From b6aa06de7757667bac88997a8807b143b8436035 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 29 Jun 2020 14:24:43 +0200 Subject: ASoC: qcom: Drop HAS_DMA dependency to fix link failure When building on allyesconfig kernel for a NO_DMA=y platform (e.g. Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n, leading to a link failure: sound/soc/qcom/common.o: In function `qcom_snd_parse_of': common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port' While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845 don't, so the following warning is seen: WARNING: unmet direct dependencies detected for SND_SOC_QDSP6 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n] Selected by [y]: - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y] Until recently, this warning was harmless (from a compile-testing point-of-view), but the new user of q6afe_is_rx_port() turned this into a hard failure. As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is safe to increase compile testing coverage. Hence fix the link failure by dropping the HAS_DMA dependency of SND_SOC_QDSP6. Fixes: a2120089251f1fe2 ("ASoC: qcom: common: set correct directions for dailinks") Fixes: 6b1687bf76ef84cb ("ASoC: qcom: add sdm845 sound card support") Fixes: a6f933f63f2ffdb2 ("ASoC: qcom: apq8096: Add db820c machine driver") Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20200629122443.21736-1-geert@linux-m68k.org Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d..92f51d0e9fe2 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" - depends on QCOM_APR && HAS_DMA + depends on QCOM_APR select SND_SOC_QDSP6_COMMON select SND_SOC_QDSP6_CORE select SND_SOC_QDSP6_AFE -- cgit From 0ceb8a36d023d4bb4ffca3474a452fb1dfaa0ef2 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 28 Jun 2020 17:52:26 +0200 Subject: ASoC: Intel: cht_bsw_rt5672: Change bus format to I2S 2 channel The default mode for SSP configuration is TDM 4 slot and so far we were using this for the bus format on cht-bsw-rt56732 boards. One board, the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The second piggy-backed, output-only codec is inside the keyboard-dock (which has extra speakers). Unlike the main rt5672 codec, we cannot configure this codec, it is hard coded to use 2 channel 24 bit I2S. Using 4 channel TDM leads to the dock speakers codec (which listens in on the data send from the SSP to the rt5672 codec) emiting horribly distorted sound. Since we only support 2 channels anyways, there is no need for TDM on any cht-bsw-rt5672 designs. So we can simply use I2S 2ch everywhere. This commit fixes the Lenovo Miix 2 10 dock speakers issue by changing the bus format set in cht_codec_fixup() to I2S 2 channel. This change has been tested on the following devices with a rt5672 codec: Lenovo Miix 2 10 Lenovo Thinkpad 8 Lenovo Thinkpad 10 (gen 1) Signed-off-by: Hans de Goede Cc: stable@vger.kernel.org BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723 Link: https://lore.kernel.org/r/20200628155231.71089-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/cht_bsw_rt5672.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 7a43c70a1378..22e432768edb 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,21 +253,20 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot + * Default mode for SSP configuration is TDM 4 slot. One board/design, + * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The + * second piggy-backed, output-only codec is inside the keyboard-dock + * (which has extra speakers). Unlike the main rt5672 codec, we cannot + * configure this codec, it is hard coded to use 2 channel 24 bit I2S. + * Since we only support 2 channels anyways, there is no need for TDM + * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. */ - ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_IB_NF | + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) { - dev_err(rtd->dev, "can't set format to TDM %d\n", ret); - return ret; - } - - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; } -- cgit From 5cacc6f5764e94fa753b2c1f5f7f1f3f74286e82 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 28 Jun 2020 17:52:27 +0200 Subject: ASoC: rt5670: Correct RT5670_LDO_SEL_MASK The RT5670_PWR_ANLG1 register has 3 bits to select the LDO voltage, so the correct mask is 0x7 not 0x3. Because of this wrong mask we were programming the ldo bits to a setting of binary 001 (0x05 & 0x03) instead of binary 101 when moving to SND_SOC_BIAS_PREPARE. According to the datasheet 