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

[PATCH v2 0/2] ASoC topology header parsing refinement
by Keyon Jie 27 May '20

27 May '20

This small series is to optimize the header logging during the topology parsing. This is verified work fine on both SOF and SST drivers (bxt_rt298 with dfw_sst.bin). Change History: v2: - Change the internal used array to be 'static' to fix the issue reported by: kbuild test robot <lkp(a)intel.com> - Add testing coverage including Intel SST driver also. v1: - Initial version. Keyon Jie (2): ASoC: topology: refine and log the header in the correct pass ASoC: topology: remove the redundant pass checks sound/soc/soc-topology.c | 99 ++++++++++++++++++---------------------- 1 file changed, 44 insertions(+), 55 deletions(-) -- 2.25.1

3 10

[PATCH] ASoC: Intel: sst: Fix runtime PM imbalance in sst_power_control
by Dinghao Liu 27 May '20

27 May '20

When sst_load_fw() returns an error code, a pairing runtime PM usage counter decrement is needed to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 762495385d5c..3897985b254f 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -150,6 +150,7 @@ static int sst_power_control(struct device *dev, bool state) if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { ret = sst_load_fw(ctx); if (ret) { + pm_runtime_put_sync(dev); dev_err(dev, "FW download fail %d\n", ret); sst_set_fw_state_locked(ctx, SST_RESET); ret = sst_pm_runtime_put(ctx); -- 2.17.1

3 2

ucm2: Add profile for Chromebook Asus C300
by GitHub pull_request - opened 26 May '20

26 May '20

alsa-project/alsa-ucm-conf pull request #26 was opened from tiwai: ASUS Chromebook C300 alias Google QUAWKS is an Intel Baytrail platform with max98090 codec. This patch adds the basic UCM snippet for the max98090 codec and HiFi.conf for this model. Note that MIC2 is used for the headset mic. If another model with this codec uses a different ADC port, we'd need to create another profile snippet and rename the device. BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1171492 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/26 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/26.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] [v2] ASoC: ti: Fix runtime PM imbalance in omap2_mcbsp_set_clks_src
by Dinghao Liu 26 May '20

26 May '20

When clk_set_parent() returns an error code, a pairing runtime PM usage counter increment is needed to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- Changelog: v2: - Fix this in a cleaner way. --- sound/soc/ti/omap-mcbsp.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 3d41ca2238d4..d7406cc3c2a5 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -77,18 +77,15 @@ static int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) pm_runtime_put_sync(mcbsp->dev); r = clk_set_parent(mcbsp->fclk, fck_src); - if (r) { + if (r) dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", src); - clk_put(fck_src); - return r; - } pm_runtime_get_sync(mcbsp->dev); clk_put(fck_src); - return 0; + return r; } static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) -- 2.17.1

3 2

[PATCH] ASoC: fix semicolon.cocci warnings
by kbuild test robot 26 May '20

26 May '20

From: kbuild test robot <lkp(a)intel.com> sound/soc/codecs/zl38060.c:298:2-3: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Fixes: 52e8a94baf90 ("ASoC: Add initial ZL38060 driver") CC: Sven Van Asbroeck <thesven73(a)gmail.com> Signed-off-by: kbuild test robot <lkp(a)intel.com> --- tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.8 head: f202272cabf276441174dc05ad8b94d3c1174877 commit: 52e8a94baf9026276fcdc9ff21a50dc2ca0bc94b [26/131] ASoC: Add initial ZL38060 driver zl38060.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/codecs/zl38060.c +++ b/sound/soc/codecs/zl38060.c @@ -295,7 +295,7 @@ static int zl38_hw_params(struct snd_pcm break; default: return -EINVAL; - }; + } err = regmap_update_bits(priv->regmap, REG_TDMA_CFG_CLK, CFG_CLK_FSRATE_MASK, fsrate);

2 1

[PATCH] ASoC: tas2552: Fix runtime PM imbalance in tas2552_component_probe
by Dinghao Liu 26 May '20

