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[alsa-devel] Applied "ASoC: rt5645: Make a few struct const" to the asoc tree
by Mark Brown 19 Sep '17

19 Sep '17

The patch ASoC: rt5645: Make a few struct const has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 72d7a16bb07582f8416bbcdc65fe2a1e63f11294 Mon Sep 17 00:00:00 2001 From: Jean Delvare <jdelvare(a)suse.de> Date: Fri, 15 Sep 2017 10:27:10 +0200 Subject: [PATCH] ASoC: rt5645: Make a few struct const These dmi_system_id structures and associated platform data are never modified so they can be marked const. Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> -- sound/soc/codecs/rt5645.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5645.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a98647ac497c..4b2dbf0c3409 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3637,14 +3637,14 @@ static const struct dmi_system_id dmi_platform_gpd_win[] = { {} }; -static struct rt5645_platform_data general_platform_data2 = { +static const struct rt5645_platform_data general_platform_data2 = { .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, .dmic2_data_pin = RT5645_DMIC2_DISABLE, .jd_mode = 3, .inv_jd1_1 = true, }; -static struct dmi_system_id dmi_platform_asus_t100ha[] = { +static const struct dmi_system_id dmi_platform_asus_t100ha[] = { { .ident = "ASUS T100HAN", .matches = { @@ -3655,11 +3655,11 @@ static struct dmi_system_id dmi_platform_asus_t100ha[] = { { } }; -static struct rt5645_platform_data minix_z83_4_platform_data = { +static const struct rt5645_platform_data minix_z83_4_platform_data = { .jd_mode = 3, }; -static struct dmi_system_id dmi_platform_minix_z83_4[] = { +static const struct dmi_system_id dmi_platform_minix_z83_4[] = { { .ident = "MINIX Z83-4", .matches = { -- 2.14.1

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[alsa-devel] [PATCH] ASoC: zte: spdif: remove duplicate initialization of dma_data
by Colin King 19 Sep '17

19 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> dma_data is being initialized twice, remove the unused first initialization and use the latter one instead. Fixed clang warning: "warning: Value stored to 'dma_data' during its initialization is never read" Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/zte/zx-spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c index b143f9f682d2..17b6ce35037a 100644 --- a/sound/soc/zte/zx-spdif.c +++ b/sound/soc/zte/zx-spdif.c @@ -139,11 +139,11 @@ static int zx_spdif_hw_params(struct snd_pcm_substream *substream, { struct zx_spdif_info *zx_spdif = dev_get_drvdata(socdai->dev); struct zx_spdif_info *spdif = snd_soc_dai_get_drvdata(socdai); - struct snd_dmaengine_dai_dma_data *dma_data = &zx_spdif->dma_data; + struct snd_dmaengine_dai_dma_data *dma_data = + snd_soc_dai_get_dma_data(socdai, substream); u32 val, ch_num, rate; int ret; - dma_data = snd_soc_dai_get_dma_data(socdai, substream); dma_data->addr_width = params_width(params) >> 3; val = readl_relaxed(zx_spdif->reg_base + ZX_CTRL); -- 2.14.1

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[alsa-devel] [PATCH] ASoC: qcom: remove duplicate initializations of dma_ch and v
by Colin King 19 Sep '17

19 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> dma_ch is being initialized to zero and then immediately afterwards being updated again, so the first initialization is redundant and can be remove. Likewise, the pointer v is being initialized twice, so remove the second duplicate initialization. Cleans up two clang build warnings: warning: Value stored to 'dma_ch' is never read warning: Value stored to 'v' during its initialization is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/lpass-platform.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index e1945e1772cd..caf71aab8196 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -74,7 +74,6 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) data->i2s_port = cpu_dai->driver->id; runtime->private_data = data; - dma_ch = 0; if (v->alloc_dma_channel) dma_ch = v->alloc_dma_channel(drvdata, dir); else @@ -122,7 +121,6 @@ static int lpass_platform_pcmops_close(struct snd_pcm_substream *substream) struct lpass_pcm_data *data; data = runtime->private_data; - v = drvdata->variant; drvdata->substream[data->dma_ch] = NULL; if (v->free_dma_channel) v->free_dma_channel(drvdata, data->dma_ch); -- 2.14.1

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[alsa-devel] [PATCH] ASoC: hdmi-codec: use sizeof_field rather than declaring hcp
by Colin King 19 Sep '17

19 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> Use FIELD_SIZEOF rather than declaring and initializing hcp. Remove unused variables. Cleans up clang warning: warning: Value stored to 'hcp' during its initialization is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 3abf82563408..5672e516bec3 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -303,11 +303,8 @@ enum { static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); - uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; - uinfo->count = sizeof(hcp->eld); + uinfo->count = FIELD_SIZEOF(struct hdmi_codec_priv, eld); return 0; } -- 2.14.1

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[alsa-devel] [PATCH v3] ASoC: rt5663: Add the function of impedance sensing
by Oder Chiou 19 Sep '17

19 Sep '17

Support the function of impedance sensing. It could be set the matrix row number of the impedance sensing table and the related parameters in the DTS. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5663.txt | 16 ++ include/sound/rt5663.h | 3 + sound/soc/codecs/rt5663.c | 218 ++++++++++++++++++++- 3 files changed, 233 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index ff38171..497bcfc 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -19,6 +19,22 @@ Optional properties: Based on the different PCB layout, add the manual offset value to compensate the DC offset for each L and R channel, and they are different between headphone and headset. +- "realtek,impedance_sensing_num" + The matrix row number of the impedance sensing table. + If the value is 0, it means the impedance sensing is not supported. +- "realtek,impedance_sensing_table" + The matrix rows of the impedance sensing table are consisted by impedance + minimum, impedance maximun, volume, DC offset w/o and w/ mic of each L and + R channel accordingly. Example is shown as following. + < 0 300 7 0xffd160 0xffd1c0 0xff8a10 0xff8ab0 + 301 65535 4 0xffe470 0xffe470 0xffb8e0 0xffb8e0> + The first and second column are defined for the impedance range. If the + detected impedance value is in the range, then the volume value of the + third column will be set to codec. In our codec design, each volume value + should compensate different DC offset to avoid the pop sound, and it is + also different between headphone and headset. In the example, the + "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of + impedance in the impedance sensing function. Pins on the device (for linking into audio routes) for RT5663: diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h index 7d00e58..7b90a8f 100644 --- a/include/sound/rt5663.h +++ b/include/sound/rt5663.h @@ -16,6 +16,9 @@ struct rt5663_platform_data { unsigned int dc_offset_r_manual; unsigned int dc_offset_l_manual_mic; unsigned int dc_offset_r_manual_mic; + + unsigned int impedance_sensing_num; + unsigned int *impedance_sensing_table; }; #endif diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ab9e0eb..767f219 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -38,6 +38,16 @@ enum { CODEC_VER_0, }; +struct impedance_mapping_table { + unsigned int imp_min; + unsigned int imp_max; + unsigned int vol; + unsigned int dc_offset_l_manual; + unsigned int dc_offset_r_manual; + unsigned int dc_offset_l_manual_mic; + unsigned int dc_offset_r_manual_mic; +}; + struct rt5663_priv { struct snd_soc_codec *codec; struct rt5663_platform_data pdata; @@ -45,6 +55,7 @@ struct rt5663_priv { struct delayed_work jack_detect_work; struct snd_soc_jack *hs_jack; struct timer_list btn_check_timer; + struct impedance_mapping_table *imp_table; int codec_ver; int sysclk; @@ -1575,6 +1586,9 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) rt5663->jack_type = SND_JACK_HEADSET; rt5663_enable_push_button_irq(codec, true); + if (rt5663->pdata.impedance_sensing_num) + break; + if (rt5663->pdata.dc_offset_l_manual_mic) { regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, rt5663->pdata.dc_offset_l_manual_mic >> @@ -1596,6 +1610,9 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) default: rt5663->jack_type = SND_JACK_HEADPHONE; + if (rt5663->pdata.impedance_sensing_num) + break; + if (rt5663->pdata.dc_offset_l_manual) { regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, rt5663->pdata.dc_offset_l_manual >> 16); @@ -1623,6 +1640,177 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) return rt5663->jack_type; } +static int rt5663_impedance_sensing(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int value, i, reg84, reg26, reg2fa, reg91, reg10, reg80; + + for (i = 0; i < rt5663->pdata.impedance_sensing_num; i++) { + if (rt5663->imp_table[i].vol == 7) + break; + } + + if (rt5663->jack_type == SND_JACK_HEADSET) { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual_mic & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual_mic & 0xffff); + } else { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual & 0xffff); + } + + reg84 = snd_soc_read(codec, RT5663_ASRC_2); + reg26 = snd_soc_read(codec, RT5663_STO1_ADC_MIXER); + reg2fa = snd_soc_read(codec, RT5663_DUMMY_1); + reg91 = snd_soc_read(codec, RT5663_HP_CHARGE_PUMP_1); + reg10 = snd_soc_read(codec, RT5663_RECMIX); + reg80 = snd_soc_read(codec, RT5663_GLB_CLK); + + snd_soc_update_bits(codec, RT5663_STO_DRE_1, 0x8000, 0); + snd_soc_write(codec, RT5663_ASRC_2, 0); + snd_soc_write(codec, RT5663_STO1_ADC_MIXER, 0x4040); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK | + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_VREF1 | RT5663_PWR_VREF2); + usleep_range(10000, 10005); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_FV1 | RT5663_PWR_FV2); + snd_soc_update_bits(codec, RT5663_GLB_CLK, RT5663_SCLK_SRC_MASK, + RT5663_SCLK_SRC_RCCLK); + snd_soc_update_bits(codec, RT5663_RC_CLK, RT5663_DIG_25M_CLK_MASK, + RT5663_DIG_25M_CLK_EN); + snd_soc_update_bits(codec, RT5663_ADDA_CLK_1, RT5663_I2S_PD1_MASK, 0); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0xff00); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0xfffc); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_1, 0x1232); + snd_soc_write(codec, RT5663_HP_LOGIC_2, 0x0005); + snd_soc_write(codec, RT5663_DEPOP_2, 0x3003); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0x0030); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0x0003); + snd_soc_update_bits(codec, RT5663_PWR_DIG_2, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F); + snd_soc_update_bits(codec, RT5663_PWR_DIG_1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_MASK | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_ON | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1); + msleep(40); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2); + msleep(30); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_2, 0x1371); + snd_soc_write(codec, RT5663_STO_DAC_MIXER, 0); + snd_soc_write(codec, RT5663_BYPASS_STO_DAC, 0x000c); + snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); + snd_soc_write(codec, RT5663_CHARGE_PUMP_1, 0x2224); + snd_soc_write(codec, RT5663_HP_OUT_EN, 0x8088); + snd_soc_write(codec, RT5663_CHOP_ADC, 0x3000); + snd_soc_write(codec, RT5663_ADDA_RST, 0xc000); + snd_soc_write(codec, RT5663_STO1_HPF_ADJ1, 0x3320); + snd_soc_write(codec, RT5663_HP_CALIB_2, 0x00c9); + snd_soc_write(codec, RT5663_DUMMY_1, 0x004c); + snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7733); + snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_write(codec, RT5663_STO_DRE_9, 0x0007); + snd_soc_write(codec, RT5663_STO_DRE_10, 0x0007); + snd_soc_write(codec, RT5663_DUMMY_2, 0x02a4); + snd_soc_write(codec, RT5663_RECMIX, 0x0005); + snd_soc_write(codec, RT5663_HP_IMP_SEN_1, 0x4334); + snd_soc_update_bits(codec, RT5663_IRQ_3, 0x0004, 0x0004); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0x2200); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0x6200); + + for (i = 0; i < 100; i++) { + msleep(20); + if (snd_soc_read(codec, RT5663_INT_ST_1) & 0x2) + break; + } + + value = snd_soc_read(codec, RT5663_HP_IMP_SEN_4); + + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0); + snd_soc_write(codec, RT5663_INT_ST_1, 0); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0); + snd_soc_update_bits(codec, RT5663_RC_CLK, RT5663_DIG_25M_CLK_MASK, + RT5663_DIG_25M_CLK_DIS); + snd_soc_write(codec, RT5663_GLB_CLK, reg80); + snd_soc_write(codec, RT5663_RECMIX, reg10); + snd_soc_write(codec, RT5663_DUMMY_2, 0x00a4); + snd_soc_write(codec, RT5663_DUMMY_1, reg2fa); + snd_soc_write(codec, RT5663_HP_CALIB_2, 0x00c8); + snd_soc_write(codec, RT5663_STO1_HPF_ADJ1, 0xb320); + snd_soc_write(codec, RT5663_ADDA_RST, 0xe400); + snd_soc_write(codec, RT5663_CHOP_ADC, 0x2000); + snd_soc_write(codec, RT5663_HP_OUT_EN, 0x0008); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2, 0); + snd_soc_update_bits(codec, RT5663_PWR_DIG_1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_MASK | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1, 0); + snd_soc_update_bits(codec, RT5663_PWR_DIG_2, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F, 0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0); + snd_soc_write(codec, RT5663_HP_LOGIC_2, 0); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_1, reg91); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK, 0); + snd_soc_write(codec, RT5663_STO1_ADC_MIXER, reg26); + snd_soc_write(codec, RT5663_ASRC_2, reg84); + + for (i = 0; i < rt5663->pdata.impedance_sensing_num; i++) { + if (value >= rt5663->imp_table[i].imp_min && + value <= rt5663->imp_table[i].imp_max) + break; + } + + snd_soc_update_bits(codec, RT5663_STO_DRE_9, RT5663_DRE_GAIN_HP_MASK, + rt5663->imp_table[i].vol); + snd_soc_update_bits(codec, RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_MASK, + rt5663->imp_table[i].vol); + + if (rt5663->jack_type == SND_JACK_HEADSET) { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual_mic & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual_mic & 0xffff); + } else { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual & 0xffff); + } + + return 0; +} + static int rt5663_button_detect(struct snd_soc_codec *codec) { int btn_type, val; @@ -1701,6 +1889,8 @@ static void rt5663_jack_detect_work(struct work_struct *work) break; case CODEC_VER_0: report = rt5663_jack_detect(rt5663->codec, 1); + if (rt5663->pdata.impedance_sensing_num) + rt5663_impedance_sensing(rt5663->codec); break; default: dev_err(codec->dev, "Unknown CODEC Version\n"); @@ -1796,10 +1986,6 @@ static const struct snd_kcontrol_new rt5663_v2_specific_controls[] = { }; static const struct snd_kcontrol_new rt5663_specific_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, - RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, - rt5663_hp_vol_tlv), /* Mic Boost Volume*/ SOC_SINGLE_TLV("IN1 Capture Volume", RT5663_CBJ_2, RT5663_GAIN_BST1_SHIFT, 8, 0, in_bst_tlv), @@ -1807,6 +1993,13 @@ static const struct snd_kcontrol_new rt5663_specific_controls[] = { SOC_ENUM("IF1 ADC Data Swap", rt5663_if1_adc_enum), }; +static const struct snd_kcontrol_new rt5663_hpvol_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, + RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, + rt5663_hp_vol_tlv), +}; + static int rt5663_is_sys_clk_from_pll(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_widget *sink) { @@ -2889,6 +3082,10 @@ static int rt5663_probe(struct snd_soc_codec *codec) ARRAY_SIZE(rt5663_specific_dapm_routes)); snd_soc_add_codec_controls(codec, rt5663_specific_controls, ARRAY_SIZE(rt5663_specific_controls)); + + if (!rt5663->imp_table) + snd_soc_add_codec_controls(codec, rt5663_hpvol_controls, + ARRAY_SIZE(rt5663_hpvol_controls)); break; } @@ -3177,6 +3374,8 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) { + int table_size; + device_property_read_u32(dev, "realtek,dc_offset_l_manual", &rt5663->pdata.dc_offset_l_manual); device_property_read_u32(dev, "realtek,dc_offset_r_manual", @@ -3185,6 +3384,17 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) &rt5663->pdata.dc_offset_l_manual_mic); device_property_read_u32(dev, "realtek,dc_offset_r_manual_mic", &rt5663->pdata.dc_offset_r_manual_mic); + device_property_read_u32(dev, "realtek,impedance_sensing_num", + &rt5663->pdata.impedance_sensing_num); + + if (rt5663->pdata.impedance_sensing_num) { + table_size = sizeof(struct impedance_mapping_table) * + rt5663->pdata.impedance_sensing_num; + rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); + device_property_read_u32_array(dev, + "realtek,impedance_sensing_table", + (u32 *)rt5663->imp_table, table_size); + } return 0; } -- 2.7.4

2 1

[alsa-devel] [PATCH v11 1/2] ASoC: rt5514: Add devicetree binding support for rt5514-spi
by Jeffy Chen 19 Sep '17

19 Sep '17

Add devicetree binding support for rt5514 spi dsp codec. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v11: Use same compatible for rt5514-i2c & rt5514-spi. Changes in v10: Put rt5514-i2c doc & rt5514-spi doc together. Changes in v9: Address comments from Brian. Documentation/devicetree/bindings/sound/rt5514.txt | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 929ca6756b02..4f33b0d96afe 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -1,22 +1,27 @@ RT5514 audio CODEC -This device supports I2C only. +This device supports both I2C and SPI. Required properties: - compatible : "realtek,rt5514". -- reg : The I2C address of the device. +- reg : the I2C address of the device for I2C, the chip select + number for SPI. Optional properties: - clocks: The phandle of the master clock to the CODEC - clock-names: Should be "mclk" +- interrupt-parent: The phandle for the interrupt controller. +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + - realtek,dmic-init-delay-ms - Set the DMIC initial delay (ms) to wait it ready. + Set the DMIC initial delay (ms) to wait it ready for I2C. -Pins on the device (for linking into audio routes) for RT5514: +Pins on the device (for linking into audio routes) for I2C: * DMIC1L * DMIC1R -- 2.11.0

4 4

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Caculate bit clock rate using slot number and width
by Nicolin Chen 19 Sep '17

19 Sep '17

The set_sysclk() now is used to override the output bit clock rate. But this is not a common way to implement a set_dai_sysclk(). And this creates a problem when a general machine driver (simple-card for example) tries to do set_dai_sysclk() by passing an input clock rate for the baud clock instead of setting the bit clock rate as fsl_ssi driver expected. So this patch solves this problem by firstly removing set_sysclk() since the hw_params() can calculate the bit clock rate. Secondly, in order not to break those TDM use cases which previously might have been using set_sysclk() to override the bit clock rate, this patch changes the driver to calculate the bit clock rate using the slot number and the slot width from the via set_tdm_slot(). The patch also removes an obsolete comment of the dir parameter. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 46 ++++++++++++++++++++++++++-------------------- 1 file changed, 26 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 64598d1..f2f51e06 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -197,12 +197,13 @@ struct fsl_ssi_soc_data { * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for both FIFOs used * @fifo_deph: Depth of the SSI FIFOs + * @slot_width: width of each DAI slot + * @slots: number of slots * @rxtx_reg_val: Specific register settings for receive/transmit configuration * * @clk: SSI clock * @baudclk: SSI baud clock for master mode * @baudclk_streams: Active streams that are using baudclk - * @bitclk_freq: bitclock frequency set by .set_dai_sysclk * * @dma_params_tx: DMA transmit parameters * @dma_params_rx: DMA receive parameters @@ -233,12 +234,13 @@ struct fsl_ssi_private { bool use_dual_fifo; bool has_ipg_clk_name; unsigned int fifo_depth; + unsigned int slot_width; + unsigned int slots; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; struct clk *clk; struct clk *baudclk; unsigned int baudclk_streams; - unsigned int bitclk_freq; /* regcache for volatile regs */ u32 regcache_sfcsr; @@ -700,8 +702,8 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, * Note: This function can be only called when using SSI as DAI master * * Quick instruction for parameters: - * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels - * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + * freq: Output BCLK frequency = samplerate * slots * slot_width + * (In 2-channel I2S Master mode, slot_width is fixed 32) */ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai, @@ -712,15 +714,21 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; unsigned long clkrate, baudrate, tmprate; + unsigned int slots = params_channels(hw_params); + unsigned int slot_width = 32; u64 sub, savesub = 100000; unsigned int freq; bool baudclk_is_used; - /* Prefer the explicitly set bitclock frequency */ - if (ssi_private->bitclk_freq) - freq = ssi_private->bitclk_freq; - else - freq = params_channels(hw_params) * 32 * params_rate(hw_params); + /* Override slots and slot_width if being specifically set... */ + if (ssi_private->slots) + slots = ssi_private->slots; + /* ...but keep 32 bits if slots is 2 -- I2S Master mode */ + if (ssi_private->slot_width && slots != 2) + slot_width = ssi_private->slot_width; + + /* Generate bit clock based on the slot number and slot width */ + freq = slots * slot_width * params_rate(hw_params); /* Don't apply it to any non-baudclk circumstance */ if (IS_ERR(ssi_private->baudclk)) @@ -805,16 +813,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, return 0; } -static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - - ssi_private->bitclk_freq = freq; - - return 0; -} - /** * fsl_ssi_hw_params - program the sample size * @@ -1095,6 +1093,12 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, struct regmap *regs = ssi_private->regs; u32 val; + /* The word length should be 8, 10, 12, 16, 18, 20, 22 or 24 */ + if (slot_width & 1 || slot_width < 8 || slot_width > 24) { + dev_err(cpu_dai->dev, "invalid slot width: %d\n", slot_width); + return -EINVAL; + } + /* The slot number should be >= 2 if using Network mode or I2S mode */ regmap_read(regs, CCSR_SSI_SCR, &val); val &= CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET; @@ -1121,6 +1125,9 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val); + ssi_private->slot_width = slot_width; + ssi_private->slots = slots; + return 0; } @@ -1191,7 +1198,6 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, - .set_sysclk = fsl_ssi_set_dai_sysclk, .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; -- 2.7.4

2 1

[alsa-devel] [PATCH v2 1/3] Asoc: rockchip: Add dapm route for HDMI
by Jeffy Chen 19 Sep '17

19 Sep '17

Add dapm route for DP codec. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v2: Separate HDMI/DMIC patches. sound/soc/rockchip/rk3399_gru_sound.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 0513fe480353..b050510be65f 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -47,6 +47,7 @@ static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_LINE("HDMI", NULL), }; static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { @@ -59,6 +60,7 @@ static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { {"Headphones", NULL, "HPL"}, {"Headphones", NULL, "HPR"}, {"Speakers", NULL, "Speaker"}, + {"HDMI", NULL, "TX"}, }; static const struct snd_kcontrol_new rockchip_controls[] = { @@ -66,6 +68,7 @@ static const struct snd_kcontrol_new rockchip_controls[] = { SOC_DAPM_PIN_SWITCH("Speakers"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("HDMI"), }; static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, -- 2.11.0

2 5

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix an error handling path in 'davinci_mcasp_probe()'
by Christophe JAILLET 19 Sep '17

19 Sep '17

All error handling paths in this function 'goto err' except this one. If one of the 2 previous memory allocations fails, we should go through the existing error handling path. Otherwise there is an unbalanced pm_runtime_enable()/pm_runtime_disable(). Fixes: dd55ff8346a9 ("ASoC: davinci-mcasp: Add set_tdm_slots() support") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/davinci/davinci-mcasp.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f395bbc7c354..23b0da7df1f2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1982,8 +1982,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) GFP_KERNEL); if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || - !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) - return -ENOMEM; + !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) { + ret = -ENOMEM; + goto err; + } ret = davinci_mcasp_set_ch_constraints(mcasp); if (ret) -- 2.11.0

3 2

[alsa-devel] Applied "ASoC: Intel: Headset button support in kabylake machine driver" to the asoc tree
by Mark Brown 19 Sep '17

19 Sep '17

The patch ASoC: Intel: Headset button support in kabylake machine driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ae09a4783b9caf9307f303ef039f8297ce0371fe Mon Sep 17 00:00:00 2001 From: Naveen M <naveen.m(a)intel.com> Date: Fri, 15 Sep 2017 13:43:21 +0530 Subject: [PATCH] ASoC: Intel: Headset button support in kabylake machine driver Patch adds headset button support for kabylake machine driver (kbl_rt5663_max98927). Signed-off-by: Naveen M <naveen.m(a)intel.com> Acked-By: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7f7607420706..e45b6feca076 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -17,6 +17,7 @@ * GNU General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -208,6 +209,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_jack *jack; /* * Headset buttons map to the google Reference headset. @@ -221,6 +223,13 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; } + + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + rt5663_set_jack_detect(codec, &ctx->kabylake_headset); return ret; } -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: Intel: atom: make sst_platform_compr_ops const
by Bhumika Goyal 19 Sep '17

19 Sep '17

Make this const as it not modified in the file referencing it. It only stored in the const field 'compr_ops' of a snd_soc_platform_driver structure. Also, add const to the declaration in the header file. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/intel/atom/sst-mfld-platform-compress.c | 2 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-compress.c b/sound/soc/intel/atom/sst-mfld-platform-compress.c index 1bead81..1dbcab5 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-compress.c +++ b/sound/soc/intel/atom/sst-mfld-platform-compress.c @@ -259,7 +259,7 @@ static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, return stream->compr_ops->set_metadata(sst->dev, stream->id, metadata); } -struct snd_compr_ops sst_platform_compr_ops = { +const struct snd_compr_ops sst_platform_compr_ops = { .open = sst_platform_compr_open, .free = sst_platform_compr_free, diff --git a/sound/soc/intel/atom/sst-mfld-platform.h b/sound/soc/intel/atom/sst-mfld-platform.h index cb32cc7..31a58c2 100644 --- a/sound/soc/intel/atom/sst-mfld-platform.h +++ b/sound/soc/intel/atom/sst-mfld-platform.h @@ -25,7 +25,7 @@ #include "sst-atom-controls.h" extern struct sst_device *sst; -extern struct snd_compr_ops sst_platform_compr_ops; +extern const struct snd_compr_ops sst_platform_compr_ops; #define SST_MONO 1 #define SST_STEREO 2 -- 1.9.1

2 1

[alsa-devel] [PATCH v3 0/5] This series of patches fix the boot time dapm route warnings(without rt5514):
by Jeffy Chen 19 Sep '17

19 Sep '17

[ 2.475501] rk3399-gru-sound sound: ASoC: no sink widget found for DMIC1L [ 2.475503] rk3399-gru-sound sound: ASoC: Failed to add route Int Mic -> direct -> DMIC1L [ 2.475505] rk3399-gru-sound sound: ASoC: no sink widget found for DMIC1R [ 2.475507] rk3399-gru-sound sound: ASoC: Failed to add route Int Mic -> direct -> DMIC1R And support detect rt5514-dsp codec and rt5514 codec by checking bus_type. Changes in v3: Rebase onto for-next 20170919 Changes in v2: Separate HDMI/DMIC patches. Fix DMic typo. Jeffy Chen (5): ASoC: rockchip: Add dapm route for HDMI ASoC: rockchip: Add dapm route for DMic ASoC: rockchip: Init dapm routes dynamically ASoC: rockchip: Use bus_type to distinguish rt5514 dsp from rt5514 codec ASoC: rt5514: Add devicetree binding support for rt5514-spi Documentation/devicetree/bindings/sound/rt5514.txt | 13 +- sound/soc/rockchip/rk3399_gru_sound.c | 149 +++++++++++++++++---- 2 files changed, 132 insertions(+), 30 deletions(-) -- 2.11.0

2 6

[alsa-devel] [PATCH 0/5] BYT/CHT-Realtek remaining fixes
by Pierre-Louis Bossart 19 Sep '17

19 Sep '17

This last batch for the week is about simplification of the MCLK-related code for byt/cht-realtek machine drivers and fixes for rt5651. This is mostly a v2 of patches I submitted earlier, with improvements suggested by Andy Shevchenko, a fixed mistake on quirk management flagged by Vinod Koul and better commit messages. No new functionality otherwise. Pierre-Louis Bossart (5): ASoC: Intel: boards: use devm_clk_get() unconditionally ASoC: Intel: bytcr-rt5651: fix capture routes ASoC: Intel: bytcr_rt5651: add MCLK support and fix quirks ASoC: Intel: bytcr_rt5651: filter codec name ASoC: Intel: bytcr_rt5640: simplify MCLK quirk tests sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 277 ++++++++++++++++++++++++++++---- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 27 +--- 4 files changed, 265 insertions(+), 61 deletions(-) -- 2.9.3

4 11

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Don't error out if ENOTSUPP
by Nicolin Chen 19 Sep '17

19 Sep '17

The snd_soc_component_set_sysclk() and snd_soc_dai_set_tdm_slot() in the soc-core.c will return -ENOTSUPP if there is no function implementation for them in the dai and component drivers. So this patch tries to ignore this errno. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2db4d0c..3772abb 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -166,7 +166,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], cpu_priv->sysclk_freq[tx], cpu_priv->sysclk_dir[tx]); - if (ret) { + if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); return ret; } @@ -174,7 +174,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, if (cpu_priv->slot_width) { ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, cpu_priv->slot_width); - if (ret) { + if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); return ret; } @@ -270,7 +270,7 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, pll_out, SND_SOC_CLOCK_IN); - if (ret) { + if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set SYSCLK: %d\n", ret); return ret; } @@ -283,7 +283,7 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); - if (ret) { + if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to switch away from FLL: %d\n", ret); return ret; } @@ -459,7 +459,7 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); - if (ret) { + if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk in %s\n", __func__); return ret; } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/7] BYT/CHT clocks/max98090 updates
by Pierre-Louis Bossart 19 Sep '17

19 Sep '17

This patchset corrects issues with an earlier submission (missing signoffs, error case not consistent) and enables the cht_bsw_max98090 machine driver on Baytrail. This should help distributions solve problems with Baytrail Chromebooks which required a specific build. These patches were tested on Lenovo 100s ('orco'/Baytrail) and Acer R11 ('cyan'/Cherrytrail), using the same build based on CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH=m. Playback on speakers/headphone and capture from internal mic work fine. UCM files are available at https://github.com/plbossart/UCM/tree/master/byt-max98090 Thanks to Jarkko Nikula and Andy Shevchenko for their comments on an initial version reviewed internally at Intel. Fang, Yang A (1): ASoC: ts3a227e: add acpi table Pierre-Louis Bossart (5): ASoC: max98090: reduce verbosity on PLL unlock ASoC: Intel: cht_bsw_max98090: Fix I2S config + unused code ASoC: Intel: cht_bsw_max98090: add support for Baytrail ASoC: Intel: atom: use cht_bsw_max98090 for Baytrail Chromebooks ASoC: Intel: cht_bsw_max98090: add gpio-based jack detection Thierry Escande (1): ASoc: Intel: cht_bsw_max98090_ti: Fix jack initialization sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/ts3a227e.c | 10 ++ sound/soc/intel/atom/sst/sst_acpi.c | 8 ++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 185 ++++++++++++++++++++++++--- 4 files changed, 186 insertions(+), 19 deletions(-) -- 2.9.3

4 19

[alsa-devel] [PATCH 0/2] Add ACPI table for the ts3a227e driver
by Thierry Escande 19 Sep '17

19 Sep '17

Hi, This series adds ACPI match table for the ts3a227e driver. The ts3a227e is an Autonomous Audio Accessory Detection and Configuration Switch used to detect audio accessory features like jack plugging and unplugging, mic presence, and button pressed events. This series also fixes a null pointer dereference occurring in the TI max9890 driver when the ts3a227e hardware is present and its driver probed. The jack detection was enabled in the ts3a227e driver before the jack structure was allocated, causing the null pointer access. Regards, Thierry Fang, Yang A (1): ASoC: ts3a227e: add acpi table Thierry Escande (1): ASoc: cht_bsw_max98090_ti: Fix jack initialization sound/soc/codecs/ts3a227e.c | 10 ++++++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 52 +++++++++++++++++----------- 2 files changed, 41 insertions(+), 21 deletions(-) -- 2.7.4

3 6

[alsa-devel] [PATCH v12 1/2] ASoC: rt5514: Add devicetree binding support for rt5514-spi
by Jeffy Chen 19 Sep '17

19 Sep '17

Add devicetree binding support for rt5514 spi dsp codec. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v12: None Changes in v11: Use same compatible for rt5514-i2c & rt5514-spi. Changes in v10: Put rt5514-i2c doc & rt5514-spi doc together. Changes in v9: Address comments from Brian. Documentation/devicetree/bindings/sound/rt5514.txt | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 929ca6756b02..4f33b0d96afe 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -1,22 +1,27 @@ RT5514 audio CODEC -This device supports I2C only. +This device supports both I2C and SPI. Required properties: - compatible : "realtek,rt5514". -- reg : The I2C address of the device. +- reg : the I2C address of the device for I2C, the chip select + number for SPI. Optional properties: - clocks: The phandle of the master clock to the CODEC - clock-names: Should be "mclk" +- interrupt-parent: The phandle for the interrupt controller. +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + - realtek,dmic-init-delay-ms - Set the DMIC initial delay (ms) to wait it ready. + Set the DMIC initial delay (ms) to wait it ready for I2C. -Pins on the device (for linking into audio routes) for RT5514: +Pins on the device (for linking into audio routes) for I2C: * DMIC1L * DMIC1R -- 2.11.0

3 3

[alsa-devel] [RESEND PATCH v2 1/3] ASoC: rockchip: Add dapm route for HDMI
by Jeffy Chen 19 Sep '17

19 Sep '17

Add dapm route for DP codec. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v2: Separate HDMI/DMIC patches. sound/soc/rockchip/rk3399_gru_sound.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 0513fe480353..b050510be65f 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -47,6 +47,7 @@ static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_LINE("HDMI", NULL), }; static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { @@ -59,6 +60,7 @@ static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { {"Headphones", NULL, "HPL"}, {"Headphones", NULL, "HPR"}, {"Speakers", NULL, "Speaker"}, + {"HDMI", NULL, "TX"}, }; static const struct snd_kcontrol_new rockchip_controls[] = { @@ -66,6 +68,7 @@ static const struct snd_kcontrol_new rockchip_controls[] = { SOC_DAPM_PIN_SWITCH("Speakers"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("HDMI"), }; static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, -- 2.11.0

3 6

[alsa-devel] [PATCH 2/4] ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm
by anish kumar 19 Sep '17

19 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> refactor snd_soc_dapm_new_pcm to reduce the size of this function to facilitate further refactoring. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f51f613..d55cac6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3821,19 +3821,15 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kasprintf(card->dev, GFP_KERNEL, "Anonymous Configuration %d", count); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, strlen(config->stream_name) + 1, GFP_KERNEL); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + } + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; } config++; } -- 2.5.4 (Apple Git-61)

2 1

[alsa-devel] Resend: [PATCH 1/4] ASoC: dapm: fix error path in snd_soc_dapm_new_pcm
by anish kumar 19 Sep '17

19 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Before freeing the memory, check if it was allcoated. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..f51f613 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3907,9 +3907,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); -outfree_w_param: for (count = 0 ; count < num_params; count++) devm_kfree(card->dev, (void *)w_param_text[count]); +outfree_w_param: devm_kfree(card->dev, w_param_text); return ret; -- 2.5.4 (Apple Git-61)

2 1

[alsa-devel] [RFCv2][PATCH 00/38] alsa-utils: axfer: rewrite aplay
by Takashi Sakamoto 19 Sep '17

19 Sep '17

Hi, This large patchset is to introduce a new command, named as 'axfer' (ALSA transfer). This comes from my work to rewrite aplay. Initial patchset was already posted as RFCv1. [alsa-devel][PATCH 00/23] alsa-utils: rewrite aplay http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124132.html Unlike the initial patchset, this patchset includes support for the most of options in aplay, and simple unit tests for functionalities to handle data frames for files. At patch 17, transmission of data frame with blocking mode is supported. You can actually test playback/capture. * axfer: add a sub-command to transfer data frames At patch 24, transmission with non-blocking mode is supported. * axfer: add support for non-blocking operation At patch 25, transmission by MMAP operation is supported. * axfer: add support for MMAP PCM operation Between patch 30-36, some minor features are supported, but they includes own issues. I'd like to abandon them in future patchset. * axfer: add an option to support volume unit meter * axfer: add a unit test for vumeter calculation * axfer: add an option for formatted filename * axfer: add an option to handle key events * axfer: add a parser for channel map API * axfer: add a feature to generate a file for process id * axfer: obsolete some test options At patch 37/38, an integration to select waiter type is added. You can select one of poll(2) and epoll(2) to wait for I/O event notification. At present, I found some problems of this program: * When MMAP operation is used for PCM nodes with 'plug' plugin, XRUN occurs for capture and noisy sound for playback. I don't found this in the other operations such as blocking. * When using two axfer processes for capture/playback with pipe, some frames are truncated in the end of playback. I leave some tasks such as documentation and localization, but this patchset covers major use cases, I think. Regards Takashi Sakamoto (38): axfer: add an entry point for this command axfer: add a sub-command to print list of PCMs/devices axfer: add a common interface to handle a file with audio-specific data format axfer: add support for a container of Microsoft/IBM RIFF/Wave format axfer: add support for a container of Sparc AU format axfer: add support for a container of Creative Tech. voice format axfer: add support for a container of raw data axfer: add unit test for container interface axfer: add a common interface to align data frames on different layout axfer: add support for a mapper for single target axfer: add support for a mapper for multiple target axfer: add a unit test for mapper interface axfer: add a parser for command-line options axfer: add a common interface to transfer data frames axfer: add support to transfer data frames by alsa-lib PCM APIs axfer: add support for blocking data transmission operation of alsa-lib PCM API axfer: add a sub-command to transfer data frames axfer: add informative output and an option to suppress it axfer: add an option to dump available hardware parameters axfer: add options related to duration and obsolete '--max-file-size' option axfer: add an option to finish transmission at XRUN axfer: add a common interface of waiter for I/O event notification axfer: add support of waiter for poll(2) axfer: add support for non-blocking operation axfer: add support for MMAP PCM operation axfer: add an option to suppress event waiting axfer: add options for buffer arrangement axfer: add options for software parameters of PCM substream axfer: add options for plugins in alsa-lib axfer: add an option to support volume unit meter axfer: add a unit test for vumeter calculation axfer: add an option for formatted filename axfer: add an option to handle key events axfer: add a parser for channel map API axfer: add a feature to generate a file for process id axfer: obsolete some test options axfer: add an implementation of waiter for epoll(7) axfer: add an option for waiter type Makefile.am | 2 +- axfer/Makefile.am | 60 +++ axfer/container-au.c | 207 +++++++++ axfer/container-raw.c | 66 +++ axfer/container-riff-wave.c | 581 ++++++++++++++++++++++++ axfer/container-voc.c | 843 +++++++++++++++++++++++++++++++++++ axfer/container.c | 470 ++++++++++++++++++++ axfer/container.h | 126 ++++++ axfer/key-event.c | 121 +++++ axfer/key-event.h | 19 + axfer/main.c | 194 ++++++++ axfer/mapper-multiple.c | 271 +++++++++++ axfer/mapper-single.c | 195 ++++++++ axfer/mapper.c | 152 +++++++ axfer/mapper.h | 87 ++++ axfer/misc.h | 16 + axfer/options.c | 963 ++++++++++++++++++++++++++++++++++++++++ axfer/options.h | 83 ++++ axfer/subcmd-list.c | 234 ++++++++++ axfer/subcmd-transfer.c | 520 ++++++++++++++++++++++ axfer/subcmd.h | 18 + axfer/test/Makefile.am | 39 ++ axfer/test/container-test.c | 299 +++++++++++++ axfer/test/generator.c | 260 +++++++++++ axfer/test/generator.h | 47 ++ axfer/test/mapper-test.c | 491 ++++++++++++++++++++ axfer/test/vumeter-test.c | 128 ++++++ axfer/vumeter.c | 565 +++++++++++++++++++++++ axfer/vumeter.h | 56 +++ axfer/waiter-epoll.c | 81 ++++ axfer/waiter-poll.c | 68 +++ axfer/waiter.c | 84 ++++ axfer/waiter.h | 52 +++ axfer/xfer-libasound-irq-mmap.c | 224 ++++++++++ axfer/xfer-libasound-irq-rw.c | 518 +++++++++++++++++++++ axfer/xfer-libasound.c | 682 ++++++++++++++++++++++++++++ axfer/xfer-libasound.h | 57 +++ axfer/xfer.c | 191 ++++++++ axfer/xfer.h | 76 ++++ configure.ac | 3 +- 40 files changed, 9147 insertions(+), 2 deletions(-) create mode 100644 axfer/Makefile.am create mode 100644 axfer/container-au.c create mode 100644 axfer/container-raw.c create mode 100644 axfer/container-riff-wave.c create mode 100644 axfer/container-voc.c create mode 100644 axfer/container.c create mode 100644 axfer/container.h create mode 100644 axfer/key-event.c create mode 100644 axfer/key-event.h create mode 100644 axfer/main.c create mode 100644 axfer/mapper-multiple.c create mode 100644 axfer/mapper-single.c create mode 100644 axfer/mapper.c create mode 100644 axfer/mapper.h create mode 100644 axfer/misc.h create mode 100644 axfer/options.c create mode 100644 axfer/options.h create mode 100644 axfer/subcmd-list.c create mode 100644 axfer/subcmd-transfer.c create mode 100644 axfer/subcmd.h create mode 100644 axfer/test/Makefile.am create mode 100644 axfer/test/container-test.c create mode 100644 axfer/test/generator.c create mode 100644 axfer/test/generator.h create mode 100644 axfer/test/mapper-test.c create mode 100644 axfer/test/vumeter-test.c create mode 100644 axfer/vumeter.c create mode 100644 axfer/vumeter.h create mode 100644 axfer/waiter-epoll.c create mode 100644 axfer/waiter-poll.c create mode 100644 axfer/waiter.c create mode 100644 axfer/waiter.h create mode 100644 axfer/xfer-libasound-irq-mmap.c create mode 100644 axfer/xfer-libasound-irq-rw.c create mode 100644 axfer/xfer-libasound.c create mode 100644 axfer/xfer-libasound.h create mode 100644 axfer/xfer.c create mode 100644 axfer/xfer.h -- 2.11.0

1 38

[alsa-devel] [PATCH] ASoC: Intel: Atom: Remove unneeded linux/miscdevice.h include
by Corentin Labbe 18 Sep '17

18 Sep '17

No file in sound/soc/intel/ use any miscdevice. This patch remove this uncessary include. Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 0e928d54305d..4166925a689b 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -23,7 +23,6 @@ #include <linux/interrupt.h> #include <linux/slab.h> #include <linux/io.h> -#include <linux/miscdevice.h> #include <linux/platform_device.h> #include <linux/firmware.h> #include <linux/pm_runtime.h> -- 2.13.5

1 0

[alsa-devel] [PATCH] ALSA: oxygen: Xonar DG(X): make model_xonar_dg const
by Bhumika Goyal 18 Sep '17