001 is a reserved value, so no idea what it did, since the driver was working fine before I guess we got lucky and it does something which is ok. Fixes: 5e8351de740d ("ASoC: add RT5670 CODEC driver") Signed-off-by: Hans de Goede Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20200628155231.71089-3-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index a8c3e44770b8..de0203369b7c 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -757,7 +757,7 @@ #define RT5670_PWR_VREF2_BIT 4 #define RT5670_PWR_FV2 (0x1 << 3) #define RT5670_PWR_FV2_BIT 3 -#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_MASK (0x7) #define RT5670_LDO_SEL_SFT 0 /* Power Management for Analog 2 (0x64) */ -- cgit From 85ca6b17e2bb96b19caac3b02c003d670b66de96 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 28 Jun 2020 17:52:28 +0200 Subject: ASoC: rt5670: Add new gpio1_is_ext_spk_en quirk and enable it on the Lenovo Miix 2 10 The Lenovo Miix 2 10 has a keyboard dock with extra speakers in the dock. Rather then the ACL5672's GPIO1 pin being used as IRQ to the CPU, it is actually used to enable the amplifier for these speakers (the IRQ to the CPU comes directly from the jack-detect switch). Add a quirk for having an ext speaker-amplifier enable pin on GPIO1 and replace the Lenovo Miix 2 10's dmi_system_id table entry's wrong GPIO_DEV quirk (which needs to be renamed to GPIO1_IS_IRQ) with the new RT5670_GPIO1_IS_EXT_SPK_EN quirk, so that we enable the external speaker-amplifier as necessary. Also update the ident field for the dmi_system_id table entry, the Miix models are not Thinkpads. Fixes: 67e03ff3f32f ("ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk") Signed-off-by: Hans de Goede BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723 Link: https://lore.kernel.org/r/20200628155231.71089-4-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 71 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 56 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 70fee6849ab0..f21181734170 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -31,18 +31,19 @@ #include "rt5670.h" #include "rt5670-dsp.h" -#define RT5670_DEV_GPIO BIT(0) -#define RT5670_IN2_DIFF BIT(1) -#define RT5670_DMIC_EN BIT(2) -#define RT5670_DMIC1_IN2P BIT(3) -#define RT5670_DMIC1_GPIO6 BIT(4) -#define RT5670_DMIC1_GPIO7 BIT(5) -#define RT5670_DMIC2_INR BIT(6) -#define RT5670_DMIC2_GPIO8 BIT(7) -#define RT5670_DMIC3_GPIO5 BIT(8) -#define RT5670_JD_MODE1 BIT(9) -#define RT5670_JD_MODE2 BIT(10) -#define RT5670_JD_MODE3 BIT(11) +#define RT5670_DEV_GPIO BIT(0) +#define RT5670_IN2_DIFF BIT(1) +#define RT5670_DMIC_EN BIT(2) +#define RT5670_DMIC1_IN2P BIT(3) +#define RT5670_DMIC1_GPIO6 BIT(4) +#define RT5670_DMIC1_GPIO7 BIT(5) +#define RT5670_DMIC2_INR BIT(6) +#define RT5670_DMIC2_GPIO8 BIT(7) +#define RT5670_DMIC3_GPIO5 BIT(8) +#define RT5670_JD_MODE1 BIT(9) +#define RT5670_JD_MODE2 BIT(10) +#define RT5670_JD_MODE3 BIT(11) +#define RT5670_GPIO1_IS_EXT_SPK_EN BIT(12) static unsigned long rt5670_quirk; static unsigned int quirk_override; @@ -1447,6 +1448,33 @@ static int rt5670_hp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5670_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component); + + if (!rt5670->pdata.gpio1_is_ext_spk_en) + return 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_HI); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_LO); + break; + + default: + return 0; + } + + return 0; +} + static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1860,7 +1888,9 @@ static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = { - SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + rt5670_spk_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), SND_SOC_DAPM_OUTPUT("SPORP"), @@ -2857,14 +2887,14 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { }, { .callback = rt5670_quirk_cb, - .ident = "Lenovo Thinkpad Tablet 10", + .ident = "Lenovo Miix 2 10", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"), }, .driver_data = (unsigned long *)(RT5670_DMIC_EN | RT5670_DMIC1_IN2P | - RT5670_DEV_GPIO | + RT5670_GPIO1_IS_EXT_SPK_EN | RT5670_JD_MODE2), }, { @@ -2924,6 +2954,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, rt5670->pdata.dev_gpio = true; dev_info(&i2c->dev, "quirk dev_gpio\n"); } + if (rt5670_quirk & RT5670_GPIO1_IS_EXT_SPK_EN) { + rt5670->pdata.gpio1_is_ext_spk_en = true; + dev_info(&i2c->dev, "quirk GPIO1 is external speaker enable\n"); + } if (rt5670_quirk & RT5670_IN2_DIFF) { rt5670->pdata.in2_diff = true; dev_info(&i2c->dev, "quirk IN2_DIFF\n"); @@ -3023,6 +3057,13 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); } + if (rt5670->pdata.gpio1_is_ext_spk_en) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_GPIO1); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); + } + if (rt5670->pdata.jd_mode) { regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK, RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); -- cgit From 3f31f7d9b5404a10648abe536c8b408bfb4502e1 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 28 Jun 2020 17:52:29 +0200 Subject: ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10 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. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20200628155231.71089-5-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index f21181734170..dfbc0ca38ff7 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -603,9 +603,9 @@ int rt5670_set_jack_detect(struct snd_soc_component *component, EXPORT_SYMBOL_GPL(rt5670_set_jack_detect); 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 */ -- cgit From bc4be656471b39af8f2ad57ee372012c55da1da7 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 1 Jul 2020 15:16:45 +0800 Subject: ASoC: rt5682: cancel jack_detect_work if hs_jack is set to null even soundwire mode Base on https://patchwork.kernel.org/patch/11237953/ Soundwire mode also should follow it. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200701071645.32061-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7d6670abdb08..dd741835e4d0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -992,16 +992,17 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, rt5682->hs_jack = hs_jack; - if (!rt5682->is_sdw) { - if (!hs_jack) { - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - cancel_delayed_work_sync(&rt5682->jack_detect_work); - return 0; - } + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + return 0; + } + + if (!rt5682->is_sdw) { switch (rt5682->pdata.jd_src) { case RT5682_JD1: snd_soc_component_update_bits(component, -- cgit From ad922ca199b38974dbe392e2faeba3aadf461ac2 Mon Sep 17 00:00:00 2001 From: Ravulapati Vishnu vardhan rao Date: Tue, 30 Jun 2020 14:52:38 +0530 Subject: ASoC: amd: Rectifying Unbalanced pm_runtime_enable! issue When snd_pci_acp3x driver loads we see: WARNING kernel:snd_pci_acp3x 0000:04:00.5: Unbalanced pm_runtime_enable! at boot time. same can be observed in /var/log/messages/. Modifying pm runtime sequence for fixing unbalanced pm issue. Signed-off-by: Ravulapati Vishnu vardhan rao Link: https://lore.kernel.org/r/20200630092242.7799-1-Vishnuvardhanrao.Ravulapati@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/raven/pci-acp3x.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index f25ce50f1a90..ebf4388b6262 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -232,9 +232,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pm_runtime_set_autosuspend_delay(&pci->dev, 2000); pm_runtime_use_autosuspend(&pci->dev); - pm_runtime_set_active(&pci->dev); pm_runtime_put_noidle(&pci->dev); - pm_runtime_enable(&pci->dev); pm_runtime_allow(&pci->dev); return 0; @@ -303,7 +301,7 @@ static void snd_acp3x_remove(struct pci_dev *pci) ret = acp3x_deinit(adata->acp3x_base); if (ret) dev_err(&pci->dev, "ACP de-init failed\n"); - pm_runtime_disable(&pci->dev); + pm_runtime_forbid(&pci->dev); pm_runtime_get_noresume(&pci->dev); pci_disable_msi(pci); pci_release_regions(pci); -- cgit From 1eb96c198aff13162de4857e19f9488d59c4acb1 Mon Sep 17 00:00:00 2001 From: Yu-Hsuan Hsu Date: Tue, 30 Jun 2020 17:16:15 +0800 Subject: ASoC: rockchip: add format and rate constraints on rk3399 S8 and S24 formats does not work on this machine driver so force to use S16_LE instead. In addition, add constraint to limit the max value of rate because the rate higher than 96000(172000, 192000) is not stable either. Signed-off-by: Yu-Hsuan Hsu Link: https://lore.kernel.org/r/20200630091615.4020059-1-yuhsuan@chromium.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rk3399_gru_sound.