26 May '20

pm_runtime_get_sync() increments the runtime PM usage counter even when it returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/codecs/tas2552.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 56671f21cfe5..d90e5f2b6f27 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -602,6 +602,7 @@ static int tas2552_component_probe(struct snd_soc_component *component) return 0; probe_fail: + pm_runtime_put_noidle(component->dev); gpiod_set_value(tas2552->enable_gpio, 0); regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), -- 2.17.1

2 1

[PATCH] ALSA: hwdep: fix a left shifting 1 by 31 UB bug
by Changming Liu 26 May '20

26 May '20

The "info.index" variable can be 31 in "1 << info.index". This might trigger an undefined behavior since 1 is signed. Fix this by casting 1 to 1u just to be sure "1u << 31" is defined. Signed-off-by: Changming Liu <liu.changm(a)northeastern.edu> --- sound/core/hwdep.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index b412d3b3d5ff..21edb8ac95eb 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -216,12 +216,12 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, if (info.index >= 32) return -EINVAL; /* check whether the dsp was already loaded */ - if (hw->dsp_loaded & (1 << info.index)) + if (hw->dsp_loaded & (1u << info.index)) return -EBUSY; err = hw->ops.dsp_load(hw, &info); if (err < 0) return err; - hw->dsp_loaded |= (1 << info.index); + hw->dsp_loaded |= (1u << info.index); return 0; } -- 2.17.1

2 3

[PATCH] ALSA: hda/realtek - Add a model for Thinkpad T570 without DAC workaround
by Takashi Iwai 26 May '20

26 May '20

We fixed the regression of the speaker volume for some Thinkpad models (e.g. T570) by the commit 54947cd64c1b ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570"). Essentially it fixes the DAC / pin pairing by a static table. It was confirmed and merged to stable kernel later. Now, interestingly, we got another regression report for the very same model (T570) about the similar problem, and the commit above was the culprit. That is, by some reason, there are devices that prefer the DAC1, and another device DAC2! Unfortunately those have the same ID and we have no idea what can differentiate, in this patch, a new fixup model "tpt470-dock-fix" is provided, so that users with such a machine can apply it manually. When model=tpt470-dock-fix option is passed to snd-hda-intel module, it avoids the fixed DAC pairing and the DAC1 is assigned to the speaker like the earlier versions. Fixes: 54947cd64c1b ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570") BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1172017 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++++++++++++++---------- 1 file changed, 26 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 041d2a32059b..92c6e58c3862 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5484,18 +5484,9 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, { 0x19, 0x21a11010 }, /* dock mic */ { } }; - /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise - * the speaker output becomes too low by some reason on Thinkpads with - * ALC298 codec - */ - static const hda_nid_t preferred_pairs[] = { - 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, - 0 - }; struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.preferred_dacs = preferred_pairs; spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; snd_hda_apply_pincfgs(codec, pincfgs); } else if (action == HDA_FIXUP_ACT_INIT) { @@ -5508,6 +5499,23 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, } } +static void alc_fixup_tpt470_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise + * the speaker output becomes too low by some reason on Thinkpads with + * ALC298 codec + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.preferred_dacs = preferred_pairs; +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -6063,6 +6071,7 @@ enum { ALC700_FIXUP_INTEL_REFERENCE, ALC274_FIXUP_DELL_BIND_DACS, ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + ALC298_FIXUP_TPT470_DOCK_FIX, ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, @@ -6994,12 +7003,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC274_FIXUP_DELL_BIND_DACS }, - [ALC298_FIXUP_TPT470_DOCK] = { + [ALC298_FIXUP_TPT470_DOCK_FIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_tpt470_dock, .chained = true, .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE }, + [ALC298_FIXUP_TPT470_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt470_dacs, + .chained = true, + .chain_id = ALC298_FIXUP_TPT470_DOCK_FIX + }, [ALC255_FIXUP_DUMMY_LINEOUT_VERB] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7638,6 +7653,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, + {.id = ALC298_FIXUP_TPT470_DOCK_FIX, .name = "tpt470-dock-fix"}, {.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, -- 2.16.4