18 Sep '17

Make this const as it not modified anywhere. It is only used during a copy operation. Also, add const to the declaration in header. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/oxygen/xonar_dg.h | 2 +- sound/pci/oxygen/xonar_dg_mixer.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/xonar_dg.h b/sound/pci/oxygen/xonar_dg.h index d461df3..5a07cda 100644 --- a/sound/pci/oxygen/xonar_dg.h +++ b/sound/pci/oxygen/xonar_dg.h @@ -51,6 +51,6 @@ void dump_cs4245_registers(struct oxygen *chip, void dg_resume(struct oxygen *chip); void dg_cleanup(struct oxygen *chip); -extern struct oxygen_model model_xonar_dg; +extern const struct oxygen_model model_xonar_dg; #endif diff --git a/sound/pci/oxygen/xonar_dg_mixer.c b/sound/pci/oxygen/xonar_dg_mixer.c index b885dac..d22fbe8 100644 --- a/sound/pci/oxygen/xonar_dg_mixer.c +++ b/sound/pci/oxygen/xonar_dg_mixer.c @@ -449,7 +449,7 @@ static int dg_mixer_init(struct oxygen *chip) return 0; } -struct oxygen_model model_xonar_dg = { +const struct oxygen_model model_xonar_dg = { .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: compress: Remove unused variable
by Guneshwor Singh 18 Sep '17

18 Sep '17

Commit 04c5d5a430fc ("ALSA: compress: Embed struct device") removed the statement that used 'str' but didn't remove the variable itself. So remove it. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/core/compress_offload.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index fec1dfdb14ad..4490a699030b 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -948,14 +948,13 @@ static const struct file_operations snd_compr_file_ops = { static int snd_compress_dev_register(struct snd_device *device) { int ret = -EINVAL; - char str[16]; struct snd_compr *compr; if (snd_BUG_ON(!device || !device->device_data)) return -EBADFD; compr = device->device_data; - pr_debug("reg %s for device %s, direction %d\n", str, compr->name, + pr_debug("reg device %s, direction %d\n", compr->name, compr->direction); /* register compressed device */ ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS, -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: hwdep: prevent a harmless shift wrapping bug
by Dan Carpenter 18 Sep '17

18 Sep '17

The "info.index" variable represents a bit in hw->dsp_loaded which is an unsigned int. If it's higher than 31 we hit a shift wrapping bug. This seems harmless, but I wanted to silence the static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index a73baa1242be..8faae3d1455d 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -228,6 +228,8 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, memset(&info, 0, sizeof(info)); if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; + if (info.index >= 32) + return -EINVAL; /* check whether the dsp was already loaded */ if (hw->dsp_loaded & (1 << info.index)) return -EBUSY;

2 1

[alsa-devel] [PATCH v2] pcm: softvol: add support for S24_LE
by Jörg Krause 18 Sep '17

18 Sep '17

Tested with the Wolfson WM8524 DAC on a i.MX6UL board and the following ALSA configuration file using the pcm test utility from alsa-lib: """ $ cat /etc/asound.conf pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "hw:0" } control { name "Master" card 0 } } ctl.!default { type hw card 0 } ctl.softvol { type hw card 0 } $ pcm -D softvol -o S24_LE -c 2 -r 48000 """ The data in the Synchronous Audio Interface (SAI) of the i.MX6UL is aligned the following way: """ 31 30 29 28 | 27 26 25 24 | 23 22 21 20 | .. | 3 2 1 0 ## ## ## ## ## ## ## ## [ DATA[23:0] ] """ Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- v2: * add CONVERT_AREA_S24_LE algorithm instead of using the same one as S32_LE --- src/pcm/pcm_softvol.c | 56 ++++++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 53 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 1fe5784d..68a35b56 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -251,6 +251,44 @@ static inline short MULTI_DIV_short(short a, unsigned int b, int swap) } \ } \ } while (0) + +#define CONVERT_AREA_S24_LE() do { \ + unsigned int ch, fr; \ + int *src, *dst; \ + int tmp; \ + for (ch = 0; ch < channels; ch++) { \ + src_area = &src_areas[ch]; \ + dst_area = &dst_areas[ch]; \ + src = snd_pcm_channel_area_addr(src_area, src_offset); \ + dst = snd_pcm_channel_area_addr(dst_area, dst_offset); \ + src_step = snd_pcm_channel_area_step(src_area) \ + / sizeof(int); \ + dst_step = snd_pcm_channel_area_step(dst_area) \ + / sizeof(int); \ + GET_VOL_SCALE; \ + fr = frames; \ + if (! vol_scale) { \ + while (fr--) { \ + *dst = 0; \ + dst += dst_step; \ + } \ + } else if (vol_scale == 0xffff) { \ + while (fr--) { \ + *dst = *src; \ + src += dst_step; \ + dst += src_step; \ + } \ + } else { \ + while (fr--) { \ + tmp = *src << 8; \ + tmp = (signed int) tmp >> 8; \ + *dst = MULTI_DIV_24(tmp, vol_scale); \ + src += dst_step; \ + dst += src_step; \ + } \ + } \ + } \ +} while (0) #define GET_VOL_SCALE \ switch (ch) { \ @@ -315,6 +353,10 @@ static void softvol_convert_stereo_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(int, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples */ + CONVERT_AREA_S24_LE(); + break; case SND_PCM_FORMAT_S24_3LE: CONVERT_AREA_S24_3LE(); break; @@ -366,6 +408,10 @@ static void softvol_convert_mono_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(int, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples */ + CONVERT_AREA_S24_LE(); + break; case SND_PCM_FORMAT_S24_3LE: CONVERT_AREA_S24_3LE(); break; @@ -422,6 +468,7 @@ static int snd_pcm_softvol_hw_refine_cprepare(snd_pcm_t *pcm, { (1ULL << SND_PCM_FORMAT_S16_LE) | (1ULL << SND_PCM_FORMAT_S16_BE) | + (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S32_LE) | (1ULL << SND_PCM_FORMAT_S32_BE), (1ULL << (SND_PCM_FORMAT_S24_3LE - 32)) @@ -577,10 +624,11 @@ static int snd_pcm_softvol_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * param if (slave->format != SND_PCM_FORMAT_S16_LE && slave->format != SND_PCM_FORMAT_S16_BE && slave->format != SND_PCM_FORMAT_S24_3LE && + slave->format != SND_PCM_FORMAT_S24_LE && slave->format != SND_PCM_FORMAT_S32_LE && slave->format != SND_PCM_FORMAT_S32_BE) { - SNDERR("softvol supports only S16_LE, S16_BE, S24_3LE, S32_LE " - " or S32_BE"); + SNDERR("softvol supports only S16_LE, S16_BE, S24_LE, S24_3LE, " + "S32_LE or S32_BE"); return -EINVAL; } svol->sformat = slave->format; @@ -863,6 +911,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) return -EINVAL; @@ -1082,9 +1131,10 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) { - SNDERR("only S16_LE, S16_BE, S24_3LE, S32_LE or S32_BE format is supported"); + SNDERR("only S16_LE, S16_BE, S24_LE, S24_3LE, S32_LE or S32_BE format is supported"); snd_config_delete(sconf); return -EINVAL; } -- 2.14.1

2 1

[alsa-devel] [RESENT PATCH v9 1/2] ASoC: rt5514: Add devicetree bindings for rt5514-spi
by Jeffy Chen 18 Sep '17

18 Sep '17

Add devicetree bindings documentation file for rt5514 spi dsp codec. Also update rt5514 i2c dt-binding's compatible to distinguish it from rt5514 spi. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v9: Address comments from Brian. .../devicetree/bindings/sound/rt5514-spi.txt | 29 ++++++++++++++++++++++ Documentation/devicetree/bindings/sound/rt5514.txt | 4 +-- 2 files changed, 31 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5514-spi.txt diff --git a/Documentation/devicetree/bindings/sound/rt5514-spi.txt b/Documentation/devicetree/bindings/sound/rt5514-spi.txt new file mode 100644 index 000000000000..7911d1adbd29 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5514-spi.txt @@ -0,0 +1,29 @@ +RT5514 SPI audio CODEC + +This device supports SPI only. + +Required properties: + +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-spi". + +- reg : The SPI address of the device. + +Optional properties: + +- interrupt-parent: Should be the phandle for the interrupt controller +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + +Example: + +codec: rt5514-spi@0 { + compatible = "realtek,rt5514", "realtek,rt5514-spi"; + reg = <0>; + + pinctrl-names = "default"; + pinctrl-0 = <&mic_int>; + + interrupt-parent = <&gpio1>; + interrupts = <13 IRQ_TYPE_LEVEL_HIGH>; + wakeup-source; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 929ca6756b02..841a1159e234 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "realtek,rt5514". +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-i2c". - reg : The I2C address of the device. @@ -28,6 +28,6 @@ Pins on the device (for linking into audio routes) for RT5514: Example: codec: rt5514@57 { - compatible = "realtek,rt5514"; + compatible = "realtek,rt5514", "realtek,rt5514-i2c"; reg = <0x57>; }; -- 2.11.0

4 4

[alsa-devel] Problems with Chromebook Pixel 2015 audio
by Tom Rini 18 Sep '17

18 Sep '17

Hey folks, Entirely unrelated to the regression on this board that we dealt with the other week, I have (and have had for a bit), a problem with audio on this laptop. After a while of use, I'll have a problem like this: Sep 12 20:59:30 oliver kernel: haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 (everything is fine, audio works, is in use) Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: error: audio DSP boot timeout IPCD 0x0 IPCX 0x0 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: ipc: --message timeout-- ipcx 0x83000000 isr 0x00000000 ipcd 0x00000000 imrx 0x7fff0000 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: error: stream commit failed Sep 12 21:24:04 oliver kernel: System PCM: error: failed to commit stream -110 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: ASoC: haswell-pcm-audio hw params failed: -110 Sep 12 21:24:04 oliver kernel: System PCM: ASoC: hw_params FE failed -110 With those last 4 messages repeating constantly (~3 sets per second, and the above are from v4.13, but I've had similar for as long as the driver has been mainline I think) and it only stops when I reboot. Rebooting may or may not clear the problem, last night I had to power off to get audio to come back up. I'm not quite sure where to begin debugging this issue. Does anyone have any suggestions? Thanks! -- Tom

2 1

Re: [alsa-devel] [PATCH v2] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Vinod Koul 18 Sep '17

18 Sep '17

On Sun, Sep 10, 2017 at 11:03:20AM +0530, Naveen M wrote: > rt5663 needs mclk/sclk early to synchronize its internal clocks. > Clocks are enabled in hw_params and disabled in platform_clk_control > as the codec requires mclk/bclk need to turned ON before set_sysclk > call for a successful clock synchronization. ah there was a v2, sorry missed that. > > Depends-on: patch series ("ASoC: Intel: Skylake: Add a clk driver to > enable ssp clks early") > ASoC: Intel: kbl: Enable mclk and ssp sclk early > ASoC: Intel: Skylake: Modify skl_dsp_set_dma_control API arguments > ASoC: Intel: Skylake: Parse nhlt to populate clock information > ASoC: Intel: Skylake: Prepare DMA control IPC to enable/disable clock > ASoC: Intel: Skylake: Register clock device and ops > ASoC: Intel: Skylake: Add ssp clock driver this is good info but is useless for applying > > Signed-off-by: Naveen M <naveen.m(a)intel.com> > Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> > > V2: Add Depends-on you have removed the dashes after s-o-b, so when I apply this patch the depends on, V2 also get applied. It is go d practice to apply the .patch file yourself after hand editing and seeing if the output is something you really want. Can you fix this along with comments on V1. -- ~Vinod

1 0

Re: [alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Vinod Koul 18 Sep '17

18 Sep '17

On Sun, Sep 10, 2017 at 08:32:59AM +0530, Naveen M wrote: > rt5663 needs mclk/sclk early to synchronize its internal clocks. > Clocks are enabled in hw_params and disabled in platform_clk_control > as the codec requires mclk/bclk need to turned ON before set_sysclk > call for a successful clock synchronization. This would be dependent on the series Subhransu posted right? If so please do mention that > +static int kabylake_enable_ssp_clks(struct snd_soc_card *card) > +{ > + we don't need the empty line here > + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); > + int ret; > + > + /* Enable MCLK */ > + ret = clk_set_rate(priv->mclk, 24000000); > + if (ret < 0) { > + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", ret); > + return ret; > + } > + > + ret = clk_prepare_enable(priv->mclk); > + if (ret < 0) { > + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); > + return ret; > + } > + > + /* Enable SCLK */ > + ret = clk_set_rate(priv->sclk, 3072000); why do we need both mclk and sclk for this codec? One should be enough for codec to sync > + if (ret < 0) { > + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", ret); > + clk_disable_unprepare(priv->mclk); > + return ret; > + } > + > + ret = clk_prepare_enable(priv->sclk); > + if (ret < 0) { > + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); > + clk_disable_unprepare(priv->mclk); > + } > + > + return 0; this looks wrong to me, shouldn't this be return ret; > @@ -647,6 +710,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) > { > struct kbl_codec_private *ctx; > struct skl_machine_pdata *pdata; > + int ret = 0; > > ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); > if (!ctx) > @@ -662,6 +726,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) > dmic_constraints = pdata->dmic_num == 2 ? > &constraints_dmic_2ch : &constraints_dmic_channels; > > + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); > + if (IS_ERR(ctx->mclk)) { > + ret = PTR_ERR(ctx->mclk); > + if (ret == -ENOENT) { > + dev_info(&pdev->dev, > + "Failed to get ssp1_mclk, defer probe\n"); can you check this, I think driver core emits warnings as well, so do we need it here? > + return -EPROBE_DEFER; > + } > + > + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", > + ret); > + return ret; > + } > + > + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); > + if (IS_ERR(ctx->sclk)) { > + ret = PTR_ERR(ctx->sclk); > + if (ret == -ENOENT) { > + dev_info(&pdev->dev, > + "Failed to get ssp1_sclk, defer probe\n"); shouldn't the mclk ref be given back? -- ~Vinod

1 0

[alsa-devel] [PATCH] ASoC: Intel: Headset button support in kabylake machine driver
by Naveen M 18 Sep '17

18 Sep '17

Patch adds headset button support for kabylake machine driver (kbl_rt5663_max98927). Signed-off-by: Naveen M <naveen.m(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7f76074..e45b6fe 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -17,6 +17,7 @@ * GNU General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -208,6 +209,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_jack *jack; /* * Headset buttons map to the google Reference headset. @@ -221,6 +223,13 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; } + + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + rt5663_set_jack_detect(codec, &ctx->kabylake_headset); return ret; } -- 1.9.1

2 1

[alsa-devel] [PATCH] Asoc: rockchip: Init dapm routes dynamically
by Jeffy Chen 18 Sep '17

18 Sep '17

Currently we are using a fixed list of dapm routes. Init dapm routes dynamically when parsing dailinks, since we are supporting optional codecs. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 88 +++++++++++++++++++++++++++++------ 1 file changed, 74 insertions(+), 14 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 0513fe480353..c537781f8054 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -47,18 +47,7 @@ static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), -}; - -static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { - /* Input Lines */ - {"MIC", NULL, "Headset Mic"}, - {"DMIC1L", NULL, "Int Mic"}, - {"DMIC1R", NULL, "Int Mic"}, - - /* Output Lines */ - {"Headphones", NULL, "HPL"}, - {"Headphones", NULL, "HPR"}, - {"Speakers", NULL, "Speaker"}, + SND_SOC_DAPM_LINE("HDMI", NULL), }; static const struct snd_kcontrol_new rockchip_controls[] = { @@ -66,6 +55,7 @@ static const struct snd_kcontrol_new rockchip_controls[] = { SOC_DAPM_PIN_SWITCH("Speakers"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("HDMI"), }; static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, @@ -314,8 +304,6 @@ static struct snd_soc_card rockchip_sound_card = { .owner = THIS_MODULE, .dapm_widgets = rockchip_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), - .dapm_routes = rockchip_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), .controls = rockchip_controls, .num_controls = ARRAY_SIZE(rockchip_controls), }; @@ -391,6 +379,65 @@ static const struct snd_soc_dai_link rockchip_dais[] = { }, }; +static const struct snd_soc_dapm_route rockchip_sound_cdndp_routes[] = { + /* Output */ + {"HDMI", NULL, "TX"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_da7219_routes[] = { + /* Output */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + + /* Input */ + {"MIC", NULL, "Headset Mic"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_dmic_routes[] = { + /* Input */ + {"Dmic", NULL, "Int Mic"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_max98357a_routes[] = { + /* Output */ + {"Speakers", NULL, "Speaker"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_rt5514_routes[] = { + /* Input */ + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, +}; + +struct rockchip_sound_route { + const struct snd_soc_dapm_route *routes; + int num_routes; +}; + +static const struct rockchip_sound_route rockchip_routes[] = { + [DAILINK_CDNDP] = { + .routes = rockchip_sound_cdndp_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_cdndp_routes), + }, + [DAILINK_DA7219] = { + .routes = rockchip_sound_da7219_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_da7219_routes), + }, + [DAILINK_DMIC] = { + .routes = rockchip_sound_dmic_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_dmic_routes), + }, + [DAILINK_MAX98357A] = { + .routes = rockchip_sound_max98357a_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_max98357a_routes), + }, + [DAILINK_RT5514] = { + .routes = rockchip_sound_rt5514_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_rt5514_routes), + }, + [DAILINK_RT5514_DSP] = {}, +}; + static int rockchip_sound_codec_node_match(struct device_node *np_codec) { int i; @@ -408,6 +455,7 @@ static int rockchip_sound_of_parse_dais(struct device *dev, struct device_node *np_cpu, *np_cpu0, *np_cpu1; struct device_node *np_codec; struct snd_soc_dai_link *dai; + struct snd_soc_dapm_route *routes; int i, index; card->dai_link = devm_kzalloc(dev, sizeof(rockchip_dais), @@ -415,9 +463,16 @@ static int rockchip_sound_of_parse_dais(struct device *dev, if (!card->dai_link) return -ENOMEM; + routes = devm_kzalloc(dev, sizeof(rockchip_routes), + GFP_KERNEL); + if (!routes) + return -ENOMEM; + card->dapm_routes = routes; + np_cpu0 = of_parse_phandle(dev->of_node, "rockchip,cpu", 0); np_cpu1 = of_parse_phandle(dev->of_node, "rockchip,cpu", 1); + card->num_dapm_routes = 0; card->num_links = 0; for (i = 0; i < ARRAY_SIZE(rockchip_dais); i++) { np_codec = of_parse_phandle(dev->of_node, @@ -445,6 +500,11 @@ static int rockchip_sound_of_parse_dais(struct device *dev, dai->codec_of_node = np_codec; dai->platform_of_node = np_cpu; dai->cpu_of_node = np_cpu; + + memcpy(routes + card->num_dapm_routes, + rockchip_routes[index].routes, + rockchip_routes[index].num_routes * sizeof(*routes)); + card->num_dapm_routes += rockchip_routes[index].num_routes; } return 0; -- 2.11.0

3 2

[alsa-devel] [PATCH] sound: tegra20-spdif: remove "default m"
by Michał Mirosław 17 Sep '17

17 Sep '17

Default != n for drivers is frowned upon. This is the only one among tegra sound drivers that was different. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> --- sound/soc/tegra/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 6875fc39a575..6c893c677110 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -38,7 +38,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" depends on SND_SOC_TEGRA - default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support -- 2.11.0

1 0

[alsa-devel] DICE Stereo-192-DSD-DAC issues since 2016-02-08 / kernel >= ~4.6
by Melvin Vermeeren 17 Sep '17

17 Sep '17

Hi all, Sometime in 2016, I believe with kernel ~4.6, my DAC started having weird behaviour. I found some mails about heavy firewire/DICE dev at the time so I sticked with 4.4 LTS for a while. I decided to do some proper research now and found out some details which hopefully will help solving the problem. The hardware, problem is with DAC only: https://www.siig.com/it-products/firewire/firewire-400/pcie/dp-firewire-2-p… https://mytekdigital.com/hifi/products/stereo-192dsd-dac/ https://us.focusrite.com/firewire-audio-interfaces/saffire-pro-24 The Saffire works perfectly for all modules I tried. I believe my DAC's firmware may have some quirks with control/notification messages. Behaviour comparison -------------------- Old modules (2eb65d67afbf9364b525b657f1475d1a2cbc27de): * Playback is instant, but DAC very rarely misses the lock. Need to restart the stream (stop, play MPD or restart jack) in that case. DAC's display still displays the samplerate change correctly, so probably firmware inconsistency. * Songs with different sample rate (e.g. 44100 -> 96000) switch instantly, unless DAC misses lock like above. DAC's display updates sample rate in 1-5 sec or so. * Sound itself is perfect, no cracks, pops or other issues. * Stop ALSA stream and DAC will indicate "No Lock", but procfs says locked. * dmesg is silent about firewire/dice/dac. New modules (4.1.1.12 and 4.13): * Takes ~2 sec before playback starts, but always works. * When MPD changes music track to track with different format it will always hang ~2 sec and fail to change format anyway. * Sound has very consistent popping in it. Very noticeable on higher frequencies, some timing issue? Besides that no other issues. * Power cycle the DAC and it will be "unlocked". After that the first stream opened will perma-lock the format. Changing to other frequency will not work. * Stop ALSA stream and DAC will indicate "No Lock", but procfs says locked. * dmesg shows only: "snd_dice fw1.0: Invalid CIP header for AMDTP: 00000000:00000000" probably when it's trying to change frequency. No other messages. I attached /proc/asound/DAC/dice output for old and new when idle, 44100 and 96000. global/notification and sample/lock change but that's basically it. Module details -------------- To test modules on my current kernel, which is 4.11.12-1-rt13-rt, I git checkout desired version of /sound/firewire and make those modules. After that I insmod snd-firewire-lib and snd-dice. The "old" modules refer to: 2eb65d67afbf9364b525b657f1475d1a2cbc27de ALSA: dice: expand timeout to wait for Dice notification These work like kernel ~4.4 did on the new kernel, just needed to include <linux/sched/signal.h> to get it to compile. New modules are as shipped as in kernel 4.11.12, might contain realtime patches. Also tried with mainline kernel 4.13's modules with same results. The commit that breaks it for my DAC is the one directly after "old" when it comes to the DICE module: 0d5ee195b1091e6ebcc121c091ce8646e95a48b6 ALSA: dice: limit to current sampling transfer frequency I don't notice any difference between the commit that "breaks" and current stable (4.13) when it comes to behaviour. Suspected cause --------------- I haven't read too much of the source code and all the development over time, so this is just a guess. Before the module would probably just send the audio stream and hope it works whereas the newer versions try to do it properly, perhaps violating some standard. The fact that according to procfs DAC's status is locked even when not playing might be why the current module will refuse to play at any different frequency. At least, that is what I think the description of 0d5ee195b1091e6ebcc121c091ce8646e95a48b6 is describing. So probably some quirk in my DAC's firmware is causing this. I remember that the Windows drivers always locked even without playback. Changing frequency from the driver software caused DAC display to update instantly with the lock changing shortly after. Though I believe Focusrite's software did the same thing for the mic interface. lspci -vv of my firewire card ----------------------------- Doubt this is related, but just in case. 03:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) (prog-if 10 [OHCI]) Subsystem: AFAVLAB Technology Inc Device 0000 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 32 (500ns min, 1000ns max), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 24 Region 0: Memory at fb704000 (32-bit, non-prefetchable) [size=2K] Region 1: Memory at fb700000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME+ Kernel driver in use: firewire_ohci Kernel modules: firewire_ohci P.S. Seems my DAC has dedicated stereo channels for DSD128 (probably also for DSD64) besides the PCM stereo channels. Any idea if/how that may work? Melvin Vermeeren.

1 0

[alsa-devel] [PATCH] ASoC: rt5514: expose Hotword Model control
by Hsin-Yu Chao 15 Sep '17

15 Sep '17

This change exposes mixer control 'Hotword Model' for switching hotword model in runtime. This new function requires updated firmware to load hotword model at address 0x4ff80000. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 52 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 3 +++ 2 files changed, 55 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a2722177470e..7482df8ff737 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -299,6 +299,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); struct snd_soc_codec *codec = rt5514->codec; const struct firmware *fw = NULL; + int ret = 0; if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) return 0; @@ -335,6 +336,27 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } + if (rt5514->model_buf && rt5514->model_len) { + ret = rt5514_spi_burst_write(0x4ff80000, + rt5514->model_buf, + ((rt5514->model_len / 8) + 1) * 8); + if (ret) { + dev_err(codec->dev, + "Model load failed %d\n", ret); + return ret; + } + } else { + request_firmware(&fw, RT5514_FIRMWARE3, + codec->dev); + if (fw) { + rt5514_spi_burst_write(0x4ff80000, + fw->data, + ((fw->size/8)+1)*8); + release_firmware(fw); + fw = NULL; + } + } + /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -349,6 +371,34 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } +static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, + const unsigned int __user *bytes, unsigned int size) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + int ret = 0; + + if (rt5514->model_buf || rt5514->model_len < size) { + if (rt5514->model_buf) + devm_kfree(codec->dev, rt5514->model_buf); + rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); + if (!rt5514->model_buf) { + ret = -ENOMEM; + goto done; + } + } + + /* Skips the TLV header. */ + bytes += 2; + + if (copy_from_user(rt5514->model_buf, bytes, size)) + ret = -EFAULT; +done: + rt5514->model_len = (ret ? 0 : size); + return ret; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -360,6 +410,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), + SND_SOC_BYTES_TLV("Hotword Model", 0x8504, + NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212a86d9..07475a9918d2 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -236,6 +236,7 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" +#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -262,6 +263,8 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; + u8 *model_buf; + unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.12.2

4 6

[alsa-devel] Applied "ASoC: rt5514: Revert Hotword Model control" to the asoc tree
by Mark Brown 15 Sep '17

15 Sep '17

The patch ASoC: rt5514: Revert Hotword Model control has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 37128514d708457fc060a2309aaa78b057a00043 Mon Sep 17 00:00:00 2001 From: Hsin-Yu Chao <hychao(a)chromium.org> Date: Fri, 15 Sep 2017 12:32:00 +0800 Subject: [PATCH] ASoC: rt5514: Revert Hotword Model control This reverts commit eb33869c7206 ("ASoC: rt5514: Guard Hotword Model bytes loading") and commit d18420b0a0b8 ("ASoC: rt5514: expose Hotword Model control") It is discouraged to use SND_SOC_BYTES_TLV to load arbitrary bytes from userspace to driver. Removing the 'Hotword Model' control until we have a good way to verify the content of hotword model blobs. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 63 ----------------------------------------------- sound/soc/codecs/rt5514.h | 3 --- 2 files changed, 66 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 0945d212b8dc..d7956ababd11 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -338,39 +338,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } - if (rt5514->model_buf && rt5514->model_len) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - int ret; - - ret = rt5514_spi_burst_write(0x4ff80000, - rt5514->model_buf, - ((rt5514->model_len / 8) + 1) * 8); - if (ret) { - dev_err(codec->dev, - "Model load failed %d\n", ret); - return ret; - } -#else - dev_err(codec->dev, - "No SPI driver for loading firmware\n"); -#endif - } else { - request_firmware(&fw, RT5514_FIRMWARE3, - codec->dev); - if (fw) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - rt5514_spi_burst_write(0x4ff80000, - fw->data, - ((fw->size/8)+1)*8); -#else - dev_err(codec->dev, - "No SPI driver to load fw\n"); -#endif - release_firmware(fw); - fw = NULL; - } - } - /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -385,34 +352,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } -static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, - const unsigned int __user *bytes, unsigned int size) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); - struct snd_soc_codec *codec = rt5514->codec; - int ret = 0; - - if (rt5514->model_buf || rt5514->model_len < size) { - if (rt5514->model_buf) - devm_kfree(codec->dev, rt5514->model_buf); - rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); - if (!rt5514->model_buf) { - ret = -ENOMEM; - goto done; - } - } - - /* Skips the TLV header. */ - bytes += 2; - - if (copy_from_user(rt5514->model_buf, bytes, size)) - ret = -EFAULT; -done: - rt5514->model_len = (ret ? 0 : size); - return ret; -} - static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -424,8 +363,6 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), - SND_SOC_BYTES_TLV("Hotword Model", 0x8504, - NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 803311cb7e2a..2dc40e6d8b3f 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -255,7 +255,6 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" -#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -282,8 +281,6 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; - u8 *model_buf; - unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.13.2

1 0

[alsa-devel] [PATCH] ASoC: rt5514: Revert Hotword Model control
by Hsin-Yu Chao 15 Sep '17

15 Sep '17

This reverts commit eb33869c7206 ("ASoC: rt5514: Guard Hotword Model bytes loading") and commit d18420b0a0b8 ("ASoC: rt5514: expose Hotword Model control") It is discouraged to use SND_SOC_BYTES_TLV to load arbitrary bytes from userspace to driver. Removing the 'Hotword Model' control until we have a good way to verify the content of hotword model blobs. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 63 ----------------------------------------------- sound/soc/codecs/rt5514.h | 3 --- 2 files changed, 66 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 250581082e9d..a2722177470e 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -335,39 +335,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } - if (rt5514->model_buf && rt5514->model_len) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - int ret; - - ret = rt5514_spi_burst_write(0x4ff80000, - rt5514->model_buf, - ((rt5514->model_len / 8) + 1) * 8); - if (ret) { - dev_err(codec->dev, - "Model load failed %d\n", ret); - return ret; - } -#else - dev_err(codec->dev, - "No SPI driver for loading firmware\n"); -#endif - } else { - request_firmware(&fw, RT5514_FIRMWARE3, - codec->dev); - if (fw) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - rt5514_spi_burst_write(0x4ff80000, - fw->data, - ((fw->size/8)+1)*8); -#else - dev_err(codec->dev, - "No SPI driver to load fw\n"); -#endif - release_firmware(fw); - fw = NULL; - } - } - /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -382,34 +349,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } -static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, - const unsigned int __user *bytes, unsigned int size) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); - struct snd_soc_codec *codec = rt5514->codec; - int ret = 0; - - if (rt5514->model_buf || rt5514->model_len < size) { - if (rt5514->model_buf) - devm_kfree(codec->dev, rt5514->model_buf); - rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); - if (!rt5514->model_buf) { - ret = -ENOMEM; - goto done; - } - } - - /* Skips the TLV header. */ - bytes += 2; - - if (copy_from_user(rt5514->model_buf, bytes, size)) - ret = -EFAULT; -done: - rt5514->model_len = (ret ? 0 : size); - return ret; -} - static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -421,8 +360,6 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), - SND_SOC_BYTES_TLV("Hotword Model", 0x8504, - NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 07475a9918d2..02bc212a86d9 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -236,7 +236,6 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" -#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -263,8 +262,6 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; - u8 *model_buf; - unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.12.2

3 2

Re: [alsa-devel] [PATCH v2] ASoC: rt5663: Add the function of impedance sensing
by Mark Brown 15 Sep '17

15 Sep '17

On Fri, Sep 15, 2017 at 02:20:09AM +0000, Oder Chiou wrote: > Sorry for my poor explanation. That's OK. > The impedance table should like the following setting in the DTS. > < 0 300 7 0xffd160 0xffd1c0 0xff8a10 0xff8ab0 > 301 65535 4 0xffe470 0xffe470 0xffb8e0 0xffb8e0> > The first and second column are defined for the impedance range. If the > detected impedance value is in the range, then the volume value of the third > column will be set to codec. In our codec design, each volume value should > compensate different DC offset to avoid the pop sound, and it is also > different between headphone and headset. In the example, the > "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of > impedance in the impedance sensing function, and it also can be defined the > device whether can support this function or not. This stuff looks fairly good, can you put it in the binding document please?

1 0

[alsa-devel] Need help in getting ALSA information
by Nagendra Kumar 15 Sep '17

15 Sep '17

Hi, My project needs ALSA device driver integration. Can you please help in in getting:1. Demo application(may be sound recorder and player) using ALSA device driver. 2. in knowing function call traces about packets coming from sound card and being delivering to demo(during recording).3. in knowing function call traces about packets going from application to ALSA till sound card (during playing sound). I just want to understand the function/call flow from/to application to/from sound card. My project have requirement to change the packet flow and add some code and test in the packet flow path like delay the sound packet, modify the sound packet from app till sound card and vice versa. Please kindly provide some pointers. Thanks-Nagendra

1 1

[alsa-devel] [PATCH] ASoC: add null_snd_soc_ops and reduce NULL ops check
by Kuninori Morimoto 15 Sep '17

15 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Double NULL pointer check for ops and ops->func is difficult to read and might be forget to check it if new func was add. This patch adds new null_snd_soc_ops and use it if rtd->dai_link didn't have it to avoid NULL ops, and reduces ops NULL check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 5 +++++ sound/soc/soc-pcm.c | 16 ++++++++-------- 2 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f8696dc..100be5c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -614,6 +614,8 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); +static const struct snd_soc_ops null_snd_soc_ops; + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -626,6 +628,9 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( INIT_LIST_HEAD(&rtd->component_list); rtd->card = card; rtd->dai_link = dai_link; + if (!rtd->dai_link->ops) + rtd->dai_link->ops = &null_snd_soc_ops; + rtd->codec_dais = kzalloc(sizeof(struct snd_soc_dai *) * dai_link->num_codecs, GFP_KERNEL); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e906b7c..0d6f913 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -543,7 +543,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->rx_mask = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + if (rtd->dai_link->ops->startup) { ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", @@ -617,7 +617,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); machine_err: @@ -739,7 +739,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->driver->ops->shutdown(substream, codec_dai); } - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); if (platform && platform->driver->ops && platform->driver->ops->close) @@ -814,7 +814,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + if (rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine prepare error:" @@ -944,7 +944,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret) goto out; - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine hw_params" @@ -1074,7 +1074,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->rate = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); mutex_unlock(&rtd->pcm_mutex); @@ -1122,7 +1122,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); /* free any DMA resources */ @@ -1206,7 +1206,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (rtd->dai_link->ops && rtd->dai_link->ops->trigger) { + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: rt5645: Make a few struct const
by Jean Delvare 15 Sep '17

15 Sep '17

These dmi_system_id structures and associated platform data are never modified so they can be marked const. Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> -- sound/soc/codecs/rt5645.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) --- linux-4.14-rc0.orig/sound/soc/codecs/rt5645.c 2017-09-14 11:08:35.328003671 +0200 +++ linux-4.14-rc0/sound/soc/codecs/rt5645.c 2017-09-14 11:38:34.014328130 +0200 @@ -3637,14 +3637,14 @@ static const struct dmi_system_id dmi_pl {} }; -static struct rt5645_platform_data general_platform_data2 = { +static const struct rt5645_platform_data general_platform_data2 = { .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, .dmic2_data_pin = RT5645_DMIC2_DISABLE, .jd_mode = 3, .inv_jd1_1 = true, }; -static struct dmi_system_id dmi_platform_asus_t100ha[] = { +static const struct dmi_system_id dmi_platform_asus_t100ha[] = { { .ident = "ASUS T100HAN", .matches = { @@ -3655,11 +3655,11 @@ static struct dmi_system_id dmi_platform { } }; -static struct rt5645_platform_data minix_z83_4_platform_data = { +static const struct rt5645_platform_data minix_z83_4_platform_data = { .jd_mode = 3, }; -static struct dmi_system_id dmi_platform_minix_z83_4[] = { +static const struct dmi_system_id dmi_platform_minix_z83_4[] = { { .ident = "MINIX Z83-4", .matches = { -- Jean Delvare SUSE L3 Support

1 0

[alsa-devel] [PATCH v2] ASoC: rt5670: add set_bclk_ratio in dai ops
by Bard Liao 15 Sep '17

15 Sep '17

We need to set a specific bit for 50 bclk rate. So add set_bclk_ratio function to set the bit. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2: * add define for tdm data format select bit. --- sound/soc/codecs/rt5670.c | 19 +++++++++++++++++++ sound/soc/codecs/rt5670.h | 4 ++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9545764..da7614c 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2582,6 +2582,24 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5670_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (dai->id != RT5670_AIF1) + return 0; + + if (ratio == 50) + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_50FS); + else + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_NOR); + + return 0; +} + static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2712,6 +2730,7 @@ static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_fmt = rt5670_set_dai_fmt, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, + .set_bclk_ratio = rt5670_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5670_dai[] = { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 5ba485c..265df80 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1816,6 +1816,10 @@ #define RT5670_ZCD_HP_DIS (0x0 << 15) #define RT5670_ZCD_HP_EN (0x1 << 15) +/* General Control 3 (0xfc) */ +#define RT5670_TDM_DATA_MODE_SEL (0x1 << 11) +#define RT5670_TDM_DATA_MODE_NOR (0x0 << 11) +#define RT5670_TDM_DATA_MODE_50FS (0x1 << 11) /* Codec Private Register definition */ /* 3D Speaker Control (0x63) */ -- 2.7.4

2 6

[alsa-devel] [PATCH] ASoC: remove unneeded dai->driver check
by Kuninori Morimoto 15 Sep '17

15 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> On soc_add_dai(), it is checking dai->driver->xxx, This means, dai->driver is mandatory, never been NULL. dai->driver check is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0d170a0..f8696dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2634,7 +2634,7 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->driver && dai->driver->ops->set_sysclk) + if (dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); return snd_soc_component_set_sysclk(dai->component, clk_id, 0, @@ -2702,7 +2702,7 @@ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->driver && dai->driver->ops->set_clkdiv) + if (dai->driver->ops->set_clkdiv) return dai->driver->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; @@ -2722,7 +2722,7 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - if (dai->driver && dai->driver->ops->set_pll) + if (dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); @@ -2788,7 +2788,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, */ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) { - if (dai->driver && dai->driver->ops->set_bclk_ratio) + if (dai->driver->ops->set_bclk_ratio) return dai->driver->ops->set_bclk_ratio(dai, ratio); else return -EINVAL; @@ -2862,7 +2862,7 @@ static int snd_soc_xlate_tdm_slot_mask(unsigned int slots, int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - if (dai->driver && dai->driver->ops->xlate_tdm_slot_mask) + if (dai->driver->ops->xlate_tdm_slot_mask) dai->driver->ops->xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); else @@ -2871,7 +2871,7 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, dai->tx_mask = tx_mask; dai->rx_mask = rx_mask; - if (dai->driver && dai->driver->ops->set_tdm_slot) + if (dai->driver->ops->set_tdm_slot) return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); else @@ -2895,7 +2895,7 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot) { - if (dai->driver && dai->driver->ops->set_channel_map) + if (dai->driver->ops->set_channel_map) return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot, rx_num, rx_slot); else @@ -2912,7 +2912,7 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->driver && dai->driver->ops->set_tristate) + if (dai->driver->ops->set_tristate) return dai->driver->ops->set_tristate(dai, tristate); else return -EINVAL; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: remove unneeded dai->driver->ops check
by Kuninori Morimoto 15 Sep '17

15 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> On soc_add_dai(), it uses null_dai_ops if driver doesn't have its own ops. This means, dai->driver->ops never been NULL. dai->driver->ops check is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-dapm.c | 8 ++++---- sound/soc/soc-pcm.c | 31 +++++++++++++++---------------- 2 files changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..9d4748e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3681,7 +3681,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; - if (source->driver->ops && source->driver->ops->startup) { + if (source->driver->ops->startup) { ret = source->driver->ops->startup(&substream, source); if (ret < 0) { dev_err(source->dev, @@ -3695,7 +3695,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, goto out; substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - if (sink->driver->ops && sink->driver->ops->startup) { + if (sink->driver->ops->startup) { ret = sink->driver->ops->startup(&substream, sink); if (ret < 0) { dev_err(sink->dev, @@ -3725,13 +3725,13 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = 0; source->active--; - if (source->driver->ops && source->driver->ops->shutdown) { + if (source->driver->ops->shutdown) { substream.stream = SNDRV_PCM_STREAM_CAPTURE; source->driver->ops->shutdown(&substream, source); } sink->active--; - if (sink->driver->ops && sink->driver->ops->shutdown) { + if (sink->driver->ops->shutdown) { substream.stream = SNDRV_PCM_STREAM_PLAYBACK; sink->driver->ops->shutdown(&substream, sink); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index da55e4e..e906b7c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -483,7 +483,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ - if (cpu_dai->driver->ops && cpu_dai->driver->ops->startup) { + if (cpu_dai->driver->ops->startup) { ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: can't open interface" @@ -526,7 +526,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { + if (codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { @@ -853,7 +853,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { + if (codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { @@ -865,7 +865,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { + if (cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, @@ -911,7 +911,7 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, { int ret; - if (dai->driver->ops && dai->driver->ops->hw_params) { + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n", @@ -1060,7 +1060,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, platform->driver->ops->hw_free(substream); platform_err: - if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: @@ -1069,7 +1069,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_err: while (--i >= 0) { struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_dai->rate = 0; } @@ -1147,11 +1147,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* now free hw params for the DAIs */ for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); mutex_unlock(&rtd->pcm_mutex); @@ -1170,7 +1170,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { + if (codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); if (ret < 0) @@ -1200,7 +1200,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->trigger) { + if (cpu_dai->driver->ops->trigger) { ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -1225,8 +1225,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && - codec_dai->driver->ops->bespoke_trigger) { + if (codec_dai->driver->ops->bespoke_trigger) { ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); if (ret < 0) @@ -1234,7 +1233,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, } } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->bespoke_trigger) { + if (cpu_dai->driver->ops->bespoke_trigger) { ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -1279,12 +1278,12 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) break; } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) + if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->delay) + if (codec_dai->driver->ops->delay) codec_delay = max(codec_delay, codec_dai->driver->ops->delay(substream, codec_dai)); -- 1.9.1

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[alsa-devel] [PATCH 4/4] ASoC: dapm: Remove the redundant check
by anish kumar 15 Sep '17

15 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Remove this redundant check, the check for creating the control in the case of codec to codec dai link is moved to snd_soc_dapm_new_pcm function. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3c493b3..fcad345 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1032,10 +1032,6 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; - /* create control for links with > 1 config */ - if (w->num_params <= 1) - return 0; - /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, -- 2.5.4 (Apple Git-61)

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[alsa-devel] [PATCH 3/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 15 Sep '17

15 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 114 +++++++++++++++++++++++++++++---------------------- 1 file changed, 66 insertions(+), 48 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..3c493b3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,12 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3792,16 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; + const char **w_param_text; + unsigned long private_value; + int count; w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); + sizeof(char *), GFP_KERNEL); if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + return NULL; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,24 +3818,14 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), @@ -3852,22 +3833,66 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } kcontrol_dai_link[0].private_value = private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; goto outfree_private_value; } + return kcontrol_news; + +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto outfree_link_name; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3924,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

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[alsa-devel] Resend: [patch 0/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 15 Sep '17

