March 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: davinci-mcasp: dai format runtime reconfiguration
by Peter Ujfalusi 13 Mar '16

13 Mar '16

In case when the dai format is set via the dai_link the format configuration happens once when the links are probed. If the McASP lose context after this, the information will be lost and McASP will not going to work correctly. To overcome this issue, we save the fmt and set it within hw_params as well. Reported-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1bd86c8e7bc8..b0d1b9f93fa1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -78,6 +78,7 @@ struct davinci_mcasp { u32 fifo_base; struct device *dev; struct snd_pcm_substream *substreams[2]; + unsigned int dai_fmt; /* McASP specific data */ int tdm_slots; @@ -399,6 +400,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, bool fs_pol_rising; bool inv_fs = false; + if (!fmt) + return 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: @@ -530,6 +534,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); } + + mcasp->dai_fmt = fmt; out: pm_runtime_put(mcasp->dev); return ret; @@ -1048,6 +1054,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); + if (ret) + return ret; + /* * If mcasp is BCLK master, and a BCLK divider was not provided by * the machine driver, we need to calculate the ratio. -- 2.7.2

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[alsa-devel] [PATCH] ASoC: davinci: Kconfig: Update the edma-pcm section's dependency and help
by Peter Ujfalusi 13 Mar '16

13 Mar '16

Instead of depending on individual SoCs make the edma-pcm depend on the eDMA dmaengine driver (TI_EDMA). Update the help text and add DRA7xx family since they have eDMA integrated as well along with sDMA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index e0f6581494bf..a834558fd01e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -5,7 +5,7 @@ config SND_DAVINCI_SOC config SND_EDMA_SOC tristate "SoC Audio for Texas Instruments chips using eDMA" - depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI + depends on TI_EDMA select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. @@ -13,6 +13,7 @@ config SND_EDMA_SOC - daVinci devices - AM335x - AM437x/AM438x + - DRA7xx family config SND_DAVINCI_SOC_I2S tristate -- 2.7.2

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[alsa-devel] [PATCH] ASoC: qcom: fix build error
by Sudip Mukherjee 13 Mar '16

13 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building m32r allmodconfig the build failed with: ERROR: "bad_dma_ops" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! ERROR: "dma_common_mmap" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! To satisfy the dependency CONFIG_SND_SOC_LPASS_PLATFORM should depend on HAS_DMA. Some other configs also needs the dependency on HAS_DMA as they are directly or indirectly selecting SND_SOC_LPASS_PLATFORM. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- m32r build log at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/115245240 sound/soc/qcom/Kconfig | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3cc252e..8ec9a07 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -11,21 +11,24 @@ config SND_SOC_LPASS_CPU config SND_SOC_LPASS_PLATFORM tristate + depends on HAS_DMA select REGMAP_MMIO config SND_SOC_LPASS_IPQ806X tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_APQ8016 tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" - depends on SND_SOC_QCOM + depends on SND_SOC_QCOM && HAS_DMA select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -34,7 +37,7 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" - depends on SND_SOC_QCOM + depends on SND_SOC_QCOM && HAS_DMA select SND_SOC_LPASS_APQ8016 help Support for Qualcomm Technologies LPASS audio block in -- 1.9.1

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[alsa-devel] [PATCH v4] ASoC: nau8825: fix issue that pop noise when start playback
by Ben Zhang 13 Mar '16