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index f45e5aaa4b30..9539b0d024fe 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -219,19 +219,32 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, return 0; } +static int rockchip_sound_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + return snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, + 8000, 96000); +} + static const struct snd_soc_ops rockchip_sound_max98357a_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_max98357a_hw_params, }; static const struct snd_soc_ops rockchip_sound_rt5514_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_rt5514_hw_params, }; static const struct snd_soc_ops rockchip_sound_da7219_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_da7219_hw_params, }; static const struct snd_soc_ops rockchip_sound_dmic_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_dmic_hw_params, }; -- cgit From 58f30150ffd6d95efa524ff05bbcee4e95bfa870 Mon Sep 17 00:00:00 2001 From: Maxime Ripard Date: Tue, 7 Jul 2020 09:42:37 +0200 Subject: ASoC: core: Remove only the registered component in devm functions The ASoC devm_ functions that register a component (devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will clean their component by running snd_soc_unregister_component. snd_soc_unregister_component will then remove all the components for the device that was used to register the component in the first place. However, some drivers register several components (such as a DAI and a dmaengine PCM) on the same device, and if the dmaengine PCM is registered first, then the DAI will be cleaned up first and snd_dmaengine_pcm_unregister will be called next. snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component on the device, and if there's one unregister that component and release its dmaengine channels. That doesn't happen in practice though since the first call to snd_soc_unregister_component removed all the components, so we never get the chance to release the dmaengine channels. In order to fix this, instead of removing all the components for a given device, we can simply remove the component that was registered in the first place. We should have the same number of component registration than we have components, so it should work just fine. Signed-off-by: Maxime Ripard Link: https://lore.kernel.org/r/20200707074237.287171-1-maxime@cerno.tech Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 27 +++++++++++++++++++++++++++ sound/soc/soc-devres.c | 8 +++++--- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 3 files changed, 33 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0f30f5aabaa8..2b8abf88ec60 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2572,6 +2572,33 @@ int snd_soc_register_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_register_component); +/** + * snd_soc_unregister_component_by_driver - Unregister component using a given driver + * from the ASoC core + * + * @dev: The device to unregister + * @component_driver: The component driver to unregister + */ +void snd_soc_unregister_component_by_driver(struct device *dev, + const struct snd_soc_component_driver *component_driver) +{ + struct snd_soc_component *component; + + if (!component_driver) + return; + + mutex_lock(&client_mutex); + component = snd_soc_lookup_component_nolocked(dev, component_driver->name); + if (!component) + goto out; + + snd_soc_del_component_unlocked(component); + +out: + mutex_unlock(&client_mutex); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver); + /** * snd_soc_unregister_component - Unregister all related component * from the ASoC core diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 11e5d7962370..4534a1c03e8e 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -48,7 +48,9 @@ EXPORT_SYMBOL_GPL(devm_snd_soc_register_dai); static void devm_component_release(struct device *dev, void *res) { - snd_soc_unregister_component(*(struct device **)res); + const struct snd_soc_component_driver **cmpnt_drv = res; + + snd_soc_unregister_component_by_driver(dev, *cmpnt_drv); } /** @@ -65,7 +67,7 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai) { - struct device **ptr; + const struct snd_soc_component_driver **ptr; int ret; ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); @@ -74,7 +76,7 @@ int devm_snd_soc_register_component(struct device *dev, ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); if (ret == 0) { - *ptr = dev; + *ptr = cmpnt_drv; devres_add(dev, ptr); } else { devres_free(ptr); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 80a4e71f2d95..61844403f181 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -478,7 +478,7 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_component_to_pcm(component); - snd_soc_unregister_component(dev); + snd_soc_unregister_component_by_driver(dev, component->driver); dmaengine_pcm_release_chan(pcm); kfree(pcm); } -- cgit From 25612477d20b522a3203707ff23575b99f639fff Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Jul 2020 16:04:37 -0500 Subject: ASoC: soc-dai: set dai_link dpcm_ flags with a helper Add a helper to walk through all the DAIs and set dpcm_playback and dpcm_capture flags based on the DAIs capabilities, and use this helper to avoid setting these flags arbitrarily in generic cards. The commit referenced in the Fixes tag did not introduce the configuration issue but will prevent the card from probing when detecting invalid configurations. Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20200707210439.115300-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 4 ++-- sound/soc/generic/simple-card.c | 4 ++-- sound/soc/soc-dai.c | 38 ++++++++++++++++++++++++++++++++++++ 3 files changed, 42 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 9ad35d9940fe..97b4f5480a31 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -317,8 +317,8 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (ret < 0) goto out_put_node; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + snd_soc_dai_link_set_capabilities(dai_link); + dai_link->ops = &graph_ops; dai_link->init = asoc_simple_dai_init; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 55e9f8800b3e..04d4d28ed511 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -231,8 +231,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (ret < 0) goto out_put_node; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + snd_soc_dai_link_set_capabilities(dai_link); + dai_link->ops = &simple_ops; dai_link->init = asoc_simple_dai_init; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index b05e18b63a1c..457159975b01 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -391,6 +391,44 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) return stream->channels_min; } +/* + * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs + */ +void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_component *cpu; + struct snd_soc_dai_link_component *codec; + struct snd_soc_dai *dai; + bool supported[SNDRV_PCM_STREAM_LAST + 1]; + int direction; + int i; + + for_each_pcm_streams(direction) { + supported[direction] = true; + + for_each_link_cpus(dai_link, i, cpu) { + dai = snd_soc_find_dai(cpu); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + if (!supported[direction]) + continue; + for_each_link_codecs(dai_link, i, codec) { + dai = snd_soc_find_dai(codec); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + } + + dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; + dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE]; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities); + void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action) { -- cgit From fffebe8a8339c7e56db4126653a3bc0c0c5592cf Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Jul 2020 16:04:38 -0500 Subject: ASoC: Intel: bdw-rt5677: fix non BE conversion When SOF is used, the normal links are converted into DPCM ones. This generates an error [ 58.276668] bdw-rt5677 bdw-rt5677: CPU DAI spi-RT5677AA:00 for rtd Wake on Voice does not support playback [ 58.276676] bdw-rt5677 bdw-rt5677: ASoC: can't create pcm Wake on Voice :-22 Fix by forcing the capture direction. Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Guennadi Liakhovetski Reviewed-by: Curtis Malainey Link: https://lore.kernel.org/r/20200707210439.115300-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 5f96d7ac0a22..bed4d5f73d9c 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -354,6 +354,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { { .name = "Codec DSP", .stream_name = "Wake on Voice", + .capture_only = 1, .ops = &bdw_rt5677_dsp_ops, SND_SOC_DAILINK_REG(dsp), }, -- cgit From 4e7f8cac1171ba369a9209a8d949732a4d3b939a Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 7 Jul 2020 16:04:39 -0500 Subject: ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M This is identical with change for Intel platforms done with commit 8c05246c0b58 ("ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell") and fixes a regression on i.MX8/i.MX8M: [ 25.705750] esai-Codec: ASoC: no backend playback stream [ 27.923378] esai-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Signed-off-by: Daniel Baluta Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20200707210439.115300-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 8 ++++++++ sound/soc/sof/imx/imx8m.c | 8 ++++++++ 2 files changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 63f9c20a1bac..a4fa8451d8cb 