15 Sep '17

Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. This code change doesn't do much except just refactor snd_soc_dapm_new_pcm function. Now the part of allocating memory for kcontrol is allocated to a new function and it is called only in the case it is needed. anish kumar (4): ASoC: dapm: fix error path in snd_soc_dapm_new_pcm ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm ASoC: dapm: Avoid creating kcontrol for params ASoC: dapm: Remove the redundant check sound/soc/soc-dapm.c | 123 ++++++++++++++++++++++++++++----------------------- 1 file changed, 67 insertions(+), 56 deletions(-) -- 2.10.1 (Apple Git-78)

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[alsa-devel] [PATCH 1/3] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 15 Sep '17

15 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changelog: Added support for DSP_A and DSP_B format Added 'max98927_dai_tdm_slot' function to set proper slot configuration. Moved max98927->iface out of switch statement to avoid line duplication. Added variable 'tdm_mode' to avoid BCLK overwrite after 'max98927_dai_tdm_slot' configure BCLK value. sound/soc/codecs/max98927.c | 155 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/max98927.h | 7 +- 2 files changed, 129 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe7..a1d3935 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + } return 0; } @@ -245,6 +250,21 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,23 +290,20 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: - return -EINVAL; + if (!max98927->tdm_mode) { + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + value); } - regmap_update_bits(max98927->regmap, - MAX98927_R0022_PCM_CLK_SETUP, - MAX98927_PCM_CLK_SETUP_BSEL_MASK, - value); return 0; } @@ -386,6 +403,78 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + max98927->tdm_mode = true; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +494,7 @@ static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, @@ -414,6 +504,9 @@ static int max98927_dac_event(struct snd_soc_dapm_widget *w, struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + max98927->tdm_mode = 0; + break; case SND_SOC_DAPM_POST_PMU: regmap_update_bits(max98927->regmap, MAX98927_R003A_AMP_EN, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..9ea8397 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) @@ -268,5 +270,6 @@ struct max98927_priv { unsigned int iface; unsigned int master; unsigned int digital_gain; + bool tdm_mode; }; #endif -- 1.9.1

2 3

[alsa-devel] [PATCH 2/2] ASoC: topology: Fix a potential memory leak in 'soc_tplg_dapm_widget_denum_create()'
by Christophe JAILLET 14 Sep '17

14 Sep '17

If this sanity check fails, we must free the memory that has already been allocated. So we must go to 'err' as in the other error handling parth of this function. Fixes: 1a7dd6e2f192 ("ASoC: topology: Allow a widget to have multiple enum controls") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 2499b54a57e8..01a50413c66f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1301,7 +1301,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( /* validate kcontrol */ if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - return NULL; + goto err; se = kzalloc(sizeof(*se), GFP_KERNEL); if (se == NULL) -- 2.11.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: topology: Fix a potential NULL pointer dereference in 'soc_tplg_dapm_widget_denum_create()'
by Christophe JAILLET 14 Sep '17

14 Sep '17

if 'se = kzalloc()' fails in the 'for' loop, we will branch to 'err'. But in this case, 'kc[i].private_value' will still be NULL. A NULL pointer dereference will then occur is the error handling path. In such a case, it is safe to just 'continue' in order to skip this entry and free the other ones. Fixes: 1a7dd6e2f192 ("ASoC: topology: Allow a widget to have multiple enum controls") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/soc-topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index dd471d2c0266..2499b54a57e8 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1378,6 +1378,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( for (; i >= 0; i--) { /* free values and texts */ se = (struct soc_enum *)kc[i].private_value; + if (!se) + continue; + kfree(se->dobj.control.dvalues); for (j = 0; j < ec->items; j++) kfree(se->dobj.control.dtexts[j]); -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Decrease loglevel for topology loading
by Chintan Patel 14 Sep '17

14 Sep '17

In case of failure in loading customize topology firmware, dfw_sst.bin gets loaded. However, current log provides this message as error even after successfully falling back to default topology "dfw_sst.bin". Hence to convey proper message, changing log level and message. Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22f768ca3c73..14dafa9f5b35 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3404,7 +3404,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_err(bus->dev, "tplg fw %s load failed with %d\n", + dev_warn(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", skl->tplg_name, ret); ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { -- 1.9.1

3 3

[alsa-devel] [PATCH v2] ASoC: rt5663: Add the function of impedance sensing
by Oder Chiou 14 Sep '17

14 Sep '17

Support the function of impedance sensing. It could be set the matrix row number of the impedance sensing table and the related parameters in the DTS. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5663.txt | 6 + include/sound/rt5663.h | 3 + sound/soc/codecs/rt5663.c | 218 ++++++++++++++++++++- 3 files changed, 223 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index ff38171..4f52fac 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -19,6 +19,12 @@ Optional properties: Based on the different PCB layout, add the manual offset value to compensate the DC offset for each L and R channel, and they are different between headphone and headset. +- "realtek,impedance_sensing_num" + The matrix row number of the impedance sensing table. +- "realtek,impedance_sensing_table" + The matrix rows of the impedance sensing table are consisted by impedance + minimum, impedance maximun, volume, DC offset w/o and w/ mic of each L and + R channel accordingly. Pins on the device (for linking into audio routes) for RT5663: diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h index 7d00e58..7b90a8f 100644 --- a/include/sound/rt5663.h +++ b/include/sound/rt5663.h @@ -16,6 +16,9 @@ struct rt5663_platform_data { unsigned int dc_offset_r_manual; unsigned int dc_offset_l_manual_mic; unsigned int dc_offset_r_manual_mic; + + unsigned int impedance_sensing_num; + unsigned int *impedance_sensing_table; }; #endif diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ab9e0eb..767f219 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -38,6 +38,16 @@ enum { CODEC_VER_0, }; +struct impedance_mapping_table { + unsigned int imp_min; + unsigned int imp_max; + unsigned int vol; + unsigned int dc_offset_l_manual; + unsigned int dc_offset_r_manual; + unsigned int dc_offset_l_manual_mic; + unsigned int dc_offset_r_manual_mic; +}; + struct rt5663_priv { struct snd_soc_codec *codec; struct rt5663_platform_data pdata; @@ -45,6 +55,7 @@ struct rt5663_priv { struct delayed_work jack_detect_work; struct snd_soc_jack *hs_jack; struct timer_list btn_check_timer; + struct impedance_mapping_table *imp_table; int codec_ver; int sysclk; @@ -1575,6 +1586,9 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) rt5663->jack_type = SND_JACK_HEADSET; rt5663_enable_push_button_irq(codec, true); + if (rt5663->pdata.impedance_sensing_num) + break; + if (rt5663->pdata.dc_offset_l_manual_mic) { regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, rt5663->pdata.dc_offset_l_manual_mic >> @@ -1596,6 +1610,9 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) default: rt5663->jack_type = SND_JACK_HEADPHONE; + if (rt5663->pdata.impedance_sensing_num) + break; + if (rt5663->pdata.dc_offset_l_manual) { regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, rt5663->pdata.dc_offset_l_manual >> 16); @@ -1623,6 +1640,177 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) return rt5663->jack_type; } +static int rt5663_impedance_sensing(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int value, i, reg84, reg26, reg2fa, reg91, reg10, reg80; + + for (i = 0; i < rt5663->pdata.impedance_sensing_num; i++) { + if (rt5663->imp_table[i].vol == 7) + break; + } + + if (rt5663->jack_type == SND_JACK_HEADSET) { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual_mic & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual_mic & 0xffff); + } else { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual & 0xffff); + } + + reg84 = snd_soc_read(codec, RT5663_ASRC_2); + reg26 = snd_soc_read(codec, RT5663_STO1_ADC_MIXER); + reg2fa = snd_soc_read(codec, RT5663_DUMMY_1); + reg91 = snd_soc_read(codec, RT5663_HP_CHARGE_PUMP_1); + reg10 = snd_soc_read(codec, RT5663_RECMIX); + reg80 = snd_soc_read(codec, RT5663_GLB_CLK); + + snd_soc_update_bits(codec, RT5663_STO_DRE_1, 0x8000, 0); + snd_soc_write(codec, RT5663_ASRC_2, 0); + snd_soc_write(codec, RT5663_STO1_ADC_MIXER, 0x4040); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK | + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_VREF1 | RT5663_PWR_VREF2); + usleep_range(10000, 10005); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_FV1 | RT5663_PWR_FV2); + snd_soc_update_bits(codec, RT5663_GLB_CLK, RT5663_SCLK_SRC_MASK, + RT5663_SCLK_SRC_RCCLK); + snd_soc_update_bits(codec, RT5663_RC_CLK, RT5663_DIG_25M_CLK_MASK, + RT5663_DIG_25M_CLK_EN); + snd_soc_update_bits(codec, RT5663_ADDA_CLK_1, RT5663_I2S_PD1_MASK, 0); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0xff00); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0xfffc); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_1, 0x1232); + snd_soc_write(codec, RT5663_HP_LOGIC_2, 0x0005); + snd_soc_write(codec, RT5663_DEPOP_2, 0x3003); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0x0030); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0x0003); + snd_soc_update_bits(codec, RT5663_PWR_DIG_2, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F); + snd_soc_update_bits(codec, RT5663_PWR_DIG_1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_MASK | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_ON | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1); + msleep(40); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2); + msleep(30); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_2, 0x1371); + snd_soc_write(codec, RT5663_STO_DAC_MIXER, 0); + snd_soc_write(codec, RT5663_BYPASS_STO_DAC, 0x000c); + snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); + snd_soc_write(codec, RT5663_CHARGE_PUMP_1, 0x2224); + snd_soc_write(codec, RT5663_HP_OUT_EN, 0x8088); + snd_soc_write(codec, RT5663_CHOP_ADC, 0x3000); + snd_soc_write(codec, RT5663_ADDA_RST, 0xc000); + snd_soc_write(codec, RT5663_STO1_HPF_ADJ1, 0x3320); + snd_soc_write(codec, RT5663_HP_CALIB_2, 0x00c9); + snd_soc_write(codec, RT5663_DUMMY_1, 0x004c); + snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7733); + snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_write(codec, RT5663_STO_DRE_9, 0x0007); + snd_soc_write(codec, RT5663_STO_DRE_10, 0x0007); + snd_soc_write(codec, RT5663_DUMMY_2, 0x02a4); + snd_soc_write(codec, RT5663_RECMIX, 0x0005); + snd_soc_write(codec, RT5663_HP_IMP_SEN_1, 0x4334); + snd_soc_update_bits(codec, RT5663_IRQ_3, 0x0004, 0x0004); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0x2200); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0x6200); + + for (i = 0; i < 100; i++) { + msleep(20); + if (snd_soc_read(codec, RT5663_INT_ST_1) & 0x2) + break; + } + + value = snd_soc_read(codec, RT5663_HP_IMP_SEN_4); + + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0); + snd_soc_write(codec, RT5663_INT_ST_1, 0); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0); + snd_soc_update_bits(codec, RT5663_RC_CLK, RT5663_DIG_25M_CLK_MASK, + RT5663_DIG_25M_CLK_DIS); + snd_soc_write(codec, RT5663_GLB_CLK, reg80); + snd_soc_write(codec, RT5663_RECMIX, reg10); + snd_soc_write(codec, RT5663_DUMMY_2, 0x00a4); + snd_soc_write(codec, RT5663_DUMMY_1, reg2fa); + snd_soc_write(codec, RT5663_HP_CALIB_2, 0x00c8); + snd_soc_write(codec, RT5663_STO1_HPF_ADJ1, 0xb320); + snd_soc_write(codec, RT5663_ADDA_RST, 0xe400); + snd_soc_write(codec, RT5663_CHOP_ADC, 0x2000); + snd_soc_write(codec, RT5663_HP_OUT_EN, 0x0008); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2, 0); + snd_soc_update_bits(codec, RT5663_PWR_DIG_1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_MASK | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1, 0); + snd_soc_update_bits(codec, RT5663_PWR_DIG_2, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F, 0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0); + snd_soc_write(codec, RT5663_HP_LOGIC_2, 0); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_1, reg91); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK, 0); + snd_soc_write(codec, RT5663_STO1_ADC_MIXER, reg26); + snd_soc_write(codec, RT5663_ASRC_2, reg84); + + for (i = 0; i < rt5663->pdata.impedance_sensing_num; i++) { + if (value >= rt5663->imp_table[i].imp_min && + value <= rt5663->imp_table[i].imp_max) + break; + } + + snd_soc_update_bits(codec, RT5663_STO_DRE_9, RT5663_DRE_GAIN_HP_MASK, + rt5663->imp_table[i].vol); + snd_soc_update_bits(codec, RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_MASK, + rt5663->imp_table[i].vol); + + if (rt5663->jack_type == SND_JACK_HEADSET) { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual_mic & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual_mic & 0xffff); + } else { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual & 0xffff); + } + + return 0; +} + static int rt5663_button_detect(struct snd_soc_codec *codec) { int btn_type, val; @@ -1701,6 +1889,8 @@ static void rt5663_jack_detect_work(struct work_struct *work) break; case CODEC_VER_0: report = rt5663_jack_detect(rt5663->codec, 1); + if (rt5663->pdata.impedance_sensing_num) + rt5663_impedance_sensing(rt5663->codec); break; default: dev_err(codec->dev, "Unknown CODEC Version\n"); @@ -1796,10 +1986,6 @@ static const struct snd_kcontrol_new rt5663_v2_specific_controls[] = { }; static const struct snd_kcontrol_new rt5663_specific_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, - RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, - rt5663_hp_vol_tlv), /* Mic Boost Volume*/ SOC_SINGLE_TLV("IN1 Capture Volume", RT5663_CBJ_2, RT5663_GAIN_BST1_SHIFT, 8, 0, in_bst_tlv), @@ -1807,6 +1993,13 @@ static const struct snd_kcontrol_new rt5663_specific_controls[] = { SOC_ENUM("IF1 ADC Data Swap", rt5663_if1_adc_enum), }; +static const struct snd_kcontrol_new rt5663_hpvol_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, + RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, + rt5663_hp_vol_tlv), +}; + static int rt5663_is_sys_clk_from_pll(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_widget *sink) { @@ -2889,6 +3082,10 @@ static int rt5663_probe(struct snd_soc_codec *codec) ARRAY_SIZE(rt5663_specific_dapm_routes)); snd_soc_add_codec_controls(codec, rt5663_specific_controls, ARRAY_SIZE(rt5663_specific_controls)); + + if (!rt5663->imp_table) + snd_soc_add_codec_controls(codec, rt5663_hpvol_controls, + ARRAY_SIZE(rt5663_hpvol_controls)); break; } @@ -3177,6 +3374,8 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) { + int table_size; + device_property_read_u32(dev, "realtek,dc_offset_l_manual", &rt5663->pdata.dc_offset_l_manual); device_property_read_u32(dev, "realtek,dc_offset_r_manual", @@ -3185,6 +3384,17 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) &rt5663->pdata.dc_offset_l_manual_mic); device_property_read_u32(dev, "realtek,dc_offset_r_manual_mic", &rt5663->pdata.dc_offset_r_manual_mic); + device_property_read_u32(dev, "realtek,impedance_sensing_num", + &rt5663->pdata.impedance_sensing_num); + + if (rt5663->pdata.impedance_sensing_num) { + table_size = sizeof(struct impedance_mapping_table) * + rt5663->pdata.impedance_sensing_num; + rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); + device_property_read_u32_array(dev, + "realtek,impedance_sensing_table", + (u32 *)rt5663->imp_table, table_size); + } return 0; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rockchip: i2s: fix unbalanced clk_disable
by John Keeping 14 Sep '17

14 Sep '17

mclk is enabled and disabled only in i2s_runtime_{resume,suspend}() and we ensure that the device is runtime suspended before reaching this clk_disable_unprepare() call, so it is wrong to call it again here. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/rockchip_i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b6590467fd14..908211e1d6fc 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -692,7 +692,6 @@ static int rockchip_i2s_remove(struct platform_device *pdev) if (!pm_runtime_status_suspended(&pdev->dev)) i2s_runtime_suspend(&pdev->dev); - clk_disable_unprepare(i2s->mclk); clk_disable_unprepare(i2s->hclk); return 0; -- 2.14.1

2 1

[alsa-devel] [bug report] ASoC: soc-core: add snd_soc_rtdcom_xxx()
by Dan Carpenter 14 Sep '17

14 Sep '17

Hello Kuninori Morimoto, This is a semi-automatic email about new static checker warnings. The patch a0ac44115223: "ASoC: soc-core: add snd_soc_rtdcom_xxx()" from Aug 8, 2017, leads to the following Smatch complaint: sound/soc/soc-core.c:644 soc_free_pcm_runtime() error: we previously assumed 'rtd' could be null (see line 642) sound/soc/soc-core.c 641 { 642 if (rtd && rtd->codec_dais) ^^^ Check for NULL. 643 kfree(rtd->codec_dais); 644 snd_soc_rtdcom_del_all(rtd); ^^^ The check adds a new unchecked dereference inside the function. It's weird that we call INIT_LIST_HEAD(&rtd->component_list), btw. Don't we need to free the elements in the list? 645 kfree(rtd); 646 } regards, dan carpenter

3 5

[alsa-devel] [PATCH] ASoC: rt5514-spi: check irq status to schedule data copy
by Hsin-Yu Chao 13 Sep '17

13 Sep '17

For wake on voice use case, we need to copy data from DSP buffer to PCM stream when system wakes up by voice. However the edge triggered IRQ could be missed when system wakes up, in that case the irq function will not be called. Fix that by checking the irq status bit and schedule data copy accordingly. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514-spi.c | 17 +++++++++++++++-- sound/soc/codecs/rt5514-spi.h | 3 +++ 2 files changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 60f61af9ced3..f932d866056d 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -145,9 +145,8 @@ done: mutex_unlock(&rt5514_dsp->dma_lock); } -static irqreturn_t rt5514_spi_irq(int irq, void *data) +static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) { - struct rt5514_dsp *rt5514_dsp = data; u8 buf[8]; rt5514_dsp->get_size = 0; @@ -180,6 +179,13 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data) if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && rt5514_dsp->buf_rp && rt5514_dsp->buf_size) schedule_delayed_work(&rt5514_dsp->copy_work, 0); +} + +static irqreturn_t rt5514_spi_irq(int irq, void *data) +{ + struct rt5514_dsp *rt5514_dsp = data; + + rt5514_schedule_copy(rt5514_dsp); return IRQ_HANDLED; } @@ -199,12 +205,19 @@ static int rt5514_spi_hw_params(struct snd_pcm_substream *substream, struct rt5514_dsp *rt5514_dsp = snd_soc_platform_get_drvdata(rtd->platform); int ret; + u8 buf[8]; mutex_lock(&rt5514_dsp->dma_lock); ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); rt5514_dsp->substream = substream; rt5514_dsp->dma_offset = 0; + + /* Read IRQ status and schedule copy accordingly. */ + rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, sizeof(buf)); + if (buf[0] & RT5514_IRQ_STATUS_BIT) + rt5514_schedule_copy(rt5514_dsp); + mutex_unlock(&rt5514_dsp->dma_lock); return ret; diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h index a6434ee6ff03..c1a36647c119 100644 --- a/sound/soc/codecs/rt5514-spi.h +++ b/sound/soc/codecs/rt5514-spi.h @@ -20,6 +20,9 @@ #define RT5514_BUFFER_VOICE_BASE 0x18000200 #define RT5514_BUFFER_VOICE_LIMIT 0x18000204 #define RT5514_BUFFER_VOICE_WP 0x1800020c +#define RT5514_IRQ_CTRL 0x18002094 + +#define RT5514_IRQ_STATUS_BIT (0x1 << 5) /* SPI Command */ enum { -- 2.12.2

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fix ADG flags
by Kuninori Morimoto 13 Sep '17

13 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ADG driver is over-writing flags. This patch fixes it. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 938baff..e11ce50 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -479,10 +479,10 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, } if (req_rate[0] % 48000 == 0) - adg->flags = AUDIO_OUT_48; + adg->flags |= AUDIO_OUT_48; if (of_get_property(np, "clkout-lr-asynchronous", NULL)) - adg->flags = LRCLK_ASYNC; + adg->flags |= LRCLK_ASYNC; /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Add the function of impedance sensing
by Oder Chiou 13 Sep '17

13 Sep '17

Support the function of impedance sensing. It could be set the range number and the related parameters in the DTS. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5663.txt | 4 + include/sound/rt5663.h | 3 + sound/soc/codecs/rt5663.c | 218 ++++++++++++++++++++- 3 files changed, 221 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index ff38171..4dcd181 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -19,6 +19,10 @@ Optional properties: Based on the different PCB layout, add the manual offset value to compensate the DC offset for each L and R channel, and they are different between headphone and headset. +- "realtek,impedance_sensing_num" + The range number of impedance sensing. +- "realtek,impedance_sensing_table" + The parameters table of impedance sensing. Pins on the device (for linking into audio routes) for RT5663: diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h index 7d00e58..7b90a8f 100644 --- a/include/sound/rt5663.h +++ b/include/sound/rt5663.h @@ -16,6 +16,9 @@ struct rt5663_platform_data { unsigned int dc_offset_r_manual; unsigned int dc_offset_l_manual_mic; unsigned int dc_offset_r_manual_mic; + + unsigned int impedance_sensing_num; + unsigned int *impedance_sensing_table; }; #endif diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ab9e0eb..767f219 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -38,6 +38,16 @@ enum { CODEC_VER_0, }; +struct impedance_mapping_table { + unsigned int imp_min; + unsigned int imp_max; + unsigned int vol; + unsigned int dc_offset_l_manual; + unsigned int dc_offset_r_manual; + unsigned int dc_offset_l_manual_mic; + unsigned int dc_offset_r_manual_mic; +}; + struct rt5663_priv { struct snd_soc_codec *codec; struct rt5663_platform_data pdata; @@ -45,6 +55,7 @@ struct rt5663_priv { struct delayed_work jack_detect_work; struct snd_soc_jack *hs_jack; struct timer_list btn_check_timer; + struct impedance_mapping_table *imp_table; int codec_ver; int sysclk; @@ -1575,6 +1586,9 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) rt5663->jack_type = SND_JACK_HEADSET; rt5663_enable_push_button_irq(codec, true); + if (rt5663->pdata.impedance_sensing_num) + break; + if (rt5663->pdata.dc_offset_l_manual_mic) { regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, rt5663->pdata.dc_offset_l_manual_mic >> @@ -1596,6 +1610,9 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) default: rt5663->jack_type = SND_JACK_HEADPHONE; + if (rt5663->pdata.impedance_sensing_num) + break; + if (rt5663->pdata.dc_offset_l_manual) { regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, rt5663->pdata.dc_offset_l_manual >> 16); @@ -1623,6 +1640,177 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) return rt5663->jack_type; } +static int rt5663_impedance_sensing(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int value, i, reg84, reg26, reg2fa, reg91, reg10, reg80; + + for (i = 0; i < rt5663->pdata.impedance_sensing_num; i++) { + if (rt5663->imp_table[i].vol == 7) + break; + } + + if (rt5663->jack_type == SND_JACK_HEADSET) { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual_mic & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual_mic & 0xffff); + } else { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual & 0xffff); + } + + reg84 = snd_soc_read(codec, RT5663_ASRC_2); + reg26 = snd_soc_read(codec, RT5663_STO1_ADC_MIXER); + reg2fa = snd_soc_read(codec, RT5663_DUMMY_1); + reg91 = snd_soc_read(codec, RT5663_HP_CHARGE_PUMP_1); + reg10 = snd_soc_read(codec, RT5663_RECMIX); + reg80 = snd_soc_read(codec, RT5663_GLB_CLK); + + snd_soc_update_bits(codec, RT5663_STO_DRE_1, 0x8000, 0); + snd_soc_write(codec, RT5663_ASRC_2, 0); + snd_soc_write(codec, RT5663_STO1_ADC_MIXER, 0x4040); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK | + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_VREF1 | RT5663_PWR_VREF2); + usleep_range(10000, 10005); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_FV1 | RT5663_PWR_FV2); + snd_soc_update_bits(codec, RT5663_GLB_CLK, RT5663_SCLK_SRC_MASK, + RT5663_SCLK_SRC_RCCLK); + snd_soc_update_bits(codec, RT5663_RC_CLK, RT5663_DIG_25M_CLK_MASK, + RT5663_DIG_25M_CLK_EN); + snd_soc_update_bits(codec, RT5663_ADDA_CLK_1, RT5663_I2S_PD1_MASK, 0); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0xff00); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0xfffc); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_1, 0x1232); + snd_soc_write(codec, RT5663_HP_LOGIC_2, 0x0005); + snd_soc_write(codec, RT5663_DEPOP_2, 0x3003); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0x0030); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0x0003); + snd_soc_update_bits(codec, RT5663_PWR_DIG_2, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F); + snd_soc_update_bits(codec, RT5663_PWR_DIG_1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_MASK | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_ON | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1); + msleep(40); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2); + msleep(30); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_2, 0x1371); + snd_soc_write(codec, RT5663_STO_DAC_MIXER, 0); + snd_soc_write(codec, RT5663_BYPASS_STO_DAC, 0x000c); + snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); + snd_soc_write(codec, RT5663_CHARGE_PUMP_1, 0x2224); + snd_soc_write(codec, RT5663_HP_OUT_EN, 0x8088); + snd_soc_write(codec, RT5663_CHOP_ADC, 0x3000); + snd_soc_write(codec, RT5663_ADDA_RST, 0xc000); + snd_soc_write(codec, RT5663_STO1_HPF_ADJ1, 0x3320); + snd_soc_write(codec, RT5663_HP_CALIB_2, 0x00c9); + snd_soc_write(codec, RT5663_DUMMY_1, 0x004c); + snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7733); + snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_write(codec, RT5663_STO_DRE_9, 0x0007); + snd_soc_write(codec, RT5663_STO_DRE_10, 0x0007); + snd_soc_write(codec, RT5663_DUMMY_2, 0x02a4); + snd_soc_write(codec, RT5663_RECMIX, 0x0005); + snd_soc_write(codec, RT5663_HP_IMP_SEN_1, 0x4334); + snd_soc_update_bits(codec, RT5663_IRQ_3, 0x0004, 0x0004); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0x2200); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0x6200); + + for (i = 0; i < 100; i++) { + msleep(20); + if (snd_soc_read(codec, RT5663_INT_ST_1) & 0x2) + break; + } + + value = snd_soc_read(codec, RT5663_HP_IMP_SEN_4); + + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0); + snd_soc_write(codec, RT5663_INT_ST_1, 0); + snd_soc_write(codec, RT5663_HP_LOGIC_1, 0); + snd_soc_update_bits(codec, RT5663_RC_CLK, RT5663_DIG_25M_CLK_MASK, + RT5663_DIG_25M_CLK_DIS); + snd_soc_write(codec, RT5663_GLB_CLK, reg80); + snd_soc_write(codec, RT5663_RECMIX, reg10); + snd_soc_write(codec, RT5663_DUMMY_2, 0x00a4); + snd_soc_write(codec, RT5663_DUMMY_1, reg2fa); + snd_soc_write(codec, RT5663_HP_CALIB_2, 0x00c8); + snd_soc_write(codec, RT5663_STO1_HPF_ADJ1, 0xb320); + snd_soc_write(codec, RT5663_ADDA_RST, 0xe400); + snd_soc_write(codec, RT5663_CHOP_ADC, 0x2000); + snd_soc_write(codec, RT5663_HP_OUT_EN, 0x0008); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5663_PWR_RECMIX1 | RT5663_PWR_RECMIX2, 0); + snd_soc_update_bits(codec, RT5663_PWR_DIG_1, + RT5663_PWR_DAC_L1 | RT5663_PWR_DAC_R1 | + RT5663_PWR_LDO_DACREF_MASK | RT5663_PWR_ADC_L1 | + RT5663_PWR_ADC_R1, 0); + snd_soc_update_bits(codec, RT5663_PWR_DIG_2, + RT5663_PWR_ADC_S1F | RT5663_PWR_DAC_S1F, 0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0); + snd_soc_write(codec, RT5663_HP_LOGIC_2, 0); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_1, reg91); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK, 0); + snd_soc_write(codec, RT5663_STO1_ADC_MIXER, reg26); + snd_soc_write(codec, RT5663_ASRC_2, reg84); + + for (i = 0; i < rt5663->pdata.impedance_sensing_num; i++) { + if (value >= rt5663->imp_table[i].imp_min && + value <= rt5663->imp_table[i].imp_max) + break; + } + + snd_soc_update_bits(codec, RT5663_STO_DRE_9, RT5663_DRE_GAIN_HP_MASK, + rt5663->imp_table[i].vol); + snd_soc_update_bits(codec, RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_MASK, + rt5663->imp_table[i].vol); + + if (rt5663->jack_type == SND_JACK_HEADSET) { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual_mic & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual_mic >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual_mic & 0xffff); + } else { + snd_soc_write(codec, RT5663_MIC_DECRO_2, + rt5663->imp_table[i].dc_offset_l_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_3, + rt5663->imp_table[i].dc_offset_l_manual & 0xffff); + snd_soc_write(codec, RT5663_MIC_DECRO_5, + rt5663->imp_table[i].dc_offset_r_manual >> 16); + snd_soc_write(codec, RT5663_MIC_DECRO_6, + rt5663->imp_table[i].dc_offset_r_manual & 0xffff); + } + + return 0; +} + static int rt5663_button_detect(struct snd_soc_codec *codec) { int btn_type, val; @@ -1701,6 +1889,8 @@ static void rt5663_jack_detect_work(struct work_struct *work) break; case CODEC_VER_0: report = rt5663_jack_detect(rt5663->codec, 1); + if (rt5663->pdata.impedance_sensing_num) + rt5663_impedance_sensing(rt5663->codec); break; default: dev_err(codec->dev, "Unknown CODEC Version\n"); @@ -1796,10 +1986,6 @@ static const struct snd_kcontrol_new rt5663_v2_specific_controls[] = { }; static const struct snd_kcontrol_new rt5663_specific_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, - RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, - rt5663_hp_vol_tlv), /* Mic Boost Volume*/ SOC_SINGLE_TLV("IN1 Capture Volume", RT5663_CBJ_2, RT5663_GAIN_BST1_SHIFT, 8, 0, in_bst_tlv), @@ -1807,6 +1993,13 @@ static const struct snd_kcontrol_new rt5663_specific_controls[] = { SOC_ENUM("IF1 ADC Data Swap", rt5663_if1_adc_enum), }; +static const struct snd_kcontrol_new rt5663_hpvol_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, + RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, + rt5663_hp_vol_tlv), +}; + static int rt5663_is_sys_clk_from_pll(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_widget *sink) { @@ -2889,6 +3082,10 @@ static int rt5663_probe(struct snd_soc_codec *codec) ARRAY_SIZE(rt5663_specific_dapm_routes)); snd_soc_add_codec_controls(codec, rt5663_specific_controls, ARRAY_SIZE(rt5663_specific_controls)); + + if (!rt5663->imp_table) + snd_soc_add_codec_controls(codec, rt5663_hpvol_controls, + ARRAY_SIZE(rt5663_hpvol_controls)); break; } @@ -3177,6 +3374,8 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) { + int table_size; + device_property_read_u32(dev, "realtek,dc_offset_l_manual", &rt5663->pdata.dc_offset_l_manual); device_property_read_u32(dev, "realtek,dc_offset_r_manual", @@ -3185,6 +3384,17 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) &rt5663->pdata.dc_offset_l_manual_mic); device_property_read_u32(dev, "realtek,dc_offset_r_manual_mic", &rt5663->pdata.dc_offset_r_manual_mic); + device_property_read_u32(dev, "realtek,impedance_sensing_num", + &rt5663->pdata.impedance_sensing_num); + + if (rt5663->pdata.impedance_sensing_num) { + table_size = sizeof(struct impedance_mapping_table) * + rt5663->pdata.impedance_sensing_num; + rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); + device_property_read_u32_array(dev, + "realtek,impedance_sensing_table", + (u32 *)rt5663->imp_table, table_size); + } return 0; } -- 2.7.4

2 1

[alsa-devel] [PATCH] alsa-lib: topology: fix usage of SND_TPLG_INDEX_ALL when checking routes
by Liam Girdwood 12 Sep '17

12 Sep '17

Make sure SND_TPLG_INDEX_ALL is used correctly when checking routes so that connecting routes of different indexes does not emit any warnings. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/dapm.c | 4 ++-- src/topology/elem.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 66892a6..0ddbf96 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -279,7 +279,7 @@ int tplg_build_routes(snd_tplg_t *tplg) } if (!tplg_elem_lookup(&tplg->widget_list, route->sink, - SND_TPLG_TYPE_DAPM_WIDGET, elem->index)) { + SND_TPLG_TYPE_DAPM_WIDGET, SND_TPLG_INDEX_ALL)) { SNDERR("warning: undefined sink widget/stream '%s'\n", route->sink); } @@ -302,7 +302,7 @@ int tplg_build_routes(snd_tplg_t *tplg) } if (!tplg_elem_lookup(&tplg->widget_list, route->source, - SND_TPLG_TYPE_DAPM_WIDGET, elem->index)) { + SND_TPLG_TYPE_DAPM_WIDGET, SND_TPLG_INDEX_ALL)) { SNDERR("warning: Undefined source widget/stream '%s'\n", route->source); } diff --git a/src/topology/elem.c b/src/topology/elem.c index 89a4ac9..9a7c7b7 100644 --- a/src/topology/elem.c +++ b/src/topology/elem.c @@ -123,7 +123,7 @@ struct tplg_elem *tplg_elem_lookup(struct list_head *base, const char* id, return elem; /* SND_TPLG_INDEX_ALL is the default value "0" and applicable for all use cases */ - if ((elem->index != SND_TPLG_INDEX_ALL) + if ((index != SND_TPLG_INDEX_ALL) && (elem->index > index)) break; } -- 1.9.1

2 1

[alsa-devel] [PATCH] alsa-lib: topology: Fix private data for BEs
by Liam Girdwood 12 Sep '17

12 Sep '17

Private data was not being added to BEs. Fix this. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/data.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/src/topology/data.c b/src/topology/data.c index 5b83b9f..f96ff9b 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -50,7 +50,9 @@ struct snd_soc_tplg_private *get_priv_data(struct tplg_elem *elem) case SND_TPLG_TYPE_DAI: priv = &elem->dai->priv; break; - + case SND_TPLG_TYPE_BE: + priv = &elem->link->priv; + break; default: SNDERR("error: '%s': no support for private data for type %d\n", elem->id, elem->type); -- 1.9.1

1 1

[alsa-devel] [PATCH] pcm: softvol: add support for S24_LE
by Jörg Krause 12 Sep '17

12 Sep '17

Tested with a Wolfson WM8524 DAC on a i.MX6UL board running Linux version 4.13.0-next-20170907. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- src/pcm/pcm_softvol.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 1fe5784d..d159fb1f 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -309,6 +309,8 @@ static void softvol_convert_stereo_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(short, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples, fallthrough */ case SND_PCM_FORMAT_S32_LE: case SND_PCM_FORMAT_S32_BE: /* 32bit samples */ @@ -360,6 +362,8 @@ static void softvol_convert_mono_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(short, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples, fallthrough */ case SND_PCM_FORMAT_S32_LE: case SND_PCM_FORMAT_S32_BE: /* 32bit samples */ @@ -422,6 +426,7 @@ static int snd_pcm_softvol_hw_refine_cprepare(snd_pcm_t *pcm, { (1ULL << SND_PCM_FORMAT_S16_LE) | (1ULL << SND_PCM_FORMAT_S16_BE) | + (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S32_LE) | (1ULL << SND_PCM_FORMAT_S32_BE), (1ULL << (SND_PCM_FORMAT_S24_3LE - 32)) @@ -577,10 +582,11 @@ static int snd_pcm_softvol_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * param if (slave->format != SND_PCM_FORMAT_S16_LE && slave->format != SND_PCM_FORMAT_S16_BE && slave->format != SND_PCM_FORMAT_S24_3LE && + slave->format != SND_PCM_FORMAT_S24_LE && slave->format != SND_PCM_FORMAT_S32_LE && slave->format != SND_PCM_FORMAT_S32_BE) { - SNDERR("softvol supports only S16_LE, S16_BE, S24_3LE, S32_LE " - " or S32_BE"); + SNDERR("softvol supports only S16_LE, S16_BE, S24_LE, S24_3LE, " + "S32_LE or S32_BE"); return -EINVAL; } svol->sformat = slave->format; @@ -863,6 +869,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) return -EINVAL; @@ -1082,9 +1089,10 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) { - SNDERR("only S16_LE, S16_BE, S24_3LE, S32_LE or S32_BE format is supported"); + SNDERR("only S16_LE, S16_BE, S24_LE, S24_3LE, S32_LE or S32_BE format is supported"); snd_config_delete(sconf); return -EINVAL; } -- 2.14.1

4 9

[alsa-devel] usb/midi: use-after-free in snd_rawmidi_dev_seq_free
by Andrey Konovalov 12 Sep '17

12 Sep '17

Hi! While working on adding USB fuzzing support to syzkaller, I got the following crash. On commit 81a84ad3cb5711cec79f4dd53a4ce026b092c432 (4.13+). My .config is attached. I don't have a reproducer for this bug. ================================================================== BUG: KASAN: use-after-free in snd_rawmidi_dev_seq_free+0x69/0x70 sound/core/rawmidi.c:1617 Write of size 8 at addr ffff88006c611d90 by task kworker/2:1/567 CPU: 2 PID: 567 Comm: kworker/2:1 Not tainted 4.13.0+ #29 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: events autoload_drivers Call Trace: __dump_stack lib/dump_stack.c:16 [inline] dump_stack+0x192/0x22c lib/dump_stack.c:52 print_address_description+0x78/0x280 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 [inline] kasan_report+0x230/0x340 mm/kasan/report.c:409 __asan_report_store8_noabort+0x1c/0x20 mm/kasan/report.c:435 snd_rawmidi_dev_seq_free+0x69/0x70 sound/core/rawmidi.c:1617 snd_seq_dev_release+0x4f/0x70 sound/core/seq_device.c:192 device_release+0x13f/0x210 drivers/base/core.c:814 kobject_cleanup lib/kobject.c:648 [inline] kobject_release lib/kobject.c:677 [inline] kref_put include/linux/kref.h:70 [inline] kobject_put+0x145/0x240 lib/kobject.c:694 put_device+0x25/0x30 drivers/base/core.c:1799 klist_devices_put+0x36/0x40 drivers/base/bus.c:827 klist_next+0x264/0x4a0 lib/klist.c:403 next_device drivers/base/bus.c:270 [inline] bus_for_each_dev+0x17e/0x210 drivers/base/bus.c:312 autoload_drivers+0x3b/0x50 sound/core/seq_device.c:117 process_one_work+0x9fb/0x1570 kernel/workqueue.c:2097 worker_thread+0x1e4/0x1350 kernel/workqueue.c:2231 kthread+0x324/0x3f0 kernel/kthread.c:231 ret_from_fork+0x25/0x30 arch/x86/entry/entry_64.S:425 Allocated by task 567: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551 kmem_cache_alloc_trace+0x66/0x170 mm/slub.c:2739 kmalloc include/linux/slab.h:493 [inline] kzalloc include/linux/slab.h:666 [inline] snd_rawmidi_new+0x111/0x500 sound/core/rawmidi.c:1535 snd_usbmidi_create_rawmidi sound/usb/midi.c:2221 [inline] __snd_usbmidi_create+0x5cb/0x2700 sound/usb/midi.c:2450 snd_usbmidi_create sound/usb/midi.h:53 [inline] create_any_midi_quirk+0x54/0x70 sound/usb/quirks.c:103 snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:561 usb_audio_probe+0xf98/0x2aa0 sound/usb/card.c:598 usb_probe_interface+0x351/0x8d0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:385 [inline] driver_probe_device+0x610/0xa00 drivers/base/dd.c:529 __device_attach_driver+0x230/0x290 drivers/base/dd.c:625 bus_for_each_drv+0x15e/0x210 drivers/base/bus.c:463 __device_attach+0x269/0x3c0 drivers/base/dd.c:682 device_initial_probe+0x1f/0x30 drivers/base/dd.c:729 bus_probe_device+0x1da/0x280 drivers/base/bus.c:523 device_add+0xcf9/0x1640 drivers/base/core.c:1703 usb_set_configuration+0x1064/0x1890 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:385 [inline] driver_probe_device+0x610/0xa00 drivers/base/dd.c:529 __device_attach_driver+0x230/0x290 drivers/base/dd.c:625 bus_for_each_drv+0x15e/0x210 drivers/base/bus.c:463 __device_attach+0x269/0x3c0 drivers/base/dd.c:682 device_initial_probe+0x1f/0x30 drivers/base/dd.c:729 bus_probe_device+0x1da/0x280 drivers/base/bus.c:523 device_add+0xcf9/0x1640 drivers/base/core.c:1703 usb_new_device+0x91d/0x17a0 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4890 [inline] hub_port_connect_change drivers/usb/core/hub.c:4996 [inline] port_event drivers/usb/core/hub.c:5102 [inline] hub_event+0x261c/0x3960 drivers/usb/core/hub.c:5186 process_one_work+0x9fb/0x1570 kernel/workqueue.c:2097 worker_thread+0x1e4/0x1350 kernel/workqueue.c:2231 kthread+0x324/0x3f0 kernel/kthread.c:231 ret_from_fork+0x25/0x30 arch/x86/entry/entry_64.S:425 Freed by task 1454: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_slab_free+0x72/0xc0 mm/kasan/kasan.c:524 slab_free_hook mm/slub.c:1357 [inline] slab_free_freelist_hook mm/slub.c:1379 [inline] slab_free mm/slub.c:2955 [inline] kfree+0x96/0x1a0 mm/slub.c:3883 release_rawmidi_device+0x1e/0x30 sound/core/rawmidi.c:1502 device_release+0x13f/0x210 drivers/base/core.c:814 kobject_cleanup lib/kobject.c:648 [inline] kobject_release lib/kobject.c:677 [inline] kref_put include/linux/kref.h:70 [inline] kobject_put+0x145/0x240 lib/kobject.c:694 put_device+0x25/0x30 drivers/base/core.c:1799 snd_rawmidi_free.part.15+0x14b/0x190 sound/core/rawmidi.c:1603 snd_rawmidi_free sound/core/rawmidi.c:1589 [inline] snd_rawmidi_dev_free+0x43/0x60 sound/core/rawmidi.c:1610 __snd_device_free+0x1e5/0x380 sound/core/device.c:91 snd_device_free_all+0x8f/0xe0 sound/core/device.c:244 snd_card_do_free sound/core/init.c:463 [inline] release_card_device+0x26/0x150 sound/core/init.c:181 device_release+0x13f/0x210 drivers/base/core.c:814 kobject_cleanup lib/kobject.c:648 [inline] kobject_release lib/kobject.c:677 [inline] kref_put include/linux/kref.h:70 [inline] kobject_put+0x145/0x240 lib/kobject.c:694 put_device+0x25/0x30 drivers/base/core.c:1799 snd_card_free_when_closed+0x34/0x50 sound/core/init.c:491 usb_audio_disconnect+0x5fb/0x860 sound/usb/card.c:684 usb_unbind_interface+0x216/0xa90 drivers/usb/core/driver.c:423 __device_release_driver drivers/base/dd.c:833 [inline] device_release_driver_internal+0x4da/0x5a0 drivers/base/dd.c:863 device_release_driver+0x1e/0x30 drivers/base/dd.c:888 bus_remove_device+0x2da/0x490 drivers/base/bus.c:565 device_del+0x5b1/0xaa0 drivers/base/core.c:1853 usb_disable_device+0x1e9/0x680 drivers/usb/core/message.c:1170 usb_disconnect+0x2ca/0x830 drivers/usb/core/hub.c:2124 hub_port_connect drivers/usb/core/hub.c:4741 [inline] hub_port_connect_change drivers/usb/core/hub.c:4996 [inline] port_event drivers/usb/core/hub.c:5102 [inline] hub_event+0x14d8/0x3960 drivers/usb/core/hub.c:5186 process_one_work+0x9fb/0x1570 kernel/workqueue.c:2097 worker_thread+0x1e4/0x1350 kernel/workqueue.c:2231 kthread+0x324/0x3f0 kernel/kthread.c:231 ret_from_fork+0x25/0x30 arch/x86/entry/entry_64.S:425 The buggy address belongs to the object at ffff88006c611980 which belongs to the cache kmalloc-2048 of size 2048 The buggy address is located 1040 bytes inside of 2048-byte region [ffff88006c611980, ffff88006c612180) The buggy address belongs to the page: page:ffffea0001b18400 count:1 mapcount:0 mapping: (null) index:0xffff88006c617700 compound_mapcount: 0 flags: 0x500000000008100(slab|head) raw: 0500000000008100 0000000000000000 ffff88006c617700 00000001800f0007 raw: ffffea0001b10400 0000000300000003 ffff88003e803040 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff88006c611c80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff88006c611d00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >ffff88006c611d80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff88006c611e00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff88006c611e80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ==================================================================