13 Mar '16

From: John Hsu <KCHSU0(a)nuvoton.com> Reduce pop noise in power up and down sequence when playback. The DAPM widgets graph is reconstructed to ensure the register write sequence at playback matches exactly to the v5 clickless sequence provided by Nuvoton. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 169 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 16 ++++- 2 files changed, 141 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c1b87c5..1c87299 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -84,6 +84,7 @@ static const struct nau8825_fll_attr fll_pre_scalar[] = { static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_IIC_ADDR_SET, 0x0 }, { NAU8825_REG_CLK_DIVIDER, 0x0050 }, { NAU8825_REG_FLL1, 0x0 }, { NAU8825_REG_FLL2, 0x3126 }, @@ -158,8 +159,7 @@ static const struct reg_default nau8825_reg_defaults[] = { static bool nau8825_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { - case NAU8825_REG_ENA_CTRL: - case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_ENA_CTRL ... NAU8825_REG_FLL_VCO_RSV: case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: @@ -184,8 +184,7 @@ static bool nau8825_readable_reg(struct device *dev, unsigned int reg) static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) { switch (reg) { - case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: - case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_RESET ... NAU8825_REG_FLL_VCO_RSV: case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: case NAU8825_REG_INTERRUPT_MASK: case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: @@ -227,10 +226,42 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) static int nau8825_pump_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: /* Prevent startup click by letting charge pump to ramp up */ msleep(10); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, NAU8825_JAMNODCLOW); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8825_output_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disables the TESTDAC to let DAC signal pass through. */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDAC_EN); break; default: return -EINVAL; @@ -316,10 +347,10 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, NAU8825_SAR_ADC_EN_SFT, 0), - SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), - SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), - SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL Clock", 3, NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR Clock", 3, NAU8825_REG_RDAC, 9, 0, NULL, 0), SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR_SFT, 0), @@ -330,29 +361,48 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), - SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, - 0), + SND_SOC_DAPM_PGA_S("HP amp L", 0, + NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP amp R", 0, + NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, - nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("Charge Pump", 1, NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA("Output Driver R Stage 1", + SND_SOC_DAPM_PGA_S("Output Driver R Stage 1", 4, NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver L Stage 1", + SND_SOC_DAPM_PGA_S("Output Driver L Stage 1", 4, NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver R Stage 2", + SND_SOC_DAPM_PGA_S("Output Driver R Stage 2", 5, NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver L Stage 2", + SND_SOC_DAPM_PGA_S("Output Driver L Stage 2", 5, NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 6, NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 6, NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), - SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACL", 7, + NAU8825_REG_CHARGE_PUMP, 8, 1, nau8825_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("Output DACR", 7, + NAU8825_REG_CHARGE_PUMP, 9, 1, nau8825_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* HPOL/R are ungrounded by disabling 16 Ohm pull-downs on playback */ + SND_SOC_DAPM_PGA_S("HPOL Pulldown", 8, + NAU8825_REG_HSD_CTRL, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("HPOR Pulldown", 8, + NAU8825_REG_HSD_CTRL, 1, 1, NULL, 0), + + /* High current HPOL/R boost driver */ + SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, + NAU8825_REG_BOOST, 9, 1, NULL, 0), + + /* Class G operation control*/ + SND_SOC_DAPM_PGA_S("Class G", 10, + NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), @@ -375,24 +425,27 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"DACR Mux", "DACR", "DDACR"}, {"HP amp L", NULL, "DACL Mux"}, {"HP amp R", NULL, "DACR Mux"}, - {"HP amp L", NULL, "HP amp power"}, - {"HP amp R", NULL, "HP amp power"}, - {"ADACL", NULL, "HP amp L"}, - {"ADACR", NULL, "HP amp R"}, - {"ADACL", NULL, "ADACL Clock"}, - {"ADACR", NULL, "ADACR Clock"}, - {"Output Driver L Stage 1", NULL, "ADACL"}, - {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Charge Pump", NULL, "HP amp L"}, + {"Charge Pump", NULL, "HP amp R"}, + {"ADACL", NULL, "Charge Pump"}, + {"ADACR", NULL, "Charge Pump"}, + {"ADACL Clock", NULL, "ADACL"}, + {"ADACR Clock", NULL, "ADACR"}, + {"Output Driver L Stage 1", NULL, "ADACL Clock"}, + {"Output Driver R Stage 1", NULL, "ADACR