March 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.11 10/44] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1414ad15d01c..a5674c227b3a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ -- 2.30.1

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[PATCH AUTOSEL 5.11 09/44] ASoC: rt1015: fix i2c communication error
by Sasha Levin 25 Mar '21

25 Mar '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 9e0bdaa9fcb8c64efc1487a7fba07722e7bc515e ] Remove 0x100 cache re-sync to solve i2c communication error. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 32e6bcf763d1..4607039a16e7 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -209,6 +209,7 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_VENDOR_ID: case RT1015_DEVICE_ID: case RT1015_PRO_ALT: + case RT1015_MAN_I2C: case RT1015_DAC3: case RT1015_VBAT_TEST_OUT1: case RT1015_VBAT_TEST_OUT2: -- 2.30.1

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[PATCH V2] ALSA: pcm: Fix couple of typos
by Bhaskar Chowdhury 25 Mar '21

25 Mar '21

s/unconditonally/unconditionally/ s/succesful/successful/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- Changes from V1: Randy's finding incorporated ,plus the subject line adjusted. sound/core/pcm_native.c | 82 ++++++++++++++++++++--------------------- 1 file changed, 41 insertions(+), 41 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 17a85f4815d5..afb670d7fd53 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1425,7 +1425,7 @@ static int snd_pcm_do_stop(struct snd_pcm_substream *substream, substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP); substream->runtime->stop_operating = true; } - return 0; /* unconditonally stop all substreams */ + return 0; /* unconditionally stop all substreams */ } static void snd_pcm_post_stop(struct snd_pcm_substream *substream, @@ -1469,7 +1469,7 @@ EXPORT_SYMBOL(snd_pcm_stop); * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. * - * Return: Zero if succesful, or a negative error code. + * Return: Zero if successful, or a negative error code. -- 2.30.1

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[PATCH 1/2] ALSA: usb-audio: Carve out connector value checking into a helper
by Kai-Heng Feng 25 Mar '21

25 Mar '21

This is preparation for next patch, no functional change intended. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/mixer.c | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b004b2e63a5d..98f5417a70e4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1446,13 +1446,11 @@ static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, return 0; } -/* get the connectors status and report it as boolean type */ -static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int get_connector_value(struct usb_mixer_elem_info *cval, + char *name, int *val) { - struct usb_mixer_elem_info *cval = kcontrol->private_data; struct snd_usb_audio *chip = cval->head.mixer->chip; - int idx = 0, validx, ret, val; + int idx = 0, validx, ret; validx = cval->control << 8 | 0; @@ -1467,21 +1465,24 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac2_conn, sizeof(uac2_conn)); - val = !!uac2_conn.bNrChannels; + if (val) + *val = !!uac2_conn.bNrChannels; } else { /* UAC_VERSION_3 */ struct uac3_insertion_ctl_blk uac3_conn; ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, &uac3_conn, sizeof(uac3_conn)); - val = !!uac3_conn.bmConInserted; + if (val) + *val = !!uac3_conn.bmConInserted; } snd_usb_unlock_shutdown(chip); if (ret < 0) { - if (strstr(kcontrol->id.name, "Speaker")) { - ucontrol->value.integer.value[0] = 1; + if (name && strstr(name, "Speaker")) { + if (val) + *val = 1; return 0; } error: @@ -1491,6 +1492,21 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, return filter_error(cval, ret); } + return ret; +} + +/* get the connectors status and report it as boolean type */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + int ret, val; + + ret = get_connector_value(cval, kcontrol->id.name, &val); + + if (ret) + return ret; + ucontrol->value.integer.value[0] = val; return 0; } -- 2.30.2

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[PATCH] ALSA: hda/hdmi: fix max DP-MST dev_num for Intel TGL+ platforms
by Kai Vehmanen 25 Mar '21