2 4

[alsa-devel] [PATCH] ALSA: msnd: Optimize / harden DSP and MIDI loops
by grygorii tertychnyi 12 Sep '17

12 Sep '17

Hi Greg, Could you please apply it for 4.4-stable. This fixes https://nvd.nist.gov/vuln/detail/CVE-2017-9985 Takashi Iwai (1): ALSA: msnd: Optimize / harden DSP and MIDI loops sound/isa/msnd/msnd_midi.c | 30 +++++++++++++++--------------- sound/isa/msnd/msnd_pinnacle.c | 23 ++++++++++++----------- 2 files changed, 27 insertions(+), 26 deletions(-) -- 2.10.3.dirty

5 7

[alsa-devel] Question: alsa-lib snd_pcm_htimestamp inner working and precision
by gianluca a 12 Sep '17

12 Sep '17

Hi all, I am trying to understand the precision of htimestamp for sample captured and I am not sure if I understood well, because I have found old informations togheter with new informations and I've the doubt that something changed. I am interested in this because of a project for my degree thesis. snd_pcm_hw_params_supports_audio_wallclock_ts returns 0 on capture, so my hardware doesn't support wallclock timestamps on capture. I've understood that SND_PCM_TSTAMP_ENABLE is equivalent to SND_PCM_TSTAMP_MMAP, so it updates timestamp at every position update on DMA from the driver. I've read elsewhere that this timestamp is updated every period-size ( equal interrupt), is it correct? I've digged in the code to understand well inner working of htimestamp, but I've found that snd_pcm_htimestamp in pcm.c calls: err = pcm->fast_ops->htimestamp(pcm->fast_op_arg, avail, tstamp); And I have not found where fast_ops is initialized, maybe in the driver? Anyway I thougth that maybe in my case the called function is snd_pcm_hw_htimestamp in pcm_hw.c : "while (1) { avail1 = snd_pcm_hw_avail_update(pcm); if (avail1 < 0) return avail1; if (ok && (snd_pcm_uframes_t)avail1 == *avail) break; *avail = avail1; *tstamp = snd_pcm_hw_fast_tstamp(pcm); ok = 1; }" that calls snd_pcm_hw_fast_tstamp, that calls finally: "#define FAST_PCM_TSTAMP(hw) \ ((hw)->mmap_status->tstamp) " And finally is this the timestamp that is updated every period-size? Am I correct that in the case that hardware doesn't support wallclock timestamp is snd_pcm_hw_htimestamp in pcm_hw.c the called function? Thanks in advance

1 0

[alsa-devel] [PATCH v5 00/17] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 12 Sep '17

12 Sep '17

The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Summary of changes since V4: - Fixed review comments on MFD code - Fixed dependency error in core codec driver - minor bugfix to core codec driver Richard Fitzgerald (17): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add DT bindings for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add DT bindings for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 Documentation/devicetree/bindings/mfd/madera.txt | 102 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 99 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 67 + MAINTAINERS | 23 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 301 ++ drivers/mfd/Kconfig | 48 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1610 +++++++ drivers/mfd/cs47l85-tables.c | 3010 +++++++++++++ drivers/mfd/cs47l90-tables.c | 2675 ++++++++++++ drivers/mfd/madera-core.c | 599 +++ drivers/mfd/madera-i2c.c | 139 + drivers/mfd/madera-spi.c | 138 + drivers/mfd/madera.h | 44 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1074 +++++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 27 + include/linux/irqchip/irq-madera-pdata.h | 23 + include/linux/irqchip/irq-madera.h | 98 + include/linux/mfd/madera/core.h | 196 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 3987 ++++++++++++++++++ include/sound/madera-pdata.h | 61 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1748 ++++++++ sound/soc/codecs/cs47l85.c | 2722 ++++++++++++ sound/soc/codecs/cs47l90.c | 2661 ++++++++++++ sound/soc/codecs/madera.c | 4439 ++++++++++++++++++++ sound/soc/codecs/madera.h | 470 +++ 46 files changed, 27066 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 1.9.1

2 18

[alsa-devel] [PATCH] ALSA: FireWire-MOTU: Use common error handling code in snd_motu_stream_start_duplex()
by SF Markus Elfring 12 Sep '17

12 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 13:30:14 +0200 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/firewire/motu/motu-stream.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index dc5541c8b359..73e7a5e527fc 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -253,24 +253,21 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu, unsigned int rate) if (err < 0) { dev_err(&motu->unit->device, "fail to start isochronous comm: %d\n", err); - stop_both_streams(motu); - return err; + goto stop_streams; } err = start_isoc_ctx(motu, &motu->rx_stream); if (err < 0) { dev_err(&motu->unit->device, "fail to start IT context: %d\n", err); - stop_both_streams(motu); - return err; + goto stop_streams; } err = protocol->switch_fetching_mode(motu, true); if (err < 0) { dev_err(&motu->unit->device, "fail to enable frame fetching: %d\n", err); - stop_both_streams(motu); - return err; + goto stop_streams; } } @@ -281,12 +278,15 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu, unsigned int rate) dev_err(&motu->unit->device, "fail to start IR context: %d", err); amdtp_stream_stop(&motu->rx_stream); - stop_both_streams(motu); - return err; + goto stop_streams; } } return 0; + +stop_streams: + stop_both_streams(motu); + return err; } void snd_motu_stream_stop_duplex(struct snd_motu *motu) -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: rt5670: add set_bclk_ratio in dai ops
by Bard Liao 12 Sep '17

12 Sep '17

We need to set a specific bit for 50 bclk rate. So add set_bclk_ratio function to set the bit. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9545764..dbdd01c 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2582,6 +2582,22 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5670_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (dai->id != RT5670_AIF1) + return 0; + + if (ratio == 50) + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x0800, 0x0800); + else + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x0800, 0x0); + + return 0; +} + static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2712,6 +2728,7 @@ static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_fmt = rt5670_set_dai_fmt, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, + .set_bclk_ratio = rt5670_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5670_dai[] = { -- 2.7.4

2 1

[alsa-devel] [PATCH 3/3] ASoC: max98927: Removed obsolete variables
by Ryan Lee 11 Sep '17

11 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changelog: Removed obsolete variables that have never been referred from driver before. sound/soc/codecs/max98927.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index 98b5f17..d215f60 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -263,17 +263,13 @@ struct max98927_priv { struct regmap *regmap; struct snd_soc_codec *codec; - struct max98927_pdata *pdata; - unsigned int spk_gain; unsigned int sysclk; unsigned int v_l_slot; unsigned int i_l_slot; bool interleave_mode; unsigned int ch_size; - unsigned int rate; unsigned int iface; unsigned int master; - unsigned int digital_gain; bool tdm_mode; }; #endif -- 1.9.1

1 0

[alsa-devel] [PATCH v3 00/11] Add STM32 DFSDM support
by Arnaud Pouliquen 11 Sep '17

11 Sep '17

Hello, I kept it as RFC as i reworked the global design (but also because not full validated...). Regarding Mark and Jonathan's comments, I decided to try an implementation with DMA support in IIO instead of handling it in ASOC. Solution seems more straightforward but as DMA cyclic mode and associated buffer block API is not part of IIO today, I still need some hacks... To be honest, I tried to integrate DMA cyclic management and associated buffer block Consumer API. but no enough expertise on IIO to success... :( Advantage vs V2: - No more data processing need in ASoC , done in IIO - ASoC uses IIO consumer interface ( except for buffer callback) Drawback : - Need to create a "SPI bus" trigger to allo to use Buffer consumer API => Need to find a solution to set this trigger by default in IIO audio driver - No standard buffer management - Implement DMA cyclic in DFSDM IIO driver. - Define specific API to handle buffer callback per period instead of callback per samples generated by iio_push_to_buffers_with_timestamp. V2-V3 delta: - New patches to support ASoC DMA codec in DT - ASoC: Add Bindins for DMIC codec driver. - ASoC: codec: add DT support in dmic codec. - New patches to allow in-kernel set of IIO buffer size and watermark - IIO: consumer: allow to set buffer sizes. - IIO DFSDM drivers - Split audio and ADC support in 2 drivers. - Implement DMA cyclic mode. - Add SPI bus Trigger for buffer management. - IIO sigma delta adc drivers - Suppress "simple and rename driver. - ASoC driver - Suppress DMA engine. - Suppress copy ops. - Use IIO consmumer interface to enable/Disable DFSDM. Regards, Arnaud Arnaud Pouliquen (11): iio: Add hardware consumer support IIO: Add DT bindings for sigma delta adc modulator IIO: ADC: add sigma delta modulator support IIO: add DT bindings for stm32 DFSDM filter IIO: ADC: add stm32 DFSDM support for Sigma delta ADC IIO: ADC: add stm32 DFSDM support for PDM microphone IIO: consumer: allow to set buffer sizes ASoC: Add Bindins for DMIC codec driver ASoC: codec: add DT support in dmic codec ASoC: add bindings for stm32 DFSDM filter ASoC: stm32: add DFSDM DAI support .../bindings/iio/adc/sigma-delta-modulator.txt | 13 + .../bindings/iio/adc/st,stm32-dfsdm-adc.txt | 120 ++++ Documentation/devicetree/bindings/sound/dmic.txt | 11 + .../devicetree/bindings/sound/st,stm32-adfsdm.txt | 41 ++ drivers/iio/Kconfig | 6 + drivers/iio/Makefile | 1 + drivers/iio/adc/Kconfig | 50 ++ drivers/iio/adc/Makefile | 4 + drivers/iio/adc/sd_adc_modulator.c | 98 +++ drivers/iio/adc/stm32-dfsdm-adc.c | 419 ++++++++++++ drivers/iio/adc/stm32-dfsdm-audio.c | 720 +++++++++++++++++++++ drivers/iio/adc/stm32-dfsdm-core.c | 658 +++++++++++++++++++ drivers/iio/adc/stm32-dfsdm.h | 372 +++++++++++ drivers/iio/buffer/industrialio-buffer-cb.c | 12 + drivers/iio/hw_consumer.c | 156 +++++ include/linux/iio/adc/stm32-dfsdm-audio.h | 41 ++ include/linux/iio/consumer.h | 13 + include/linux/iio/hw_consumer.h | 12 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/dmic.c | 8 + sound/soc/stm/Kconfig | 10 + sound/soc/stm/Makefile | 2 + sound/soc/stm/stm32_adfsdm.c | 362 +++++++++++ 24 files changed, 3131 insertions(+) create mode 100644 Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt create mode 100644 Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt create mode 100644 Documentation/devicetree/bindings/sound/dmic.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt create mode 100644 drivers/iio/adc/sd_adc_modulator.c create mode 100644 drivers/iio/adc/stm32-dfsdm-adc.c create mode 100644 drivers/iio/adc/stm32-dfsdm-audio.c create mode 100644 drivers/iio/adc/stm32-dfsdm-core.c create mode 100644 drivers/iio/adc/stm32-dfsdm.h create mode 100644 drivers/iio/hw_consumer.c create mode 100644 include/linux/iio/adc/stm32-dfsdm-audio.h create mode 100644 include/linux/iio/hw_consumer.h create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_adfsdm.c V2: Patch-set associated to this RFC proposes an implementation of the DFSDM features shared between ASoC and IIO frameworks. Patch-set is only a Skeleton of the drivers, so a base to discuss and validate a design. It contains minimum code to allow probing (with DT) and to expose the ASoC and IIO ABI. Hope that is sufficent in a first step to allow to focus on APIs. In this patch-set there are two new APIs used: - IIO in-kern API: based on hw_customer API proposed by Lars - ASOC <-> IIO API inspired by API defined for hdmi-codec for ASoC/DRM interconnect. API is dedicated to DFSDM only. Notice also that this design is based on following assumption: - Audio stream ABI interface is ASoC, no access to data through IIO ABI for PDM. - ASoC DMA should be used for audio transfer as designed for real time stream. - Need some runtime parameters exchange between ASoC and IIO due to the correlation between the sample frequency, the DFSDM decimation factor and the associated scaling. - "ASoC: dmaengine_pcm: add copy support" patch: I added a patch in ASoC that allows to implement a copy function to process data after DMA transfer. Requested, as DFSDM samples captured contain channel ID on 8-LSB bits and need also a potential rescale to present DAT on 24-bits. - "IIO: ADC: add sigma delta modulator support" patch: Simple dummy driver created to support external Sigma delta modulator. It is binded to DFSDM driver through hw_customer API. - 1.9.1

5 35

[alsa-devel] Different audio sample rate between in and out(44100 Hz resp. 44100 Hz)
by Joakim Tjernlund 11 Sep '17

11 Sep '17

When starting a audio chat I often have problems with my microphon and recenly I found that the sample rate is different for in and out: # > cat /proc/asound/card1/stream0 Logitech Inc Logitech Stereo H650e at usb-0000:00:1d.0-1.7, full speed : USB Audio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Capture: Status: Running Interface = 1 Altset = 1 Packet Size = 200 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 This seems odd to me. Adding some debug to kernel I see: [ 129.858365] usb 1-2: 1:1: setting freq 44100 to ep 0x81 [ 129.860514] usb 1-2: 1:1: setting freq 44100 to ep 0x81 [ 131.884314] usb 1-2: 2:1: setting freq 44100 to ep 0x1 [ 131.886567] usb 1-2: 2:1: setting freq 44100 to ep 0x1 [ 131.898714] usb 1-2: 2:1: setting freq 48000 to ep 0x1 [ 131.900847] usb 1-2: 2:1: setting freq 48000 to ep 0x1 Is this expected or should I try make both the same? Jocke

1 1

[alsa-devel] [PATCH] ASoC: cs43130: Add break keyword to switch case
by Li Xu 10 Sep '17

10 Sep '17

Add break keyword to all switch case unless the case is returning. Also remove gpio check for consistency. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- This patch addresses stylistic issues in driver. sound/soc/codecs/cs43130.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 643e37f..5ba0edc 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -909,6 +909,7 @@ static int cs43130_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_XSP << CS43130_DSD_SRC_SHIFT); + break; } if (!sclk && cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM) @@ -1039,6 +1040,7 @@ static int cs43130_pcm_ch_put(struct snd_kcontrol *kcontrol, else regmap_multi_reg_write(cs43130->regmap, pcm_ch_dis_seq, ARRAY_SIZE(pcm_ch_dis_seq)); + break; } return snd_soc_put_enum_double(kcontrol, ucontrol); @@ -1152,6 +1154,7 @@ static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, dsd_seq, ARRAY_SIZE(dsd_seq)); + break; } break; case SND_SOC_DAPM_POST_PMU: @@ -1162,6 +1165,7 @@ static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, unmute_seq, ARRAY_SIZE(unmute_seq)); + break; } break; case SND_SOC_DAPM_PRE_PMD: @@ -1184,6 +1188,7 @@ static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_1, CS43130_MUTE_MASK, CS43130_MUTE_EN); + break; } break; default: @@ -1206,6 +1211,7 @@ static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, pcm_seq, ARRAY_SIZE(pcm_seq)); + break; } break; case SND_SOC_DAPM_POST_PMU: @@ -1216,6 +1222,7 @@ static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, unmute_seq, ARRAY_SIZE(unmute_seq)); + break; } break; case SND_SOC_DAPM_PRE_PMD: @@ -1238,6 +1245,7 @@ static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs43130->regmap, CS43130_PCM_PATH_CTL_1, CS43130_MUTE_MASK, CS43130_MUTE_EN); + break; } break; default: @@ -1277,6 +1285,7 @@ static int cs43130_dac_event(struct snd_soc_dapm_widget *w, case CS43198_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, pop_free_seq2, ARRAY_SIZE(pop_free_seq2)); + break; } break; case SND_SOC_DAPM_POST_PMU: @@ -1301,6 +1310,7 @@ static int cs43130_dac_event(struct snd_soc_dapm_widget *w, case CS43198_CHIP_ID: usleep_range(12000, 12010); regmap_write(cs43130->regmap, CS43130_DXD13, 0); + break; } regmap_write(cs43130->regmap, CS43130_DXD1, 0); @@ -1311,6 +1321,7 @@ static int cs43130_dac_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, dac_postpmd_seq, ARRAY_SIZE(dac_postpmd_seq)); + break; } break; default: @@ -2133,6 +2144,7 @@ static void cs43130_imp_meas(struct work_struct *wk) cs43130_hpload_proc(cs43130, hp_dis_cal_seq2, ARRAY_SIZE(hp_dis_cal_seq2), CS43130_HPLOAD_OFF_INT, ac_idx); + break; } regmap_multi_reg_write(cs43130->regmap, hp_cln_seq, @@ -2543,6 +2555,7 @@ static int cs43130_i2c_probe(struct i2c_client *client, digital_hp_routes; soc_codec_dev_cs43130.component_driver.num_dapm_routes = ARRAY_SIZE(digital_hp_routes); + break; } ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_cs43130, @@ -2586,8 +2599,7 @@ static int cs43130_i2c_remove(struct i2c_client *client) device_remove_file(&client->dev, &dev_attr_hpload_ac_r); } - if (cs43130->reset_gpio) - gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); pm_runtime_disable(&client->dev); regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: adau17x1: Workaround for noise bug in ADC
by Ricard Wanderlof 10 Sep '17

10 Sep '17

The ADC in the ADAU1361 (and possibly other Analog Devices codecs) exhibits a cyclic variation in the noise floor (in our test setup between -87 and -93 dB), a new value being attained within this range whenever a new capture stream is started. The cycle repeats after about 10 or 11 restarts. The workaround recommended by the manufacturer is to toggle the ADOSR bit in the Converter Control 0 register each time a new capture stream is started. I have verified that the patch fixes this problem on the ADAU1361, and according to the manufacturer toggling the bit in question in this manner will at least have no detrimental effect on other chips served by this driver. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 2c1bd27..6758f78 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -90,6 +90,27 @@ static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, return 0; } +static int adau17x1_adc_fixup(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 adau *adau = snd_soc_codec_get_drvdata(codec); + + /* + * If we are capturing, toggle the ADOSR bit in Converter Control 0 to + * avoid losing SNR (workaround from ADI). This must be done after + * the ADC(s) have been enabled. According to the data sheet, it is + * normally illegal to set this bit when the sampling rate is 96 kHz, + * but according to ADI it is acceptable for this workaround. + */ + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_ADOSR, ADAU17X1_CONVERTER0_ADOSR); + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_ADOSR, 0); + + return 0; +} + static const char * const adau17x1_mono_stereo_text[] = { "Stereo", "Mono Left Channel (L+R)", @@ -121,7 +142,8 @@ static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = { SND_SOC_DAPM_MUX("Right DAC Mode Mux", SND_SOC_NOPM, 0, 0, &adau17x1_dac_mode_mux), - SND_SOC_DAPM_ADC("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0), + SND_SOC_DAPM_ADC_E("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0, + adau17x1_adc_fixup, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_ADC("Right Decimator", NULL, ADAU17X1_ADC_CONTROL, 1, 0), SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0), SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0), diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index bf04b7ef..db35003 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -129,5 +129,7 @@ bool adau17x1_has_dsp(struct adau *adau); #define ADAU17X1_CONVERTER0_CONVSR_MASK 0x7 +#define ADAU17X1_CONVERTER0_ADOSR BIT(3) + #endif -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ALSA: msnd: Optimize / harden DSP and MIDI loops
by grygorii tertychnyi 08 Sep '17

08 Sep '17

From: Takashi Iwai <tiwai(a)suse.de> commit 20e2b791796bd68816fa115f12be5320de2b8021 upstream. The ISA msnd drivers have loops fetching the ring-buffer head, tail and size values inside the loops. Such codes are inefficient and fragile. This patch optimizes it, and also adds the sanity check to avoid the endless loops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196131 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196133 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Cc: xe-linux-external(a)cisco.com Signed-off-by: grygorii tertychnyi <gtertych(a)cisco.com> --- sound/isa/msnd/msnd_midi.c | 30 +++++++++++++++--------------- sound/isa/msnd/msnd_pinnacle.c | 23 ++++++++++++----------- 2 files changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index ffc67fd80c23..58e59cd3c95c 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -120,24 +120,24 @@ void snd_msndmidi_input_read(void *mpuv) unsigned long flags; struct snd_msndmidi *mpu = mpuv; void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + u16 head, tail, size; spin_lock_irqsave(&mpu->input_lock, flags); - while (readw(mpu->dev->MIDQ + JQS_wTail) != - readw(mpu->dev->MIDQ + JQS_wHead)) { - u16 wTmp, val; - val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead)); - - if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, - &mpu->mode)) - snd_rawmidi_receive(mpu->substream_input, - (unsigned char *)&val, 1); - - wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1; - if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize)) - writew(0, mpu->dev->MIDQ + JQS_wHead); - else - writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + head = readw(mpu->dev->MIDQ + JQS_wHead); + tail = readw(mpu->dev->MIDQ + JQS_wTail); + size = readw(mpu->dev->MIDQ + JQS_wSize); + if (head > size || tail > size) + goto out; + while (head != tail) { + unsigned char val = readw(pwMIDQData + 2 * head); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, &val, 1); + if (++head > size) + head = 0; + writew(head, mpu->dev->MIDQ + JQS_wHead); } + out: spin_unlock_irqrestore(&mpu->input_lock, flags); } EXPORT_SYMBOL(snd_msndmidi_input_read); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 4c072666115d..a31ea6c22d19 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -170,23 +170,24 @@ static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) { struct snd_msnd *chip = dev_id; void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + u16 head, tail, size; /* Send ack to DSP */ /* inb(chip->io + HP_RXL); */ /* Evaluate queued DSP messages */ - while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) { - u16 wTmp; - - snd_msnd_eval_dsp_msg(chip, - readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead))); - - wTmp = readw(chip->DSPQ + JQS_wHead) + 1; - if (wTmp > readw(chip->DSPQ + JQS_wSize)) - writew(0, chip->DSPQ + JQS_wHead); - else - writew(wTmp, chip->DSPQ + JQS_wHead); + head = readw(chip->DSPQ + JQS_wHead); + tail = readw(chip->DSPQ + JQS_wTail); + size = readw(chip->DSPQ + JQS_wSize); + if (head > size || tail > size) + goto out; + while (head != tail) { + snd_msnd_eval_dsp_msg(chip, readw(pwDSPQData + 2 * head)); + if (++head > size) + head = 0; + writew(head, chip->DSPQ + JQS_wHead); } + out: /* Send ack to DSP */ inb(chip->io + HP_RXL); return IRQ_HANDLED; -- 2.10.3.dirty

1 0

[alsa-devel] [bug] Missing "front" definition for byt-max98090
by Tanu Kaskinen 08 Sep '17

08 Sep '17

PulseAudio 11 stopped using "hw:" as a fallback for analog stereo if "front:" doesn't work. This caused a regression on "Toshiba Chromebook 2 (Swanky)", because PulseAudio doesn't any more do automatic mixer adjustments when plugging in headphones. Apparently on this machine "hw:1,0" works fine as a substitute for "front:1,0", but we're not going to revert the change in PulseAudio, so could someone add the "front" definition for this machine in alsa-lib? Unfortunately I'm not proficient enough in the alsa-lib configuration system to make a patch myself. alsa-info here: https://bugs.freedesktop.org/attachment.cgi?id=133993 [1] https://bugs.freedesktop.org/show_bug.cgi?id=102560 -- Tanu https://www.patreon.com/tanuk

2 5

[alsa-devel] [PATCH 23/25] ALSA/dummy: Replace tasklet with softirq hrtimer
by Anna-Maria Gleixner 08 Sep '17

08 Sep '17

From: Thomas Gleixner <tglx(a)linutronix.de> The tasklet is used to defer the execution of snd_pcm_period_elapsed() to the softirq context. Using the CLOCK_MONOTONIC_SOFT base invokes the timer callback in softirq context as well which renders the tasklet useless. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Anna-Maria Gleixner <anna-maria(a)linutronix.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org --- sound/drivers/dummy.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -376,17 +376,9 @@ struct dummy_hrtimer_pcm { ktime_t period_time; atomic_t running; struct hrtimer timer; - struct tasklet_struct tasklet; struct snd_pcm_substream *substream; }; -static void dummy_hrtimer_pcm_elapsed(unsigned long priv) -{ - struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv; - if (atomic_read(&dpcm->running)) - snd_pcm_period_elapsed(dpcm->substream); -} - static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) { struct dummy_hrtimer_pcm *dpcm; @@ -394,7 +386,8 @@ static enum hrtimer_restart dummy_hrtime dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer); if (!atomic_read(&dpcm->running)) return HRTIMER_NORESTART; - tasklet_schedule(&dpcm->tasklet); + + snd_pcm_period_elapsed(dpcm->substream); hrtimer_forward_now(timer, dpcm->period_time); return HRTIMER_RESTART; } @@ -421,7 +414,6 @@ static int dummy_hrtimer_stop(struct snd static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) { hrtimer_cancel(&dpcm->timer); - tasklet_kill(&dpcm->tasklet); } static snd_pcm_uframes_t @@ -466,12 +458,10 @@ static int dummy_hrtimer_create(struct s if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC_SOFT, HRTIMER_MODE_REL); dpcm->timer.function = dummy_hrtimer_callback; dpcm->substream = substream; atomic_set(&dpcm->running, 0); - tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed, - (unsigned long)dpcm); return 0; }

4 13

[alsa-devel] [PATCH] sound: soc: fsl: Do not set DAI sysclk when it is equal to system freq
by Lukasz Majewski 08 Sep '17

08 Sep '17

The problem is visible in the following setup (on the imx6q): "simple-audio-card" -> ssi2 -> I2S + I2C -> codec The function call log (simple-card probe -> CONFIG_SND_SIMPLE_CARD): asoc_simple_card_init_dai() @ sound/soc/generic/simple-card-utils.c snd_soc_dai_set_sysclk() fsl_ssi_set_dai_sysclk() @ sound/soc/fsl/fsl_ssi.c The last call is changing the bit clock (BCLK) frequency to SSI's IP block clock (ipg = 66 MHz) [1]. This is wrong, since IMX SSI block requires the I2S BCLK to be less than 1/5 of [1]. As a result the driver initialization passes without any errors, but the speaker-test test case breaks. This commit checks if the fsl_ssi_set_dai_sysclk() frequency passed is not equal to [1]. Signed-off-by: Lukasz Majewski <lukma(a)denx.de> --- sound/soc/fsl/fsl_ssi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 173cb84..1186fa9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -809,6 +809,8 @@ static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + if (clk_get_rate(ssi_private->clk) == freq) + return 0; ssi_private->bitclk_freq = freq; -- 2.1.4

6 27

[alsa-devel] [patch 0/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 08 Sep '17

08 Sep '17

Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. This code change doesn't do much except just refactor snd_soc_dapm_new_pcm function. Now the part of allocating memory for kcontrol is allocated to a new function and it is called only in the case it is needed. anish kumar (4): ASoC: dapm: fix error path in snd_soc_dapm_new_pcm ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm ASoC: dapm: Avoid creating kcontrol for params ASoC: dapm: Remove the redundant check sound/soc/soc-dapm.c | 123 ++++++++++++++++++++++++++++----------------------- 1 file changed, 67 insertions(+), 56 deletions(-) -- 2.10.1 (Apple Git-78)

1 1

[alsa-devel] bootstrap.sh
by Russell Treleaven 07 Sep '17

07 Sep '17

sorry wrong list. -- Sincerely, Russell Treleaven sip:rtreleaven@sip.bunnykick.ca

1 0

[alsa-devel] autopoint should be required for bootstrap.sh
by Russell Treleaven 07 Sep '17

07 Sep '17

I attempted to bootstrap pulseaudio and got a non obvious error. log is here https://pastebin.com/CctNdLwD installing autopoint fixed the problem the following patch makes bootstrap.sh fail if autopoint is not available. diff --git a/bootstrap.sh b/bootstrap.sh index 57494fc..0a05a3e 100755 --- a/bootstrap.sh +++ b/bootstrap.sh @@ -32,13 +32,17 @@ fi # configure file faulty. if ! pkg-config --version &>/dev/null; then echo "pkg-config is required to bootstrap this program" - DIE=1 + exit 1 fi - # Other necessary programs -intltoolize --version >/dev/null || DIE=1 -test "$DIE" = 1 && exit 1 - +if ! autopoint --version &>/dev/null ; then + echo "autopoint is required to bootstrap this program" + exit 1 +fi +if ! intltoolize --version >/dev/null ; then + echo "intltoolize is required to bootstrap this program" + exit 1 +fi autopoint --force AUTOPOINT='intltoolize --automake --copy' autoreconf --force --install --verbose -- Sincerely, Russell Treleaven sip:rtreleaven@sip.bunnykick.ca

1 0

[alsa-devel] SOC_DAPM_MUX implementation, routes missing and do I need more dapm definitions?
by Oleksandr Müller 07 Sep '17

07 Sep '17

Hello guys, I began with the implementation of a dapm mux and am almost done but I don't get how to configure the routes and I am not sure if I need more dapm definitions. As the capture stream is the source into the SE_IN_L/SE_IN_R I would say this should be the source for the left/right mux (right side of the route definition). Then I would need a switch in the middle and on the left side as the output should be the ADC1 (left side). { "ADC1", ADC1VINSWITCHLEFT, "SE_IN_L" }, I would think it should look something like this. But then I would need some SOC_DAPM_SINGLE, I think you can see I am clearly confused right now :) I would like to have a mux where I can switch the vin source from vin1 to vin4 in alsamixer and then its saved as a default. Because in the layout plan the vin4 is connected and needs to be selected... My code so far looks like this: static const char * const pcm1863_adc_vinl_src[] = { "VIN1l", "VIN4l" }; static const char * const pcm1863_adc_vinr_src[] = { "VIN1r", "VIN4r" }; /*Stereo ADC Source VIN1 left/right */ static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin1l_enum, PCM1863_ADC1_INPUT_SEL_L, PCM1863_SEL_L_VIN1_SFT, pcm1863_adc_vinl_src); static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin1r_enum, PCM1863_ADC1_INPUT_SEL_R, PCM1863_SEL_R_VIN1_SFT, pcm1863_adc_vinr_src); /*Stereo ADC Source VIN4 left/right */ static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin4l_enum, PCM1863_ADC1_INPUT_SEL_L, PCM1863_SEL_L_VIN4_SFT, pcm1863_adc_vinl_src); static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin4r_enum, PCM1863_ADC1_INPUT_SEL_R, PCM1863_SEL_R_VIN4_SFT, pcm1863_adc_vinr_src); static const struct snd_kcontrol_new pcm1863_adc_vin1l_mux = SOC_DAPM_ENUM("ADC VIN1l Source", pcm1863_adc_vin1l_enum); static const struct snd_kcontrol_new pcm1863_adc_vin1r_mux = SOC_DAPM_ENUM("ADC VIN1r Source", pcm1863_adc_vin1r_enum); static const struct snd_kcontrol_new pcm1863_adc_vin4l_mux = SOC_DAPM_ENUM("ADC VIN4l Source", pcm1863_adc_vin4l_enum); static const struct snd_kcontrol_new pcm1863_adc_vin4r_mux = SOC_DAPM_ENUM("ADC VIN4r Source", pcm1863_adc_vin4r_enum); static const struct snd_soc_dapm_widget pcm1863_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC1", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_INPUT("SE_IN_L"), SND_SOC_DAPM_INPUT("SE_IN_R"), SND_SOC_DAPM_MUX("ADC VIN1l Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin1l_mux), SND_SOC_DAPM_MUX("ADC VIN1r Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin1r_mux), SND_SOC_DAPM_MUX("ADC VIN1l Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin4l_mux), SND_SOC_DAPM_MUX("ADC VIN1r Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin4r_mux), }; static const struct snd_soc_dapm_route pcm1863_dapm_routes[] = { { "ADC1", NULL, "SE_IN_L" }, { "ADC1", NULL, "SE_IN_R" }, { "SE_IN_L", NULL, "Capture" }, { "SE_IN_R", NULL, "Capture" }, };

1 0

[alsa-devel] [PATCH] ASoC: All bytes readable in 6 byte PLL register in ADAU17x1
by Ricard Wanderlof 07 Sep '17

07 Sep '17

The PLL register in the ADAU17x1 is 6 bytes long, so mark all bytes as readable. This is of no importance for existing code, but is useful when instrumenting the code for debugging purposes. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 2c1bd27..b324c6f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -759,7 +759,13 @@ bool adau17x1_readable_register(struct device *dev, unsigned int reg) switch (reg) { case ADAU17X1_CLOCK_CONTROL: + /* The PLL register is 6 bytes long */ case ADAU17X1_PLL_CONTROL: + case ADAU17X1_PLL_CONTROL + 1: + case ADAU17X1_PLL_CONTROL + 2: + case ADAU17X1_PLL_CONTROL + 3: + case ADAU17X1_PLL_CONTROL + 4: + case ADAU17X1_PLL_CONTROL + 5: case ADAU17X1_REC_POWER_MGMT: case ADAU17X1_MICBIAS: case ADAU17X1_SERIAL_PORT0: -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] How to apply a SOC_DAPM_MUX if two there are two different input register for left and right channel?
by Oleksandr Müller 07 Sep '17

07 Sep '17

Hello guys, I have some trouble I want to implement a mux within the mixer application for the VIN source of my adc. Taking a look into the datasheet of the TI-PCM1863 adc I get the information that the VIN left register is at the 0x06 and the VIN right register is at the 0x07 address. Right now my code would look like this and here comes the problem it seems that there is only a SOC_ENUM_SINGLE_DECL or a SOC_ENUM_DOUBLE_DECL but no SOC_ENUM_DOUBLE_R_DECL because the DOUBLE_DECL only allows shiftl and shiftr as the input parameters but I would need a regl regr parameter. Is the only option to duplicate the code and define a VIN_L_MUX and a VIN_R_MUX? static const char * const pcm1863_adc_vin_src[] = { "VIN1", "VIN2", "VIN3", "VIN4" }; static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin_enum, PCM1863_ADC1_INPUT_SEL_L, PCM1863_SEL_L_SHIFT, pcm1863_vin_src); static const struct snd_kcontrol_new pcm1863_adc_vin_mux = SOC_DAPM_ENUM("ADC VIN Source", pcm1863_adc_vin_enum); static const struct snd_soc_dapm_widget pcm1863_dapm_widgets[] = { [...] SND_SOC_DAPM_MUX("ADC VIN Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin_mux), }; - Sanj3k

2 1

[alsa-devel] [PATCH] ALSA: asihpi: Kill BUG_ON() usages
by Takashi Iwai 07 Sep '17

07 Sep '17

BUG_ON() is the worst choice for a trivial sanity check. Either it should be removed or replaced with a softer one like WARN_ON() if still really needed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/asihpi/asihpi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 976a3d23557e..70d023a85bf5 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -558,12 +558,10 @@ static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream) struct snd_card_asihpi_pcm *dpcm; struct snd_card_asihpi *card; - BUG_ON(!substream); - dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data; card = snd_pcm_substream_chip(substream); - BUG_ON(in_interrupt()); + WARN_ON(in_interrupt()); tasklet_disable(&card->t); card->llmode_streampriv = dpcm; tasklet_enable(&card->t); @@ -578,8 +576,6 @@ static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream) struct snd_card_asihpi_pcm *dpcm; struct snd_card_asihpi *card; - BUG_ON(!substream); - dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data; card = snd_pcm_substream_chip(substream); -- 2.14.1

1 0

[alsa-devel] [PATCH 14/14] sound/core: Use %pS printk format for direct addresses
by Helge Deller 07 Sep '17

07 Sep '17

The debug functions uses wrongly the %pF instead of the %pS printk format specifier for printing symbols for the address returned by _builtin_return_address(0). Fix it for the ia64, ppc64 and parisc64 architectures. Signed-off-by: Helge Deller <deller(a)gmx.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/core/device.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/device.c b/sound/core/device.c index 8918838..cb0e46f 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -128,7 +128,7 @@ void snd_device_disconnect(struct snd_card *card, void *device_data) if (dev) __snd_device_disconnect(dev); else - dev_dbg(card->dev, "device disconnect %p (from %pF), not found\n", + dev_dbg(card->dev, "device disconnect %p (from %pS), not found\n", device_data, __builtin_return_address(0)); } EXPORT_SYMBOL_GPL(snd_device_disconnect); @@ -152,7 +152,7 @@ void snd_device_free(struct snd_card *card, void *device_data) if (dev) __snd_device_free(dev); else - dev_dbg(card->dev, "device free %p (from %pF), not found\n", + dev_dbg(card->dev, "device free %p (from %pS), not found\n", device_data, __builtin_return_address(0)); } EXPORT_SYMBOL(snd_device_free); -- 2.1.0

2 1

[alsa-devel] [PATCH 0/3] ALSA: ymfpci: Fine-tuning for some function implementations
by SF Markus Elfring 07 Sep '17

07 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 21:37:32 +0200 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Use common error handling code in snd_card_ymfpci_probe() Use common error handling code in snd_ymfpci_create() Adjust 17 checks for null pointers sound/pci/ymfpci/ymfpci.c | 62 ++++++++++++++++++------------------ sound/pci/ymfpci/ymfpci_main.c | 72 ++++++++++++++++++++++-------------------- 2 files changed, 67 insertions(+), 67 deletions(-) -- 2.14.1

3 9

[alsa-devel] [PATCH] ALSA: 6Fire: Use common error handling code in usb6fire_chip_probe()
by SF Markus Elfring 07 Sep '17

07 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 16:26:29 +0200 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/usb/6fire/chip.c | 31 +++++++++++++------------------ 1 file changed, 13 insertions(+), 18 deletions(-) diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index bc2a24f7a791..c7641cb50616 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -143,37 +143,32 @@ static int usb6fire_chip_probe(struct usb_interface *intf, chip->card = card; ret = usb6fire_comm_init(chip); - if (ret < 0) { - usb6fire_chip_destroy(chip); - return ret; - } + if (ret < 0) + goto destroy_chip; ret = usb6fire_midi_init(chip); - if (ret < 0) { - usb6fire_chip_destroy(chip); - return ret; - } + if (ret < 0) + goto destroy_chip; ret = usb6fire_pcm_init(chip); - if (ret < 0) { - usb6fire_chip_destroy(chip); - return ret; - } + if (ret < 0) + goto destroy_chip; ret = usb6fire_control_init(chip); - if (ret < 0) { - usb6fire_chip_destroy(chip); - return ret; - } + if (ret < 0) + goto destroy_chip; ret = snd_card_register(card); if (ret < 0) { dev_err(&intf->dev, "cannot register card."); - usb6fire_chip_destroy(chip); - return ret; + goto destroy_chip; } usb_set_intfdata(intf, chip); return 0; + +destroy_chip: + usb6fire_chip_destroy(chip); + return ret; } static void usb6fire_chip_disconnect(struct usb_interface *intf) -- 2.14.1

2 1

[alsa-devel] [PATCH 0/2] ALSA: usx2y: Fine-tuning for two function implementations
by SF Markus Elfring 07 Sep '17

07 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 15:10:12 +0200 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Use common error handling code in us122l_create_card() Use common error handling code in submit_urbs() sound/usb/usx2y/us122l.c | 10 ++++++---- sound/usb/usx2y/usb_stream.c | 18 ++++++++++-------- 2 files changed, 16 insertions(+), 12 deletions(-) -- 2.14.1

2 4

[alsa-devel] [PATCH 0/2] ALSA: rme9652: Adjustments for two function implementations
by SF Markus Elfring 07 Sep '17

07 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 5 Sep 2017 22:30:22 +0200 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Use common code in hdsp_get_iobox_version() Use common error handling code in snd_hdspm_probe() sound/pci/rme9652/hdsp.c | 26 +++++++++++--------------- sound/pci/rme9652/hdspm.c | 16 ++++++++-------- 2 files changed, 19 insertions(+), 23 deletions(-) -- 2.14.1

2 4

[alsa-devel] [PATCH] ALSA: maestro3: Use common error handling code in two functions
by SF Markus Elfring 07 Sep '17

07 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 15:00:27 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/maestro3.c | 55 +++++++++++++++++++++++++--------------------------- 1 file changed, 26 insertions(+), 29 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index cafea6dc5c01..759c3f6ad7aa 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2622,22 +2622,18 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, err = request_firmware(&chip->assp_kernel_image, "ess/maestro3_assp_kernel.fw", &pci->dev); - if (err < 0) { - snd_m3_free(chip); - return err; - } + if (err) + goto free_chip; err = request_firmware(&chip->assp_minisrc_image, "ess/maestro3_assp_minisrc.fw", &pci->dev); - if (err < 0) { - snd_m3_free(chip); - return err; - } + if (err) + goto free_chip; + + err = pci_request_regions(pci, card->driver); + if (err) + goto free_chip; - if ((err = pci_request_regions(pci, card->driver)) < 0) { - snd_m3_free(chip); - return err; - } chip->iobase = pci_resource_start(pci, 0); /* just to be sure */ @@ -2655,8 +2651,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_m3_free(chip); - return -ENOMEM; + err = -ENOMEM; + goto free_chip; } chip->irq = pci->irq; @@ -2666,10 +2662,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, dev_warn(card->dev, "can't allocate apm buffer\n"); #endif - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_m3_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err) + goto free_chip; if ((err = snd_m3_mixer(chip)) < 0) return err; @@ -2699,6 +2694,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, *chip_ret = chip; return 0; +free_chip: + snd_m3_free(chip); + return err; } /* @@ -2741,23 +2739,19 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) break; } - if ((err = snd_m3_create(card, pci, - external_amp[dev], - amp_gpio[dev], - &chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_m3_create(card, pci, external_amp[dev], amp_gpio[dev], &chip); + if (err) + goto free_card; + card->private_data = chip; sprintf(card->shortname, "ESS %s PCI", card->driver); sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, chip->iobase, chip->irq); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err) + goto free_card; #if 0 /* TODO: not supported yet */ /* TODO enable MIDI IRQ and I/O */ @@ -2772,6 +2766,9 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) pci_set_drvdata(pci, card); dev++; return 0; +free_card: + snd_card_free(card); + return err; } static void snd_m3_remove(struct pci_dev *pci) -- 2.14.0

3 3

[alsa-devel] [PATCH] ASoC: cs4270: Use common error handling code in two functions
by SF Markus Elfring 06 Sep '17

06 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 19:09:31 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/cs4270.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 84f86745c30e..aa70d471cc57 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -362,10 +362,8 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg |= cs4270_mode_ratios[i].speed_mode; ret = snd_soc_write(codec, CS4270_MODE, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } + if (ret < 0) + goto report_failure; /* Set the DAI format */ @@ -385,11 +383,13 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } ret = snd_soc_write(codec, CS4270_FORMAT, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } + if (ret < 0) + goto report_failure; + + return ret; +report_failure: + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -512,10 +512,8 @@ static int cs4270_probe(struct snd_soc_codec *codec) * re-enabled it by using the controls. */ ret = snd_soc_update_bits(codec, CS4270_MUTE, CS4270_MUTE_AUTO, 0); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } + if (ret < 0) + goto report_failure; /* Disable automatic volume control. The hardware enables, and it * causes volume change commands to be delayed, sometimes until after @@ -524,15 +522,17 @@ static int cs4270_probe(struct snd_soc_codec *codec) */ ret = snd_soc_update_bits(codec, CS4270_TRANS, CS4270_TRANS_SOFT | CS4270_TRANS_ZERO, 0); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } + if (ret < 0) + goto report_failure; ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return ret; + +report_failure: + dev_err(codec->dev, "i2c write failed\n"); + return ret; } /** -- 2.14.1

1 0

[alsa-devel] ALSA MIDI exclusive mode ?
by happy musicmaker 06 Sep '17

06 Sep '17

I ready somewhere that some applications, like Bitwig, allocate(?) MIDI ports in "exclusive" mode so that other application cannot connect to the MIDI ports. I used Bitwig and confirmed that after starting Bitwig in qjackctrl you cannot connect any of the MIDI write/read ports. ( qjackctrl MIDI option is set to "none") I could not find any documentation on such "exclusive" feature though ?