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -375,6 +375,14 @@ static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev, static struct snd_soc_dai_driver imx8_dai[] = { { .name = "esai-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index fa86a9e2990f..287114a37688 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -240,6 +240,14 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev, static struct snd_soc_dai_driver imx8m_dai[] = { { .name = "sai-port", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, }, }; -- cgit From 6f0307df83f2aa6bdf656c2219c89ce96502d20e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Jul 2020 15:37:45 -0500 Subject: ASoC: topology: fix kernel oops on route addition error When errors happens while loading graph components, the kernel oopses while trying to remove all topology components. This can be root-caused to a list pointing to memory that was already freed on error. remove_route() is already called on errors and will perform the required cleanups so there's no need to free the route memory in soc_tplg_dapm_graph_elems_load() if the route was added to the list. We do however want to free the routes allocated but not added to the list. Fixes: 7df04ea7a31ea ('ASoC: topology: modify dapm route loading routine and add dapm route unloading') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200707203749.113883-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 43e5745b06aa..f336a9cfc16f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1261,17 +1261,29 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list); ret = soc_tplg_add_route(tplg, routes[i]); - if (ret < 0) + if (ret < 0) { + /* + * this route was added to the list, it will + * be freed in remove_route() so increment the + * counter to skip it in the error handling + * below. + */ + i++; break; + } /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, routes[i], 1); } - /* free memory allocated for all dapm routes in case of error */ - if (ret < 0) - for (i = 0; i < count ; i++) - kfree(routes[i]); + /* + * free memory allocated for all dapm routes not added to the + * list in case of error + */ + if (ret < 0) { + while (i < count) + kfree(routes[i++]); + } /* * free pointer to array of dapm routes as this is no longer needed. -- cgit From 8edac489e7c3fce44208373bb3e7b5835a672c66 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Jul 2020 15:37:46 -0500 Subject: ASoC: topology: fix tlvs in error handling for widget_dmixer we need to free all allocated tlvs, not just the one allocated in the loop before releasing kcontrols - other the tlvs references will leak. Fixes: 9f90af3a995298 ('ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200707203749.113883-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f336a9cfc16f..6eaa00c21011 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1371,7 +1371,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( if (err < 0) { dev_err(tplg->dev, "ASoC: failed to init %s\n", mc->hdr.name); - soc_tplg_free_tlv(tplg, &kc[i]); goto err_sm; } } @@ -1379,6 +1378,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( err_sm: for (; i >= 0; i--) { + soc_tplg_free_tlv(tplg, &kc[i]); sm = (struct soc_mixer_control *)kc[i].private_value; kfree(sm); kfree(kc[i].name); -- cgit From a53bacc04d7e2b813ebe0ca4dae38716c00d7953 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Tue, 7 Jul 2020 15:57:36 -0500 Subject: ASoC: codecs: max98373: Removed superfluous volume control from chip default Volume control in probe function is not necessary. Signed-off-by: Ryan Lee Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200707205740.114927-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 96718e3a1ad0..ec247491e5a9 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -779,13 +779,6 @@ static int max98373_probe(struct snd_soc_component *component) regmap_write(max98373->regmap, MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2, 0x1); - /* Set inital volume (0dB) */ - regmap_write(max98373->regmap, - MAX98373_R203D_AMP_DIG_VOL_CTRL, - 0x00); - regmap_write(max98373->regmap, - MAX98373_R203E_AMP_PATH_GAIN, - 0x00); /* Enable DC blocker */ regmap_write(max98373->regmap, MAX98373_R203F_AMP_DSP_CFG, -- cgit From 0fd3935ef888b7231fde87eba3fdf613c4923b4a Mon Sep 17 00:00:00 2001 From: randerwang Date: Tue, 7 Jul 2020 15:57:37 -0500 Subject: ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend Idle_bias_on is used to decide bias on/off in standby state by dapm. When Idle_bias_on