Clock"}, {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, {"Output DACL", NULL, "Output Driver L Stage 3"}, {"Output DACR", NULL, "Output Driver R Stage 3"}, - {"HPOL", NULL, "Output DACL"}, - {"HPOR", NULL, "Output DACR"}, - {"HPOL", NULL, "Charge Pump"}, - {"HPOR", NULL, "Charge Pump"}, + {"HPOL Pulldown", NULL, "Output DACL"}, + {"HPOR Pulldown", NULL, "Output DACR"}, + {"HP Boost Driver", NULL, "HPOL Pulldown"}, + {"HP Boost Driver", NULL, "HPOR Pulldown"}, + {"Class G", NULL, "HP Boost Driver"}, + {"HPOL", NULL, "Class G"}, + {"HPOR", NULL, "Class G"}, }; static int nau8825_hw_params(struct snd_pcm_substream *substream, @@ -659,11 +712,10 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) break; } - if (type & SND_JACK_HEADPHONE) { - /* Unground HPL/R */ - regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); - } - + /* Leaving HPOL/R grounded after jack insert by default. They will be + * ungrounded as part of the widget power up sequence at the beginning + * of playback to reduce pop. + */ return type; } @@ -768,6 +820,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) { struct regmap *regmap = nau8825->regmap; + /* Latch IIC LSB value */ + regmap_write(regmap, NAU8825_REG_IIC_ADDR_SET, 0x0001); /* Enable Bias/Vmid */ regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); @@ -780,10 +834,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825) nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); /* Disable Boost Driver, Automatic Short circuit protection enable */ regmap_update_bits(regmap, NAU8825_REG_BOOST, - NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | - NAU8825_SHORT_SHUTDOWN_EN, - NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | - NAU8825_SHORT_SHUTDOWN_EN); + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_DIS | + NAU8825_HP_BOOST_G_DIS | NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_DIS | + NAU8825_HP_BOOST_G_DIS | NAU8825_SHORT_SHUTDOWN_EN); regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, NAU8825_JKDET_OUTPUT_EN, @@ -822,6 +876,35 @@ static void nau8825_init_regs(struct nau8825 *nau8825) NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); + /* Disable DACR/L power */ + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL); + /* Enable TESTDAC. This sets the analog DAC inputs to a '0' input + * signal to avoid any glitches due to power up transients in both + * the analog and digital DAC circuit. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDAC_EN); + /* CICCLP off */ + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_CLIP_OFF, NAU8825_DAC_CLIP_OFF); + + /* Class AB bias current to 2x, DAC Capacitor enable MSB/LSB */ + regmap_update_bits(regmap, NAU8825_REG_ANALOG_CONTROL_2, + NAU8825_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8825_DAC_CAPACITOR_MSB | NAU8825_DAC_CAPACITOR_LSB, + NAU8825_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8825_DAC_CAPACITOR_MSB | NAU8825_DAC_CAPACITOR_LSB); + /* Class G timer 64ms */ + regmap_update_bits(regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_TIMER_MASK, + 0x20 << NAU8825_CLASSG_TIMER_SFT); + /* DAC clock delay 2ns, VREF */ + regmap_update_bits(regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_CLK_DELAY_MASK | NAU8825_RDAC_VREF_MASK, + (0x2 << NAU8825_RDAC_CLK_DELAY_SFT) | + (0x3 << NAU8825_RDAC_VREF_SFT)); } static const struct regmap_config nau8825_regmap_config = { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..8ceb5f3 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -14,6 +14,7 @@ #define NAU8825_REG_RESET 0x00 #define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_IIC_ADDR_SET 0x02 #define NAU8825_REG_CLK_DIVIDER 0x03 #define NAU8825_REG_FLL1 0x04 #define NAU8825_REG_FLL2 0x05 @@ -129,7 +130,7 @@ /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -/* 0 - short to GND, 1 - open */ +/* 0 - open, 1 - short to GND */ #define NAU8825_SPKR_DWN1R (1 << 1) #define NAU8825_SPKR_DWN1L (1 << 0) @@ -251,12 +252,18 @@ /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +/* CLASSG_CTRL (0x50) */ +#define NAU8825_CLASSG_TIMER_SFT 8 +#define NAU8825_CLASSG_TIMER_MASK (0x3f << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_EN (1 << 0) + /* I2C_DEVICE_ID (0x58) */ #define NAU8825_GPIO2JD1 (1 << 7) #define NAU8825_SOFTWARE_ID_MASK 0x3 #define NAU8825_SOFTWARE_ID_NAU8825 0x0 /* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_TESTDAC_EN (0x3 << 8) #define NAU8825_BIAS_VMID (1 << 6) #define NAU8825_BIAS_VMID_SEL_SFT 4 #define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) @@ -274,6 +281,12 @@ #define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) #define NAU8825_POWERUP_ADCL (1 << 6) +/* RDAC (0x73) */ +#define NAU8825_RDAC_CLK_DELAY_SFT 4 +#define NAU8825_RDAC_CLK_DELAY_MASK (0x7 << NAU8825_RDAC_CLK_DELAY_SFT) +#define NAU8825_RDAC_VREF_SFT 2 +#define NAU8825_RDAC_VREF_MASK (0x3 << NAU8825_RDAC_VREF_SFT) + /* MIC_BIAS (0x74) */ #define NAU8825_MICBIAS_JKSLV (1 << 14) #define NAU8825_MICBIAS_JKR2 (1 << 12) @@ -284,6 +297,7 @@ /* BOOST (0x76) */ #define NAU8825_PRECHARGE_DIS (1 << 13) #define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 9) #define NAU8825_HP_BOOST_G_DIS (1 << 8) #define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) -- 2.7.0.rc3.207.g0ac5344