2 2

[alsa-devel] USB dB data
by Jack Bates 06 Sep '17

06 Sep '17

I confirmed with Wireshark that my USB sound card responds to GET_MIN with 0x0000 and GET_MAX with 0x6300 .. The audio device class definition [1] says the units are 1/256 dB (section 5.2.2.2.3: Mixer Control) ... So by my calculation, the min and max are 0 and 99 dB (0x6300 / 256 == 99) ... Why then does alsactl print: > range '0 - 99' > dbmin 0 > dbmax 38 (/usr/sbin/aslactl -f - store) I originally asked about this on the pulseaudio-discuss list [2] because it uses "dB data" [3]. The "0 - 99" makes sense -- it lines up with my calculation ... Where does the 38 come from? (I expect it's obvious, I just haven't found the explanation?) Background: I just got a Bose SoundLink Revolve ... When I connect it with USB, the volume is silent from 0 to 99% and deafening at 100% (using PulseAudio). Directly adjusting the volume with alsamixer works properly: It increases smoothly from 0 to 100% -- changes that are reflected in the PulseAudio volume as fine adjustments in between 99 and 100% (64587 to 65536 PulseAudio volume). From what I've read there's a common problem where especially USB devices report the wrong dB data (don't follow the spec [4]) -- this has previously been handled by patching the USB audio driver [5] ... However I haven't figured out how my device deviates from the spec (0x0000 to 0x6300 or 0 to 99 dB seem plausible) -- my understanding is incomplete: I haven't figured out where the value 38 comes from ... [1] http://www.usb.org/developers/docs/devclass_docs/audio10.pdf [2] https://lists.freedesktop.org/archives/pulseaudio-discuss/2017-August/02860… [3] http://0pointer.de/blog/projects/decibel-data [4] https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/637066 [5] https://bugs.freedesktop.org/show_bug.cgi?id=93193#c14

2 3

Re: [alsa-devel] [PATCH v8 1/2] ASoC: rt5514: Add devicetree bindings for rt5514-spi
by Brian Norris 06 Sep '17

06 Sep '17

On Tue, Sep 05, 2017 at 12:24:59PM +0800, Jeffy Chen wrote: > Add devicetree bindings documentation file for rt5514 spi dsp codec. > > Also update rt5514 i2c dt-binding's compatible. You should probably describe *why*? > Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> > --- > > Changes in v8: None > > .../devicetree/bindings/sound/rt5514-spi.txt | 32 ++++++++++++++++++++++ > Documentation/devicetree/bindings/sound/rt5514.txt | 2 +- > 2 files changed, 33 insertions(+), 1 deletion(-) > create mode 100644 Documentation/devicetree/bindings/sound/rt5514-spi.txt > > diff --git a/Documentation/devicetree/bindings/sound/rt5514-spi.txt b/Documentation/devicetree/bindings/sound/rt5514-spi.txt > new file mode 100644 > index 000000000000..8a63d9423dd9 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/rt5514-spi.txt > @@ -0,0 +1,32 @@ > +RT5514 SPI audio CODEC > + > +This device supports SPI only. > + > +Required properties: > + > +- compatible : "realtek,rt5514". You list one string here, but there's 2 in the example. The point of this exercise was to document (not just in an Example) the strings you're using. > + > +- reg : The SPI address of the device. > + > +Optional properties: > + > +- interrupt-parent: Should be the phandle for the interrupt controller > +- interrupts: The interrupt number to the cpu. The interrupt specifier format > + depends on the interrupt controller. > + > +Example: > + > +codec: rt5514-spi@0 { > + compatible = "realtek,rt5514", "realtek,rt5514-spi"; > + reg = <0>; > + > + /* May run faster once verified. */ Really? That was barely useful in the rk3399-gru DTSI, but it's definitely not worth sticking in an example. Also, you switch between tabs and spaces for indentation. Please be consistent. (And as with C code, I think the right answer is tabs?) > + spi-max-frequency = <10000000>; > + > + pinctrl-names = "default"; > + pinctrl-0 = <&mic_int>; > + > + interrupt-parent = <&gpio1>; > + interrupts = <13 IRQ_TYPE_LEVEL_HIGH>; > + wakeup-source; > +}; > diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt > index 929ca6756b02..12a8c93a67cc 100644 > --- a/Documentation/devicetree/bindings/sound/rt5514.txt > +++ b/Documentation/devicetree/bindings/sound/rt5514.txt > @@ -28,6 +28,6 @@ Pins on the device (for linking into audio routes) for RT5514: > Example: > > codec: rt5514@57 { > - compatible = "realtek,rt5514"; > + compatible = "realtek,rt5514", "realtek,rt5514-i2c"; Example != Documentation. Please document the string properly. Brian > reg = <0x57>; > }; > -- > 2.11.0 > >

2 1

[alsa-devel] [PATCH v9 1/2] ASoC: rt5514: Add devicetree bindings for rt5514-spi
by Jeffy Chen 06 Sep '17

06 Sep '17

Add devicetree bindings documentation file for rt5514 spi dsp codec. Also update rt5514 i2c dt-binding's compatible to distinguish it from rt5514 spi. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v9: Address comments from Brian. .../devicetree/bindings/sound/rt5514-spi.txt | 29 ++++++++++++++++++++++ Documentation/devicetree/bindings/sound/rt5514.txt | 4 +-- 2 files changed, 31 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5514-spi.txt diff --git a/Documentation/devicetree/bindings/sound/rt5514-spi.txt b/Documentation/devicetree/bindings/sound/rt5514-spi.txt new file mode 100644 index 000000000000..7911d1adbd29 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5514-spi.txt @@ -0,0 +1,29 @@ +RT5514 SPI audio CODEC + +This device supports SPI only. + +Required properties: + +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-spi". + +- reg : The SPI address of the device. + +Optional properties: + +- interrupt-parent: Should be the phandle for the interrupt controller +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + +Example: + +codec: rt5514-spi@0 { + compatible = "realtek,rt5514", "realtek,rt5514-spi"; + reg = <0>; + + pinctrl-names = "default"; + pinctrl-0 = <&mic_int>; + + interrupt-parent = <&gpio1>; + interrupts = <13 IRQ_TYPE_LEVEL_HIGH>; + wakeup-source; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 929ca6756b02..841a1159e234 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "realtek,rt5514". +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-i2c". - reg : The I2C address of the device. @@ -28,6 +28,6 @@ Pins on the device (for linking into audio routes) for RT5514: Example: codec: rt5514@57 { - compatible = "realtek,rt5514"; + compatible = "realtek,rt5514", "realtek,rt5514-i2c"; reg = <0x57>; }; -- 2.11.0

1 0

[alsa-devel] alsa-utils uses incorrect format for 24 bit audio?
by Ricard Wanderlof 06 Sep '17

06 Sep '17

When attempting to use arecord to capture audio from a 24 bit source, and create a 24-bit .wav file I ran into the problem that the resulting file does not seem valid. Looking closer at the contents, it appears that arecord writes the 32-bit S24_LE samples verbatim to the file, and sets the number of bits per sample specifier to 24 in the .wav header, at the same time as it sets the 'block align' to 4 which I suppose is logical, although normally the block align corresponds to the frame size (in bytes) for the sample size specified in the bits per sample specifier. A number of utilities and applications I tried refused to play the file back properly, including Audacity, VLC, sox and kwave, so I'm guessing the format written by arecord is not valid in this case. 24 bit audio as exported from Audacity for instance in reality corresponds to the ALSA S24_3LE format, i.e. the data is sent with no padding (resulting in a .wav 'block align' specifier of 3 for a mono stream). Is this a known problem with arecord/aplay, or is it the other applications which have been less than accurate when implementing .wav file standard? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ASoC: improve RT5514 dependencies
by Arnd Bergmann 05 Sep '17

05 Sep '17

With SND_SOC_RT5514_SPI=m and SND_SOC_RT5514=y, we get this link error: sound/soc/codecs/rt5514.o: In function `rt5514_dsp_voice_wake_up_put': rt5514.c:(.text+0x2068): undefined reference to `rt5514_spi_burst_write' rt5514.c:(.text+0x20c8): undefined reference to `rt5514_spi_burst_write' This adds another Kconfig symbol to work around the link error, forcing the main driver to be a loadable module as well if some other driver selects the I2C side to be built-in. Fixes: 2a18483a7fb4 ("ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- This can now lead to a configuration with SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH=y and SND_SOC_RT5514=m. I don't know if that is a problem or not. An alternative approach would be to flip the dependency between rt5514_spi.ko and rt5514.ko and avoid having the rt5514_spi_burst_write symbol exported. --- sound/soc/codecs/Kconfig | 7 ++++++- sound/soc/intel/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 2 +- sound/soc/rockchip/Kconfig | 1 - 4 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 024ddc9938ed..115c6144ca2a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -117,7 +117,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C - select SND_SOC_RT5514 if I2C + select SND_SOC_RT5514_I2C if I2C select SND_SOC_RT5616 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -738,6 +738,11 @@ config SND_SOC_RT298 config SND_SOC_RT5514 tristate + default m if SND_SOC_RT5514_SPI=m + default SND_SOC_RT5514_I2C || SND_SOC_RT5514_SPI + +config SND_SOC_RT5514_I2C + tristate config SND_SOC_RT5514_SPI tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b301bfff1c09..f5ef912190ed 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -259,7 +259,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 - select SND_SOC_RT5514 + select SND_SOC_RT5514_I2C select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 5c68797f36c4..6ccf73ae77ca 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -67,7 +67,7 @@ config SND_SOC_MT8173_RT5650_RT5514 tristate "ASoC Audio driver for MT8173 with RT5650 RT5514 codecs" depends on SND_SOC_MT8173 && I2C select SND_SOC_RT5645 - select SND_SOC_RT5514 + select SND_SOC_RT5514_I2C help This adds ASoC driver for Mediatek MT8173 boards with the RT5650 and RT5514 codecs. diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index c84487805876..c1139aecdfbe 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -65,7 +65,6 @@ config SND_SOC_RK3399_GRU_SOUND depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP && SPI select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98357A - select SND_SOC_RT5514 select SND_SOC_DA7219 select SND_SOC_RT5514_SPI help -- 2.9.0

3 4

[alsa-devel] [PATCH v6 00/12] AC97 device/driver model revamp
by Robert Jarzmik 05 Sep '17

05 Sep '17

Hi Lars, Mark, Charles, Lee, This is a revision for Lee, only targetted at wm97xx-core changes suggested by Lee. I hope I have every concern covered Lee, just le me know if it suits you. For easier spotting, I included in [1] the diff from the last serie. I kept Charles ack for patch 6/12 following Lee's review, if you don't agree Charles, I'll remove it. Cheers. -- Robert Robert Jarzmik (12): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++++++----- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 379 ++++++++++++++++++++++ include/linux/mfd/wm97xx.h | 25 ++ include/sound/ac97/codec.h | 118 +++++++ include/sound/ac97/compat.h | 20 ++ include/sound/ac97/controller.h | 85 +++++ include/sound/ac97/regs.h | 262 ++++++++++++++++ include/sound/ac97_codec.h | 239 +------------- include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 16 + sound/ac97/bus.c | 539 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 108 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 37 ++- sound/arm/pxa2xx-ac97.c | 35 ++- sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 61 ++-- sound/soc/codecs/wm9712.c | 39 ++- sound/soc/codecs/wm9713.c | 39 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 46 ++- 29 files changed, 1978 insertions(+), 414 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.11.0 [1] Diff from the previous serie ---8>--- diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 5ffe4f4dfa10..6d175698a49a 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -1635,7 +1635,6 @@ config MFD_WM97xx select AC97_BUS_COMPAT depends on AC97_BUS_NEW help - The WM9705, WM9712 and WM9713 is a highly integrated hi-fi CODEC designed for smartphone applications. As well as audio functionality it has on board GPIO and a touchscreen functionality which is diff --git a/drivers/mfd/wm97xx-core.c b/drivers/mfd/wm97xx-core.c index 473bbf9510e5..66e477fffd43 100644 --- a/drivers/mfd/wm97xx-core.c +++ b/drivers/mfd/wm97xx-core.c @@ -1,7 +1,7 @@ /* * Wolfson WM97xx -- Core device * - * Copyright (C) 2016 Robert Jarzmik + * Copyright (C) 2017 Robert Jarzmik * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -14,14 +14,13 @@ * - a GPIO block */ -#include <linux/module.h> - #include <linux/device.h> #include <linux/mfd/core.h> #include <linux/mfd/wm97xx.h> -#include <linux/wm97xx.h> +#include <linux/module.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/wm97xx.h> #include <sound/ac97/codec.h> #include <sound/ac97/compat.h> @@ -34,6 +33,7 @@ struct wm97xx_priv { struct regmap *regmap; struct snd_ac97 *ac97; struct device *dev; + struct wm97xx_platform_data codec_pdata; }; static bool wm97xx_readable_reg(struct device *dev, unsigned int reg) @@ -108,33 +108,14 @@ static const struct regmap_config wm9705_regmap_config = { .writeable_reg = wm97xx_writeable_reg, }; -static int wm9705_register(struct wm97xx_priv *wm97xx, - struct wm97xx_pdata *pdata) -{ - static struct wm97xx_platform_data codec_pdata; - static const struct mfd_cell cells[] = { - { - .name = "wm9705-codec", - .platform_data = &codec_pdata, - .pdata_size = sizeof(codec_pdata), - }, - { - .name = "wm97xx-ts", - .platform_data = &codec_pdata, - .pdata_size = sizeof(codec_pdata), - }, - }; - - codec_pdata.ac97 = wm97xx->ac97; - codec_pdata.regmap = devm_regmap_init_ac97(wm97xx->ac97, - &wm9705_regmap_config); - codec_pdata.batt_pdata = pdata->batt_pdata; - if (IS_ERR(codec_pdata.regmap)) - return PTR_ERR(codec_pdata.regmap); - - return devm_mfd_add_devices(wm97xx->dev, PLATFORM_DEVID_NONE, cells, - ARRAY_SIZE(cells), NULL, 0, NULL); -} +static struct mfd_cell wm9705_cells[] = { + { + .name = "wm9705-codec", + }, + { + .name = "wm97xx-ts", + }, +}; static bool wm9712_volatile_reg(struct device *dev, unsigned int reg) { @@ -199,33 +180,14 @@ static const struct regmap_config wm9712_regmap_config = { .writeable_reg = wm97xx_writeable_reg, }; -static int wm9712_register(struct wm97xx_priv *wm97xx, - struct wm97xx_pdata *pdata) -{ - static struct wm97xx_platform_data codec_pdata; - static const struct mfd_cell cells[] = { - { - .name = "wm9712-codec", - .platform_data = &codec_pdata, - .pdata_size = sizeof(codec_pdata), - }, - { - .name = "wm97xx-ts", - .platform_data = &codec_pdata, - .pdata_size = sizeof(codec_pdata), - }, - }; - - codec_pdata.ac97 = wm97xx->ac97; - codec_pdata.regmap = devm_regmap_init_ac97(wm97xx->ac97, - &wm9712_regmap_config); - codec_pdata.batt_pdata = pdata->batt_pdata; - if (IS_ERR(codec_pdata.regmap)) - return PTR_ERR(codec_pdata.regmap); - - return devm_mfd_add_devices(wm97xx->dev, PLATFORM_DEVID_NONE, cells, - ARRAY_SIZE(cells), NULL, 0, NULL); -} +static struct mfd_cell wm9712_cells[] = { + { + .name = "wm9712-codec", + }, + { + .name = "wm97xx-ts", + }, +}; static const struct reg_default wm9713_reg_defaults[] = { { 0x02, 0x8080 }, /* Speaker Output Volume */ @@ -293,39 +255,23 @@ static const struct regmap_config wm9713_regmap_config = { .writeable_reg = wm97xx_writeable_reg, }; -static int wm9713_register(struct wm97xx_priv *wm97xx, - struct wm97xx_pdata *pdata) -{ - static struct wm97xx_platform_data codec_pdata; - static const struct mfd_cell cells[] = { - { - .name = "wm9713-codec", - .platform_data = &codec_pdata, - .pdata_size = sizeof(codec_pdata), - }, - { - .name = "wm97xx-ts", - .platform_data = &codec_pdata, - .pdata_size = sizeof(codec_pdata), - }, - }; - - codec_pdata.ac97 = wm97xx->ac97; - codec_pdata.regmap = devm_regmap_init_ac97(wm97xx->ac97, - &wm9713_regmap_config); - codec_pdata.batt_pdata = pdata->batt_pdata; - if (IS_ERR(codec_pdata.regmap)) - return PTR_ERR(codec_pdata.regmap); - - return devm_mfd_add_devices(wm97xx->dev, PLATFORM_DEVID_NONE, cells, - ARRAY_SIZE(cells), NULL, 0, NULL); -} +static struct mfd_cell wm9713_cells[] = { + { + .name = "wm9713-codec", + }, + { + .name = "wm97xx-ts", + }, +}; static int wm97xx_ac97_probe(struct ac97_codec_device *adev) { struct wm97xx_priv *wm97xx; - int ret; - void *pdata = snd_ac97_codec_get_platdata(adev); + const struct regmap_config *config; + struct wm97xx_platform_data *codec_pdata; + struct mfd_cell *cells; + int ret = 0, nb_cells, i; + struct wm97xx_pdata *pdata = snd_ac97_codec_get_platdata(adev); wm97xx = devm_kzalloc(ac97_codec_dev2dev(adev), sizeof(*wm97xx), GFP_KERNEL); @@ -342,20 +288,48 @@ static int wm97xx_ac97_probe(struct ac97_codec_device *adev) dev_info(wm97xx->dev, "wm97xx core found, id=0x%x\n", adev->vendor_id); + codec_pdata = &wm97xx->codec_pdata; + codec_pdata->ac97 = wm97xx->ac97; + codec_pdata->batt_pdata = pdata->batt_pdata; + switch (adev->vendor_id) { case WM9705_VENDOR_ID: - ret = wm9705_register(wm97xx, pdata); + config = &wm9705_regmap_config; + cells = wm9705_cells; + nb_cells = ARRAY_SIZE(wm9705_cells); break; case WM9712_VENDOR_ID: - ret = wm9712_register(wm97xx, pdata); + config = &wm9712_regmap_config; + cells = wm9712_cells; + nb_cells = ARRAY_SIZE(wm9712_cells); break; case WM9713_VENDOR_ID: - ret = wm9713_register(wm97xx, pdata); + config = &wm9713_regmap_config; + cells = wm9713_cells; + nb_cells = ARRAY_SIZE(wm9713_cells); break; default: + config = NULL; + } + + for (i = 0; i < nb_cells; i++) { + cells[i].platform_data = codec_pdata; + cells[i].pdata_size = sizeof(*codec_pdata); + } + + if (config) { + codec_pdata->regmap = + devm_regmap_init_ac97(wm97xx->ac97, config); + if (IS_ERR(codec_pdata->regmap)) + ret = PTR_ERR(codec_pdata->regmap); + } else { ret = -ENODEV; } + if (!ret) + ret = devm_mfd_add_devices(wm97xx->dev, PLATFORM_DEVID_NONE, + cells, nb_cells, NULL, 0, NULL); + if (ret) snd_ac97_compat_release(wm97xx->ac97); diff --git a/include/linux/mfd/wm97xx.h b/include/linux/mfd/wm97xx.h index 627322f14d48..45fb54f19d09 100644 --- a/include/linux/mfd/wm97xx.h +++ b/include/linux/mfd/wm97xx.h @@ -1,17 +1,12 @@ /* * wm97xx client interface * - * Copyright (C) 2016 Robert Jarzmik + * Copyright (C) 2017 Robert Jarzmik * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. */ #ifndef __LINUX_MFD_WM97XX_H @@ -27,5 +22,4 @@ struct wm97xx_platform_data { struct wm97xx_batt_pdata *batt_pdata; }; - #endif

5 18

[alsa-devel] [PATCH v2 1/2] ASoC: codec: use enable pin to control dmic start and stop
by Lin Huang 05 Sep '17

05 Sep '17

From: huang lin <hl(a)rock-chips.com> on some board use enable pin to control dmic start and stop, so add this feature in dmic driver. Signed-off-by: Lin Huang <hl(a)rock-chips.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/dmic.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 010811e..d98233b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 - select SND_SOC_DMIC + select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 12e07f9..b88a1ee 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,8 @@ * */ +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +29,34 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> +static int dmic_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); + + if (!dmic_en) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + gpiod_set_value(dmic_en, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(dmic_en, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops dmic_dai_ops = { + .trigger = dmic_daiops_trigger, +}; + static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -38,8 +68,23 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &dmic_dai_ops, }; +static int dmic_codec_probe(struct snd_soc_codec *codec) +{ + struct gpio_desc *dmic_en; + + dmic_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic_en)) + return PTR_ERR(dmic_en); + + snd_soc_codec_set_drvdata(codec, dmic_en); + + return 0; +} + static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, SND_SOC_NOPM, 0, 0), @@ -51,6 +96,7 @@ static const struct snd_soc_dapm_route intercon[] = { }; static const struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_codec_probe, .component_driver = { .dapm_widgets = dmic_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), -- 2.7.4

4 4

[alsa-devel] [RFC PATCH 0/7] AC97 device/driver model revamp
by Robert Jarzmik 04 Sep '17

04 Sep '17

It all started in the pxa device-tree submission here : https://lkml.org/lkml/2016/2/25/965 It will be maintained in : git fetch https://github.com/rjarzmik/linux.git work/ac97 And now it transformed into this RFC, which would bring a ground for AC'97 devices closer to the linux device/driver model. This is just an RFC to see if we understand each other Mark. Amongst the driving ideas behind this RFC : - use device/driver model for AC'97 device/driver binding is based on vendor_id{1,2} - use auto-probe of the AC'97 bus to enumerate codecs and create devices - be compatible with previous platform_data model for codecs => this enables a smooth transition, where a codec (here wm9713) can be used either through platform_device or automatic discovery - struct snd_ac97 is not used This structure is really heavy, and doesn't represent an AC97 device, but rather an AC'97 codec internals IIUC. I think it's right place would be in include/sound/ac97/codec.h. This is also a problem for regmap adherence, hence the include/sound/ac97/compat.h. - a new AC'97 bus Kconfig is created This was done for a smooth transition ... let's see if it's a good idea. - split the AC'97 into : - the codec - the digital controller - the bus Amongst the things that are not all touched yet : - ac97_codec.c The generic AC'97 codec in sound/pci/ac97/ac97_codec.c has really a lot features, which are not at all PCI related. They could be transfered to sound/ac97, without the pci specific field. => this deserves a deep ahead thinking, which I'll do if the current approach is accepted by the communauty. The most important patch is 2/7 "ALSA: ac97: add an ac97 bus". This is the main evaluation point for the serie, the others are here so that I can test it all. Well, this is a long term effort, which might need a complete rewrite according to the comments it'll get. Let's expose it for comments and see how I can progress with it. Cheers. -- Robert Robert Jarzmik (7): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: wm9713: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support ARM: pxa: mioa701 remove wm9713 from platform devices ASoC: mioa701_wm9713: convert to new ac97 bus ASoC: add new ac97 bus support arch/arm/mach-pxa/mioa701.c | 1 - include/sound/ac97/codec.h | 98 ++++++++++++ include/sound/ac97/compat.h | 21 +++ include/sound/ac97/controller.h | 39 +++++ include/sound/ac97/regs.h | 262 +++++++++++++++++++++++++++++++ include/sound/ac97_codec.h | 239 +---------------------------- include/sound/pxa2xx-lib.h | 14 +- sound/Kconfig | 7 + sound/Makefile | 1 + sound/ac97/Kconfig | 9 ++ sound/ac97/Makefile | 5 + sound/ac97/ac97_core.h | 10 ++ sound/ac97/bus.c | 330 ++++++++++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 ++ sound/ac97/snd_ac97_compat.c | 104 +++++++++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 31 ++-- sound/soc/Kconfig | 4 + sound/soc/codecs/wm9713.c | 62 +++++++- sound/soc/pxa/Kconfig | 4 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/pxa2xx-ac97.c | 12 +- 22 files changed, 1000 insertions(+), 273 deletions(-) create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.1.4

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[alsa-devel] [PATCH v5 00/12] AC97 device/driver model revamp
by Robert Jarzmik 04 Sep '17

04 Sep '17

Hi Lars, Mark, Charles, Lee, This is a minor revision after v4, mainly reviews from Takashi, Charles. I extracted the diff from the v3 of sound/ac97 which is in [1] for easier spotting of what has globally changed. I tentatively kept your Ack Charles for patch 11/12, please shout if you disagree (there is an #ifdef added). I would still need a review from you, Lee, for the mfd patch (patch 6/12). Happy review. -- Robert Robert Jarzmik (12): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++++++----- drivers/mfd/Kconfig | 15 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 405 ++++++++++++++++++++++++ include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 118 +++++++ include/sound/ac97/compat.h | 20 ++ include/sound/ac97/controller.h | 85 +++++ include/sound/ac97/regs.h | 262 ++++++++++++++++ include/sound/ac97_codec.h | 239 +------------- include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 16 + sound/ac97/bus.c | 539 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 108 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 37 ++- sound/arm/pxa2xx-ac97.c | 35 ++- sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 61 ++-- sound/soc/codecs/wm9712.c | 39 ++- sound/soc/codecs/wm9713.c | 39 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 46 ++- 29 files changed, 2011 insertions(+), 414 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.11.0 [1] sound diff from last revision (git diff submit/ac97-3..submit/ac97-4 sound/ac97 include/sound/ac97) diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h index d876464bf7e4..1351cba40048 100644 --- a/include/sound/ac97/compat.h +++ b/include/sound/ac97/compat.h @@ -13,7 +13,6 @@ #define AC97_COMPAT_H #include <sound/ac97_codec.h> -#include <sound/soc.h> struct snd_ac97 *snd_ac97_compat_alloc(struct ac97_codec_device *adev); void snd_ac97_compat_release(struct snd_ac97 *ac97); diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h index 18c699925a85..b36ecdd64f14 100644 --- a/include/sound/ac97/controller.h +++ b/include/sound/ac97/controller.h @@ -8,12 +8,13 @@ #ifndef AC97_CONTROLLER_H #define AC97_CONTROLLER_H +#include <linux/device.h> #include <linux/list.h> #define AC97_BUS_MAX_CODECS 4 #define AC97_SLOTS_AVAILABLE_ALL 0xf -struct device; +struct ac97_controller_ops; /** * struct ac97_controller - The AC97 controller of the AC-Link @@ -21,8 +22,8 @@ struct device; * @controllers: linked list of all existing controllers. * @adap: the shell device ac97-%d, ie. ac97 adapter * @nr: the number of the shell device - * @parent: the device providing the AC97 controller. * @slots_available: the mask of accessible/scanable codecs. + * @parent: the device providing the AC97 controller. * @codecs: the 4 possible AC97 codecs (NULL if none found). * @codecs_pdata: platform_data for each codec (NULL if no pdata). * @@ -34,8 +35,8 @@ struct ac97_controller { struct list_head controllers; struct device adap; int nr; - struct device *parent; unsigned short slots_available; + struct device *parent; struct ac97_codec_device *codecs[AC97_BUS_MAX_CODECS]; void *codecs_pdata[AC97_BUS_MAX_CODECS]; }; diff --git a/sound/ac97/ac97_core.h b/sound/ac97/ac97_core.h index 219fec0d52e0..08441a4fda7c 100644 --- a/sound/ac97/ac97_core.h +++ b/sound/ac97/ac97_core.h @@ -8,3 +8,9 @@ unsigned int snd_ac97_bus_scan_one(struct ac97_controller *ac97, unsigned int codec_num); + +static inline bool ac97_ids_match(unsigned int id1, unsigned int id2, + unsigned int mask) +{ + return (id1 & mask) == (id2 & mask); +} diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index af618a0e89cb..31f858eceffc 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -105,6 +105,7 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, return 0; err_free_codec: + put_device(&codec->dev); kfree(codec); ac97_ctrl->codecs[idx] = NULL; @@ -171,14 +172,10 @@ static int ac97_bus_reset(struct ac97_controller *ac97_ctrl) */ int snd_ac97_codec_driver_register(struct ac97_codec_driver *drv) { - int ret; - drv->driver.bus = &ac97_bus_type; - ret = driver_register(&drv->driver); - - return ret; + return driver_register(&drv->driver); } -EXPORT_SYMBOL(snd_ac97_codec_driver_register); +EXPORT_SYMBOL_GPL(snd_ac97_codec_driver_register); /** * snd_ac97_codec_driver_unregister - unregister an AC97 codec driver @@ -190,7 +187,7 @@ void snd_ac97_codec_driver_unregister(struct ac97_codec_driver *drv) { driver_unregister(&drv->driver); } -EXPORT_SYMBOL(snd_ac97_codec_driver_unregister); +EXPORT_SYMBOL_GPL(snd_ac97_codec_driver_unregister); /** * snd_ac97_codec_get_platdata - get platform_data @@ -208,7 +205,7 @@ void *snd_ac97_codec_get_platdata(const struct ac97_codec_device *adev) return ac97_ctrl->codecs_pdata[adev->num]; } -EXPORT_SYMBOL(snd_ac97_codec_get_platdata); +EXPORT_SYMBOL_GPL(snd_ac97_codec_get_platdata); static void ac97_ctrl_codecs_unregister(struct ac97_controller *ac97_ctrl) { @@ -359,7 +356,7 @@ struct ac97_controller *snd_ac97_controller_register( kfree(ac97_ctrl); return ERR_PTR(ret); } -EXPORT_SYMBOL(snd_ac97_controller_register); +EXPORT_SYMBOL_GPL(snd_ac97_controller_register); /** * snd_ac97_controller_unregister - unregister an ac97 controller @@ -370,7 +367,7 @@ void snd_ac97_controller_unregister(struct ac97_controller *ac97_ctrl) { ac97_del_adapter(ac97_ctrl); } -EXPORT_SYMBOL(snd_ac97_controller_unregister); +EXPORT_SYMBOL_GPL(snd_ac97_controller_unregister); #ifdef CONFIG_PM static int ac97_pm_runtime_suspend(struct device *dev) @@ -466,7 +463,7 @@ static int ac97_bus_match(struct device *dev, struct device_driver *drv) return false; do { - if ((id[i].id & id->mask) == (adev->vendor_id & id[i].mask)) + if (ac97_ids_match(id[i].id, adev->vendor_id, id[i].mask)) return true; } while (id[i++].id); diff --git a/sound/ac97/snd_ac97_compat.c b/sound/ac97/snd_ac97_compat.c index ac8d835c1513..61544e0d8de4 100644 --- a/sound/ac97/snd_ac97_compat.c +++ b/sound/ac97/snd_ac97_compat.c @@ -88,16 +88,19 @@ int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, { struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); struct ac97_controller *actrl = adev->ac97_ctrl; + unsigned int scanned; if (try_warm) { compat_ac97_warm_reset(ac97); - if (snd_ac97_bus_scan_one(actrl, adev->num) == adev->vendor_id) + scanned = snd_ac97_bus_scan_one(actrl, adev->num); + if (ac97_ids_match(scanned, adev->vendor_id, id_mask)) return 1; } compat_ac97_reset(ac97); compat_ac97_warm_reset(ac97); - if (snd_ac97_bus_scan_one(actrl, adev->num) == adev->vendor_id) + scanned = snd_ac97_bus_scan_one(actrl, adev->num); + if (ac97_ids_match(scanned, adev->vendor_id, id_mask)) return 0; return -ENODEV;

3 16

[alsa-devel] [PATCH v4 00/12] AC97 device/driver model revamp
by Robert Jarzmik 04 Sep '17

04 Sep '17

Hi Lars, Mark, Charles, Lee, This is a minor revision after v3, mainly reviews from Takashi, Charles. I did more extensive testing of load/unload of the whole serie, and apart from a bug in wm97xx-core in touchscreen tree (for a which a patch will be sent apart), it seems to work properly. I extracted the diff from the v3 of sound/ac97 which is in [1] for easier spotting of what has globally changed. I would still need a review from you, Lee, for the mfd patch (patch 6/12). Happy review. -- Robert Robert Jarzmik (12): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++++++----- drivers/mfd/Kconfig | 15 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 405 ++++++++++++++++++++++++ include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 118 +++++++ include/sound/ac97/compat.h | 21 ++ include/sound/ac97/controller.h | 84 +++++ include/sound/ac97/regs.h | 262 +++++++++++++++ include/sound/ac97_codec.h | 239 +------------- include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 10 + sound/ac97/bus.c | 542 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 105 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 37 ++- sound/arm/pxa2xx-ac97.c | 35 ++- sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 59 ++-- sound/soc/codecs/wm9712.c | 37 ++- sound/soc/codecs/wm9713.c | 39 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 46 ++- 29 files changed, 2001 insertions(+), 414 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.11.0 [1] sound diff from last revision (git diff submit/ac97-2..submit/ac97-3 sound/ac97 include/sound/ac97) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7cda8175341e..af618a0e89cb 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -76,7 +76,6 @@ static void ac97_codec_release(struct device *dev) adev = to_ac97_device(dev); ac97_ctrl = adev->ac97_ctrl; ac97_ctrl->codecs[adev->num] = NULL; - sysfs_remove_link(&dev->kobj, "ac97_controller"); kfree(adev); } @@ -269,8 +268,29 @@ static const struct attribute_group *ac97_adapter_groups[] = { NULL, }; +static void ac97_del_adapter(struct ac97_controller *ac97_ctrl) +{ + mutex_lock(&ac97_controllers_mutex); + ac97_ctrl_codecs_unregister(ac97_ctrl); + list_del(&ac97_ctrl->controllers); + mutex_unlock(&ac97_controllers_mutex); + + device_unregister(&ac97_ctrl->adap); +} + +static void ac97_adapter_release(struct device *dev) +{ + struct ac97_controller *ac97_ctrl; + + ac97_ctrl = to_ac97_controller(dev); + idr_remove(&ac97_adapter_idr, ac97_ctrl->nr); + dev_dbg(&ac97_ctrl->adap, "adapter unregistered by %s\n", + dev_name(ac97_ctrl->parent)); +} + static const struct device_type ac97_adapter_type = { .groups = ac97_adapter_groups, + .release = ac97_adapter_release, }; static int ac97_add_adapter(struct ac97_controller *ac97_ctrl) @@ -298,16 +318,6 @@ static int ac97_add_adapter(struct ac97_controller *ac97_ctrl) return ret; } -static void ac97_del_adapter(struct ac97_controller *ac97_ctrl) -{ - mutex_lock(&ac97_controllers_mutex); - ac97_ctrl_codecs_unregister(ac97_ctrl); - list_del(&ac97_ctrl->controllers); - idr_remove(&ac97_adapter_idr, ac97_ctrl->nr); - mutex_unlock(&ac97_controllers_mutex); - put_device(&ac97_ctrl->adap); -} - /** * snd_ac97_controller_register - register an ac97 controller * @ops: the ac97 bus operations @@ -508,7 +518,7 @@ static int ac97_bus_remove(struct device *dev) } static struct bus_type ac97_bus_type = { - .name = "ac97", + .name = "ac97bus", .dev_groups = ac97_dev_groups, .match = ac97_bus_match, .pm = &ac97_pm, @@ -522,5 +532,11 @@ static int __init ac97_bus_init(void) } subsys_initcall(ac97_bus_init); +static void __exit ac97_bus_exit(void) +{ + bus_unregister(&ac97_bus_type); +} +module_exit(ac97_bus_exit); + MODULE_LICENSE("GPL"); MODULE_AUTHOR("Robert Jarzmik <robert.jarzmik(a)free.fr>");

4 19

[alsa-devel] [GIT PULL] cs43130 fixes
by Mark Brown 04 Sep '17

04 Sep '17

The following changes since commit 465c925695593b43695e2396f7279879a72275eb: ASoC: cs43130: Add devicetree bindings for CS43130 (2017-09-01 11:45:16 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.14-cs43130 for you to fetch changes up to 7051334672e54fae67e02d5d3296fb62b3343be7: ASoC: cs43130: Fix unused compiler warnings for PM runtime (2017-09-04 15:45:12 +0100) ---------------------------------------------------------------- ASoC: Fixes for the CS43130 driver A couple of warning fixes for the newly added CS43130 driver. ---------------------------------------------------------------- Colin Ian King (1): ASoC: cs43130: fix spelling mistake: "irq_occurrance" -> "irq_occurrence" Takashi Iwai (2): ASoC: cs43130: Fix possible Oops with invalid dev_id ASoC: cs43130: Fix unused compiler warnings for PM runtime sound/soc/codecs/cs43130.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-)

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: cs43130: Fix possible Oops with invalid dev_id
by Takashi Iwai 04 Sep '17

04 Sep '17

As compiler spotted out, there is the potential NULL-dereference in the code when dc-measure OF is given for other than 43130/43131: sound/soc/codecs/cs43130.c:2089:18: warning: ‘hpload_seq’ may be used uninitialized in this function [-Wmaybe-uninitialized] Warn it and return before triggering Oops. Fixes: 8f1e5bf9b440 ("ASoC: cs43130: Add support for CS43130 codec") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/cs43130.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 220e30199c5b..3fe3b8d644b7 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2079,6 +2079,10 @@ static void cs43130_imp_meas(struct work_struct *wk) case CS43131_CHIP_ID: hpload_seq = hpload_seq2; seq_size = ARRAY_SIZE(hpload_seq2); + break; + default: + WARN(1, "Invalid dev_id for meas: %d", cs43130->dev_id); + return; } i = 0; -- 2.14.1

3 6

[alsa-devel] [PATCH] ALSA: hda - Fix memory leak at error path
by Takashi Iwai 04 Sep '17

04 Sep '17

The CA0132 codec driver doesn't call the free function at its error path of the probe, which leaves the allocated memory. Call ca0132_free() properly at the error handling. Fixes: a73d511c4867 ("ALSA: hda/ca0132: Add unsol handler for DSP and jack detection") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index a148176c16a9..3e73d5c6ccfc 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4774,13 +4774,17 @@ static int patch_ca0132(struct hda_codec *codec) err = ca0132_prepare_verbs(codec); if (err < 0) - return err; + goto error; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); if (err < 0) - return err; + goto error; return 0; + + error: + ca0132_free(codec); + return err; } /* -- 2.14.1

1 0

[alsa-devel] [PATCH v2] ALSA: hda: Fix forget to free resource in error handling code path in hda_codec_driver_probe
by Wang YanQing 04 Sep '17

04 Sep '17

When hda_codec_driver_probe meet error and return failure, we need to free resource with patch_ops.free, or we will get resource leak. Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- Changes v1-v2: 1: Fix calling patch_ops.free after module_put, reported by Takashi Iwai. Hi! Takashi Iwai Because the only one calling site of patch_ops.free is in hda_codec_driver_remove, and if hda_codec_driver_probe return failure, we will no chance to call the hda_codec_driver_remove, so I think we don't need to care below case: "Or, for safety, we may put an internal flag to indicate that the codec free got already called, and check it at before calling patch_ops.free, too." Thanks! sound/pci/hda/hda_bind.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 6efadbf..d361bb7 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -100,7 +100,7 @@ static int hda_codec_driver_probe(struct device *dev) if (patch) { err = patch(codec); if (err < 0) - goto error_module; + goto error_module_put; } err = snd_hda_codec_build_pcms(codec); @@ -120,6 +120,9 @@ static int hda_codec_driver_probe(struct device *dev) return 0; error_module: + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + error_module_put: module_put(owner); error: -- 1.8.5.6.2.g3d8a54e.dirty

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs43130: Fix possible Oops with invalid dev_id
by Takashi Iwai 04 Sep '17