is set to one, dapm will keep max98373 active at idle time. Max98373 is doing nothing in this state, so remove idle_bias_on setting to let max98373 get suspended when it is idle. Signed-off-by: randerwang Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ryan Lee Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200707205740.114927-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index ec247491e5a9..d87402a86c88 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -862,7 +862,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98373 = { .num_dapm_widgets = ARRAY_SIZE(max98373_dapm_widgets), .dapm_routes = max98373_audio_map, .num_dapm_routes = ARRAY_SIZE(max98373_audio_map), - .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, -- cgit From eceb5437ed0d41be5d12af3add58b3be2d5719e5 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 7 Jul 2020 15:40:27 -0500 Subject: ASoC: SOF: core: fix null-ptr-deref bug during device removal The DSP should be notified for device removal only if the probe was successful. Fixes the following KASAN bug: BUG: KASAN: null-ptr-deref in sof_ipc_tx_message+0x80/0x160 [snd_sof] Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200707204027.114169-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 339c4930b0c0..adc7c37145d6 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev) struct snd_sof_pdata *pdata = sdev->pdata; int ret; - ret = snd_sof_dsp_power_down_notify(sdev); - if (ret < 0) - dev_warn(dev, "error: %d failed to prepare DSP for device removal", - ret); - if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) { + ret = snd_sof_dsp_power_down_notify(sdev); + if (ret < 0) + dev_warn(dev, "error: %d failed to prepare DSP for device removal", + ret); + snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); -- cgit From 574ea5c80eb18edd0d93864985650efec63347c0 Mon Sep 17 00:00:00 2001 From: Puyou Lu Date: Thu, 2 Jul 2020 10:30:25 +0800 Subject: ASoC: wm8974: fix Boost Mixer Aux Switch Clear BIT6 of INPPGA means not muted (Switch On). Signed-off-by: Puyou Lu Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1593657025-4903-1-git-send-email-puyou.lu@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 06ba36595ddd..764bf93fb58a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0), /* Boost mixer */ static const struct snd_kcontrol_new wm8974_boost_mixer[] = { -SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1), }; /* Input PGA */ -- cgit From 01283d56f0ea0040b64dc785542f3ad3fb8b3e68 Mon Sep 17 00:00:00 2001 From: Puyou Lu Date: Thu, 2 Jul 2020 10:30:56 +0800 Subject: ASoC: wm8974: remove unsupported clock mode In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie. inverted frame clock is not support in this mode). Signed-off-by: Puyou Lu Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1593657056-4989-1-git-send-email-puyou.lu@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 764bf93fb58a..7cfc89602fc3 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -474,6 +474,10 @@ static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0008; break; case SND_SOC_DAIFMT_DSP_A: + if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF || + (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) { + return -EINVAL; + } iface |= 0x00018; break; default: -- cgit From 12eb3ad0638c2a6af72de866e9d7837de16ee82f Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Thu, 9 Jul 2020 18:13:45 +0800 Subject: ASoC: rt286: fix unexpected interrupt happens The HV/VREF should not turn off if the headphone jack plug-in. This patch could solve the unexpected interrupt issue in some devices. Signed-off-by: Shuming Fan Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200709101345.11449-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9593a9a27bf8..e8d14eefc41b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - if (!*mic) { + + if (!