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

[alsa-devel] [PATCH] ALSA: hda: Add Clevo W65_67SB the power_save blacklist
by Hans de Goede 26 Jan '20

26 Jan '20

Using HDA power-saving on the Clevo W65_67SB causes the first 0.5 seconds of audio to be missing every time audio starts playing. This commit adds the Clevo W65_67SB the power_save blacklist to avoid this issue. Cc: stable(a)vger.kernel.org BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5b92f290cbb0..54d9ea1750f9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2185,6 +2185,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1558, 0x6504, "Clevo W65_67SB", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ -- 2.23.0

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[alsa-devel] [PATCH v2] ASoC: cros_ec_codec: Support setting bclk ratio
by Yu-Hsuan Hsu 26 Jan '20

26 Jan '20

Support setting bclk ratio from machine drivers. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 6a24f570c5e86f..d3dc42aa682565 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -45,6 +45,9 @@ struct cros_ec_codec_priv { /* DMIC */ atomic_t dmic_probed; + /* I2S_RX */ + uint32_t i2s_rx_bclk_ratio; + /* WoV */ bool wov_enabled; uint8_t *wov_audio_shm_p; @@ -259,6 +262,7 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, snd_soc_component_get_drvdata(component); struct ec_param_ec_codec_i2s_rx p; enum ec_codec_i2s_rx_sample_depth depth; + uint32_t bclk; int ret; if (params_rate(params) != 48000) @@ -284,15 +288,29 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - dev_dbg(component->dev, "set bclk to %u\n", - snd_soc_params_to_bclk(params)); + if (priv->i2s_rx_bclk_ratio) + bclk = params_rate(params) * priv->i2s_rx_bclk_ratio; + else + bclk = snd_soc_params_to_bclk(params); + + dev_dbg(component->dev, "set bclk to %u\n", bclk); p.cmd = EC_CODEC_I2S_RX_SET_BCLK; - p.set_bclk_param.bclk = snd_soc_params_to_bclk(params); + p.set_bclk_param.bclk = bclk; return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); } +static int i2s_rx_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct cros_ec_codec_priv *priv = + snd_soc_component_get_drvdata(component); + + priv->i2s_rx_bclk_ratio = ratio; + return 0; +} + static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; @@ -340,6 +358,7 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops i2s_rx_dai_ops = { .hw_params = i2s_rx_hw_params, .set_fmt = i2s_rx_set_fmt, + .set_bclk_ratio = i2s_rx_set_bclk_ratio, }; static int i2s_rx_event(struct snd_soc_dapm_widget *w, -- 2.25.0.341.g760bfbb309-goog

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[alsa-devel] [PATCH v2] ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
by Nicola Lunghi 26 Jan '20

26 Jan '20

With firmware 2.82 Line6 changed the usb id of some of the Helix devices but the quirks is still needed. Add it to the quirk list for line6 helix family of devices. Thanks to Jens for pointing out the missing ids. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db71305f6..53922f73467f 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -296,6 +296,9 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0E41, 0x4248): /* Line6 Helix >= fw 2.82 */ + case USB_ID(0x0E41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ + case USB_ID(0x0E41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ /* supported rates: 48Khz */ kfree(fp->rate_table); fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: add quirks for Line6 Helix fw 2.82
by Nicola Lunghi 25 Jan '20

25 Jan '20

Line6 Helix fw 2.82 still needs a quirk to set the audio rate to 48khz but in respect to older firmware has changed the USB ID of the device. Add it to the quirk list for line6 helix family of devices. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db71305f6..e2b98f181d05 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -296,6 +296,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0E41, 0x4248): /* Line6 Helix >= fw 2.82 */ /* supported rates: 48Khz */ kfree(fp->rate_table); fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); -- 2.20.1

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Re: [alsa-devel] Line 6 Helix quirks
by nick83ola 25 Jan '20