04 Sep '17

As compiler spotted out, there is the potential NULL-dereference in the code when dc-measure OF is given for other than 43130/43131: sound/soc/codecs/cs43130.c:2089:18: warning: ‘hpload_seq’ may be used uninitialized in this function [-Wmaybe-uninitialized] Warn it and return before triggering Oops. Fixes: 8f1e5bf9b440 ("ASoC: cs43130: Add support for CS43130 codec") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/cs43130.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 220e30199c5b..3fe3b8d644b7 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2079,6 +2079,10 @@ static void cs43130_imp_meas(struct work_struct *wk) case CS43131_CHIP_ID: hpload_seq = hpload_seq2; seq_size = ARRAY_SIZE(hpload_seq2); + break; + default: + WARN(1, "Invalid dev_id for meas: %d", cs43130->dev_id); + return; } i = 0; -- 2.14.1

2 3

[alsa-devel] [GIT PULL] ASoC updates for v4.14
by Mark Brown 04 Sep '17

04 Sep '17

The following changes since commit cc4a41fe5541a73019a864883297bd5043aa6d98: Linux 4.13-rc7 (2017-08-27 17:20:40 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.14 for you to fetch changes up to f5fd4a67bef5f2c5c2c9eb0cb2900e12d192ae23: Merge remote-tracking branches 'asoc/topic/wm8524', 'asoc/topic/wm8804' and 'asoc/topic/zte' into asoc-next (2017-09-01 12:13:19 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.14 This is quite a large release by volume of patches and diff, a lot of that is mechanical cleanup patches but it's great to also see a range of vendors actively working on adding new features and fixing issues in their drivers. Intel and Realtek have been especially active here. - Continued work towards moving everything to the component model from Morimoto-san. - Use of devres for jack detection GPIOs, eliminating some potential resource leaks. - Jack detection support for Qualcomm MSM8916. - Support for Allwinner H3, Cirrus Logic CS43130, Intel Kabylake systems with RT5663, Realtek RT274, TI TLV320AIC32x6 and Wolfson WM8523. ---------------------------------------------------------------- Andy Shevchenko (3): ASoC: rt5677: Hide platform data in the module sources ASoC: rt5677: Refactor code to avoid comparison unsigned >= 0 ASoC: rt5677: Remove never used variables Anton Vasilyev (1): ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove Arnaud Pouliquen (2): ASoC: Add bindings for DMIC codec driver ASoC: codec: add DT support in dmic codec Arnd Bergmann (1): ASoC: codecs: fix wm8524 build error Arvind Yadav (27): ASoC: rt5514: constify acpi_device_id. ASoC: rt5663: constify acpi_device_id. ASoC: rt5659: constify acpi_device_id. ASoC: rt5665: constify acpi_device_id. ASoC: samsung: s3c2412: Handle return value of clk_prepare_enable. ASoC: samsung: s3c24xx: Handle return value of clk_prepare_enable. ASoC: samsung: pcm: Handle return value of clk_prepare_enable. ASoC: samsung: i2s: Handle return value of clk_prepare_enable. ASoC: samsung: spdif: Handle return value of clk_prepare_enable. ASoC: mxs-saif: Handle return value of clk_prepare_enable/clk_prepare. ASoC: jz4740: Handle return value of clk_prepare_enable. ASoC: sun4i-spdif: Handle return value of clk_prepare_enable. ASoC: atmel: ac97c: Handle return value of clk_prepare_enable. ASoC: Intel: constify pci_device_id. ASoC: samsung: constify snd_pcm_ops structures ASoC: au1x: constify snd_pcm_ops structures ASoC: blackfin: constify snd_pcm_ops structures ASoC: fsl: constify snd_pcm_ops structures ASoC: omap: constify snd_pcm_ops structures ASoC: pxa: constify snd_pcm_ops structures ASoC: txx9: constify snd_pcm_ops structures ASoC: intel: constify snd_pcm_ops structures ASoC: nuc900: constify snd_pcm_ops structures ASoC: sh: constify snd_pcm_ops structures ASoC: codecs: constify snd_soc_dai_ops structures ASoC: codecs: constify snd_soc_dai_ops structures ASoC: ux500: constify snd_soc_dai_ops structures Axel Lin (3): ASoC: wm8523: Fix array size for bclk_ratios ASoC: wm8523: Constfiy lrclk_ratios and bclk_ratios ASoC: wm8524: Don't use dev_err to show supported sample rate Bard Liao (9): ASoC: rt5665: force using PLL if MCLK is not suitable ASoC: rt274: add rt274 codec driver ASoC: rt274: correct comment style ASoC: rt5665: add clcok control for master mode ASoC: rt5665: add clock sync control for master mode ASoC: rt5665: add MONOVOL Playback Volume control ASoC: rt5670: add symmetric_rates flag ASoC: rt5670: merge ADC L/R Mux ASoC: rt274: add acpi id Bhumika Goyal (31): ASoC: sunxi: make snd_soc_codec_driver structures as const ASoC: codecs: add const to snd_soc_codec_driver structures ASoC: samsung: make snd_soc_platform_driver const ASoC: fsl: make snd_soc_platform_driver const ASoC: nuc900: make snd_soc_platform_driver const ASoC: omap: make snd_soc_platform_driver const ASoC: pxa: make snd_soc_platform_driver const ASoC: qcom: make snd_soc_platform_driver const ASoC: sh: make snd_soc_platform_driver const ASoC: txx9: make snd_soc_platform_driver const ASoC: soc-utils: make snd_soc_platform_driver const ASoC: codecs: make snd_soc_platform_driver const ASoC: Intel: make snd_soc_platform_driver const ASoC: Intel: kbl: make snd_pcm_hw_constraint_list const ASoC: Intel: Skylake: make skl_dsp_fw_ops const ASoC: codecs: make snd_compr_ops const ASoC: fsl: make snd_pcm_hardware const ASoC: kirkwood: make snd_pcm_hardware const ASoC: sh: make snd_pcm_hardware const ASoC: qcom: make snd_pcm_hardware const ASoC: Intel: Atom: make snd_pcm_hardware const ASoC: Intel: Skylake: make snd_pcm_hardware const ASoC: aux1x: make snd_soc_ops const ASoC: cirrus: make snd_soc_ops const ASoC: generic: make snd_soc_ops const ASoC: rockchip: make snd_soc_ops const ASoC: samsung: make snd_soc_ops const ASoC: sh: make snd_soc_ops const ASoC: bcm: make snd_soc_dai_driver const ASoC: codecs: make snd_soc_dai_driver and snd_soc_component_driver const ASoC: rt5645: make rt5645_platform_data const Christophe Jaillet (3): ASoC: mediatek: Fix an error checking code ASoC: davinci-mcasp: check memory allocation failure ALSA: ac97c: Fix an error handling path in 'atmel_ac97c_probe()' Colin Ian King (8): ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static ASoC: rt5663: add in missing loop counter to avoid infinite loop ASoC: Intel: Skylake: fix type in debug message ASoC: max98926: make max98926_spk_tlv and max98926_current_tlv static ASoC: rt5514: make array rt5514_dai static ASoC: zx_aud96p22: make array aud96p22_dt_ids static ASoC: samsung: make tm2_dapm_widgets and tm2_pm_ops static ASoC: hdmi-codec: make a function and two arrays static Dan Carpenter (1): ASoC: codecs: msm8916-wcd-analog: always true condition Dan Murphy (2): ASoC: tlv320aic32x4: Add support for tlv320aic32x6 ASoC: tlv320aic32x4: Add gpio configuration to the codec Daniel Baluta (1): ASoC: simple-scu-card: Parse off codec widgets Dharageswari R (1): ASoC: Intel: Skylake: Add num of cores in dsp ops Donglin Peng (11): ASoC: s3c24xx_uda134x: Remove unnecessary function call ASoC: atmel: Remove unnecessary function call ASoC: atmel: Remove unnecessary function call ASoC: rockchip: Remove unnecessary function call ASoC: mxs-sgtl5000: Remove unnecessary function call ASoC: qcom: Remove useless function call ASoC: qcom: Remove unnecessary function call ASoC: sun4i-codec: Remove unnecessary function call ASoC: mediatek: Remove unnecessary function call ASoC: tegra: Remove unnecessary function call ASoC: ux500: Remove unnecessary function call Dronamraju Santosh P K (1): ASoC: Intel: bxtn: Remove code loader reference in cleanup Fabio Estevam (11): ASoC: sun4i-codec: Remove unneeded gpiod NULL check ASoC: adau1977: Remove unneeded gpiod NULL check ASoC: wm8804: Remove unneeded gpiod NULL check ASoC: cs35l33: Remove unneeded gpiod NULL check ASoC: cs35l34: Remove unneeded gpiod NULL check ASoC: cs42l42: Remove unneeded gpiod NULL check ASoC: cs53l30: Remove unneeded gpiod NULL check ASoC: sta32x: Remove unneeded gpiod NULL check ASoC: soc-pcm: Remove unused 'debugfs_dpcm_state' entry ASoC: soc-core: Remove unneeded dentry member from snd_soc_codec ASoC: soc-core: Use IS_ERR_OR_NULL() Geert Uytterhoeven (1): ASoC: rsnd: Drop unit-addresses without reg properties Guneshwor Singh (7): ASoC: Intel: Skylake: Use num_core to allocate instead of macro ASoC: Intel: cnl: Add cnl dsp functions and registers ASoC: Intel: Skylake: Move platform specific init to platform dsp_init() ASoC: Intel: cnl: Unstatify common ipc functions ASoC: Intel: cnl: Add sst library functions for cnl platform ASoC: Intel: cnl: add dsp ops for cannonlake ASoC: Intel: cnl: add pci id for cnl Gustavo A. R. Silva (32): ASoC: kirkwood-i2s: fix error return code in kirkwood_i2s_dev_probe() ASoC: spear: fix error return code in spdif_in_probe() ASoC: twl6040: fix error return code in twl6040_probe() ASoC: fsi: constify dev_pm_ops structure ASoC: psc-ac97: constify dev_pm_ops structure ASoC: au1x: psc-i2s: constify dev_pm_ops structure ASoC: Intel: Skylake: constify snd_soc_dai_ops structures ASoC: Intel: Atom: constify snd_soc_dai_ops structures ASoC: tegra: constify snd_soc_dai_ops structure ASoC: sun8i-codec: constify snd_soc_dai_ops structure ASoC: fsl_spdif: constify snd_soc_dai_ops structure ASoC: fsl_esai: constify snd_soc_dai_ops structure ASoC: fsl_asrc: constify snd_soc_dai_ops structure ASoC: hisilicon: constify snd_soc_dai_ops structure ASoC: mediatek: mt2701: constify snd_soc_dai_ops structure ASoC: mmp-sspa: constify snd_soc_dai_ops structure ASoC: zx-i2s: constify snd_soc_dai_ops structure ASoC: zx-tdm: constify snd_soc_dai_ops structure ASoC: zte: spdif: constify snd_soc_dai_ops structure ASoC: cs42l42: constify snd_soc_dai_ops structure ASoC: rt5663: constify snd_soc_dai_ops structure ASoC: rt5616: constify snd_soc_dai_ops structure ASoC: msm8916-wcd-analog: constify snd_soc_dai_ops structure ASoC: max98926: constify snd_soc_dai_ops structure ASoC: rt5514: constify snd_soc_dai_ops structure ASoC: codecs: msm8916-wcd-digital: constify snd_soc_dai_ops structure ASoC: max9867: constify snd_soc_dai_ops structure ASoC: rk3036: constify snd_soc_dai_ops structure ASoC: hdac_hdmi: constify snd_soc_dai_ops structure ASoC: dwc: constify snd_soc_dai_ops structure ASoC: tas5720: constify snd_soc_dai_ops structure ASoC: rsnd: remove unnecessary static in rsnd_ssiu_probe() Harsha Priya N (4): ASoC: Intel: Enabling 4 slot IV feedback for max98927 on Kabylake platform ASoC: Intel: kbl_rt5663_rt5514_max98927: Add rt5514 spi dailink ASoC: Intel: kbl: Enabling ASRC for RT5663 codec on kabylake platform ASoC: Intel: Headset button support in kabylake machine driver Hsin-Yu Chao (2): ASoC: rt5514: expose Hotword Model control ASoC: rt5514: Guard Hotword Model bytes loading Ian W MORRISON (1): ASoC: rt5645: Add jack detection workaround for MINIX Z83-4 based devices Jaechul Lee (1): ASoC: samsung: i2s: Support more resolution rates Jeeja KP (1): ASoC: Intel: Skylake: Update module id in pin connections Jeffy Chen (10): ASoC: soc-core: Allow searching dai driver name in snd_soc_find_dai ASoC: rockchip: Correct 'dmic-delay' property name ASoC: Add a sanity check before using dai driver name ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec ASoC: rockchip: Update description of rockchip, codec ASoC: rockchip: constify snd_soc_ops structures ASoC: hdmi-codec: Use different name for playback streams John Hsu (1): ASoC: nau8825: correct typo of semaphore comment Julia Lawall (7): ASoC: rsnd: add missing of_node_put ASoC: blackfin: constify snd_soc_dai_ops structures ASoC: rockchip: constify snd_soc_dai_ops structures ASoC: codecs: es8316: constify snd_soc_dai_ops structures ASoC: codecs: zx_aud96p22: constify snd_soc_dai_ops structures ASoC: spear: constify snd_soc_dai_ops structures ASoC: Intel: constify snd_compr_codec_caps structures Kevin Cheng (2): ASoC: Intel: Add Kabylake machine driver for RT5663 ASoC: Intel: Add Kabylake RT5663 machine driver entry Krzysztof Kozlowski (4): ASoC: samsung: Remove non-existing CONFIG_CPU_S3C2413 ASoC: samsung: Fix possible double iounmap on s3c24xx driver probe failure ASoC: samsung: Add missing prepare for iis clock of s3c24xx ASoC: samsung: Add proper error paths to s3c24xx I2S driver Kuninori Morimoto (34): ASoC: hdmi-codec: ELD control corresponds to the PCM stream ASoC: rsnd: don't use private_value on rsnd_kctrl_new() ASoC: remove cache_bypass from snd_soc_codec ASoC: rsnd: merge snd_pcm_ops::open and snd_soc_dai_ops::startup ASoC: rcar: unregister fixed rate on ADG ASoC: soc-core: rename "cmpnt" to "component" ASoC: soc-core: remove duplicate mutex_unlock from snd_soc_unregister_component() ASoC: rsnd: move rsnd_ssi_config_init() execute condition into it. ASoC: rsnd: rsnd_ssi_can_output_clk() macro ASoC: rsnd: control SSICR::EN correctly ASoC: rsnd: remove 16ch support for now ASoC: rsnd: avoid duplicate free_irq() ASoC: rsnd: call free_irq() both DMA/PIO mode ASoC: rsnd: add rsnd_ssi_flags_has/add() macro ASoC: rsnd: call request_irq/free_irq once in MIX case ASoC: soc-core: add snd_soc_rtdcom_xxx() ASoC: use snd_soc_rtdcom_add() and convert to consistent operation ASoC: soc-core: snd_soc_unregister_component() unregister all component ASoC: rsnd: tidyup comments position/space/tab ASoC: rsnd: remove unused rsnd_xxx_to_dma() ASoC: wm8524: remove unnecessary snd_soc_unregister_platform() ASoC: simple-scu-card: Add pm callbacks to platform driver ASoC: audio-graph-card: Add pm callbacks to platform driver ASoC: audio-graph-scu-card: Add pm callbacks to platform driver ASoC: use snd_soc_component_get_dapm() ASoC: remove duplicate definition of controls/num_controls ASoC: remove duplicate definition of dapm_widgets/num_dapm_widgets ASoC: remove duplicate definition of dapm_routes/num_dapm_routes ASoC: add Component level set_sysclk ASoC: add Component level set_pll ASoC: add Component level set_jack ASoC: add missing compile rule for max98371 ASoC: add missing compile rule for sirf-audio-codec ASoC: add missing compile rule for max98371 Li Xu (2): ASoC: cs43130: Add support for CS43130 codec ASoC: cs43130: Add devicetree bindings for CS43130 Lori Hikichi (1): ASoC: cygnus: Add support for 384kHz frame rates Lucas Stach (2): ASoC: fsl-asoc-card: don't print EPROBE_DEFER as error ASoC: simple_card_utils: fix fallback when "label" property isn't present Marcus Cooper (12): ASoC: sun4i-i2s: Extend quirks scope ASoC: sun4i-i2s: Add clkdiv offsets to quirks ASoC: sun4i-i2s: Add regmap config to quirks ASoC: sun4i-i2s: Add TX FIFO offset to quirks ASoC: sun4i-i2s: Add regmap fields for channels ASoC: sun4i-i2s: Add regfields for word size select and sample resolution ASoC: sun4i-i2s: bclk and lrclk polarity tidyup ASoC: sun4i-i2s: Add mclk enable regmap field ASoC: sun4i-i2s: Add regmap field to set DAI format ASoC: sun4i-i2s: Check for slave select bit ASoC: sun4i-i2s: Update global enable with bitmask ASoC: sun4i-i2s: Add support for H3 Mark Brown (40): Merge branch 'fix/rcar' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branches 'topic/rt274', 'topic/hdmi' and 'topic/msm8916' of git://git.kernel.org/.../broonie/sound into asoc-const Merge branch 'topic/qcom' of git://git.kernel.org/.../broonie/sound into asoc-msm8916 Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'topic/pcm' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'topic/sh' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branch 'topic/const' of git://git.kernel.org/.../broonie/sound into asoc-dmic Merge branches 'topic/dmic' and 'topic/qcom' of git://git.kernel.org/.../broonie/sound into asoc-msm8916 Merge tag 'v4.13-rc6' into asoc-msm8916 Merge branch 'topic/msm8916' of git://git.kernel.org/.../broonie/sound into asoc-const Merge branch 'topic/const' of git://git.kernel.org/.../broonie/sound into asoc-blackfin Merge tag 'sound-4.13-rc7' of git://git.kernel.org/.../tiwai/sound into asoc-rt5677 Merge tag 'v4.13-rc7' into asoc-component Merge remote-tracking branch 'asoc/fix/rt5670' into asoc-fixes Merge tag 'asoc-fix-v4.13-rc7' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona' and 'asoc/fix/mtk' into asoc-linus Merge remote-tracking branch 'asoc/topic/msm8916' into asoc-next Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/const' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/adau1977', 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/au1x' and 'asoc/topic/bcm' into asoc-next Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/build', 'asoc/topic/cirrus' and 'asoc/topic/compress' into asoc-next Merge remote-tracking branches 'asoc/topic/cs35l33', 'asoc/topic/cs35l34' and 'asoc/topic/cs42l42' into asoc-next Merge remote-tracking branches 'asoc/topic/cs43130', 'asoc/topic/cs53l30', 'asoc/topic/cygnus' and 'asoc/topic/davinci' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/es8316', 'asoc/topic/fsi', 'asoc/topic/fsl' and 'asoc/topic/hdmi' into asoc-next Merge remote-tracking branches 'asoc/topic/hisi', 'asoc/topic/img', 'asoc/topic/jack' and 'asoc/topic/jz4740' into asoc-next Merge remote-tracking branches 'asoc/topic/kirkwood', 'asoc/topic/max9867', 'asoc/topic/max98926' and 'asoc/topic/max98927' into asoc-next Merge remote-tracking branches 'asoc/topic/mtk', 'asoc/topic/mxs', 'asoc/topic/nau8825', 'asoc/topic/nuc900' and 'asoc/topic/of-graph' into asoc-next Merge remote-tracking branches 'asoc/topic/omap', 'asoc/topic/psc-ac97' and 'asoc/topic/pxa' into asoc-next Merge remote-tracking branches 'asoc/topic/rk3036', 'asoc/topic/rockchip', 'asoc/topic/rt274' and 'asoc/topic/rt5514' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5563', 'asoc/topic/rt5616', 'asoc/topic/rt5645', 'asoc/topic/rt5659' and 'asoc/topic/rt5663' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5665', 'asoc/topic/rt5670', 'asoc/topic/rt5677' and 'asoc/topic/samsung' into asoc-next Merge remote-tracking branches 'asoc/topic/simple', 'asoc/topic/spear', 'asoc/topic/sta32x', 'asoc/topic/stm32' and 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tas5720', 'asoc/topic/tegra', 'asoc/topic/tlv320aic32x4' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branches 'asoc/topic/twl6040', 'asoc/topic/txx9', 'asoc/topic/utils', 'asoc/topic/ux500' and 'asoc/topic/wm8523' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8524', 'asoc/topic/wm8804' and 'asoc/topic/zte' into asoc-next Markus Elfring (13): ASoC: spear: Delete an error message for a failed memory allocation in two functions ASoC: compress: Delete error messages for a failed memory allocation in snd_soc_new_compress() ASoC: rsnd: Delete an error message for a failed memory allocation in three functions ASoC: blackfin: Delete an error message for a failed memory allocation in sport_create() ASoC: blackfin: Use common error handling code in sport_create() ASoC: blackfin: Add some spaces for better code readability ASoC: dwc: Delete an error message for a failed memory allocation in dw_i2s_probe() ASoC: kirkwood: Delete an error message for a failed memory allocation in kirkwood_i2s_dev_probe() ASoC: fsi: Delete an error message for a failed memory allocation in fsi_probe() ASoC: rockchip: Delete an error message for a failed memory allocation in rockchip_i2s_probe() ASoC: DaVinci: Delete an error message for a failed memory allocation in two functions ASoC: Freescale: Delete an error message for a failed memory allocation in three functions ASoC: Medfield: Delete an error message for a failed memory allocation in snd_mfld_mc_probe() Mihai Serban (1): ASoC: codecs: add wm8524 codec driver Naveen M (2): ASoC: Intel: kbl: Add MST route change to kbl machine drivers ASoC: Intel: kbl: Add jack port initialize in kbl machine drivers Naveen Manohar (5): ASoC: Intel: Skylake: Use correct nuvoton codec ID ASoC: Intel: kbl: Add new FEs for Multi-Playback & Echo-Reference ASoC: Intel: kbl: Add DAI links for Multi-Playback & Echo-reference ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref ASoC: Intel: kbl: Add map for Maxim IV Feedback Oskar Schirmer (1): ASoC: tas2552: Fix fraction overflow in PLL calculation Pardha Saradhi K (1): ASoC: Intel: Skylake: Add dsp cores management Paul Kocialkowski (1): ASoC: tegra: Remove SoC-specific Kconfig depends and selects Peng Donglin (1): ASoC: compress: Set reasonable compress id string Peter Ujfalusi (1): ASoC: tlv320aic3x: Support for OCMV configuration Philipp Zabel (4): ASoC: img: explicitly request exclusive reset control ASoC: stm32: explicitly request exclusive reset control ASoC: sun4i: explicitly request exclusive reset control ASoC: tegra: explicitly request exclusive reset control Pierre-Louis Bossart (2): ASoC: codecs: rt5645: add quirks for Asus T100HA ASoC: codecs: rt5670: add jack detection quirk for Dell Venue 5585 Pradeep Tewani (5): ASoC: Intel: board: Remove .owner initialization in bxt_rt298 driver ASoC: hdac_hdmi: Add the vendor nid for Geminilake HDMI ASoC: Intel: board: Add Geminilake platform support ASoC: Intel: board: Fix missing sentinel for bxt_board_id ASoC: Intel: Skylake: Add IPC to configure the copier secondary pins Ramesh Babu (1): ASoC: Intel: Skylake: Parse and update module config structure Rob Herring (2): ASoC: fsl: Convert to using %pOF instead of full_name ASoC: fsl_dma: remove dma_object path member Ryan Lee (6): ASoC: max98927: Updated volatile register list ASoC: max98927: Added missing \n to end of dev_err messages ASoC: max98927: Modified chip default register values ASoC: max98927: Added PM suspend and resume function ASoC: max98927: Modified DAPM widget and map to enable/disable VI sense path ASoC: max98927: Changed device property read function Ryder Lee (2): ASoC: mediatek: Add interrupt-names property in binding text ASoC: mediatek: switch to use platform_get_irq_byname() Sapthagiri Baratam (1): arizona: anc: Correct setting of FCx_MIC_MODE_SEL Shreyas NC (5): ASoC: Intel: Skylake: Parse multiple manifest data blocks ASoC: Intel: uapi: Add new tokens for module common data ASoC: Intel: Skylake: Commonize parsing of format tokens ASoC: Intel: Skylake: Add driver structures to be filled from topology manifest ASoC: Intel: Skylake: Populate module data from topology manifest Srinivas Kandagatla (9): ASoC: jack: fix snd_soc_codec_set_jack return error ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk ASoC: qcom: apq8016-sbc: set default mclk rate ASoC: codecs: msm8916-wcd-digital: add CIC filter source selection path ASoC: codecs: msm8916-wcd-analog: move codec reset to probe ASoC: codecs: msm8916-wcd-analog: get micbias voltage from dt ASoC: codecs: msm8916-wcd-analog: add MBHC support ASoC: qcom: apq8016-sbc: Add support to Headset JACK Subhransu S. Prusty (5): ASoC: Intel: Skylake: Fix to free dsp resource on ipc_init failure ASoC: Intel: Skylake: Fix to free resources for dsp_init failure ASoC: Intel: Skylake: Fix to free correct dev id in free_irq ASoC: Intel: Skylake: Fix DSP core ref count for init failure ASoC: Intel: Skylake: Fix uninitialized return Sugar Zhang (2): ASoC: rockchip: add bindings for i2s ASoC: rockchip: separate pinctrl pins from each other Sylwester Nawrocki (1): ASoC: samsung: odroid: Drop requirement of clocks in the sound node Takashi Iwai (10): ASoC: jack: Manage gpios via devres ASoC: fsl: Remove superfluous snd_soc_jack_free_gpios() call ASoC: simple-card: Remove superfluous snd_soc_jack_free_gpios() call ASoC: intel: Remove superfluous snd_soc_jack_free_gpios() call ASoC: omap: Remove superfluous snd_soc_jack_free_gpios() call ASoC: pxa: Remove superfluous snd_soc_jack_free_gpios() call ASoC: samsung: Remove superfluous snd_soc_jack_free_gpios() call ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call ASoC: rt5670: Fix GPIO headset detection regression ASoC: tegra: Fix unused variable warning Vitaly Wool (1): ASoC: make clock direction configurable in asoc-simple guneshwor.o.singh(a)intel.com (3): ASoC: Intel: Skylake: Reset the controller in probe ASoC: Intel: Skylake: Remove return check for skl_codec_create() ASoC: Intel: Skylake: Fix potential null pointer dereference huang lin (2): dt-bindings: sound: add dmicen property in dmic driver ASoC: codec: use enable pin to control dmic start and stop oder_chiou(a)realtek.com (18): ASoC: rt5514: Move the auto disable DSP function to set_bias_level() ASoC: rt5663: Update the calibration funciton ASoC: rt5663: Optimize the Jack Type detection ASoC: rt5663: Modify the power sequence for reducing the pop sound ASoC: rt5663: Add the manual offset field to compensate the DC offset ASoC: rt5514: Support the DSP recording continuously after the hotwording triggered ASoC: rt5663: Modify the default value for ASRC function ASoC: rt5663: Correct the mixer switch setting and remove redundant routing path ASoC: rt5514: Support the TDM docking mode ASoC: rt5514: Add the I2S ASRC support ASoC: rt5514: Use the IS_ENABLED to supports the module build ASoC: rt5663: Add the delay time to correct the calibration ASoC: rt5514: Add the sanity checks of the buffer related address ASoC: rt5663: Seprate the DC offset between headphone and headset ASoC: rt5514: reset dma_offset at hw_params ASoC: rt5514: Eliminate the noise in the ASRC case ASoC: rt5663: Fine tune for the headphone output pop sound ASoC: rt5663: Add delay for jack plug in .../devicetree/bindings/sound/cs43130.txt | 67 + Documentation/devicetree/bindings/sound/dmic.txt | 16 + .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 4 +- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 18 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 68 +- .../devicetree/bindings/sound/rockchip,pdm.txt | 7 +- .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 +- .../devicetree/bindings/sound/rockchip-i2s.txt | 4 + Documentation/devicetree/bindings/sound/rt274.txt | 33 + Documentation/devicetree/bindings/sound/rt5663.txt | 8 + .../devicetree/bindings/sound/samsung,odroid.txt | 6 - .../devicetree/bindings/sound/simple-card.txt | 3 + .../devicetree/bindings/sound/simple-scu-card.txt | 1 + .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../devicetree/bindings/sound/tlv320aic32x4.txt | 13 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 5 + Documentation/devicetree/bindings/sound/wm8524.txt | 16 + include/sound/rt5663.h | 22 + include/sound/rt5677.h | 45 - include/sound/simple_card_utils.h | 1 + include/sound/soc.h | 53 +- include/sound/tlv320aic32x4.h | 23 + include/uapi/sound/snd_sst_tokens.h | 92 +- sound/atmel/ac97c.c | 9 +- sound/soc/atmel/atmel-classd.c | 1 - sound/soc/atmel/atmel-pdmic.c | 1 - sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 237 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/blackfin/bf6xx-i2s.c | 2 +- sound/soc/blackfin/bf6xx-sport.c | 38 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/Kconfig | 20 +- sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adau1977.c | 8 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ads117x.c | 2 +- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4554.c | 2 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/ak5386.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/arizona.c | 4 +- sound/soc/codecs/bt-sco.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs35l33.c | 14 +- sound/soc/codecs/cs35l34.c | 5 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l42.c | 11 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs43130.c | 2690 ++++++++++++++++++++ sound/soc/codecs/cs43130.h | 546 ++++ sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs47l24.c | 6 +- sound/soc/codecs/cs53l30.c | 17 +- sound/soc/codecs/cx20442.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/dmic.c | 54 +- sound/soc/codecs/es7134.c | 2 +- sound/soc/codecs/es8316.c | 4 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 43 +- sound/soc/codecs/hdmi-codec.c | 47 +- sound/soc/codecs/ics43432.c | 2 +- sound/soc/codecs/inno_rk3036.c | 4 +- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max98357a.c | 2 +- sound/soc/codecs/max98371.c | 14 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/max9867.c | 4 +- sound/soc/codecs/max98926.c | 8 +- sound/soc/codecs/max98927.c | 87 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 415 ++- sound/soc/codecs/msm8916-wcd-digital.c | 39 +- sound/soc/codecs/nau8540.c | 2 +- sound/soc/codecs/nau8810.c | 2 +- sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 28 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/pcm179x.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/rt274.c | 1229 +++++++++ sound/soc/codecs/rt274.h | 217 ++ sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514-spi.c | 146 +- sound/soc/codecs/rt5514-spi.h | 7 +- sound/soc/codecs/rt5514.c | 174 +- sound/soc/codecs/rt5514.h | 22 + sound/soc/codecs/rt5616.c | 4 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 45 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 4 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 264 +- sound/soc/codecs/rt5663.h | 2 + sound/soc/codecs/rt5665.c | 55 +- sound/soc/codecs/rt5665.h | 21 + sound/soc/codecs/rt5670.c | 64 +- sound/soc/codecs/rt5677.c | 82 +- sound/soc/codecs/rt5677.h | 30 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 12 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/ssm4567.c | 2 +- sound/soc/codecs/sta32x.c | 3 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/tas2552.c | 13 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tas5720.c | 4 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tlv320aic32x4-i2c.c | 2 + sound/soc/codecs/tlv320aic32x4-spi.c | 2 + sound/soc/codecs/tlv320aic32x4.c | 208 +- sound/soc/codecs/tlv320aic32x4.h | 3 + sound/soc/codecs/tlv320aic3x.c | 47 +- sound/soc/codecs/tlv320aic3x.h | 8 + sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl6040.c | 6 +- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm5110.c | 4 +- sound/soc/codecs/wm8523.c | 6 +- sound/soc/codecs/wm8524.c | 261 ++ sound/soc/codecs/wm8804.c | 3 +- sound/soc/codecs/zx_aud96p22.c | 6 +- sound/soc/davinci/davinci-mcasp.c | 6 +- sound/soc/davinci/davinci-vcif.c | 5 +- sound/soc/dwc/dwc-i2s.c | 6 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 6 +- sound/soc/fsl/fsl_dma.c | 11 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 4 +- sound/soc/fsl/imx-audmux.c | 16 +- sound/soc/fsl/imx-es8328.c | 3 - sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/mpc5200_dma.c | 4 +- sound/soc/generic/audio-graph-card.c | 3 +- sound/soc/generic/audio-graph-scu-card.c | 3 +- sound/soc/generic/simple-card-utils.c | 11 +- sound/soc/generic/simple-card.c | 10 - sound/soc/generic/simple-scu-card.c | 5 + sound/soc/hisilicon/hi6210-i2s.c | 2 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/img/img-parallel-out.c | 2 +- sound/soc/img/img-spdif-in.c | 2 +- sound/soc/img/img-spdif-out.c | 2 +- sound/soc/intel/Kconfig | 3 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 10 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 4 +- sound/soc/intel/atom/sst/sst_pci.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 86 +- sound/soc/intel/boards/byt-max98090.c | 12 - sound/soc/intel/boards/cht_bsw_rt5672.c | 10 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 310 ++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 81 +- sound/soc/intel/boards/mfld_machine.c | 4 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- sound/soc/intel/skylake/Makefile | 5 +- sound/soc/intel/skylake/bxt-sst.c | 14 +- sound/soc/intel/skylake/cnl-sst-dsp.c | 274 ++ sound/soc/intel/skylake/cnl-sst-dsp.h | 112 + sound/soc/intel/skylake/cnl-sst.c | 497 ++++ sound/soc/intel/skylake/skl-messages.c | 163 +- sound/soc/intel/skylake/skl-pcm.c | 82 +- sound/soc/intel/skylake/skl-sst-dsp.c | 6 +- sound/soc/intel/skylake/skl-sst-ipc.c | 6 +- sound/soc/intel/skylake/skl-sst-ipc.h | 12 +- sound/soc/intel/skylake/skl-sst-utils.c | 6 +- sound/soc/intel/skylake/skl-sst.c | 12 +- sound/soc/intel/skylake/skl-topology.c | 741 +++++- sound/soc/intel/skylake/skl-topology.h | 83 + sound/soc/intel/skylake/skl.c | 32 +- sound/soc/intel/skylake/skl.h | 3 + sound/soc/jz4740/jz4740-i2s.c | 21 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 11 +- sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 15 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650.c | 1 - sound/soc/mxs/mxs-saif.c | 13 +- sound/soc/mxs/mxs-sgtl5000.c | 1 - sound/soc/nuc900/nuc900-pcm.c | 4 +- sound/soc/omap/ams-delta.c | 10 - sound/soc/omap/omap-pcm.c | 4 +- sound/soc/omap/omap-twl4030.c | 13 - sound/soc/omap/rx51.c | 9 - sound/soc/pxa/hx4700.c | 8 - sound/soc/pxa/mmp-pcm.c | 4 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 4 +- sound/soc/qcom/apq8016_sbc.c | 58 +- sound/soc/qcom/lpass-platform.c | 4 +- sound/soc/qcom/storm.c | 1 - sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 270 +- sound/soc/rockchip/rockchip_i2s.c | 4 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/samsung/h1940_uda1380.c | 9 - sound/soc/samsung/i2s.c | 45 +- sound/soc/samsung/idma.c | 4 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/odroid.c | 44 +- sound/soc/samsung/pcm.c | 8 +- sound/soc/samsung/rx1950_uda1380.c | 10 - sound/soc/samsung/s3c-i2s-v2.c | 14 +- sound/soc/samsung/s3c-i2s-v2.h | 7 + sound/soc/samsung/s3c2412-i2s.c | 15 +- sound/soc/samsung/s3c24xx-i2s.c | 11 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 3 +- sound/soc/samsung/smartq_wm8987.c | 9 - sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/spdif.c | 8 +- sound/soc/samsung/tm2_wm5110.c | 4 +- sound/soc/sh/dma-sh7760.c | 6 +- sound/soc/sh/fsi.c | 12 +- sound/soc/sh/migor.c | 2 +- sound/soc/sh/rcar/adg.c | 11 +- sound/soc/sh/rcar/core.c | 109 +- sound/soc/sh/rcar/ctu.c | 5 +- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/sh/rcar/dvc.c | 5 +- sound/soc/sh/rcar/gen.c | 4 +- sound/soc/sh/rcar/mix.c | 5 +- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/src.c | 14 +- sound/soc/sh/rcar/ssi.c | 141 +- sound/soc/sh/rcar/ssiu.c | 2 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sh/siu_pcm.c | 2 +- sound/soc/soc-compress.c | 23 +- sound/soc/soc-core.c | 321 ++- sound/soc/soc-jack.c | 76 +- sound/soc/soc-pcm.c | 43 +- sound/soc/soc-utils.c | 2 +- sound/soc/spear/spdif_in.c | 12 +- sound/soc/spear/spdif_out.c | 4 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 15 +- sound/soc/sunxi/sun4i-i2s.c | 455 +++- sound/soc/sunxi/sun4i-spdif.c | 14 +- sound/soc/sunxi/sun8i-codec.c | 4 +- sound/soc/tegra/Kconfig | 42 +- sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 14 - sound/soc/tegra/tegra_max98090.c | 19 - sound/soc/tegra/tegra_rt5640.c | 14 - sound/soc/tegra/tegra_rt5677.c | 19 - sound/soc/tegra/tegra_sgtl5000.c | 1 - sound/soc/tegra/tegra_wm8753.c | 1 - sound/soc/tegra/tegra_wm8903.c | 7 - sound/soc/tegra/tegra_wm9712.c | 1 - sound/soc/tegra/trimslice.c | 1 - sound/soc/txx9/txx9aclc.c | 4 +- sound/soc/ux500/mop500.c | 1 - sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/zte/zx-i2s.c | 2 +- sound/soc/zte/zx-spdif.c | 2 +- sound/soc/zte/zx-tdm.c | 2 +- 319 files changed, 10887 insertions(+), 1921 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs43130.txt create mode 100644 Documentation/devicetree/bindings/sound/dmic.txt create mode 100644 Documentation/devicetree/bindings/sound/rt274.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8524.txt create mode 100644 include/sound/rt5663.h delete mode 100644 include/sound/rt5677.h create mode 100644 sound/soc/codecs/cs43130.c create mode 100644 sound/soc/codecs/cs43130.h create mode 100644 sound/soc/codecs/rt274.c create mode 100644 sound/soc/codecs/rt274.h create mode 100644 sound/soc/codecs/wm8524.c create mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h create mode 100644 sound/soc/intel/skylake/cnl-sst.c

2 1

[alsa-devel] [PATCH] ASoC: samsung: i2s: Fix error handling path in i2s_set_sysclk()
by Christophe JAILLET 04 Sep '17

04 Sep '17

If 'clk_prepare_enable()' fails, we must 'put' the corresponding clock. Othewise, there is a resource leak. Fixes: f5c97c7b0438 ("ASoC: samsung: i2s: Handle return value of clk_prepare_enable.") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- I'm not 100% confident in this patch. It is mostly inspired by surrounding error handling. (L533 and L550) --- sound/soc/samsung/i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 10a4da06c0a1..d7e7f4244d38 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -552,8 +552,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, } ret = clk_prepare_enable(i2s->op_clk); - if (ret) + if (ret) { + clk_put(i2s->op_clk); + i2s->op_clk = NULL; goto err; + } i2s->rclk_srcrate = clk_get_rate(i2s->op_clk); /* Over-ride the other's */ -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_spdif: make const arrays rate static
by Colin King 04 Sep '17

04 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the const arrays rate on the stack, instead make them static. Makes the object code smaller by 220 bytes: Before: text data bss dec hex filename 24385 9776 128 34289 85f1 sound/soc/fsl/fsl_spdif.o After: text data bss dec hex filename 24005 9936 128 34069 8515 sound/soc/fsl/fsl_spdif.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 7e6cc4da0088..4f7469c1864c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1110,7 +1110,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, struct clk *clk, u64 savesub, enum spdif_txrate index, bool round) { - const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; + static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; u32 sysclk_dfmin, sysclk_dfmax; @@ -1169,7 +1169,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index) { - const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; + static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; struct platform_device *pdev = spdif_priv->pdev; struct device *dev = &pdev->dev; u64 savesub = 100000, ret; -- 2.14.1

2 1

[alsa-devel] [PATCH][sound-next] ASoC: cs43130: fix spelling mistake: "irq_occurrance" -> "irq_occurrence"
by Colin King 04 Sep '17

04 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in variable name Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/cs43130.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 220e30199c5b..9f1aa0516fd0 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2147,7 +2147,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) struct cs43130_private *cs43130 = (struct cs43130_private *)data; struct snd_soc_codec *codec = cs43130->codec; unsigned int stickies[CS43130_NUM_INT]; - unsigned int irq_occurrance = 0; + unsigned int irq_occurrence = 0; unsigned int masks[CS43130_NUM_INT]; int i, j; @@ -2161,12 +2161,12 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) for (i = 0; i < ARRAY_SIZE(stickies); i++) { stickies[i] = stickies[i] & (~masks[i]); for (j = 0; j < 8; j++) - irq_occurrance += (stickies[i] >> j) & 1; + irq_occurrence += (stickies[i] >> j) & 1; } dev_dbg(codec->dev, "number of interrupts occurred (%u)\n", - irq_occurrance); + irq_occurrence); - if (!irq_occurrance) + if (!irq_occurrence) return IRQ_NONE; if (stickies[0] & CS43130_XTAL_RDY_INT) { -- 2.14.1

2 1

[alsa-devel] [PATCH v2 1/2] ALSA: atmel: convert AC97c driver to GPIO descriptor API
by Andy Shevchenko 04 Sep '17

