 ] Add alternative topology binary file name based on used machine driver and fallback to use this name after failed attempt to load topology file with name based on NHLT. This change addresses multiple issues with current mechanism, for example - there are devices without NHLT table, and that currently results in tplg_name being empty. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200427132727.24942-2-mateusz.gorski@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- This functionality is merged on upstream kernel and widely used. Merging it to LTS kernel would improve the user experience and resolve some of the problems regarding topology naming that the users are facing. sound/soc/intel/skylake/skl-topology.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..4b114ece58c6 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -14,6 +14,7 @@ #include <linux/uuid.h> #include <sound/intel-nhlt.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-topology.h> #include <uapi/sound/snd_sst_tokens.h> #include <uapi/sound/skl-tplg-interface.h> @@ -3565,8 +3566,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->mach->drv_name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3588,8 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index -- 2.17.1

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[PATCH] ALSA: usb-audio: Fix coding style problems
by Atul 08 Nov '20

08 Nov '20

From: Atul Gopinathan <leoatul12(a)gmail.com> Hi, This patch fixes some coding style warnings and errors that were generated using scripts/checkpatch.pl. Signed-off-by: Atul Gopinathan <leoatul12(a)gmail.com> --- sound/usb/pcm.c | 16 ++++++++-------- sound/usb/quirks.c | 21 +++++++++++---------- 2 files changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a860303cc522..ed5c77ce8e14 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -104,7 +104,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs) continue; } attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; - if (! found) { + if (!found) { found = fp; cur_attr = attr; continue; @@ -304,7 +304,7 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, if (altsd->bAlternateSetting != altsetting || altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC || (altsd->bInterfaceSubClass != 2 && - altsd->bInterfaceProtocol != 2 ) || + altsd->bInterfaceProtocol != 2) || altsd->bNumEndpoints < 1) return -ENOENT; epd = get_endpoint(*alts, 0); @@ -482,8 +482,9 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress - because there is a version of the SB Audigy 2 NX firmware lacking - the audio fields in the endpoint descriptors */ + * because there is a version of the SB Audigy 2 NX firmware lacking + * the audio fields in the endpoint descriptors + */ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0)) { @@ -911,7 +912,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) struct usb_interface *iface; int ret; - if (! subs->cur_audiofmt) { + if (!subs->cur_audiofmt) { dev_err(&subs->dev->dev, "no format is specified!\n"); return -ENXIO; } @@ -973,9 +974,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return ret; } -static const struct snd_pcm_hardware snd_usb_hardware = -{ - .info = SNDRV_PCM_INFO_MMAP | +static const struct snd_pcm_hardware snd_usb_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_INTERLEAVED | diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index c989ad8052ae..b6f3345db93f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -398,7 +398,7 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. */ static int create_uaxx_quirk(struct snd_usb_audio *chip, @@ -593,7 +593,7 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_NEW) { dev_dbg(&dev->dev, "sending Extigy boot sequence...\n"); /* Send message to force it to reconnect with full interface. */ - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0), + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x10, 0x43, 0x0001, 0x000a, NULL, 0); if (err < 0) dev_dbg(&dev->dev, "error sending boot message: %d\n", err); @@ -646,8 +646,9 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) "error usb_driver_set_configuration: %d\n", err); /* Always return an error, so that we stop creating a device - that will just be destroyed and recreated with a new - configuration */ + * that will just be destroyed and recreated with a new + * configuration + */ return -ENODEV; } else dev_info(&dev->dev, "Fast Track Pro config OK\n"); @@ -895,9 +896,10 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) usb_reset_device(dev); /* return -EAGAIN, so the creation of an audio interface for this - * temporary device is aborted. The device will reconnect with a - * new product ID */ - return -EAGAIN; + * temporary device is aborted. The device will reconnect with a + * new product ID + */ + return -EAGAIN; } static void mbox2_setup_48_24_magic(struct usb_device *dev) @@ -919,7 +921,6 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev) 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003); - return; } /* Digidesign Mbox 2 needs to load firmware onboard @@ -1830,8 +1831,8 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; - case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ - case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ + case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ + case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is an older model 77d:223) */ -- 2.25.1

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ThinkPad support headset button
by Kailang 07 Nov '20

07 Nov '20

Hi Takashi, Two ThinkPad want to support headset button. Attach was support patch. Many Thanks. BR, Kailang

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Re: [Sound-open-firmware] linux-next: Tree for Nov 6 (sound/soc/sof/sof-pci-dev.o)
by Pierre-Louis Bossart 06 Nov '20

06 Nov '20

> on x86_64: > > ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe': > sof-pci-dev.c:(.text+0x5c): undefined reference to `snd_intel_dsp_driver_probe' > > > Full randconfig file is attached. Nice catch, thanks Randy! Looks like we put the select SND_INTEL_DSP_CONFIG in the wrong place, it's not dependent on the HDaudio link being selected. Clearly a bug we've had for a while. The diff below makes the error go away but I have to run it past folks who are already enjoying their week-end. Will follow-up next week with a proper fix. diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index a066e08860cb..5bfc2f8b13b9 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -271,6 +271,7 @@ config SND_SOC_SOF_JASPERLAKE config SND_SOC_SOF_HDA_COMMON tristate + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_HDA_LINK_BASELINE help @@ -330,7 +331,6 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level

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[PATCH v3 1/2] ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module
by Shengjiu Wang 06 Nov '20

06 Nov '20

AUD2HTX (Audio Subsystem TO HDMI TX Subsystem) is a new IP module found on i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3: - Add additionalProperties changes in v2: - fix indentation issue - remove nodename .../bindings/sound/fsl,aud2htx.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000000..aa4be7170717 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; -- 2.27.0

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[PATCH 0/7] PinePhone BT audio bringup
by Samuel Holland 06 Nov '20

06 Nov '20

This series makes use of the additional DAIs recently added to the sun8i-codec driver to add hardware routing for BT SCO (headset) audio on the PinePhone. The BT audio connection is represented by the "dummy" bt-sco codec. The connection to the Quectel EG-25G modem via AIF2 works as well, but I do not include it here because there is no appropriate codec driver in tree. We have been using an out-of-tree "dummy" codec driver similar to bt-sco, and I'm not sure if such a driver would be desired upstream. The modem has a similar amount of configurability as the rtl8723cs BT chip. For the BT chip, the DAI format and PCM parameters are set in a configuration firmware file and loaded at driver load time. For the modem, the DAI format and PCM parameters are set by (and can be queried from) an AT command. However, this AT command requires a modem restart to take effect, so the parameters cannot feasibly be changed at runtime. With a dummy driver, we pick some "standard" set of PCM parameters, e.g. 16 bit mono at 8 or 16 kHz, and assume the hardware agrees. Similarly, we assume the DAI format pre-programmed in the remote hardware matches what is configured in the DAI link (in this case, in the device tree). Is that the right way to model this? Or does the modem (and maybe even the BT chip) deserve a more featureful driver that reads and/or programs the format and params at probe time? Because of those unanswered questions, I'm sending BT SCO support first/only. Regards, Samuel Arnaud Ferraris (1): arm64: dts: allwinner: pinephone: Set audio card name Samuel Holland (6): ASoC: dt-bindings: sun8i-codec: Increase #sound-dai-cells ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Add pinmux nodes for AIF2/AIF3 arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: pinephone: Add support for Bluetooth audio .../sound/allwinner,sun8i-a33-codec.yaml | 2 +- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 25 +++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 37 ++++++++++++++----- 4 files changed, 55 insertions(+), 13 deletions(-) -- 2.26.2

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