*hp) { snd_soc_dapm_disable_pin(dapm, "HV"); snd_soc_dapm_disable_pin(dapm, "VREF"); - } - if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + } return 0; } -- cgit From fa291331cb24bd9665096d660b917998285aae17 Mon Sep 17 00:00:00 2001 From: "derek.fang" Date: Tue, 14 Jul 2020 18:13:20 +0800 Subject: ASoC: rt5682: Enable Vref2 under using PLL2 Enable Vref2 under long term using PLL2 to avoid clock unstable. Signed-off-by: derek.fang Link: https://lore.kernel.org/r/1594721600-29994-1-git-send-email-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index dd741835e4d0..5adfaf3a7134 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -967,13 +967,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); - else + RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1609,8 +1608,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ @@ -2493,6 +2491,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, RT5682_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2, + RT5682_PWR_VREF2); + usleep_range(55000, 60000); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); @@ -2518,9 +2525,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); if (!rt5682->jack_type) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2 | RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); -- cgit From b3df80ab6d147d4738be242e1c91e5fdbb6b03ef Mon Sep 17 00:00:00 2001 From: Jing Xiangfeng Date: Tue, 14 Jul 2020 16:09:18 +0800 Subject: ASoC: Intel: bytcht_es8316: Add missed put_device() snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error path. Add the missed function call to fix it. Fixes: ba49cf6f8e4a ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect") Signed-off-by: Jing Xiangfeng Reviewed-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 9e5fc9430628..ecbc58e8a37f 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -543,8 +543,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (cnt) { ret = device_add_properties(codec_dev, props); - if (ret) + if (ret) { + put_device(codec_dev); return ret; + } } devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); -- cgit From fe0a53044b4bce947045eadd7fa1adbc4685afab Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Thu, 16 Jul 2020 11:01:23 +0800 Subject: ASoC: rt5682: Report the button event in the headset type only The irq work will be manipulated by resume function, and it will report the wrong jack type while the jack type is headphone in the button event. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20200716030123.27122-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 5adfaf3a7134..d503b5bef4ba 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1082,7 +1082,8 @@ void rt5682_jack_detect_handler(struct work_struct *work) /* jack was out, report jack type */ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 1); - } else { + } else if ((rt5682->jack_type & SND_JACK_HEADSET) == + SND_JACK_HEADSET) { /* jack is already in, report button event */ rt5682->jack_type = SND_JACK_HEADSET; btn_type = rt5682_button_detect(rt5682->component); -- cgit From 60379ba08532eca861e933b389526a4dc89e0c42 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jul 2020 10:40:23 +0200 Subject: ALSA: info: Drop WARN_ON() from buffer NULL sanity check snd_info_get_line() has a sanity check of NULL buffer -- both buffer itself being NULL and buffer->buffer being NULL. Basically both checks are valid and necessary, but the problem is that it's with snd_BUG_ON() macro that triggers WARN_ON(). The latter condition (NULL buffer->buffer) can be met arbitrarily by user since the buffer is allocated at the first write, so it means that user can trigger WARN_ON() at will. This patch addresses it by simply moving buffer->buffer NULL check out of snd_BUG_ON() so that spurious WARNING is no longer triggered. Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/info.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/info.c b/sound/core/info.c index 8c6bc5241df5..9fec3070f8ba 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -606,7 +606,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { int c; - if (snd_BUG_ON(!buffer || !buffer->buffer)) + if (snd_BUG_ON(!buffer)) + return 1; + if (!buffer->buffer) return 1; if (len <= 0 || buffer->stop || buffer->error) return 1; -- cgit From 568e4e82128aac2c62c2c359ebebb6007fd794f9 Mon Sep 17 00:00:00 2001 From: Joonho Wohn Date: Sat, 18 Jul 2020 17:23:15 +0900 Subject: ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S Fixed no headphone sound bug on laptop Samsung Notebook Pen S (950SBE-951SBE), by using existing patch in Linus' tree, commit 14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook). This laptop uses the same ALC298 but different subsystem id 0x144dc812. I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c Signed-off-by: Joonho Wohn Cc: Link: https://lore.kernel.org/r/CAHcbMh291aWDKiWSZoxXB4-Eru6OYRwGA4AVEdCZeYmVLo5ZxQ@mail.gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b06c4261248..1b2d8e56390a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7587,6 +7587,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), -- cgit