04 Sep '17

Convert the driver to use GPIO descriptor API. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/atmel/ac97c.c | 70 ++++++++--------------------------------------------- 1 file changed, 10 insertions(+), 60 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 5ffefac2fa8f..380025887aef 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -12,16 +12,15 @@ #include <linux/bitmap.h> #include <linux/device.h> #include <linux/atmel_pdc.h> +#include <linux/gpio/consumer.h> #include <linux/init.h> #include <linux/interrupt.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/mutex.h> -#include <linux/gpio.h> #include <linux/types.h> #include <linux/io.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/of_device.h> #include <sound/core.h> @@ -29,7 +28,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/ac97_codec.h> -#include <sound/atmel-ac97c.h> #include <sound/memalloc.h> #include "ac97c.h" @@ -56,7 +54,7 @@ struct atmel_ac97c { void __iomem *regs; int irq; int opened; - int reset_pin; + struct gpio_desc *reset_pin; }; #define get_chip(card) ((struct atmel_ac97c *)(card)->private_data) @@ -700,11 +698,11 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) ac97c_writel(chip, CAMR, 0); ac97c_writel(chip, COMR, 0); - if (gpio_is_valid(chip->reset_pin)) { - gpio_set_value(chip->reset_pin, 0); + if (!IS_ERR(chip->reset_pin)) { + gpiod_set_value(chip->reset_pin, 0); /* AC97 v2.2 specifications says minimum 1 us. */ udelay(2); - gpio_set_value(chip->reset_pin, 1); + gpiod_set_value(chip->reset_pin, 1); } else { ac97c_writel(chip, MR, AC97C_MR_WRST | AC97C_MR_ENA); udelay(2); @@ -712,45 +710,18 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) } } -#ifdef CONFIG_OF static const struct of_device_id atmel_ac97c_dt_ids[] = { { .compatible = "atmel,at91sam9263-ac97c", }, { } }; MODULE_DEVICE_TABLE(of, atmel_ac97c_dt_ids); -static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) -{ - struct ac97c_platform_data *pdata; - struct device_node *node = dev->of_node; - - if (!node) { - dev_err(dev, "Device does not have associated DT data\n"); - return ERR_PTR(-EINVAL); - } - - pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); - if (!pdata) - return ERR_PTR(-ENOMEM); - - pdata->reset_pin = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); - - return pdata; -} -#else -static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) -{ - dev_err(dev, "no platform data defined\n"); - return ERR_PTR(-ENXIO); -} -#endif - static int atmel_ac97c_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct snd_card *card; struct atmel_ac97c *chip; struct resource *regs; - struct ac97c_platform_data *pdata; struct clk *pclk; static struct snd_ac97_bus_ops ops = { .write = atmel_ac97c_write, @@ -765,13 +736,6 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pdata = dev_get_platdata(&pdev->dev); - if (!pdata) { - pdata = atmel_ac97c_probe_dt(&pdev->dev); - if (IS_ERR(pdata)) - return PTR_ERR(pdata); - } - irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_dbg(&pdev->dev, "could not get irq: %d\n", irq); @@ -821,17 +785,9 @@ static int atmel_ac97c_probe(struct platform_device *pdev) goto err_ioremap; } - if (gpio_is_valid(pdata->reset_pin)) { - if (gpio_request(pdata->reset_pin, "reset_pin")) { - dev_dbg(&pdev->dev, "reset pin not available\n"); - chip->reset_pin = -ENODEV; - } else { - gpio_direction_output(pdata->reset_pin, 1); - chip->reset_pin = pdata->reset_pin; - } - } else { - chip->reset_pin = -EINVAL; - } + chip->reset_pin = devm_gpiod_get_index(dev, "ac97", 2, GPIOD_OUT_HIGH); + if (IS_ERR(chip->reset_pin)) + dev_dbg(dev, "reset pin not available\n"); atmel_ac97c_reset(chip); @@ -871,9 +827,6 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return 0; err_ac97_bus: - if (gpio_is_valid(chip->reset_pin)) - gpio_free(chip->reset_pin); - iounmap(chip->regs); err_ioremap: free_irq(irq, chip); @@ -916,9 +869,6 @@ static int atmel_ac97c_remove(struct platform_device *pdev) struct snd_card *card = platform_get_drvdata(pdev); struct atmel_ac97c *chip = get_chip(card); - if (gpio_is_valid(chip->reset_pin)) - gpio_free(chip->reset_pin); - ac97c_writel(chip, CAMR, 0); ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); @@ -939,7 +889,7 @@ static struct platform_driver atmel_ac97c_driver = { .driver = { .name = "atmel_ac97c", .pm = ATMEL_AC97C_PM_OPS, - .of_match_table = of_match_ptr(atmel_ac97c_dt_ids), + .of_match_table = atmel_ac97c_dt_ids, }, }; module_platform_driver(atmel_ac97c_driver); -- 2.14.1

3 5

[alsa-devel] HBR support on DP]
by Subhransu S. Prusty 04 Sep '17

04 Sep '17

Hi Takashi/All, On skylake platforms HBR audio works fine through over HDMI1.4, but not over DP ports. Looking for information if anyone is aware of if HBR audio was ever enabled/tested over DP ports? Is there any specific setting to be done to play over DP using aplay? Regards, Subhransu --

2 2

[alsa-devel] [PATCH] ALSA: hda: Fix resource leak issue in snd_hda_codec_build_controls and snd_hda_codec_parse_pcms
by Wang YanQing 03 Sep '17

03 Sep '17

When patch_ops.init, patch_ops.build_pcms and patch_ops.build_controls return failure, we need to free resource with patch_ops.free, or we will get resource leak. Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index df6b57e..4e3e613 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2973,8 +2973,11 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) err = codec->patch_ops.init(codec); if (!err && codec->patch_ops.build_controls) err = codec->patch_ops.build_controls(codec); - if (err < 0) + if (err < 0) { + if (codec->patch_ops.free) + codec->patch_ops.free(codec); return err; + } /* we create chmaps here instead of build_pcms */ err = add_std_chmaps(codec); @@ -3170,6 +3173,8 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) if (err < 0) { codec_err(codec, "cannot build PCMs for #%d (error %d)\n", codec->core.addr, err); + if (codec->patch_ops.free) + codec->patch_ops.free(codec); return err; } -- 1.8.5.6.2.g3d8a54e.dirty

2 2

[alsa-devel] [PATCH] ALSA: hda: Fix forget to free resource in error handling code path in hda_codec_driver_probe
by Wang YanQing 03 Sep '17

03 Sep '17

When hda_codec_driver_probe meet error and return failure, we need to free resource with patch_ops.free, or we will get resource leak. Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_bind.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 6efadbf..f1b1b7e 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -123,6 +123,8 @@ static int hda_codec_driver_probe(struct device *dev) module_put(owner); error: + if (codec->patch_ops.free) + codec->patch_ops.free(codec); snd_hda_codec_cleanup_for_unbind(codec); return err; } -- 1.8.5.6.2.g3d8a54e.dirty

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix assign invalid fixed slots pcm numbers to HDA_PCM_TYPE_HDMI in get_empty_pcm_device
by Wang YanQing 03 Sep '17

03 Sep '17

The length of pcm_dev_bits in hda_bus is SNDRV_PCM_DEVICES, and the definiton of SNDRV_PCM_DEVICES is: " #if defined(CONFIG_SND_DYNAMIC_MINORS) #define SNDRV_PCM_DEVICES (SNDRV_OS_MINORS-2) #else #define SNDRV_PCM_DEVICES 8 #endif " So the max valid fixed slots pcm number is 7, and the first non-fixed slot is 8. Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3db26c4..df6b57e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3118,7 +3118,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, - [HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 }, + [HDA_PCM_TYPE_HDMI] = { 3, 7, -1 }, [HDA_PCM_TYPE_MODEM] = { 6, -1 }, }; int i; @@ -3139,7 +3139,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) #ifdef CONFIG_SND_DYNAMIC_MINORS /* non-fixed slots starting from 10 */ - for (i = 10; i < 32; i++) { + for (i = 8; i < 32; i++) { if (!test_and_set_bit(i, bus->pcm_dev_bits)) return i; } -- 1.8.5.6.2.g3d8a54e.dirty

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix regression of hdmi eld control created based on invalid pcm
by Wang YanQing 03 Sep '17

03 Sep '17

Commit fb087eaaef72 ("ALSA: hda - hdmi eld control created based on pcm") forget to filter out invalid pcm numbers, if there is only one invalid pcm number, then this issue causes we create eld control for invalid pcm silently, but when there are more than one invalid pcm numbers, then this issue bring probe error looks like below dmesg: " kernel: [ 1.647283] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops 0xc2967540) kernel: [ 1.651192] snd_hda_intel 0000:00:03.0: Too many HDMI devices kernel: [ 1.651195] snd_hda_intel 0000:00:03.0: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y kernel: [ 1.651197] snd_hda_intel 0000:00:03.0: Too many HDMI devices kernel: [ 1.651199] snd_hda_intel 0000:00:03.0: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y kernel: [ 1.651201] snd_hda_intel 0000:00:03.0: Too many HDMI devices kernel: [ 1.651203] snd_hda_intel 0000:00:03.0: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y kernel: [ 1.651676] snd_hda_intel 0000:00:03.0: control 3:0:0:ELD:0 is already present kernel: [ 1.651787] snd_hda_codec_hdmi: probe of hdaudioC0D0 failed with error -16 " This patch add invalid pcm number filter before calling hdmi_create_eld_ctl. Fixes: fb087eaaef72 ("ALSA: hda - hdmi eld control created based on pcm") Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 4 +++- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 14 ++++++++------ 3 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 821aad3..3db26c4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3213,8 +3213,10 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) continue; /* no substreams assigned */ dev = get_empty_pcm_device(bus, cpcm->pcm_type); - if (dev < 0) + if (dev < 0) { + cpcm->device = SNDRV_PCM_INVALID_DEVICE; continue; /* no fatal error */ + } cpcm->device = dev; err = snd_hda_attach_pcm_stream(bus, codec, cpcm); if (err < 0) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 60ce1cf..681c360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -164,6 +164,7 @@ enum { HDA_PCM_NTYPES }; +#define SNDRV_PCM_INVALID_DEVICE (-1) /* for PCM creation */ struct hda_pcm { char *name; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 53f9311..2b64fab 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2126,7 +2126,7 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - int err; + int dev, err; int pin_idx, pcm_idx; @@ -2154,11 +2154,13 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) return err; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - /* add control for ELD Bytes */ - err = hdmi_create_eld_ctl(codec, pcm_idx, - get_pcm_rec(spec, pcm_idx)->device); - if (err < 0) - return err; + dev = get_pcm_rec(spec, pcm_idx)->device; + if (dev != SNDRV_PCM_INVALID_DEVICE) { + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, pcm_idx, dev); + if (err < 0) + return err; + } } for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { -- 1.8.5.6.2.g3d8a54e.dirty

2 1

[alsa-devel] Raspberry PI reports no channels, period and buffer
by Petr Kulhavy 03 Sep '17

03 Sep '17

Hi, I'm writing an audio decoding program for ALSA, which among others detects audio capabilities. It uses the snd_pcm_hw_params_get_XXX() functions for that. This works well on my Ubuntu PC but on Raspberry PI 3 these calls return no information. They return success but the min and max values are meaningless. For instance the period: err = snd_pcm_hw_params_get_period_size_min(hw_params, &m_period_min, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get minimum period time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_period_size_max(hw_params, &m_period_max, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get maximum period time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_period_time_min(hw_params, &m_period_min_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get minimum period time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_period_time_max(hw_params, &m_period_max_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get maximum period time: %s\n", snd_strerror(err)); goto fail; } LOG(LOG_DEBUG, LOG_OBJ "Supported period: %lu to %lu frames (%u to %u us)\n", (unsigned long)m_period_min, (unsigned long)m_period_max, m_period_min_us, m_period_max_us); This prints: Supported period: 0 to 0 frames (1 to 0 us) // query buffer size err = snd_pcm_hw_params_get_buffer_size_min(hw_params, &m_buffer_min); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get minimum buffer time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_buffer_size_max(hw_params, &m_buffer_max); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get maximum buffer time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_buffer_time_min(hw_params, &m_buffer_min_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get minimum buffer time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_buffer_time_max(hw_params, &m_buffer_max_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get maximum buffer time: %s\n", snd_strerror(err)); goto fail; } LOG(LOG_DEBUG, LOG_OBJ "Supported buffer size: %lu to %lu frames (%u to %u us)\n", (unsigned long)m_buffer_min, (unsigned long)m_buffer_max, m_buffer_min_us, m_buffer_max_us); This prints: Supported buffer size: 0 to 0 frames (1 to 0 us) And so on. Same for the number of periods and channels. On the other hand the snd_pcm_hw_params_test_XXX() calls do work and that way I can detect the rates, formats, accesses. Why is that? Am I doing something wrong? And more importantly, how can I get the buffering and channel information? Thanks Petr

2 1

[alsa-devel] [patch 1/4] ASoC: dapm: fix error path in snd_soc_dapm_new_pcm
by anish kumar 02 Sep '17

02 Sep '17

Before freeing the memory, check if it was allcoated. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..6e10a9a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3907,9 +3907,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); -outfree_w_param: for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); + if (w_param_text[count]) + devm_kfree(card->dev, (void *)w_param_text[count]); +outfree_w_param: devm_kfree(card->dev, w_param_text); return ret; -- 2.10.1 (Apple Git-78)

2 2

[alsa-devel] [PATCH 1/9] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 02 Sep '17

02 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 141 ++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98927.h | 6 +- 2 files changed, 120 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b0380b5..951cc05 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +236,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + } return 0; } @@ -245,6 +252,21 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,19 +292,17 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: + if ((max98927->iface == SND_SOC_DAIFMT_DSP_A) || + (max98927->iface == SND_SOC_DAIFMT_DSP_B)) + return 0; + + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "BCLK %d not supported\n", blr_clk_ratio); return -EINVAL; } + regmap_update_bits(max98927->regmap, MAX98927_R0022_PCM_CLK_SETUP, MAX98927_PCM_CLK_SETUP_BSEL_MASK, @@ -386,6 +406,76 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +495,7 @@ static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..bf7a6f92 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 1.9.1

2 14

[alsa-devel] [PATCH 1/3] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 02 Sep '17

02 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changelog: Added support for DSP_A and DSP_B format Added 'max98927_dai_tdm_slot' function to configure for proper slot configuration. Moved max98927->iface out of switch statement to avoid line duplication. Added variable 'tdm_mode' to avoid BCLK overwrite after 'max98927_dai_tdm_slot' configure right BCLK value. sound/soc/codecs/max98927.c | 155 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/max98927.h | 7 +- 2 files changed, 129 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe7..a1d3935 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + } return 0; } @@ -245,6 +250,21 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,23 +290,20 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: - return -EINVAL; + if (!max98927->tdm_mode) { + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + value); } - regmap_update_bits(max98927->regmap, - MAX98927_R0022_PCM_CLK_SETUP, - MAX98927_PCM_CLK_SETUP_BSEL_MASK, - value); return 0; } @@ -386,6 +403,78 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + max98927->tdm_mode = true; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +494,7 @@ static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, @@ -414,6 +504,9 @@ static int max98927_dac_event(struct snd_soc_dapm_widget *w, struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + max98927->tdm_mode = 0; + break; case SND_SOC_DAPM_POST_PMU: regmap_update_bits(max98927->regmap, MAX98927_R003A_AMP_EN, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..9ea8397 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) @@ -268,5 +270,6 @@ struct max98927_priv { unsigned int iface; unsigned int master; unsigned int digital_gain; + bool tdm_mode; }; #endif -- 1.9.1

1 2

[alsa-devel] [PATCH v1] ALSA: atmel: Remove leftovers of AVR32 removal
by Andy Shevchenko 01 Sep '17

01 Sep '17

The ALSA related include header files are left overs after the commit 020c5260c2b1 ("ALSA: atmel: Remove AVR32 bits from the driver") Fixes: 020c5260c2b1 ("ALSA: atmel: Remove AVR32 bits from the driver") Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- include/sound/atmel-abdac.h | 23 ----------------------- include/sound/atmel-ac97c.h | 38 -------------------------------------- sound/atmel/ac97c.c | 1 - 3 files changed, 62 deletions(-) delete mode 100644 include/sound/atmel-abdac.h delete mode 100644 include/sound/atmel-ac97c.h diff --git a/include/sound/atmel-abdac.h b/include/sound/atmel-abdac.h deleted file mode 100644 index a8f735d677fa..000000000000 --- a/include/sound/atmel-abdac.h +++ /dev/null @@ -1,23 +0,0 @@ -/* - * Driver for the Atmel Audio Bitstream DAC (ABDAC) - * - * Copyright (C) 2009 Atmel Corporation - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License version 2 as published - * by the Free Software Foundation. - */ -#ifndef __INCLUDE_SOUND_ATMEL_ABDAC_H -#define __INCLUDE_SOUND_ATMEL_ABDAC_H - -#include <linux/platform_data/dma-dw.h> - -/** - * struct atmel_abdac_pdata - board specific ABDAC configuration - * @dws: DMA slave interface to use for sound playback. - */ -struct atmel_abdac_pdata { - struct dw_dma_slave dws; -}; - -#endif /* __INCLUDE_SOUND_ATMEL_ABDAC_H */ diff --git a/include/sound/atmel-ac97c.h b/include/sound/atmel-ac97c.h deleted file mode 100644 index f2a1cdc37661..000000000000 --- a/include/sound/atmel-ac97c.h +++ /dev/null @@ -1,38 +0,0 @@ -/* - * Driver for the Atmel AC97C controller - * - * Copyright (C) 2005-2009 Atmel Corporation - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License version 2 as published - * by the Free Software Foundation. - */ -#ifndef __INCLUDE_SOUND_ATMEL_AC97C_H -#define __INCLUDE_SOUND_ATMEL_AC97C_H - -#include <linux/platform_data/dma-dw.h> - -#define AC97C_CAPTURE 0x01 -#define AC97C_PLAYBACK 0x02 -#define AC97C_BOTH (AC97C_CAPTURE | AC97C_PLAYBACK) - -/** - * struct atmel_ac97c_pdata - board specific AC97C configuration - * @rx_dws: DMA slave interface to use for sound capture. - * @tx_dws: DMA slave interface to use for sound playback. - * @reset_pin: GPIO pin wired to the reset input on the external AC97 codec, - * optional to use, set to -ENODEV if not in use. AC97 layer will - * try to do a software reset of the external codec anyway. - * - * If the user do not want to use a DMA channel for playback or capture, i.e. - * only one feature is required on the board. The slave for playback or capture - * can be set to NULL. The AC97C driver will take use of this when setting up - * the sound streams. - */ -struct ac97c_platform_data { - struct dw_dma_slave rx_dws; - struct dw_dma_slave tx_dws; - int reset_pin; -}; - -#endif /* __INCLUDE_SOUND_ATMEL_AC97C_H */ diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 5ffefac2fa8f..6d085e1d1d0a 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -29,7 +29,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/ac97_codec.h> -#include <sound/atmel-ac97c.h> #include <sound/memalloc.h> #include "ac97c.h" -- 2.14.1

2 2

[alsa-devel] [asoc:topic/cs43130 1/2] sound/soc/codecs/cs43130.c:2089:18: warning: 'hpload_seq' may be used uninitialized in this function
by kbuild test robot 01 Sep '17

01 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/cs43130 head: 465c925695593b43695e2396f7279879a72275eb commit: 8f1e5bf9b4408e91942b94c76583eaf098f19382 [1/2] ASoC: cs43130: Add support for CS43130 codec config: i386-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 8f1e5bf9b4408e91942b94c76583eaf098f19382 # save the attached .config to linux build tree make ARCH=i386 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/codecs/cs43130.c: In function 'cs43130_imp_meas': >> sound/soc/codecs/cs43130.c:2089:18: warning: 'hpload_seq' may be used uninitialized in this function [-Wmaybe-uninitialized] hpload_seq[i].msk, ac_idx); ^ vim +/hpload_seq +2089 sound/soc/codecs/cs43130.c 2042 2043 static void cs43130_imp_meas(struct work_struct *wk) 2044 { 2045 unsigned int reg, seq_size; 2046 int i, ret, ac_idx; 2047 struct cs43130_private *cs43130; 2048 struct snd_soc_codec *codec; 2049 struct reg_sequences *hpload_seq; 2050 2051 cs43130 = container_of(wk, struct cs43130_private, work); 2052 codec = cs43130->codec; 2053 2054 if (!cs43130->mclk) 2055 return; 2056 2057 cs43130->hpload_done = false; 2058 2059 mutex_lock(&cs43130->clk_mutex); 2060 if (!cs43130->clk_req) { 2061 /* clk not in use */ 2062 cs43130_set_pll(codec, 0, 0, cs43130->mclk, CS43130_MCLK_22M); 2063 if (cs43130->pll_bypass) 2064 cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); 2065 else 2066 cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); 2067 } 2068 2069 cs43130->clk_req++; 2070 mutex_unlock(&cs43130->clk_mutex); 2071 2072 regmap_read(cs43130->regmap, CS43130_INT_STATUS_4, &reg); 2073 2074 switch (cs43130->dev_id) { 2075 case CS43130_CHIP_ID: 2076 hpload_seq = hpload_seq1; 2077 seq_size = ARRAY_SIZE(hpload_seq1); 2078 break; 2079 case CS43131_CHIP_ID: 2080 hpload_seq = hpload_seq2; 2081 seq_size = ARRAY_SIZE(hpload_seq2); 2082 } 2083 2084 i = 0; 2085 ac_idx = 0; 2086 while (i < seq_size) { 2087 ret = cs43130_hpload_proc(cs43130, hpload_seq[i].seq, 2088 hpload_seq[i].size, > 2089 hpload_seq[i].msk, ac_idx); 2090 if (ret < 0) 2091 goto exit; 2092 2093 cs43130_update_hpload(hpload_seq[i].msk, ac_idx, cs43130); 2094 2095 if (cs43130->ac_meas && 2096 hpload_seq[i].msk == CS43130_HPLOAD_AC_INT && 2097 ac_idx < CS43130_AC_FREQ - 1) { 2098 ac_idx++; 2099 } else { 2100 ac_idx = 0; 2101 i++; 2102 } 2103 } 2104 cs43130->hpload_done = true; 2105 2106 if (cs43130->hpload_dc[HP_LEFT] >= CS43130_LINEOUT_LOAD) 2107 snd_soc_jack_report(&cs43130->jack, CS43130_JACK_LINEOUT, 2108 CS43130_JACK_MASK); 2109 else 2110 snd_soc_jack_report(&cs43130->jack, CS43130_JACK_HEADPHONE, 2111 CS43130_JACK_MASK); 2112 2113 dev_dbg(codec->dev, "Set HP output control. DC threshold\n"); 2114 for (i = 0; i < CS43130_DC_THRESHOLD; i++) 2115 dev_dbg(codec->dev, "DC threshold[%d]: %u.\n", i, 2116 cs43130->dc_threshold[i]); 2117 2118 cs43130_set_hv(cs43130->regmap, cs43130->hpload_dc[HP_LEFT], 2119 cs43130->dc_threshold); 2120 2121 exit: 2122 switch (cs43130->dev_id) { 2123 case CS43130_CHIP_ID: 2124 cs43130_hpload_proc(cs43130, hp_dis_cal_seq, 2125 ARRAY_SIZE(hp_dis_cal_seq), 2126 CS43130_HPLOAD_OFF_INT, ac_idx); 2127 break; 2128 case CS43131_CHIP_ID: 2129 cs43130_hpload_proc(cs43130, hp_dis_cal_seq2, 2130 ARRAY_SIZE(hp_dis_cal_seq2), 2131 CS43130_HPLOAD_OFF_INT, ac_idx); 2132 } 2133 2134 regmap_multi_reg_write(cs43130->regmap, hp_cln_seq, 2135 ARRAY_SIZE(hp_cln_seq)); 2136 2137 mutex_lock(&cs43130->clk_mutex); 2138 cs43130->clk_req--; 2139 /* clk not in use */ 2140 if (!cs43130->clk_req) 2141 cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); 2142 mutex_unlock(&cs43130->clk_mutex); 2143 } 2144 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: add missing compile rule for max98371
by Kuninori Morimoto 01 Sep '17

01 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is still using old driver style, this patch also fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max98371.c | 14 ++++++++------ 2 files changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 12dc2f7..d93318e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -322,6 +322,7 @@ obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o +obj-$(CONFIG_SND_SOC_MAX98371) += snd-soc-max98371.o obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 781be9ba..7bc2a17 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -349,12 +349,14 @@ static int max98371_dai_hw_params(struct snd_pcm_substream *substream, }; static const struct snd_soc_codec_driver max98371_codec = { - .controls = max98371_snd_controls, - .num_controls = ARRAY_SIZE(max98371_snd_controls), - .dapm_routes = max98371_audio_map, - .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), - .dapm_widgets = max98371_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), + .component_driver = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), + }, }; static const struct regmap_config max98371_regmap = { -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Add IPC to configure the copier secondary pins
by Pradeep Tewani 01 Sep '17

01 Sep '17

Copier can support upto 4 output pins. However, only pin 0 is configured as a part of copier initialization. Configuring rest of pins require the separate IPC to be sent to fw. Signed-off-by: Pradeep Tewani <pradeep.d.tewani(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 53 ++++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.h | 5 ++++ 2 files changed, 53 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index ea83e8b..89f7013 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -1083,6 +1083,21 @@ int skl_unbind_modules(struct skl_sst *ctx, return ret; } +static void fill_pin_params(struct skl_audio_data_format *pin_fmt, + struct skl_module_fmt *format) +{ + pin_fmt->number_of_channels = format->channels; + pin_fmt->s_freq = format->s_freq; + pin_fmt->bit_depth = format->bit_depth; + pin_fmt->valid_bit_depth = format->valid_bit_depth; + pin_fmt->ch_cfg = format->ch_cfg; + pin_fmt->sample_type = format->sample_type; + pin_fmt->channel_map = format->ch_map; + pin_fmt->interleaving = format->interleaving_style; +} + +#define CPR_SINK_FMT_PARAM_ID 2 + /* * Once a module is instantiated it need to be 'bind' with other modules in * the pipeline. For binding we need to find the module pins which are bind @@ -1094,11 +1109,15 @@ int skl_bind_modules(struct skl_sst *ctx, struct skl_module_cfg *src_mcfg, struct skl_module_cfg *dst_mcfg) { - int ret; + int ret = 0; struct skl_ipc_bind_unbind_msg msg; int in_max = dst_mcfg->module->max_input_pins; int out_max = src_mcfg->module->max_output_pins; int src_index, dst_index; + struct skl_module_fmt *format; + struct skl_cpr_pin_fmt pin_fmt; + struct skl_module *module; + struct skl_module_iface *fmt; skl_dump_bind_info(ctx, src_mcfg, dst_mcfg); @@ -1117,6 +1136,29 @@ int skl_bind_modules(struct skl_sst *ctx, return -EINVAL; } + /* + * Copier module requires the separate large_config_set_ipc to + * configure the pins other than 0 + */ + if (src_mcfg->m_type == SKL_MODULE_TYPE_COPIER && src_index > 0) { + pin_fmt.sink_id = src_index; + module = src_mcfg->module; + fmt = &module->formats[src_mcfg->fmt_idx]; + + /* Input fmt is same as that of src module input cfg */ + format = &fmt->inputs[0].fmt; + fill_pin_params(&(pin_fmt.src_fmt), format); + + format = &fmt->outputs[src_index].fmt; + fill_pin_params(&(pin_fmt.dst_fmt), format); + ret = skl_set_module_params(ctx, (void *)&pin_fmt, + sizeof(struct skl_cpr_pin_fmt), + CPR_SINK_FMT_PARAM_ID, src_mcfg); + + if (ret < 0) + goto out; + } + msg.dst_queue = dst_index; dev_dbg(ctx->dev, "src queue = %d dst queue =%d\n", @@ -1134,11 +1176,12 @@ int skl_bind_modules(struct skl_sst *ctx, src_mcfg->m_state = SKL_MODULE_BIND_DONE; src_mcfg->m_out_pin[src_index].pin_state = SKL_PIN_BIND_DONE; dst_mcfg->m_in_pin[dst_index].pin_state = SKL_PIN_BIND_DONE; - } else { - /* error case , if IPC fails, clear the queue index */ - skl_free_queue(src_mcfg->m_out_pin, src_index); - skl_free_queue(dst_mcfg->m_in_pin, dst_index); + return ret; } +out: + /* error case , if IPC fails, clear the queue index */ + skl_free_queue(src_mcfg->m_out_pin, src_index); + skl_free_queue(dst_mcfg->m_in_pin, dst_index); return ret; } diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 4975904..2717db92 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -138,6 +138,11 @@ struct skl_cpr_cfg { struct skl_cpr_gtw_cfg gtw_cfg; } __packed; +struct skl_cpr_pin_fmt { + u32 sink_id; + struct skl_audio_data_format src_fmt; + struct skl_audio_data_format dst_fmt; +} __packed; struct skl_src_module_cfg { struct skl_base_cfg base_cfg; -- 2.7.4

3 2

[alsa-devel] [PATCH 0/6 v3] Add ASoC support for AMD Stoney APUs
by Alex Deucher 01 Sep '17

01 Sep '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. The current code is based on drm-next, but I'm happy to rebase on whatever tree this ends up going through if there are any problems applying. These patches touch both audio and drm. The entire patch set can be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=stoney_acp2 Thanks! Alex v2: - Patch 1 is already applied to the audio tree, just including it for completeness since it's required for this patch set and it's not yet in the drm tree. - New patch to share asic types between gpu and audio drivers - ACPI ID changed for rt5650 machine driver - Integrate feedback on other patches v3: - resend remaining patches that have not been applied yet Akshu Agrawal (2): drm/amdgpu Moving amdgpu asic types to a separate file ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (4): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 + drivers/gpu/drm/amd/include/amd_shared.h | 29 +--- include/drm/amd_asic_type.h | 52 +++++++ sound/soc/amd/Kconfig | 7 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 243 ++++++++++++++++++++++++------- sound/soc/amd/acp-rt5645.c | 210 ++++++++++++++++++++++++++ sound/soc/amd/acp.h | 9 ++ 8 files changed, 472 insertions(+), 82 deletions(-) create mode 100644 include/drm/amd_asic_type.h create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

5 18

[alsa-devel] [PATCH] asoc: make clock direction configurable in asoc-simple
by Vitaly Wool 01 Sep '17

01 Sep '17

Some CPU drivers (e. g. davinci-mcasp) may require the system clock to be configured as OUT, while there's no good way currently to set SND_SOC_CLK_OUT in simple-soc driver if the clock is fixed-rate. This patch makes asoc_simple_card_init_dai() initialize clock to SND_SOCK_CLK_OUT if explicitly stated in the relevant dts file. This change is transparent and doesn't change the default behavior. Signed-off-by: Vitaly Wool <vitaly.wool(a)konsulko.com> --- Documentation/devicetree/bindings/sound/simple-card.txt | 3 +++ include/sound/simple_card_utils.h | 1 + sound/soc/generic/simple-card-utils.c | 9 +++++++-- 3 files changed, 11 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c7a93931fad2..166f2290233b 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -86,6 +86,9 @@ Optional CPU/CODEC subnodes properties: in dai startup() and disabled with clk_disable_unprepare() in dai shutdown(). +- system-clock-direction-out : specifies clock direction as 'out' on + initialization. It is useful for some aCPUs with + fixed clocks. Example 1 - single DAI link: diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 42c6a6ac3ce6..7e25afce6566 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -15,6 +15,7 @@ struct asoc_simple_dai { const char *name; unsigned int sysclk; + int clk_direction; int slots; int slot_width; unsigned int tx_slot_mask; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 26d64fa40c9c..af44c8c52a19 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -196,7 +196,11 @@ int asoc_simple_card_parse_clk(struct device *dev, simple_dai->sysclk = clk_get_rate(clk); } - dev_dbg(dev, "%s : sysclk = %d\n", name, simple_dai->sysclk); + if (of_property_read_bool(node, "system-clock-direction-out")) + simple_dai->clk_direction = SND_SOC_CLOCK_OUT; + + dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name, + simple_dai->sysclk, simple_dai->clk_direction); return 0; } @@ -310,7 +314,8 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, int ret; if (simple_dai->sysclk) { - ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk, 0); + ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk, + simple_dai->clk_direction); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_sysclk error\n"); return ret; -- 2.11.0

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: cs43130: Add support for CS43130 codec
by Li Xu 01 Sep '17