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[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: Support for DRA7xx family
by Peter Ujfalusi 13 Mar '16

13 Mar '16

DRA7xx family is compatible with the OMAP5 HDMI. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-hdmi-audio.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index f83cc2bc0fc4..64425d352962 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -345,6 +345,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) dai_drv = &omap4_hdmi_dai; break; case OMAPDSS_VER_OMAP5: + case OMAPDSS_VER_DRA7xx: dai_drv = &omap5_hdmi_dai; break; default: -- 2.7.2

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[alsa-devel] [PATCH] ASoC: rsnd: add simplified module explanation
by Kuninori Morimoto 13 Mar '16

13 Mar '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound driver user needs to read its datasheet when create DT. But it is difficult to understand, because it has many modules (SRC/CTU/MIX/DVC/SSIU/SSI/AudioDMAC/AudioDMACperiperi), and many features (Asynchronous/Synchronous mode on SRC, CTU matrix, DVC volume settings feature, Multi-SSI/TDM-SSI, etc). This patch adds simplified explanation to help setting/understanding. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 340 +++++++++++++++++++++ sound/soc/sh/rcar/dvc.c | 23 ++ 2 files changed, 363 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8ee0fa9..c7b29df 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -1,6 +1,337 @@ Renesas R-Car sound +============================================= +* Modules +============================================= + +Renesas R-Car sound is constructed from below modules +(for Gen2 or later) + + SCU : Sampling Rate Converter Unit + - SRC : Sampling Rate Converter + - CMD + - CTU : Channel Transfer Unit + - MIX : Mixer + - DVC : Digital Volume and Mute Function + SSIU : Serial Sound Interface Unit + SSI : Serial Sound Interface + +See detail of each module's channels, connection, limitation on datasheet + +============================================= +* Multi channel +============================================= + +Multi channel is supported by Multi-SSI, or TDM-SSI. + + Multi-SSI : 6ch case, you can use stereo x 3 SSI + TDM-SSI : 6ch case, you can use TDM + +============================================= +* Enable/Disable each modules +============================================= + +See datasheet to check SRC/CTU/MIX/DVC connect-limitation. +DT controls enabling/disabling module. +${LINUX}/arch/arm/boot/dts/r8a7790-lager.dts can be good example. +This is example of + +Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec] +Capture: [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec] + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &src2 &dvc0>; + capture = <&ssi1 &src3 &dvc1>; + }; + }; + }; + +You can use below. +${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example. + + &src0 &ctu00 &mix0 &dvc0 &ssi0 + &src1 &ctu01 &mix1 &dvc1 &ssi1 + &src2 &ctu02 &ssi2 + &src3 &ctu03 &ssi3 + &src4 &ssi4 + &src5 &ctu10 &ssi5 + &src6 &ctu11 &ssi6 + &src7 &ctu12 &ssi7 + &src8 &ctu13 &ssi8 + &src9 &ssi9 + +============================================= +* SRC (Sampling Rate Converter) +============================================= + + [xx]Hz [yy]Hz + ------> [SRC] ------> + +SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes + + Asynchronous mode: input data / output data are based on different clocks. + you can use this mode on Playback / Capture + Synchronous mode: input data / output data are based on same clocks. + This mode will be used if system doesn't have its input clock, + for example digital TV case. + you can use this mode on Playback + +------------------ +** Asynchronous mode +------------------ + +You need to use "renesas,rsrc-card" sound card for it. +example) + + sound { + compatible = "renesas,rsrc-card"; + ... + /* + * SRC Asynchronous mode setting + * Playback: + * All input data will be converted to 48kHz + * Capture: + * Inputed 48kHz data will be converted to + * system specified Hz + */ + convert-rate = <48000>; + ... + cpu { + sound-dai = <&rcar_sound>; + }; + codec { + ... + }; + }; + +------------------ +** Synchronous mode +------------------ + + > amixer set "SRC Out Rate" on + > aplay xxxx.wav + > amixer set "SRC Out Rate" 48000 + > amixer set "SRC Out Rate" 44100 + +============================================= +* CTU (Channel Transfer Unit) +============================================= + + [xx]ch [yy]ch + ------> [CTU] --------> + +CTU can convert [xx]ch to [yy]ch, or exchange outputed channel. +CTU conversion needs matrix settings. +For more detail information, see below + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - SCU Operation + - CMD Block + - Functional Blocks in CMD + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - Register Description + - CTUn Scale Value exx Register (CTUn_SVxxR) + + ${LINUX}/sound/soc/sh/rcar/ctu.c + - comment of header + +You need to use "renesas,rsrc-card" sound card for it. +example) + + sound { + compatible = "renesas,rsrc-card"; + ... + /* + * CTU setting + * All input data will be converted to 2ch + * as output data + */ + convert-channels = <2>; + ... + cpu { + sound-dai = <&rcar_sound>; + }; + codec { + ... + }; + }; + +Ex) Exchange output channel + Input -> Output + 1ch -> 0ch + 0ch -> 1ch + + example of using matrix + output 0ch = (input 0ch x 0) + (input 1ch x 1) + output 1ch = (input 0ch x 1) + (input 1ch x 0) + + amixer set "CTU Reset" on + amixer set "CTU Pass" 9,10 + amixer set "CTU SV0" 0,4194304 + amixer set "CTU SV1" 4194304,0 + + example of changing connection + amixer set "CTU Reset" on + amixer set "CTU Pass" 2,1 + +============================================= +* MIX (Mixer) +============================================= + +MIX merges 2 sounds path. You can see 2 sound interface on system, +and these sounds will be merged by MIX. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 yyyy.wav + +You need to use "renesas,rsrc-card" sound card for it. +Ex) + [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] + | + [MEM] -> [SRC2] -> [CTU03] -+ + + sound { + compatible = "renesas,rsrc-card"; + ... + cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + codec { + ... + }; + }; + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; + }; + dai1 { + playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; + }; + }; + }; + +============================================= +* DVC (Digital Volume and Mute Function) +============================================= + +DVC controls Playback/Capture volume. + +Playback Volume + amixer set "DVC Out" 100% + +Capture Volume + amixer set "DVC In" 100% + +Playback Mute + amixer set "DVC Out Mute" on + +Capture Mute + amixer set "DVC In Mute" on + +Volume Ramp + amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + amixer set "DVC Out Ramp" on + aplay xxx.wav & + amixer set "DVC Out" 80% // Volume Down + amixer set "DVC Out" 100% // Volume Up + +============================================= +* SSIU (Serial Sound Interface Unit) +============================================= + +There is no DT settings for SSIU, because SSIU will be automatically +selected via SSI. +SSIU can avoid some under/over run error, because it has some buffer. +But you can't use it if SSI was PIO mode. +In DMA mode, you can select not to use SSIU by using "no-busif" on DT. + + &ssi0 { + no-busif; + }; + +============================================= +* SSI (Serial Sound Interface) +============================================= + +** PIO mode + +You can use PIO mode which is for connection check by using. +Note: The system will drop non-SSI modules in PIO mode +even though if DT is selecting other modules. + + &ssi0 { + pio-transfer + }; + +** DMA mode without SSIU + +You can use DMA without SSIU. +Note: under/over run, or noise are likely to occur + + &ssi0 { + no-busif; + }; + +** PIN sharing + +Each SSI can share WS pin. It is based on platform. +This is example if SSI1 want to share WS pin with SSI0 + + &ssi1 { + shared-pin; + }; + +** Multi-SSI + +You can use Multi-SSI. +This is example of SSI0/SSI1/SSI2 (= for 6ch) + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; + }; + }; + }; + +** TDM-SSI + +You can use TDM with SSI. +This is example of TDM 6ch. +Driver can automatically switches TDM <-> stereo mode in this case. + + rsnd_tdm: sound { + compatible = "simple-audio-card"; + ... + simple-audio-card,cpu { + /* system can use TDM 6ch */ + dai-tdm-slot-num = <6>; + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; + }; + + +============================================= Required properties: +============================================= + - compatible : "renesas,rcar_sound-<soctype>", fallbacks "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 @@ -64,7 +395,10 @@ DAI subnode properties: - playback : list of playback modules - capture : list of capture modules + +============================================= Example: +============================================= rcar_sound: sound@ec500000 { #sound-dai-cells = <1>; @@ -250,7 +584,9 @@ rcar_sound: sound@ec500000 { }; }; +============================================= Example: simple sound card +============================================= rsnd_ak4643: sound { compatible = "simple-audio-card"; @@ -290,7 +626,9 @@ Example: simple sound card shared-pin; }; +============================================= Example: simple sound card for TDM +============================================= rsnd_tdm: sound { compatible = "simple-audio-card"; @@ -309,7 +647,9 @@ Example: simple sound card for TDM }; }; +============================================= Example: simple sound card for Multi channel +============================================= &rcar_sound { pinctrl-0 = <&sound_pins &sound_clk_pins>; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 93b11e1..02d971f 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -8,6 +8,29 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ + +/* + * Playback Volume + * amixer set "DVC Out" 100% + * + * Capture Volume + * amixer set "DVC In" 100% + * + * Playback Mute + * amixer set "DVC Out Mute" on + * + * Capture Mute + * amixer set "DVC In Mute" on + * + * Volume Ramp + * amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + * amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + * amixer set "DVC Out Ramp" on + * aplay xxx.wav & + * amixer set "DVC Out" 80% // Volume Down + * amixer set "DVC Out" 100% // Volume Up + */ + #include "rsnd.h" #define RSND_DVC_NAME_SIZE 16 -- 1.9.1