25 Jan '20

Hi I verified on my hx_stomp with the latest driver and I can confirm tha the quirks works here Sadly I don't have an helix to verify it What model it is? you can try to recompile the kernel adding the line with you usb device to the see my patch https://lkml.org/lkml/2019/7/1/850 + case USB_ID(0x0E41, 0x4244): /* HELIX */ + case USB_ID(0x0E41, 0x4246): /* HX STOMP */ + case USB_ID(0x0E41, 0x4248): /* what kind of HELIX ???????? */ Cheers Nicola Lunghi On Sat, 19 Oct 2019 at 06:34, Carlo Calica <carlo(a)calica.com> wrote: > > Hi, > > I'm attempting to use my Helix with the newish 5.3.x kernel. It isn't working. Looks like the quirk isn't enabled for the idProduct (4248) of my Helix. My Helix's firmware is 2.81.0. I have yet to compile the kernel to see if adding the id is enough. Thanks. If you need any more info, please let me know. > > dmesg gives: > > [89014.296954] usb 3-2: new high-speed USB device number 2 using xhci_hcd > [89014.521720] usb 3-2: New USB device found, idVendor=0e41, idProduct=4248, bcdDevice= 2.00 > [89014.521723] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [89014.521724] usb 3-2: Product: HELIX > [89014.521726] usb 3-2: Manufacturer: LINE 6 > [89014.521727] usb 3-2: SerialNumber: 2796112 > [89014.555715] usb 3-2: parse_audio_format_rates_v2v3(): unable to retrieve number of sample rates (clock 16) > [89014.559724] usb 3-2: parse_audio_format_rates_v2v3(): unable to retrieve number of sample rates (clock 16) > [89014.575990] input: LINE 6 HELIX as /devices/pci0000:00/0000:00:01.3/0000:03:00.2/0000:16:08.0/0000:1c:00.0/usb3/3-2/3-2:1.5/0003:0E41:4248.000C/input/input50 > [89014.628069] hid-generic 0003:0E41:4248.000C: input,hidraw8: USB HID v1.11 Keyboard [LINE 6 HELIX ] on usb-0000:1c:00.0-2/input5 > > > -- > Carlo J. Calica

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[alsa-devel] [PATCH] ASoC: rt1015: Remove unnecessary const
by Nathan Chancellor 24 Jan '20

24 Jan '20

Clang warns: ../sound/soc/codecs/rt1015.c:392:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL(rt1015_boost_mode_enum, 0, 0, ^ ../include/sound/soc.h:355:2: note: expanded from macro 'SOC_ENUM_SINGLE_DECL' SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) ^ ../include/sound/soc.h:352:2: note: expanded from macro 'SOC_ENUM_DOUBLE_DECL' const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ^ 1 warning generated. Remove the const after static to fix it. Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver") Link: https://github.com/ClangBuiltLinux/linux/issues/845 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/codecs/rt1015.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 4a9c5b54008f..6d490e2dbc25 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -389,7 +389,7 @@ static const char * const rt1015_boost_mode[] = { "Bypass", "Adaptive", "Fixed Adaptive" }; -static const SOC_ENUM_SINGLE_DECL(rt1015_boost_mode_enum, 0, 0, +static SOC_ENUM_SINGLE_DECL(rt1015_boost_mode_enum, 0, 0, rt1015_boost_mode); static int rt1015_boost_mode_get(struct snd_kcontrol *kcontrol, -- 2.25.0

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[alsa-devel] [PATCH AUTOSEL 4.14 13/32] ASoC: sti: fix possible sleep-in-atomic
by Sasha Levin 24 Jan '20

24 Jan '20

From: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> [ Upstream commit ce780a47c3c01e1e179d0792df6b853a913928f1 ] Change mutex and spinlock management to avoid sleep in atomic issue. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Link: https://lore.kernel.org/r/20200113100400.30472-1-arnaud.pouliquen@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sti/uniperif_player.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d8b6936e544e3..908f13623f8cd 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -226,7 +226,6 @@ static void uni_player_set_channel_status(struct uniperif *player, * sampling frequency. If no sample rate is already specified, then * set one. */ - mutex_lock(&player->ctrl_lock); if (runtime) { switch (runtime->rate) { case 22050: @@ -303,7 +302,6 @@ static void uni_player_set_channel_status(struct uniperif *player, player->stream_settings.iec958.status[3 + (n * 4)] << 24; SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status); } - mutex_unlock(&player->ctrl_lock); /* Update the channel status */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -365,8 +363,10 @@ static int uni_player_prepare_iec958(struct uniperif *player, SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player); + mutex_lock(&player->ctrl_lock); /* Update the channel status */ uni_player_set_channel_status(player, runtime); + mutex_unlock(&player->ctrl_lock); /* Clear the user validity user bits */ SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); @@ -598,7 +598,6 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[1] = ucontrol->value.iec958.status[1]; iec958->status[2] = ucontrol->value.iec958.status[2]; iec958->status[3] = ucontrol->value.iec958.status[3]; - mutex_unlock(&player->ctrl_lock); spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) @@ -608,6 +607,8 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, uni_player_set_channel_status(player, NULL); spin_unlock_irqrestore(&player->irq_lock, flags); + mutex_unlock(&player->ctrl_lock); + return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 12/32] ASoC: msm8916-wcd-analog: Fix selected events for MIC BIAS External1
by Sasha Levin 24 Jan '20