25 Mar '21

Increase the device select range to 4 on platforms supporting 4 concurrent displays. This fixes a problem in scenario where total of 4 displays are active, and 3 of these are audio capable DP receivers and connected to a DP-MST hub. Due to incorrect range for device select, audio could not be played to the 3rd monitor in DP-MST hub. BugLink: https://github.com/thesofproject/linux/issues/2798 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 25 +++++++++++-------------- 1 file changed, 11 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 45ae845e82df..5de3666a7101 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1848,16 +1848,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) */ if (spec->intel_hsw_fixup) { /* - * On Intel platforms, device entries number is - * changed dynamically. If there is a DP MST - * hub connected, the device entries number is 3. - * Otherwise, it is 1. - * Here we manually set dev_num to 3, so that - * we can initialize all the device entries when - * bootup statically. + * On Intel platforms, device entries count returned + * by AC_PAR_DEVLIST_LEN is dynamic, and depends on + * the type of receiver that is connected. Allocate pin + * structures based on worst case. */ - dev_num = 3; - spec->dev_num = 3; + dev_num = spec->dev_num; } else if (spec->dyn_pcm_assign && codec->dp_mst) { dev_num = snd_hda_get_num_devices(codec, pin_nid) + 1; /* @@ -2942,7 +2938,7 @@ static int parse_intel_hdmi(struct hda_codec *codec) /* Intel Haswell and onwards; audio component with eld notifier */ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, - const int *port_map, int port_num) + const int *port_map, int port_num, int dev_num) { struct hdmi_spec *spec; int err; @@ -2957,6 +2953,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, spec->port_map = port_map; spec->port_num = port_num; spec->intel_hsw_fixup = true; + spec->dev_num = dev_num; intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); @@ -2982,12 +2979,12 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, static int patch_i915_hsw_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x08, NULL, 0); + return intel_hsw_common_init(codec, 0x08, NULL, 0, 3); } static int patch_i915_glk_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x0b, NULL, 0); + return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3); } static int patch_i915_icl_hdmi(struct hda_codec *codec) @@ -2998,7 +2995,7 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) */ static const int map[] = {0x0, 0x4, 0x6, 0x8, 0xa, 0xb}; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3); } static int patch_i915_tgl_hdmi(struct hda_codec *codec) @@ -3010,7 +3007,7 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; int ret; - ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4); if (!ret) { struct hdmi_spec *spec = codec->spec; base-commit: 950dc7fe37b99e77160154f6ce767b987816deaf -- 2.29.2

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Re: [PATCH] ALSA: usb-audio: Fix missing return assignment
by Dan Carpenter 25 Mar '21

25 Mar '21

On Thu, Mar 25, 2021 at 01:24:23AM +0500, Muhammad Usama Anjum wrote: > On Wed, 2021-03-24 at 21:50 +0300, Dan Carpenter wrote: > > > - usb_driver_claim_interface(driver, iface, (void *)-1L); > > > + err = usb_driver_claim_interface(driver, iface, (void *)-1L); > > > > This is in a loop so only the last return value is used. Which seems > > sort of weird and pointless that the last value would matter more than > > the others. > > > Correct. Lets not store the return value. To stop the static analyzers > to report the missing return assignment, can we add (void) in start of > this function call? I've not seen use of (void) this way in the > kernel. Is there any other way used in the kernel? Don't add (void). Don't add any code just to help static checkers, only do it if it helps humans. The (void) stuff is ugly. We have a __must_check annotation for functions where it's a bug not to check the return and the usb_driver_claim_interface() is not a __must_check function. Just ignore the static checker when it's wrong. When I'm reviewing static checker warnings, I only look at the new ones. Then after I've looked at them, I mark them as old. I currently have 65k old ignored warnings. regards, dan carpenter

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[PATCH] ALSA: pcm: A typo fix
by Bhaskar Chowdhury 25 Mar '21

25 Mar '21

s/unconditonally/unconditionally/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/core/pcm_native.c | 80 ++++++++++++++++++++--------------------- 1 file changed, 40 insertions(+), 40 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 17a85f4815d5..9884961bca7d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c - return 0; /* unconditonally stop all substreams */ + return 0; /* unconditionally stop all substreams */ } -- 2.30.1

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[PATCH v4 resend 00/13] MFD/extcon/ASoC: Rework arizona codec jack-detect support
by Hans de Goede 25 Mar '21