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[alsa-devel] [PATCH v4 0/6] add IEC958 channel status control helpers
by Arnaud Pouliquen 12 Mar '16

12 Mar '16

V4 - Link pcm control, defined in dai, to PCM device: Discard solution based on "pcm_new" ops for DAI initialisations replace it by an helper function that registers pcm control and links it to pcm device. - Define helpers function to handle 'IEC958 Playback Default' control . add snd_pcm_iec958_ctl_new helper for compatibility with snd_soc_add_dai_pcm_controls implementation . add substream parameter for snd_pcm_create_iec958_ctl function . fixes based on Takashi remarks. V3: - Define helpers function to handle 'IEC958 Playback Default' control - add "pcm_new" ops for DAI initialisations that need pcm runtime context. This patchset is needed to be able to associate control to the PCM device. The iec control is linked to the DAI but also to the PCM device. Furthermore, some platforms like sti support both HDMI and SPDIF outputs in parallel. Logical way to differentiate them is to link them to pcm device index. This patchset is extracted from following RFC http://permalink.gmane.org/gmane.linux.alsa.devel/149876 (sti: add audio interface to the hdmi driver) Notice that "ASoC: hdmi-codec: add IEC control" patch depends on integration of the hdmi-codec driver Patches update: - ALSA: pcm: add IEC958 channel status control helper - make mutex usage mandatory. - Set control index to pcm device id ( needed to be supported by iecset) - ASoC: core: add code to complete dai init after pcm creation - fix dai param of pcm_new ops for cpu dai. - ASoC: sti: use iec channel status control helper new patch: implementation of the helper for sti cpu-dai V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (5): ASoC: core: add snd_soc_add_dai_pcm_controls helper ASoC: sti: use snd_soc_add_dai_pcm_controls helper ALSA: pcm: add IEC958 channel status control helper ASoC: core: allow private data for snd_soc_add_dai_pcm_controls ASoC: sti: use iec channel status control helper include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 21 +++++ include/sound/soc-dai.h | 1 + include/sound/soc.h | 3 + sound/core/pcm_iec958.c | 125 ++++++++++++++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 54 ++++++++++++++++++++++++++++++++----------- sound/soc/soc-core.c | 168 +++++++++++++++++++++++++++++++++++----- sound/soc/sti/Kconfig | 1 + sound/soc/sti/sti_uniperif.c | 57 +++++++------- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 80 +++++-------------- 11 files changed, 386 insertions(+), 126 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 1/1] aplay/arecord: handle parsing errors of parameter values
by nebelbank＠posteo.de 11 Mar '16