2 2

[PATCH] ALSA: usb-audio: Quirks for Gigabyte TRX40 Aorus Master onboard audio
by Takashi Iwai 26 May '20

26 May '20

Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices, a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC (USB ID 0414:a000). The latter serves solely for the headphone output on the front panel while the former serves for the rest I/Os (mostly for the I/Os in the rear panel but also including the front mic). Both chips do work more or less with the unmodified USB-audio driver, but there are a few glitches. The ALC1220-VB returns an error for an inquiry to some jacks, as already seen on other TRX40-based mobos. However this machine has a slightly incompatible configuration, hence the existing mapping cannot be used as is. Meanwhile the ESS chip seems working without any quirk. But since both audio devices don't provide any specific names, both cards appear as "USB-Audio", and it's quite confusing for users. This patch is an attempt to overcome those issues: - The specific mapping table for ALC1220-VB is provided, reducing the non-working nodes and renaming the badly chosen controls. The connector map isn't needed here unlike other TRX40 quirks. - For both USB IDs (0414:a000 and 0414:a001), provide specific card name strings, so that user-space can identify more easily; and more importantly, UCM profile can be applied to each. Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 19 +++++++++++++++++++ sound/usb/quirks-table.h | 25 +++++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index bfdc6ad52785..9af7aa93f6fa 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -397,6 +397,21 @@ static const struct usbmix_connector_map trx40_mobo_connector_map[] = { {} }; +/* Rear panel + front mic on Gigabyte TRX40 Aorus Master with ALC1220-VB */ +static const struct usbmix_name_map aorus_master_alc1220vb_map[] = { + { 17, NULL }, /* OT, IEC958?, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Line Out" }, /* OT */ + { 22, "Line Out Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + {} +}; + /* * Control map entries */ @@ -526,6 +541,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1b1c, 0x0a42), .map = corsair_virtuoso_map, }, + { /* Gigabyte TRX40 Aorus Master (rear panel + front mic) */ + .id = USB_ID(0x0414, 0xa001), + .map = aorus_master_alc1220vb_map, + }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), .map = trx40_mobo_map, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 6313c30f5c85..eb89902a83be 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3566,4 +3566,29 @@ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP +/* Two entries for Gigabyte TRX40 Aorus Master: + * TRX40 Aorus Master has two USB-audio devices, one for the front headphone + * with ESS SABRE9218 DAC chip, while another for the rest I/O (the rear + * panel and the front mic) with Realtek ALC1220-VB. + * Here we provide two distinct names for making UCM profiles easier. + */ +{ + USB_DEVICE(0x0414, 0xa000), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Gigabyte", + .product_name = "Aorus Master Front Headphone", + .profile_name = "Gigabyte-Aorus-Master-Front-Headphone", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ + USB_DEVICE(0x0414, 0xa001), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Gigabyte", + .product_name = "Aorus Master Main Audio", + .profile_name = "Gigabyte-Aorus-Master-Main-Audio", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 2.16.4

1 0

[PATCH] ALSA: usb: mixer: volume quirk for ESS Technology Asus USB DAC
by Chris Chiu 26 May '20

26 May '20

The Asus USB DAC is a USB type-C audio dongle for connecting to the headset and headphone. The volume minimum value -23040 which is 0xa600 in hexadecimal with the resolution value 1 indicates this should be endianness issue caused by the firmware bug. Add a volume quirk to fix the volume control problem. Also fixes this warning: Warning! Unlikely big volume range (=23040), cval->res is probably wrong. [5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1 Warning! Unlikely big volume range (=23040), cval->res is probably wrong. [7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1 Signed-off-by: Chris Chiu <chiu(a)endlessm.com> --- sound/usb/mixer.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81b2db0edd5f..0b568a9fdab3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1171,6 +1171,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; + case USB_ID(0x0495, 0x3042): /* ESS Technology Asus USB DAC */ + if ((strstr(kctl->id.name, "Playback Volume") != NULL) || + strstr(kctl->id.name, "Capture Volume") != NULL) { + cval->min >>= 8; + cval->max = 0; + cval->res = 1; + } + break; } } -- 2.20.1

2 1