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

[PATCH v2] ASoC: wm8960: Remove bitclk relax condition in wm8960_configure_sysclk
by Shengjiu Wang 25 Mar '21

25 Mar '21

The call sequence in wm8960_configure_clocking is ret = wm8960_configure_sysclk(); if (ret >= 0) goto configure_clock; .... ret = wm8960_configure_pll(); configure_clock: ... wm8960_configure_sysclk is called before wm8960_configure_pll, as there is bitclk relax on both functions, so wm8960_configure_sysclk always return success, then wm8960_configure_pll() never be called. With this case: aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav the required bitclk is 48000 * 24 * 2 = 2304000, bitclk got from wm8960_configure_sysclk is 3072000, but if go to wm8960_configure_pll. it can get correct bitclk 2304000. So bitclk relax condition should be removed in wm8960_configure_sysclk, then wm8960_configure_pll can be called, and there is also bitclk relax function in wm8960_configure_pll. Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock computation") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- changes in v2: - Find the real reason for the case, just patially remove bitclk relax sound/soc/codecs/wm8960.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..847ca16b9841 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -608,10 +608,6 @@ static const int bclk_divs[] = { * - lrclk = sysclk / dac_divs * - 10 * bclk = sysclk / bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @wm8960: codec private data * @mclk: MCLK used to derive sysclk * @sysclk_idx: sysclk_divs index for found sysclk @@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, { int sysclk, bclk, lrclk; int i, j, k; - int diff, closest = mclk; + int diff; /* marker for no match */ *bclk_idx = -1; @@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, *bclk_idx = k; break; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - } } if (k != ARRAY_SIZE(bclk_divs)) break; -- 2.27.0

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[PATCH 0/3] ASoC: soc.h: small cleanups
by Kuninori Morimoto 25 Mar '21

25 Mar '21

Hi Mark These are small cleanups for soc.h Kuninori Morimoto (3): ASoC: soc.h: add asoc_link_to_cpu/codec/platform() macro ASoC: soc.h: fixup return timing for snd_soc_fixup_dai_links_platform_name() ASoC: soc.h: return error if multi platform at snd_soc_fixup_dai_links_platform_name() include/sound/soc.h | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH v2 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> --- v2: - Only return early when ret < 0. 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..5a2d9a768f70 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 < 0) + return ret; + ucontrol->value.integer.value[0] = val; return 0; } -- 2.30.2

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Directions please...
by GitHub issues - opened 25 Mar '21

25 Mar '21

alsa-project/alsa-utils issue #84 was opened from giggywithit: Hello I design and build audio equipment and would like to know where or how I would go about getting access to the sound states to control a GPIO for a raspberry Pi. What I would like ideally is to control a GPIO state based upon actual non-zero data playback. My desire is to use the GPIO to control a mute function to silence the interactive periods in DAC activity. I know little to nothing about programming but imagine the function is intelligent enough to discern the difference between an open device and actual music playback. Other methods keep open the device to keep from switching off and on, however, I am looking for a more professional and detailed method to create. Any pointers helpful thank you S Issue URL : https://github.com/alsa-project/alsa-utils/issues/84 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH AUTOSEL 4.4 05/10] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index a3dd7030f629..321b1ac52bfd 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -78,7 +78,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

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[PATCH AUTOSEL 4.4 04/10] ASoC: rt5651: 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 eee51df776bd6cac10a76b2779a9fdee3f622b2b ] 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/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 1d4031818966..883b93f0bd38 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -286,9 +286,9 @@ static bool rt5651_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 4.4 03/10] 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 b1c8bb39cdf1..db7734e45dd1 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -341,9 +341,9 @@ 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 4.9 12/13] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 635818fcda00..21a007c26407 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3389,12 +3389,17 @@ static int rt5659_set_dai_sysclk(struct snd_soc_dai *dai, struct snd_soc_codec *codec = dai->codec; struct rt5659_priv *rt5659 = snd_soc_codec_get_drvdata(codec); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

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[PATCH AUTOSEL 4.9 05/13] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0c2a1413a8f5..14e564e38f3c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -75,7 +75,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

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[PATCH AUTOSEL 4.9 04/13] ASoC: rt5651: 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 eee51df776bd6cac10a76b2779a9fdee3f622b2b ] 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/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f0c9e2562474..acd5c30694e8 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -287,9 +287,9 @@ static bool rt5651_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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