24 Jan '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit e0beec88397b163c7c4ea6fcfb67e8e07a2671dc ] MIC BIAS External1 sets pm8916_wcd_analog_enable_micbias_ext1() as event handler, which ends up in pm8916_wcd_analog_enable_micbias_ext(). But pm8916_wcd_analog_enable_micbias_ext() only handles the POST_PMU event, which is not specified in the event flags for MIC BIAS External1. This means that the code in the event handler is never actually run. Set SND_SOC_DAPM_POST_PMU as the only event for the handler to fix this. Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Link: https://lore.kernel.org/r/20200111164006.43074-2-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 969283737787f..3633eb30dd135 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -876,10 +876,10 @@ static const struct snd_soc_dapm_widget pm8916_wcd_analog_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("MIC BIAS External1", CDC_A_MICB_1_EN, 7, 0, pm8916_wcd_analog_enable_micbias_ext1, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MIC BIAS External2", CDC_A_MICB_2_EN, 7, 0, pm8916_wcd_analog_enable_micbias_ext2, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_ADC_E("ADC1", NULL, CDC_A_TX_1_EN, 7, 0, pm8916_wcd_analog_enable_adc, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 28/56] ASoC: msm8916-wcd-digital: Reset RX interpolation path after use
by Sasha Levin 24 Jan '20

24 Jan '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit 85578bbd642f65065039b1765ebe1a867d5435b0 ] For some reason, attempting to route audio through QDSP6 on MSM8916 causes the RX interpolation path to get "stuck" after playing audio a few times. In this situation, the analog codec part is still working, but the RX path in the digital codec stops working, so you only hear the analog parts powering up. After a reboot everything works again. So far I was not able to reproduce the problem when using lpass-cpu. The downstream kernel driver avoids this by resetting the RX interpolation path after use. In mainline we do something similar for the TX decimator (LPASS_CDC_CLK_TX_RESET_B1_CTL), but the interpolator reset (LPASS_CDC_CLK_RX_RESET_CTL) got lost when the msm8916-wcd driver was split into analog and digital. Fix this problem by adding the reset to msm8916_wcd_digital_enable_interpolator(). Fixes: 150db8c5afa1 ("ASoC: codecs: Add msm8916-wcd digital codec") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Link: https://lore.kernel.org/r/20200105102753.83108-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-digital.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 3063dedd21cff..6de2ab6f97068 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -357,6 +357,12 @@ static int msm8916_wcd_digital_enable_interpolator( snd_soc_component_write(component, rx_gain_reg[w->shift], snd_soc_component_read32(component, rx_gain_reg[w->shift])); break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, LPASS_CDC_CLK_RX_RESET_CTL, + 1 << w->shift, 1 << w->shift); + snd_soc_component_update_bits(component, LPASS_CDC_CLK_RX_RESET_CTL, + 1 << w->shift, 0x0); + break; } return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 27/56] ASoC: msm8916-wcd-analog: Fix MIC BIAS Internal1
by Sasha Levin 24 Jan '20

24 Jan '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit 057efcf9faea4769cf1020677d93d040db9b23f3 ] MIC BIAS Internal1 is broken at the moment because we always enable the internal rbias resistor to the TX2 line (connected to the headset microphone), rather than enabling the resistor connected to TX1. Move the RBIAS code to pm8916_wcd_analog_enable_micbias_int1/2() to fix this. Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Link: https://lore.kernel.org/r/20200111164006.43074-3-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 6083b44ce1c6c..cbdb6d4bb91ef 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -388,9 +388,6 @@ static int pm8916_wcd_analog_enable_micbias_int(struct snd_soc_component switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS, - MICB_1_INT_TX2_INT_RBIAS_EN_MASK, - MICB_1_INT_TX2_INT_RBIAS_EN_ENABLE); snd_soc_component_update_bits(component, reg, MICB_1_EN_PULL_DOWN_EN_MASK, 0); snd_soc_component_update_bits(component, CDC_A_MICB_1_EN, MICB_1_EN_OPA_STG2_TAIL_CURR_MASK, @@ -440,6 +437,14 @@ static int pm8916_wcd_analog_enable_micbias_int1(struct struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct pm8916_wcd_analog_priv *wcd = snd_soc_component_get_drvdata(component); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS, + MICB_1_INT_TX1_INT_RBIAS_EN_MASK, + MICB_1_INT_TX1_INT_RBIAS_EN_ENABLE); + break; + } + return pm8916_wcd_analog_enable_micbias_int(component, event, w->reg, wcd->micbias1_cap_mode); } @@ -550,6 +555,11 @@ static int pm8916_wcd_analog_enable_micbias_int2(struct struct pm8916_wcd_analog_priv *wcd = snd_soc_component_get_drvdata(component); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS, + MICB_1_INT_TX2_INT_RBIAS_EN_MASK, + MICB_1_INT_TX2_INT_RBIAS_EN_ENABLE); + break; case SND_SOC_DAPM_POST_PMU: pm8916_mbhc_configure_bias(wcd, true); break; -- 2.20.1

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