25 Mar '21

Hi All, Here is v4 of my series to rework the arizona codec jack-detect support to use the snd_soc_jack helpers instead of direct extcon reporting. As discussed before here is a resend rebased on 5.12-rc2, making sure that all patches this depends on are in place. Lee, can you pick-up patches 1-6 through the MFD tree and then send a pull-req to Mark so that Mark can merge the Asoc parts throught the ASoC tree ? Patches 2-6 touch drivers/extcon, these all have an Ack from Chanwoo Choi for merging these through the MFD tree. Here is some more generic info on this series from the previous cover-letter: This is done by reworking the extcon driver into an arizona-jackdet library and then modifying the codec drivers to use that directly, replacing the old separate extcon child-devices and extcon-driver. This brings the arizona-codec jack-detect handling inline with how all other ASoC codec driver do this. This was developed and tested on a Lenovo Yoga Tablet 1051L with a WM5102 codec. This was also tested by Charles Keepax, one of the Cirrus Codec folks. Regards, Hans Hans de Goede (13): mfd: arizona: Drop arizona-extcon cells extcon: arizona: Fix some issues when HPDET IRQ fires after the jack has been unplugged extcon: arizona: Fix various races on driver unbind extcon: arizona: Fix flags parameter to the gpiod_get("wlf,micd-pol") call extcon: arizona: Always use pm_runtime_get_sync() when we need the device to be awake ASoC/extcon: arizona: Move arizona jack code to sound/soc/codecs/arizona-jack.c ASoC: arizona-jack: Move jack-detect variables to struct arizona_priv ASoC: arizona-jack: Use arizona->dev for runtime-pm ASoC: arizona-jack: convert into a helper library for codec drivers ASoC: arizona-jack: Use snd_soc_jack to report jack events ASoC: arizona-jack: Cleanup logging ASoC: arizona: Make the wm5102, wm5110, wm8997 and wm8998 drivers use the new jack library ASoC: Intel: bytcr_wm5102: Add jack detect support MAINTAINERS | 3 +- drivers/extcon/Kconfig | 8 - drivers/extcon/Makefile | 1 - drivers/mfd/arizona-core.c | 20 - sound/soc/codecs/Makefile | 2 +- .../soc/codecs/arizona-jack.c | 577 +++++++----------- sound/soc/codecs/arizona.h | 44 ++ sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 12 +- sound/soc/codecs/wm8997.c | 14 +- sound/soc/codecs/wm8998.c | 9 + sound/soc/intel/boards/bytcr_wm5102.c | 28 +- 12 files changed, 325 insertions(+), 405 deletions(-) rename drivers/extcon/extcon-arizona.c => sound/soc/codecs/arizona-jack.c (76%) -- 2.30.1

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[PATCH v2 0/2] ASoC: intel: atom: 2 bug-fixes for the atom SST driver
by Hans de Goede 25 Mar '21

25 Mar '21

Hi All, Here is v2 of my bugfix series for the ASoC Intel Atom SST driver. The patches are unchanged except for adding Pierre-Louis' Acked-By and adding a fixed (subject fixed) version of the Fixes: tag which Pierre-Louis suggested. The first patch fixes a bug which is causing audio to now work with pipewire and both Fedora and Arch Linux are moving towards using pipewire as the default sound-server. As such it would be good if we can get these 2 bugfixes added to a 5.12-rc# release. Regards, Hans Hans de Goede (2): ASoC: intel: atom: Stop advertising non working S24LE support ASoC: intel: atom: Remove 44100 sample-rate from the media and deep-buffer DAI descriptions sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) -- 2.30.2

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[PATCH 1/2] ASoC: intel: atom: Stop advertising non working S24LE support
by Hans de Goede 25 Mar '21

25 Mar '21

The SST firmware's media and deep-buffer inputs are hardcoded to S16LE, the corresponding DAIs don't have a hw_params callback and their prepare callback also does not take the format into account. So far the advertising of non working S24LE support has not caused issues because pulseaudio defaults to S16LE, but changing pulse-audio's config to use S24LE will result in broken sound. Pipewire is replacing pulse now and pipewire prefers S24LE over S16LE when available, causing the problem of the broken S24LE support to come to the surface now. Cc: stable(a)vger.kernel.org BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/866 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 9e9b05883557..aa5dd590ddd5 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -488,14 +488,14 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Headset Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -506,7 +506,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { -- 2.30.1

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