11 Mar '16

From: erwin <nebelbank(a)posteo.de> Signed-off-by: erwin <nebelbank(a)posteo.de> --- It took me a while to find out why the signal from "-d hw:1,0" sounded so unexpected: was from another pair of microphones... diff --git a/aplay/aplay.c b/aplay/aplay.c index 7eacee3..7acaa83 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -427,6 +427,22 @@ enum { OPT_FATAL_ERRORS, }; +static long parse_long(const char *str, int *err) +{ + long val; + char *endptr; + + errno = 0; + val = strtol(str, &endptr, 0); + + if (errno != 0 || *endptr != '\0') + *err = -1; + else + *err = 0; + + return val; +} + int main(int argc, char *argv[]) { int option_index; @@ -558,7 +574,11 @@ int main(int argc, char *argv[]) } break; case 'c': - rhwparams.channels = strtol(optarg, NULL, 0); + rhwparams.channels = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid channels argument '%s'"), optarg); + return 1; + } if (rhwparams.channels < 1 || rhwparams.channels > 256) { error(_("value %i for channels is invalid"), rhwparams.channels); return 1; @@ -585,7 +605,11 @@ int main(int argc, char *argv[]) } break; case 'r': - tmp = strtol(optarg, NULL, 0); + tmp = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid rate argument '%s'"), optarg); + return 1; + } if (tmp < 300) tmp *= 1000; rhwparams.rate = tmp; @@ -595,32 +619,64 @@ int main(int argc, char *argv[]) } break; case 'd': - timelimit = strtol(optarg, NULL, 0); + timelimit = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid duration argument '%s'"), optarg); + return 1; + } break; case 'N': nonblock = 1; open_mode |= SND_PCM_NONBLOCK; break; case 'F': - period_time = strtol(optarg, NULL, 0); + period_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid period time argument '%s'"), optarg); + return 1; + } break; case 'B': - buffer_time = strtol(optarg, NULL, 0); + buffer_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid buffer time argument '%s'"), optarg); + return 1; + } break; case OPT_PERIOD_SIZE: - period_frames = strtol(optarg, NULL, 0); + period_frames = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid period size argument '%s'"), optarg); + return 1; + } break; case OPT_BUFFER_SIZE: - buffer_frames = strtol(optarg, NULL, 0); + buffer_frames = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid buffer size argument '%s'"), optarg); + return 1; + } break; case 'A': - avail_min = strtol(optarg, NULL, 0); + avail_min = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid min available space argument '%s'"), optarg); + return 1; + } break; case 'R': - start_delay = strtol(optarg, NULL, 0); + start_delay = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid start delay argument '%s'"), optarg); + return 1; + } break; case 'T': - stop_delay = strtol(optarg, NULL, 0); + stop_delay = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid stop delay argument '%s'"), optarg); + return 1; + } break; case 'v': verbose++; @@ -671,7 +727,11 @@ int main(int argc, char *argv[]) test_position = 1; break; case OPT_TEST_COEF: - test_coef = strtol(optarg, NULL, 0); + test_coef = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid test coef argument '%s'"), optarg); + return 1; + } if (test_coef < 1) test_coef = 1; break; @@ -679,7 +739,11 @@ int main(int argc, char *argv[]) test_nowait = 1; break; case OPT_MAX_FILE_TIME: - max_file_time = strtol(optarg, NULL, 0); + max_file_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid max file time argument '%s'"), optarg); + return 1; + } break; case OPT_PROCESS_ID_FILE: pidfile_name = optarg; -- 2.5.0

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[alsa-devel] [PATCH] ASoC: wm_adsp: Use correct local length in error message
by Charles Keepax 11 Mar '16

11 Mar '16

Fixes: 44029e9e1290 ("ASoC: wm_adsp: wm_coeff_{read|write}_control should use passed length") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3ee1c72..3c15481 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -807,7 +807,7 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, ret = regmap_raw_read(dsp->regmap, reg, scratch, len); if (ret) { adsp_err(dsp, "Failed to read %zu bytes from %x: %d\n", - ctl->len, reg, ret); + len, reg, ret); kfree(scratch); return ret; } -- 2.1.4

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[alsa-devel] [PATCH] ALSA: hda - fix the mic mute button and led problem for a Lenovo AIO
by Hui Wang 11 Mar '16

11 Mar '16

This Lenovo ThinkCentre AIO also uses Line2 as mic mute button and uses GPIO2 to control the mic mute led, so applying this quirk can make both the button and led work. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1555912 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 93d2156..4f5ca0b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5556,6 +5556,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), + SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 1.9.1

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