01 Sep '17

Add support for Cirrus Logic CS43130 codec. Support: I2S/DSP PCM playback. DoP/DSD playback. HP detection and DC/AC impedance measurement. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs43130.c | 2690 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs43130.h | 546 +++++++++ 4 files changed, 3244 insertions(+) create mode 100644 sound/soc/codecs/cs43130.c create mode 100644 sound/soc/codecs/cs43130.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 010811e..1f65336 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS43130 if I2C select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CS53L30 if I2C @@ -488,6 +489,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS43130 HiFi DAC +config SND_SOC_CS43130 + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C + # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 12dc2f7..896ef5a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -53,6 +53,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs43130-objs := cs43130.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o @@ -292,6 +293,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c new file mode 100644 index 0000000..220e301 --- /dev/null +++ b/sound/soc/codecs/cs43130.c @@ -0,0 +1,2690 @@ +/* + * cs43130.c -- CS43130 ALSA Soc Audio driver + * + * Copyright 2017 Cirrus Logic, Inc. + * + * Authors: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_runtime.h> +#include <linux/of_irq.h> +#include <linux/completion.h> +#include <linux/mutex.h> +#include <linux/workqueue.h> +#include <sound/jack.h> + +#include "cs43130.h" + +static const struct reg_default cs43130_reg_defaults[] = { + {CS43130_SYS_CLK_CTL_1, 0x06}, + {CS43130_SP_SRATE, 0x01}, + {CS43130_SP_BITSIZE, 0x05}, + {CS43130_PAD_INT_CFG, 0x03}, + {CS43130_PWDN_CTL, 0xFE}, + {CS43130_CRYSTAL_SET, 0x04}, + {CS43130_PLL_SET_1, 0x00}, + {CS43130_PLL_SET_2, 0x00}, + {CS43130_PLL_SET_3, 0x00}, + {CS43130_PLL_SET_4, 0x00}, + {CS43130_PLL_SET_5, 0x40}, + {CS43130_PLL_SET_6, 0x10}, + {CS43130_PLL_SET_7, 0x80}, + {CS43130_PLL_SET_8, 0x03}, + {CS43130_PLL_SET_9, 0x02}, + {CS43130_PLL_SET_10, 0x02}, + {CS43130_CLKOUT_CTL, 0x00}, + {CS43130_ASP_NUM_1, 0x01}, + {CS43130_ASP_NUM_2, 0x00}, + {CS43130_ASP_DEN_1, 0x08}, + {CS43130_ASP_DEN_2, 0x00}, + {CS43130_ASP_LRCK_HI_TIME_1, 0x1F}, + {CS43130_ASP_LRCK_HI_TIME_2, 0x00}, + {CS43130_ASP_LRCK_PERIOD_1, 0x3F}, + {CS43130_ASP_LRCK_PERIOD_2, 0x00}, + {CS43130_ASP_CLOCK_CONF, 0x0C}, + {CS43130_ASP_FRAME_CONF, 0x0A}, + {CS43130_XSP_NUM_1, 0x01}, + {CS43130_XSP_NUM_2, 0x00}, + {CS43130_XSP_DEN_1, 0x02}, + {CS43130_XSP_DEN_2, 0x00}, + {CS43130_XSP_LRCK_HI_TIME_1, 0x1F}, + {CS43130_XSP_LRCK_HI_TIME_2, 0x00}, + {CS43130_XSP_LRCK_PERIOD_1, 0x3F}, + {CS43130_XSP_LRCK_PERIOD_2, 0x00}, + {CS43130_XSP_CLOCK_CONF, 0x0C}, + {CS43130_XSP_FRAME_CONF, 0x0A}, + {CS43130_ASP_CH_1_LOC, 0x00}, + {CS43130_ASP_CH_2_LOC, 0x00}, + {CS43130_ASP_CH_1_SZ_EN, 0x06}, + {CS43130_ASP_CH_2_SZ_EN, 0x0E}, + {CS43130_XSP_CH_1_LOC, 0x00}, + {CS43130_XSP_CH_2_LOC, 0x00}, + {CS43130_XSP_CH_1_SZ_EN, 0x06}, + {CS43130_XSP_CH_2_SZ_EN, 0x0E}, + {CS43130_DSD_VOL_B, 0x78}, + {CS43130_DSD_VOL_A, 0x78}, + {CS43130_DSD_PATH_CTL_1, 0xA8}, + {CS43130_DSD_INT_CFG, 0x00}, + {CS43130_DSD_PATH_CTL_2, 0x02}, + {CS43130_DSD_PCM_MIX_CTL, 0x00}, + {CS43130_DSD_PATH_CTL_3, 0x40}, + {CS43130_HP_OUT_CTL_1, 0x30}, + {CS43130_PCM_FILT_OPT, 0x02}, + {CS43130_PCM_VOL_B, 0x78}, + {CS43130_PCM_VOL_A, 0x78}, + {CS43130_PCM_PATH_CTL_1, 0xA8}, + {CS43130_PCM_PATH_CTL_2, 0x00}, + {CS43130_CLASS_H_CTL, 0x1E}, + {CS43130_HP_DETECT, 0x04}, + {CS43130_HP_LOAD_1, 0x00}, + {CS43130_HP_MEAS_LOAD_1, 0x00}, + {CS43130_HP_MEAS_LOAD_2, 0x00}, + {CS43130_INT_MASK_1, 0xFF}, + {CS43130_INT_MASK_2, 0xFF}, + {CS43130_INT_MASK_3, 0xFF}, + {CS43130_INT_MASK_4, 0xFF}, + {CS43130_INT_MASK_5, 0xFF}, +}; + +static bool cs43130_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + case CS43130_HP_DC_STAT_1 ... CS43130_HP_DC_STAT_2: + case CS43130_HP_AC_STAT_1 ... CS43130_HP_AC_STAT_2: + return true; + default: + return false; + } +} + +static bool cs43130_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1: + case CS43130_SP_SRATE ... CS43130_PAD_INT_CFG: + case CS43130_PWDN_CTL: + case CS43130_CRYSTAL_SET: + case CS43130_PLL_SET_1 ... CS43130_PLL_SET_5: + case CS43130_PLL_SET_6: + case CS43130_PLL_SET_7: + case CS43130_PLL_SET_8: + case CS43130_PLL_SET_9: + case CS43130_PLL_SET_10: + case CS43130_CLKOUT_CTL: + case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF: + case CS43130_XSP_NUM_1 ... CS43130_XSP_FRAME_CONF: + case CS43130_ASP_CH_1_LOC: + case CS43130_ASP_CH_2_LOC: + case CS43130_ASP_CH_1_SZ_EN: + case CS43130_ASP_CH_2_SZ_EN: + case CS43130_XSP_CH_1_LOC: + case CS43130_XSP_CH_2_LOC: + case CS43130_XSP_CH_1_SZ_EN: + case CS43130_XSP_CH_2_SZ_EN: + case CS43130_DSD_VOL_B ... CS43130_DSD_PATH_CTL_3: + case CS43130_HP_OUT_CTL_1: + case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2: + case CS43130_CLASS_H_CTL: + case CS43130_HP_DETECT: + case CS43130_HP_STATUS: + case CS43130_HP_LOAD_1: + case CS43130_HP_MEAS_LOAD_1: + case CS43130_HP_MEAS_LOAD_2: + case CS43130_HP_DC_STAT_1: + case CS43130_HP_DC_STAT_2: + case CS43130_HP_AC_STAT_1: + case CS43130_HP_AC_STAT_2: + case CS43130_HP_LOAD_STAT: + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5: + return true; + default: + return false; + } +} + +static bool cs43130_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +struct cs43130_pll_params { + unsigned int pll_in; + u8 sclk_prediv; + u8 pll_div_int; + u32 pll_div_frac; + u8 pll_mode; + u8 pll_divout; + unsigned int pll_out; + u8 pll_cal_ratio; +}; + +static const struct cs43130_pll_params pll_ratio_table[] = { + {9600000, 0x02, 0x49, 0x800000, 0x00, 0x08, 22579200, 151}, + {9600000, 0x02, 0x50, 0x000000, 0x00, 0x08, 24576000, 164}, + + {11289600, 0x02, 0X40, 0, 0x01, 0x08, 22579200, 128}, + {11289600, 0x02, 0x44, 0x06F700, 0x0, 0x08, 24576000, 139}, + + {12000000, 0x02, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120}, + {12000000, 0x02, 0x40, 0x000000, 0x00, 0x08, 24576000, 131}, + + {12288000, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118}, + {12288000, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128}, + + {13000000, 0x02, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111}, + {13000000, 0x02, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121}, + + {19200000, 0x03, 0x49, 0x800000, 0x00, 0x08, 22579200, 151}, + {19200000, 0x03, 0x50, 0x000000, 0x00, 0x08, 24576000, 164}, + + {22579200, 0, 0, 0, 0, 0, 22579200, 0}, + {22579200, 0x03, 0x44, 0x06F700, 0x00, 0x08, 24576000, 139}, + + {24000000, 0x03, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120}, + {24000000, 0x03, 0x40, 0x000000, 0x00, 0x08, 24576000, 131}, + + {24576000, 0x03, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118}, + {24576000, 0, 0, 0, 0, 0, 24576000, 0}, + + {26000000, 0x03, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111}, + {26000000, 0x03, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121}, +}; + +static const struct cs43130_pll_params *cs43130_get_pll_table( + unsigned int freq_in, unsigned int freq_out) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { + if (pll_ratio_table[i].pll_in == freq_in && + pll_ratio_table[i].pll_out == freq_out) + return &pll_ratio_table[i]; + } + + return NULL; +} + +static int cs43130_pll_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + const struct cs43130_pll_params *pll_entry; + + dev_dbg(codec->dev, "cs43130->mclk = %u, cs43130->mclk_int = %u\n", + cs43130->mclk, cs43130->mclk_int); + + pll_entry = cs43130_get_pll_table(cs43130->mclk, cs43130->mclk_int); + if (!pll_entry) + return -EINVAL; + + if (pll_entry->pll_cal_ratio == 0) { + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, 0); + + cs43130->pll_bypass = true; + return 0; + } + + cs43130->pll_bypass = false; + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_2, + CS43130_PLL_DIV_DATA_MASK, + pll_entry->pll_div_frac >> + CS43130_PLL_DIV_FRAC_0_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_3, + CS43130_PLL_DIV_DATA_MASK, + pll_entry->pll_div_frac >> + CS43130_PLL_DIV_FRAC_1_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_4, + CS43130_PLL_DIV_DATA_MASK, + pll_entry->pll_div_frac >> + CS43130_PLL_DIV_FRAC_2_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_PLL_SET_5, + pll_entry->pll_div_int); + regmap_write(cs43130->regmap, CS43130_PLL_SET_6, pll_entry->pll_divout); + regmap_write(cs43130->regmap, CS43130_PLL_SET_7, + pll_entry->pll_cal_ratio); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_8, + CS43130_PLL_MODE_MASK, + pll_entry->pll_mode << CS43130_PLL_MODE_SHIFT); + regmap_write(cs43130->regmap, CS43130_PLL_SET_9, + pll_entry->sclk_prediv); + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, 1); + + return 0; +} + +static int cs43130_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (freq_in) { + case 9600000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 19200000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs43130->mclk = freq_in; + break; + default: + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_out) { + case 22579200: + cs43130->mclk_int = freq_out; + break; + case 24576000: + cs43130->mclk_int = freq_out; + break; + default: + dev_err(codec->dev, + "unsupported pll output ref clock: %u\n", freq_out); + return -EINVAL; + } + + ret = cs43130_pll_config(codec); + dev_dbg(codec->dev, "cs43130->pll_bypass = %d", cs43130->pll_bypass); + return ret; +} + +static int cs43130_change_clksrc(struct snd_soc_codec *codec, + enum cs43130_mclk_src_sel src) +{ + int ret; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int mclk_int_decoded; + + if (src == cs43130->mclk_int_src) { + /* clk source has not changed */ + return 0; + } + + switch (cs43130->mclk_int) { + case CS43130_MCLK_22M: + mclk_int_decoded = CS43130_MCLK_22P5; + break; + case CS43130_MCLK_24M: + mclk_int_decoded = CS43130_MCLK_24P5; + break; + default: + dev_err(codec->dev, "Invalid MCLK INT freq: %u\n", cs43130->mclk_int); + return -EINVAL; + } + + switch (src) { + case CS43130_MCLK_SRC_EXT: + cs43130->pll_bypass = true; + cs43130->mclk_int_src = CS43130_MCLK_SRC_EXT; + if (cs43130->xtal_ibias == CS43130_XTAL_UNUSED) { + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 1 << CS43130_PDN_XTAL_SHIFT); + } else { + reinit_completion(&cs43130->xtal_rdy); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 0); + ret = wait_for_completion_timeout(&cs43130->xtal_rdy, + msecs_to_jiffies(100)); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, + 1 << CS43130_XTAL_RDY_INT_SHIFT); + if (ret == 0) { + dev_err(codec->dev, "Timeout waiting for XTAL_READY interrupt\n"); + return -ETIMEDOUT; + } + } + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, + src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, + mclk_int_decoded << CS43130_MCLK_INT_SHIFT); + usleep_range(150, 200); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, + 1 << CS43130_PDN_PLL_SHIFT); + break; + case CS43130_MCLK_SRC_PLL: + cs43130->pll_bypass = false; + cs43130->mclk_int_src = CS43130_MCLK_SRC_PLL; + if (cs43130->xtal_ibias == CS43130_XTAL_UNUSED) { + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 1 << CS43130_PDN_XTAL_SHIFT); + } else { + reinit_completion(&cs43130->xtal_rdy); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 0); + ret = wait_for_completion_timeout(&cs43130->xtal_rdy, + msecs_to_jiffies(100)); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT_MASK, + 1 << CS43130_XTAL_RDY_INT_SHIFT); + if (ret == 0) { + dev_err(codec->dev, "Timeout waiting for XTAL_READY interrupt\n"); + return -ETIMEDOUT; + } + } + + reinit_completion(&cs43130->pll_rdy); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_PLL_RDY_INT_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, 0); + ret = wait_for_completion_timeout(&cs43130->pll_rdy, + msecs_to_jiffies(100)); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_PLL_RDY_INT_MASK, + 1 << CS43130_PLL_RDY_INT_SHIFT); + if (ret == 0) { + dev_err(codec->dev, "Timeout waiting for PLL_READY interrupt\n"); + return -ETIMEDOUT; + } + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, + src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, + mclk_int_decoded << CS43130_MCLK_INT_SHIFT); + usleep_range(150, 200); + break; + case CS43130_MCLK_SRC_RCO: + cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, + src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, + CS43130_MCLK_22P5 << CS43130_MCLK_INT_SHIFT); + usleep_range(150, 200); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 1 << CS43130_PDN_XTAL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, + 1 << CS43130_PDN_PLL_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid MCLK source value\n"); + return -EINVAL; + } + + return 0; +} + +static const struct cs43130_bitwidth_map cs43130_bitwidth_table[] = { + {8, CS43130_SP_BIT_SIZE_8, CS43130_CH_BIT_SIZE_8}, + {16, CS43130_SP_BIT_SIZE_16, CS43130_CH_BIT_SIZE_16}, + {24, CS43130_SP_BIT_SIZE_24, CS43130_CH_BIT_SIZE_24}, + {32, CS43130_SP_BIT_SIZE_32, CS43130_CH_BIT_SIZE_32}, +}; + +static const struct cs43130_bitwidth_map *cs43130_get_bitwidth_table( + unsigned int bitwidth) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs43130_bitwidth_table); i++) { + if (cs43130_bitwidth_table[i].bitwidth == bitwidth) + return &cs43130_bitwidth_table[i]; + } + + return NULL; +} + +static int cs43130_set_bitwidth(int dai_id, unsigned int bitwidth_dai, + struct regmap *regmap) +{ + const struct cs43130_bitwidth_map *bw_map; + + bw_map = cs43130_get_bitwidth_table(bitwidth_dai); + if (!bw_map) + return -EINVAL; + + switch (dai_id) { + case CS43130_ASP_PCM_DAI: + case CS43130_ASP_DOP_DAI: + regmap_update_bits(regmap, CS43130_ASP_CH_1_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_ASP_CH_2_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_SP_BITSIZE, + CS43130_ASP_BITSIZE_MASK, bw_map->sp_bit); + break; + case CS43130_XSP_DOP_DAI: + regmap_update_bits(regmap, CS43130_XSP_CH_1_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_XSP_CH_2_SZ_EN, + CS43130_CH_BITSIZE_MASK, bw_map->ch_bit); + regmap_update_bits(regmap, CS43130_SP_BITSIZE, + CS43130_XSP_BITSIZE_MASK, bw_map->sp_bit << + CS43130_XSP_BITSIZE_SHIFT); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct cs43130_rate_map cs43130_rate_table[] = { + {32000, CS43130_ASP_SPRATE_32K}, + {44100, CS43130_ASP_SPRATE_44_1K}, + {48000, CS43130_ASP_SPRATE_48K}, + {88200, CS43130_ASP_SPRATE_88_2K}, + {96000, CS43130_ASP_SPRATE_96K}, + {176400, CS43130_ASP_SPRATE_176_4K}, + {192000, CS43130_ASP_SPRATE_192K}, + {352800, CS43130_ASP_SPRATE_352_8K}, + {384000, CS43130_ASP_SPRATE_384K}, +}; + +static const struct cs43130_rate_map *cs43130_get_rate_table(int fs) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs43130_rate_table); i++) { + if (cs43130_rate_table[i].fs == fs) + return &cs43130_rate_table[i]; + } + + return NULL; +} + +static const struct cs43130_clk_gen *cs43130_get_clk_gen(int mclk_int, int fs, + const struct cs43130_clk_gen *clk_gen_table, int len_clk_gen_table) +{ + int i; + + for (i = 0; i < len_clk_gen_table; i++) { + if (clk_gen_table[i].mclk_int == mclk_int && + clk_gen_table[i].fs == fs) + return &clk_gen_table[i]; + } + + return NULL; +} + +static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, + struct snd_pcm_hw_params *params, + struct cs43130_private *cs43130) +{ + u16 frm_size; + u16 hi_size; + u8 frm_delay; + u8 frm_phase; + u8 frm_data; + u8 sclk_edge; + u8 lrck_edge; + u8 clk_data; + u8 loc_ch1; + u8 loc_ch2; + u8 dai_mode_val; + const struct cs43130_clk_gen *clk_gen; + + switch (cs43130->dais[dai_id].dai_format) { + case SND_SOC_DAIFMT_I2S: + hi_size = bitwidth_sclk; + frm_delay = 2; + frm_phase = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + hi_size = bitwidth_sclk; + frm_delay = 2; + frm_phase = 1; + break; + case SND_SOC_DAIFMT_DSP_A: + hi_size = 1; + frm_delay = 2; + frm_phase = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + hi_size = 1; + frm_delay = 0; + frm_phase = 1; + break; + default: + return -EINVAL; + } + + switch (cs43130->dais[dai_id].dai_mode) { + case SND_SOC_DAIFMT_CBS_CFS: + dai_mode_val = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + dai_mode_val = 1; + break; + default: + return -EINVAL; + } + + frm_size = bitwidth_sclk * params_channels(params); + sclk_edge = 1; + lrck_edge = 0; + loc_ch1 = 0; + loc_ch2 = bitwidth_sclk * (params_channels(params) - 1); + + frm_data = frm_delay & CS43130_SP_FSD_MASK; + frm_data |= (frm_phase << CS43130_SP_STP_SHIFT) & CS43130_SP_STP_MASK; + + clk_data = lrck_edge & CS43130_SP_LCPOL_IN_MASK; + clk_data |= (lrck_edge << CS43130_SP_LCPOL_OUT_SHIFT) & + CS43130_SP_LCPOL_OUT_MASK; + clk_data |= (sclk_edge << CS43130_SP_SCPOL_IN_SHIFT) & + CS43130_SP_SCPOL_IN_MASK; + clk_data |= (sclk_edge << CS43130_SP_SCPOL_OUT_SHIFT) & + CS43130_SP_SCPOL_OUT_MASK; + clk_data |= (dai_mode_val << CS43130_SP_MODE_SHIFT) & + CS43130_SP_MODE_MASK; + + switch (dai_id) { + case CS43130_ASP_PCM_DAI: + case CS43130_ASP_DOP_DAI: + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_1, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_2, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_MSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_1, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_2, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_FRAME_CONF, frm_data); + regmap_write(cs43130->regmap, CS43130_ASP_CH_1_LOC, loc_ch1); + regmap_write(cs43130->regmap, CS43130_ASP_CH_2_LOC, loc_ch2); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_1_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_2_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_CLOCK_CONF, clk_data); + break; + case CS43130_XSP_DOP_DAI: + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_PERIOD_1, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_PERIOD_2, + CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >> + CS43130_SP_LCPR_MSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_HI_TIME_1, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_LSB_DATA_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_HI_TIME_2, + CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >> + CS43130_SP_LCHI_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_FRAME_CONF, frm_data); + regmap_write(cs43130->regmap, CS43130_XSP_CH_1_LOC, loc_ch1); + regmap_write(cs43130->regmap, CS43130_XSP_CH_2_LOC, loc_ch2); + regmap_update_bits(cs43130->regmap, CS43130_XSP_CH_1_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_XSP_CH_2_SZ_EN, + CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_CLOCK_CONF, clk_data); + break; + default: + return -EINVAL; + } + + switch (frm_size) { + case 16: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_16_clk_gen, + ARRAY_SIZE(cs43130_16_clk_gen)); + break; + case 32: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_32_clk_gen, + ARRAY_SIZE(cs43130_32_clk_gen)); + break; + case 48: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_48_clk_gen, + ARRAY_SIZE(cs43130_48_clk_gen)); + break; + case 64: + clk_gen = cs43130_get_clk_gen(cs43130->mclk_int, + params_rate(params), + cs43130_64_clk_gen, + ARRAY_SIZE(cs43130_64_clk_gen)); + break; + default: + return -EINVAL; + } + + if (!clk_gen) + return -EINVAL; + + switch (dai_id) { + case CS43130_ASP_PCM_DAI: + case CS43130_ASP_DOP_DAI: + regmap_write(cs43130->regmap, CS43130_ASP_DEN_1, + (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >> + CS43130_SP_M_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_DEN_2, + (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >> + CS43130_SP_M_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_NUM_1, + (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >> + CS43130_SP_N_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_ASP_NUM_2, + (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >> + CS43130_SP_N_MSB_DATA_SHIFT); + break; + case CS43130_XSP_DOP_DAI: + regmap_write(cs43130->regmap, CS43130_XSP_DEN_1, + (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >> + CS43130_SP_M_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_DEN_2, + (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >> + CS43130_SP_M_MSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_NUM_1, + (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >> + CS43130_SP_N_LSB_DATA_SHIFT); + regmap_write(cs43130->regmap, CS43130_XSP_NUM_2, + (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >> + CS43130_SP_N_MSB_DATA_SHIFT); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs43130_pcm_dsd_mix(bool en, struct regmap *regmap) +{ + if (en) { + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_PREP_MASK, + 1 << CS43130_MIX_PCM_PREP_SHIFT); + usleep_range(6000, 6050); + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_DSD_MASK, + 1 << CS43130_MIX_PCM_DSD_SHIFT); + } else { + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_DSD_MASK, + 0 << CS43130_MIX_PCM_DSD_SHIFT); + usleep_range(1600, 1650); + regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL, + CS43130_MIX_PCM_PREP_MASK, + 0 << CS43130_MIX_PCM_PREP_SHIFT); + } + + return 0; +} + +static int cs43130_dsd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + unsigned int required_clk; + u8 dsd_speed; + + mutex_lock(&cs43130->clk_mutex); + if (!cs43130->clk_req) { + /* no DAI is currently using clk */ + if (!(CS43130_MCLK_22M % params_rate(params))) + required_clk = CS43130_MCLK_22M; + else + required_clk = CS43130_MCLK_24M; + + cs43130_set_pll(codec, 0, 0, cs43130->mclk, required_clk); + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + } + + cs43130->clk_req++; + if (cs43130->clk_req == 2) + cs43130_pcm_dsd_mix(true, cs43130->regmap); + mutex_unlock(&cs43130->clk_mutex); + + switch (params_rate(params)) { + case 176400: + dsd_speed = 0; + break; + case 352800: + dsd_speed = 1; + break; + default: + dev_err(codec->dev, "Rate(%u) not supported\n", + params_rate(params)); + return -EINVAL; + } + + if (cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM) + regmap_update_bits(cs43130->regmap, CS43130_DSD_INT_CFG, + CS43130_DSD_MASTER, CS43130_DSD_MASTER); + else + regmap_update_bits(cs43130->regmap, CS43130_DSD_INT_CFG, + CS43130_DSD_MASTER, 0); + + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SPEED_MASK, + dsd_speed << CS43130_DSD_SPEED_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_DSD << + CS43130_DSD_SRC_SHIFT); + + return 0; +} + +static int cs43130_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + const struct cs43130_rate_map *rate_map; + unsigned int sclk = cs43130->dais[dai->id].sclk; + unsigned int bitwidth_sclk; + unsigned int bitwidth_dai = (unsigned int)(params_width(params)); + unsigned int required_clk; + u8 dsd_speed; + + mutex_lock(&cs43130->clk_mutex); + if (!cs43130->clk_req) { + /* no DAI is currently using clk */ + if (!(CS43130_MCLK_22M % params_rate(params))) + required_clk = CS43130_MCLK_22M; + else + required_clk = CS43130_MCLK_24M; + + cs43130_set_pll(codec, 0, 0, cs43130->mclk, required_clk); + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + } + + cs43130->clk_req++; + if (cs43130->clk_req == 2) + cs43130_pcm_dsd_mix(true, cs43130->regmap); + mutex_unlock(&cs43130->clk_mutex); + + switch (dai->id) { + case CS43130_ASP_DOP_DAI: + case CS43130_XSP_DOP_DAI: + /* DoP bitwidth is always 24-bit */ + bitwidth_dai = 24; + sclk = params_rate(params) * bitwidth_dai * + params_channels(params); + + switch (params_rate(params)) { + case 176400: + dsd_speed = 0; + break; + case 352800: + dsd_speed = 1; + break; + default: + dev_err(codec->dev, "Rate(%u) not supported\n", + params_rate(params)); + return -EINVAL; + } + + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SPEED_MASK, + dsd_speed << CS43130_DSD_SPEED_SHIFT); + break; + case CS43130_ASP_PCM_DAI: + rate_map = cs43130_get_rate_table(params_rate(params)); + if (!rate_map) + return -EINVAL; + + regmap_write(cs43130->regmap, CS43130_SP_SRATE, rate_map->val); + break; + default: + dev_err(codec->dev, "Invalid DAI (%d)\n", dai->id); + return -EINVAL; + } + + switch (dai->id) { + case CS43130_ASP_DOP_DAI: + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_ASP << + CS43130_DSD_SRC_SHIFT); + break; + case CS43130_XSP_DOP_DAI: + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_XSP << + CS43130_DSD_SRC_SHIFT); + } + + if (!sclk && cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM) + /* Calculate SCLK in master mode if unassigned */ + sclk = params_rate(params) * bitwidth_dai * + params_channels(params); + + if (!sclk) { + /* at this point, SCLK must be set */ + dev_err(codec->dev, "SCLK freq is not set\n"); + return -EINVAL; + } + + bitwidth_sclk = (sclk / params_rate(params)) / params_channels(params); + if (bitwidth_sclk < bitwidth_dai) { + dev_err(codec->dev, "Format not supported: SCLK freq is too low\n"); + return -EINVAL; + } + + dev_dbg(codec->dev, + "sclk = %u, fs = %d, bitwidth_dai = %u\n", + sclk, params_rate(params), bitwidth_dai); + + dev_dbg(codec->dev, + "bitwidth_sclk = %u, num_ch = %u\n", + bitwidth_sclk, params_channels(params)); + + cs43130_set_bitwidth(dai->id, bitwidth_dai, cs43130->regmap); + cs43130_set_sp_fmt(dai->id, bitwidth_sclk, params, cs43130); + + return 0; +} + +static int cs43130_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + mutex_lock(&cs43130->clk_mutex); + cs43130->clk_req--; + if (!cs43130->clk_req) { + /* no DAI is currently using clk */ + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); + cs43130_pcm_dsd_mix(false, cs43130->regmap); + } + mutex_unlock(&cs43130->clk_mutex); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(pcm_vol_tlv, -12750, 50, 1); + +static const char * const pcm_ch_text[] = { + "Left-Right Ch", + "Left-Left Ch", + "Right-Left Ch", + "Right-Right Ch", +}; + +static const struct reg_sequence pcm_ch_en_seq[] = { + {CS43130_DXD1, 0x99}, + {0x180005, 0x8C}, + {0x180007, 0xAB}, + {0x180015, 0x31}, + {0x180017, 0xB2}, + {0x180025, 0x30}, + {0x180027, 0x84}, + {0x180035, 0x9C}, + {0x180037, 0xAE}, + {0x18000D, 0x24}, + {0x18000F, 0xA3}, + {0x18001D, 0x05}, + {0x18001F, 0xD4}, + {0x18002D, 0x0B}, + {0x18002F, 0xC7}, + {0x18003D, 0x71}, + {0x18003F, 0xE7}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence pcm_ch_dis_seq[] = { + {CS43130_DXD1, 0x99}, + {0x180005, 0x24}, + {0x180007, 0xA3}, + {0x180015, 0x05}, + {0x180017, 0xD4}, + {0x180025, 0x0B}, + {0x180027, 0xC7}, + {0x180035, 0x71}, + {0x180037, 0xE7}, + {0x18000D, 0x8C}, + {0x18000F, 0xAB}, + {0x18001D, 0x31}, + {0x18001F, 0xB2}, + {0x18002D, 0x30}, + {0x18002F, 0x84}, + {0x18003D, 0x9C}, + {0x18003F, 0xAE}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_pcm_ch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_soc_get_enum_double(kcontrol, ucontrol); +} + +static int cs43130_pcm_ch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (item[0] >= e->items) + return -EINVAL; + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + switch (cs43130->dev_id) { + case CS43131_CHIP_ID: + case CS43198_CHIP_ID: + if (val >= 2) + regmap_multi_reg_write(cs43130->regmap, pcm_ch_en_seq, + ARRAY_SIZE(pcm_ch_en_seq)); + else + regmap_multi_reg_write(cs43130->regmap, pcm_ch_dis_seq, + ARRAY_SIZE(pcm_ch_dis_seq)); + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static SOC_ENUM_SINGLE_DECL(pcm_ch_enum, CS43130_PCM_PATH_CTL_2, 0, + pcm_ch_text); + +static const char * const pcm_spd_texts[] = { + "Fast", + "Slow", +}; + +static SOC_ENUM_SINGLE_DECL(pcm_spd_enum, CS43130_PCM_FILT_OPT, 7, + pcm_spd_texts); + +static const char * const dsd_texts[] = { + "Off", + "BCKA Mode", + "BCKD Mode", +}; + +static const unsigned int dsd_values[] = { + CS43130_DSD_SRC_DSD, + CS43130_DSD_SRC_ASP, + CS43130_DSD_SRC_XSP, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(dsd_enum, CS43130_DSD_INT_CFG, 0, 0x03, + dsd_texts, dsd_values); + +static const struct snd_kcontrol_new cs43130_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS43130_PCM_VOL_A, CS43130_PCM_VOL_B, 0, 0xFF, 1, + pcm_vol_tlv), + SOC_DOUBLE_R_TLV("Master DSD Playback Volume", + CS43130_DSD_VOL_A, CS43130_DSD_VOL_B, 0, 0xFF, 1, + pcm_vol_tlv), + SOC_ENUM_EXT("PCM Ch Select", pcm_ch_enum, cs43130_pcm_ch_get, + cs43130_pcm_ch_put), + SOC_ENUM("PCM Filter Speed", pcm_spd_enum), + SOC_SINGLE("PCM Phase Compensation", CS43130_PCM_FILT_OPT, 6, 1, 0), + SOC_SINGLE("PCM Nonoversample Emulate", CS43130_PCM_FILT_OPT, 5, 1, 0), + SOC_SINGLE("PCM High-pass Filter", CS43130_PCM_FILT_OPT, 1, 1, 0), + SOC_SINGLE("PCM De-emphasis Filter", CS43130_PCM_FILT_OPT, 0, 1, 0), + SOC_ENUM("DSD Phase Modulation", dsd_enum), +}; + +static const struct reg_sequence pcm_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD7, 0x01}, + {CS43130_DXD8, 0}, + {CS43130_DXD9, 0x01}, + {CS43130_DXD3, 0x12}, + {CS43130_DXD4, 0}, + {CS43130_DXD10, 0x28}, + {CS43130_DXD11, 0x28}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence dsd_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD7, 0x01}, + {CS43130_DXD8, 0}, + {CS43130_DXD9, 0x01}, + {CS43130_DXD3, 0x12}, + {CS43130_DXD4, 0}, + {CS43130_DXD10, 0x1E}, + {CS43130_DXD11, 0x20}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence pop_free_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD12, 0x0A}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence pop_free_seq2[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD13, 0x20}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence mute_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD3, 0x12}, + {CS43130_DXD5, 0x02}, + {CS43130_DXD4, 0x12}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence unmute_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD3, 0x10}, + {CS43130_DXD5, 0}, + {CS43130_DXD4, 0x16}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_dsd_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 cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, dsd_seq, + ARRAY_SIZE(dsd_seq)); + } + break; + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_1, + CS43130_MUTE_MASK, 0); + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, unmute_seq, + ARRAY_SIZE(unmute_seq)); + } + break; + case SND_SOC_DAPM_PRE_PMD: + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, mute_seq, + ARRAY_SIZE(mute_seq)); + regmap_update_bits(cs43130->regmap, + CS43130_DSD_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + /* + * DSD Power Down Sequence + * According to Design, 130ms is preferred. + */ + msleep(130); + break; + case CS43131_CHIP_ID: + case CS43198_CHIP_ID: + regmap_update_bits(cs43130->regmap, + CS43130_DSD_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + } + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_pcm_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 cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, pcm_seq, + ARRAY_SIZE(pcm_seq)); + } + break; + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs43130->regmap, CS43130_PCM_PATH_CTL_1, + CS43130_MUTE_MASK, 0); + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, unmute_seq, + ARRAY_SIZE(unmute_seq)); + } + break; + case SND_SOC_DAPM_PRE_PMD: + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, mute_seq, + ARRAY_SIZE(mute_seq)); + regmap_update_bits(cs43130->regmap, + CS43130_PCM_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + /* + * PCM Power Down Sequence + * According to Design, 130ms is preferred. + */ + msleep(130); + break; + case CS43131_CHIP_ID: + case CS43198_CHIP_ID: + regmap_update_bits(cs43130->regmap, + CS43130_PCM_PATH_CTL_1, + CS43130_MUTE_MASK, CS43130_MUTE_EN); + } + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct reg_sequence dac_postpmu_seq[] = { + {CS43130_DXD9, 0x0C}, + {CS43130_DXD3, 0x10}, + {CS43130_DXD4, 0x20}, +}; + +static const struct reg_sequence dac_postpmd_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD6, 0x01}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_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 cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, pop_free_seq, + ARRAY_SIZE(pop_free_seq)); + break; + case CS43131_CHIP_ID: + case CS43198_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, pop_free_seq2, + ARRAY_SIZE(pop_free_seq2)); + } + break; + case SND_SOC_DAPM_POST_PMU: + usleep_range(10000, 10050); + + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, dac_postpmu_seq, + ARRAY_SIZE(dac_postpmu_seq)); + /* + * Per datasheet, Sec. PCM Power-Up Sequence. + * According to Design, CS43130_DXD12 must be 0 to meet + * THDN and Dynamic Range spec. + */ + msleep(1000); + regmap_write(cs43130->regmap, CS43130_DXD12, 0); + break; + case CS43131_CHIP_ID: + case CS43198_CHIP_ID: + usleep_range(12000, 12010); + regmap_write(cs43130->regmap, CS43130_DXD13, 0); + } + + regmap_write(cs43130->regmap, CS43130_DXD1, 0); + break; + case SND_SOC_DAPM_POST_PMD: + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + regmap_multi_reg_write(cs43130->regmap, dac_postpmd_seq, + ARRAY_SIZE(dac_postpmd_seq)); + } + break; + default: + dev_err(codec->dev, "Invalid DAC event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct reg_sequence hpin_prepmd_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD15, 0x64}, + {CS43130_DXD14, 0}, + {CS43130_DXD2, 0}, + {CS43130_DXD1, 0}, +}; + +static const struct reg_sequence hpin_postpmu_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD2, 1}, + {CS43130_DXD14, 0xDC}, + {CS43130_DXD15, 0xE4}, + {CS43130_DXD1, 0}, +}; + +static int cs43130_hpin_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 cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + regmap_multi_reg_write(cs43130->regmap, hpin_prepmd_seq, + ARRAY_SIZE(hpin_prepmd_seq)); + break; + case SND_SOC_DAPM_PRE_PMU: + regmap_multi_reg_write(cs43130->regmap, hpin_postpmu_seq, + ARRAY_SIZE(hpin_postpmu_seq)); + break; + default: + dev_err(codec->dev, "Invalid HPIN event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dapm_widget digital_hp_widgets[] = { + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + + SND_SOC_DAPM_AIF_IN_E("ASPIN PCM", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_ASP_SHIFT, 1, cs43130_pcm_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD)), + + SND_SOC_DAPM_AIF_IN_E("ASPIN DoP", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_ASP_SHIFT, 1, cs43130_dsd_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD)), + + SND_SOC_DAPM_AIF_IN_E("XSPIN DoP", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_XSP_SHIFT, 1, cs43130_dsd_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD)), + + SND_SOC_DAPM_AIF_IN_E("XSPIN DSD", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_DSDIF_SHIFT, 1, cs43130_dsd_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD)), + + SND_SOC_DAPM_DAC("DSD", NULL, CS43130_DSD_PATH_CTL_2, + CS43130_DSD_EN_SHIFT, 0), + + SND_SOC_DAPM_DAC_E("HiFi DAC", NULL, CS43130_PWDN_CTL, + CS43130_PDN_HP_SHIFT, 1, cs43130_dac_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD)), +}; + +static const struct snd_soc_dapm_widget analog_hp_widgets[] = { + SND_SOC_DAPM_DAC_E("Analog Playback", NULL, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_SHIFT, 0, cs43130_hpin_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), +}; + +static struct snd_soc_dapm_widget all_hp_widgets[ + ARRAY_SIZE(digital_hp_widgets) + + ARRAY_SIZE(analog_hp_widgets)]; + +static const struct snd_soc_dapm_route digital_hp_routes[] = { + {"ASPIN PCM", NULL, "ASP PCM Playback"}, + {"ASPIN DoP", NULL, "ASP DoP Playback"}, + {"XSPIN DoP", NULL, "XSP DoP Playback"}, + {"XSPIN DSD", NULL, "XSP DSD Playback"}, + {"DSD", NULL, "ASPIN DoP"}, + {"DSD", NULL, "XSPIN DoP"}, + {"DSD", NULL, "XSPIN DSD"}, + {"HiFi DAC", NULL, "ASPIN PCM"}, + {"HiFi DAC", NULL, "DSD"}, + {"HPOUTA", NULL, "HiFi DAC"}, + {"HPOUTB", NULL, "HiFi DAC"}, +}; + +static const struct snd_soc_dapm_route analog_hp_routes[] = { + {"HPOUTA", NULL, "Analog Playback"}, + {"HPOUTB", NULL, "Analog Playback"}, +}; + +static struct snd_soc_dapm_route all_hp_routes[ + ARRAY_SIZE(digital_hp_routes) + + ARRAY_SIZE(analog_hp_routes)]; + +static const unsigned int cs43130_asp_src_rates[] = { + 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs43130_asp_constraints = { + .count = ARRAY_SIZE(cs43130_asp_src_rates), + .list = cs43130_asp_src_rates, +}; + +static int cs43130_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs43130_asp_constraints); +} + +static const unsigned int cs43130_dop_src_rates[] = { + 176400, 352800, +}; + +static const struct snd_pcm_hw_constraint_list cs43130_dop_constraints = { + .count = ARRAY_SIZE(cs43130_dop_src_rates), + .list = cs43130_dop_src_rates, +}; + +static int cs43130_dop_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs43130_dop_constraints); +} + +static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM; + break; + default: + dev_err(codec->dev, "unsupported mode\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_DSP_B; + break; + default: + dev_err(codec->dev, + "unsupported audio format\n"); + return -EINVAL; + } + + dev_dbg(codec->dev, "dai_id = %d, dai_mode = %u, dai_format = %u\n", + codec_dai->id, + cs43130->dais[codec_dai->id].dai_mode, + cs43130->dais[codec_dai->id].dai_format); + + return 0; +} + +static int cs43130_dsd_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM; + break; + default: + dev_err(codec->dev, "Unsupported DAI format.\n"); + return -EINVAL; + } + + dev_dbg(codec->dev, "dai_mode = 0x%x\n", + cs43130->dais[codec_dai->id].dai_mode); + + return 0; +} + +static int cs43130_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + cs43130->dais[codec_dai->id].sclk = freq; + dev_dbg(codec->dev, "dai_id = %d, sclk = %u\n", codec_dai->id, + cs43130->dais[codec_dai->id].sclk); + + return 0; +} + +static const struct snd_soc_dai_ops cs43130_pcm_ops = { + .startup = cs43130_pcm_startup, + .hw_params = cs43130_hw_params, + .hw_free = cs43130_hw_free, + .set_sysclk = cs43130_set_sysclk, + .set_fmt = cs43130_pcm_set_fmt, +}; + +static const struct snd_soc_dai_ops cs43130_dop_ops = { + .startup = cs43130_dop_startup, + .hw_params = cs43130_hw_params, + .hw_free = cs43130_hw_free, + .set_sysclk = cs43130_set_sysclk, + .set_fmt = cs43130_pcm_set_fmt, +}; + +static const struct snd_soc_dai_ops cs43130_dsd_ops = { + .startup = cs43130_dop_startup, + .hw_params = cs43130_dsd_hw_params, + .hw_free = cs43130_hw_free, + .set_fmt = cs43130_dsd_set_fmt, +}; + +static struct snd_soc_dai_driver cs43130_dai[] = { + { + .name = "cs43130-asp-pcm", + .id = CS43130_ASP_PCM_DAI, + .playback = { + .stream_name = "ASP PCM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_PCM_FORMATS, + }, + .ops = &cs43130_pcm_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-asp-dop", + .id = CS43130_ASP_DOP_DAI, + .playback = { + .stream_name = "ASP DoP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_DOP_FORMATS, + }, + .ops = &cs43130_dop_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-xsp-dop", + .id = CS43130_XSP_DOP_DAI, + .playback = { + .stream_name = "XSP DoP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_DOP_FORMATS, + }, + .ops = &cs43130_dop_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-xsp-dsd", + .id = CS43130_XSP_DSD_DAI, + .playback = { + .stream_name = "XSP DSD Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_DOP_FORMATS, + }, + .ops = &cs43130_dsd_ops, + }, + +}; + +static int cs43130_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "clk_id = %d, source = %d, freq = %d, dir = %d\n", + clk_id, source, freq, dir); + + switch (freq) { + case CS43130_MCLK_22M: + case CS43130_MCLK_24M: + cs43130->mclk = freq; + break; + default: + dev_err(codec->dev, "Invalid MCLK INT freq: %u\n", freq); + return -EINVAL; + } + + if (source == CS43130_MCLK_SRC_EXT) { + cs43130->pll_bypass = true; + } else { + dev_err(codec->dev, "Invalid MCLK source\n"); + return -EINVAL; + } + + return 0; +} + +static inline u16 cs43130_get_ac_reg_val(u16 ac_freq) +{ + /* AC freq is counted in 5.94Hz step. */ + return ac_freq / 6; +} + +static int cs43130_show_dc(struct device *dev, char *buf, u8 ch) +{ + struct i2c_client *client = to_i2c_client(dev); + struct cs43130_private *cs43130 = i2c_get_clientdata(client); + + if (!cs43130->hpload_done) + return scnprintf(buf, PAGE_SIZE, "NO_HPLOAD\n"); + else + return scnprintf(buf, PAGE_SIZE, "%u\n", + cs43130->hpload_dc[ch]); +} + +static ssize_t cs43130_show_dc_l(struct device *dev, + struct device_attribute *attr, char *buf) +{ + return cs43130_show_dc(dev, buf, HP_LEFT); +} + +static ssize_t cs43130_show_dc_r(struct device *dev, + struct device_attribute *attr, char *buf) +{ + return cs43130_show_dc(dev, buf, HP_RIGHT); +} + +static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = { + 24, + 43, + 93, + 200, + 431, + 928, + 2000, + 4309, + 9283, + 20000, +}; + +static int cs43130_show_ac(struct device *dev, char *buf, u8 ch) +{ + int i, j = 0, tmp; + struct i2c_client *client = to_i2c_client(dev); + struct cs43130_private *cs43130 = i2c_get_clientdata(client); + + if (cs43130->hpload_done && cs43130->ac_meas) { + for (i = 0; i < ARRAY_SIZE(cs43130_ac_freq); i++) { + tmp = scnprintf(buf + j, PAGE_SIZE - j, "%u\n", + cs43130->hpload_ac[i][ch]); + if (!tmp) + break; + + j += tmp; + } + + return j; + } else { + return scnprintf(buf, PAGE_SIZE, "NO_HPLOAD\n"); + } +} + +static ssize_t cs43130_show_ac_l(struct device *dev, + struct device_attribute *attr, char *buf) +{ + return cs43130_show_ac(dev, buf, HP_LEFT); +} + +static ssize_t cs43130_show_ac_r(struct device *dev, + struct device_attribute *attr, char *buf) +{ + return cs43130_show_ac(dev, buf, HP_RIGHT); +} + +static DEVICE_ATTR(hpload_dc_l, S_IRUGO, cs43130_show_dc_l, NULL); +static DEVICE_ATTR(hpload_dc_r, S_IRUGO, cs43130_show_dc_r, NULL); +static DEVICE_ATTR(hpload_ac_l, S_IRUGO, cs43130_show_ac_l, NULL); +static DEVICE_ATTR(hpload_ac_r, S_IRUGO, cs43130_show_ac_r, NULL); + +static struct reg_sequence hp_en_cal_seq[] = { + {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, + {CS43130_HP_MEAS_LOAD_1, 0}, + {CS43130_HP_MEAS_LOAD_2, 0}, + {CS43130_INT_MASK_4, 0}, + {CS43130_DXD1, 0x99}, + {CS43130_DXD16, 0xBB}, + {CS43130_DXD12, 0x01}, + {CS43130_DXD19, 0xCB}, + {CS43130_DXD17, 0x95}, + {CS43130_DXD18, 0x0B}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x80}, +}; + +static struct reg_sequence hp_en_cal_seq2[] = { + {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, + {CS43130_HP_MEAS_LOAD_1, 0}, + {CS43130_HP_MEAS_LOAD_2, 0}, + {CS43130_INT_MASK_4, 0}, + {CS43130_HP_LOAD_1, 0x80}, +}; + +static struct reg_sequence hp_dis_cal_seq[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_DXD1, 0x99}, + {CS43130_DXD12, 0}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0}, +}; + +static struct reg_sequence hp_dis_cal_seq2[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0}, +}; + +static struct reg_sequence hp_dc_ch_l_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x0A}, + {CS43130_DXD17, 0x93}, + {CS43130_DXD18, 0x0A}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x81}, +}; + +static struct reg_sequence hp_dc_ch_l_seq2[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x81}, +}; + +static struct reg_sequence hp_dc_ch_r_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x8A}, + {CS43130_DXD17, 0x15}, + {CS43130_DXD18, 0x06}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x91}, +}; + +static struct reg_sequence hp_dc_ch_r_seq2[] = { + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x91}, +}; + +static struct reg_sequence hp_ac_ch_l_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x0A}, + {CS43130_DXD17, 0x93}, + {CS43130_DXD18, 0x0A}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x82}, +}; + +static struct reg_sequence hp_ac_ch_l_seq2[] = { + {CS43130_HP_LOAD_1, 0x80}, + {CS43130_HP_LOAD_1, 0x82}, +}; + +static struct reg_sequence hp_ac_ch_r_seq[] = { + {CS43130_DXD1, 0x99}, + {CS43130_DXD19, 0x8A}, + {CS43130_DXD17, 0x15}, + {CS43130_DXD18, 0x06}, + {CS43130_DXD1, 0}, + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x92}, +}; + +static struct reg_sequence hp_ac_ch_r_seq2[] = { + {CS43130_HP_LOAD_1, 0x90}, + {CS43130_HP_LOAD_1, 0x92}, +}; + +static struct reg_sequence hp_cln_seq[] = { + {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, + {CS43130_HP_MEAS_LOAD_1, 0}, + {CS43130_HP_MEAS_LOAD_2, 0}, +}; + +struct reg_sequences { + struct reg_sequence *seq; + int size; + unsigned int msk; +}; + +static struct reg_sequences hpload_seq1[] = { + { + .seq = hp_en_cal_seq, + .size = ARRAY_SIZE(hp_en_cal_seq), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_l_seq, + .size = ARRAY_SIZE(hp_dc_ch_l_seq), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_l_seq, + .size = ARRAY_SIZE(hp_ac_ch_l_seq), + .msk = CS43130_HPLOAD_AC_INT, + }, + { + .seq = hp_dis_cal_seq, + .size = ARRAY_SIZE(hp_dis_cal_seq), + .msk = CS43130_HPLOAD_OFF_INT, + }, + { + .seq = hp_en_cal_seq, + .size = ARRAY_SIZE(hp_en_cal_seq), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_r_seq, + .size = ARRAY_SIZE(hp_dc_ch_r_seq), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_r_seq, + .size = ARRAY_SIZE(hp_ac_ch_r_seq), + .msk = CS43130_HPLOAD_AC_INT, + }, +}; + +static struct reg_sequences hpload_seq2[] = { + { + .seq = hp_en_cal_seq2, + .size = ARRAY_SIZE(hp_en_cal_seq2), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_l_seq2, + .size = ARRAY_SIZE(hp_dc_ch_l_seq2), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_l_seq2, + .size = ARRAY_SIZE(hp_ac_ch_l_seq2), + .msk = CS43130_HPLOAD_AC_INT, + }, + { + .seq = hp_dis_cal_seq2, + .size = ARRAY_SIZE(hp_dis_cal_seq2), + .msk = CS43130_HPLOAD_OFF_INT, + }, + { + .seq = hp_en_cal_seq2, + .size = ARRAY_SIZE(hp_en_cal_seq2), + .msk = CS43130_HPLOAD_ON_INT, + }, + { + .seq = hp_dc_ch_r_seq2, + .size = ARRAY_SIZE(hp_dc_ch_r_seq2), + .msk = CS43130_HPLOAD_DC_INT, + }, + { + .seq = hp_ac_ch_r_seq2, + .size = ARRAY_SIZE(hp_ac_ch_r_seq2), + .msk = CS43130_HPLOAD_AC_INT, + }, +}; + +static int cs43130_update_hpload(unsigned int msk, int ac_idx, + struct cs43130_private *cs43130) +{ + bool left_ch = true; + unsigned int reg; + u32 addr; + u16 impedance; + struct snd_soc_codec *codec = cs43130->codec; + + switch (msk) { + case CS43130_HPLOAD_DC_INT: + case CS43130_HPLOAD_AC_INT: + break; + default: + return 0; + } + + regmap_read(cs43130->regmap, CS43130_HP_LOAD_1, &reg); + if (reg & CS43130_HPLOAD_CHN_SEL) + left_ch = false; + + if (msk == CS43130_HPLOAD_DC_INT) + addr = CS43130_HP_DC_STAT_1; + else + addr = CS43130_HP_AC_STAT_1; + + regmap_read(cs43130->regmap, addr, &reg); + impedance = reg >> 3; + regmap_read(cs43130->regmap, addr + 1, &reg); + impedance |= reg << 5; + + if (msk == CS43130_HPLOAD_DC_INT) { + if (left_ch) + cs43130->hpload_dc[HP_LEFT] = impedance; + else + cs43130->hpload_dc[HP_RIGHT] = impedance; + + dev_dbg(codec->dev, "HP DC impedance (Ch %u): %u\n", !left_ch, + impedance); + } else { + if (left_ch) + cs43130->hpload_ac[ac_idx][HP_LEFT] = impedance; + else + cs43130->hpload_ac[ac_idx][HP_RIGHT] = impedance; + + dev_dbg(codec->dev, "HP AC (%u Hz) impedance (Ch %u): %u\n", + cs43130->ac_freq[ac_idx], !left_ch, impedance); + } + + return 0; +} + +static int cs43130_hpload_proc(struct cs43130_private *cs43130, + struct reg_sequence *seq, int seq_size, + unsigned int rslt_msk, int ac_idx) +{ + int ret; + unsigned int msk; + u16 ac_reg_val; + struct snd_soc_codec *codec = cs43130->codec; + + reinit_completion(&cs43130->hpload_evt); + + if (rslt_msk == CS43130_HPLOAD_AC_INT) { + ac_reg_val = cs43130_get_ac_reg_val(cs43130->ac_freq[ac_idx]); + regmap_update_bits(cs43130->regmap, CS43130_HP_LOAD_1, + CS43130_HPLOAD_AC_START, 0); + regmap_update_bits(cs43130->regmap, CS43130_HP_MEAS_LOAD_1, + CS43130_HP_MEAS_LOAD_MASK, + ac_reg_val >> CS43130_HP_MEAS_LOAD_1_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_HP_MEAS_LOAD_2, + CS43130_HP_MEAS_LOAD_MASK, + ac_reg_val >> CS43130_HP_MEAS_LOAD_2_SHIFT); + } + + regmap_multi_reg_write(cs43130->regmap, seq, + seq_size); + + ret = wait_for_completion_timeout(&cs43130->hpload_evt, + msecs_to_jiffies(1000)); + regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); + if (!ret) { + dev_err(codec->dev, "Timeout waiting for HPLOAD interrupt\n"); + return -1; + } + + dev_dbg(codec->dev, "HP load stat: %x, INT_MASK_4: %x\n", + cs43130->hpload_stat, msk); + if ((cs43130->hpload_stat & (CS43130_HPLOAD_NO_DC_INT | + CS43130_HPLOAD_UNPLUG_INT | + CS43130_HPLOAD_OOR_INT)) || + !(cs43130->hpload_stat & rslt_msk)) { + dev_dbg(codec->dev, "HP load measure failed\n"); + return -1; + } + + return 0; +} + +static const struct reg_sequence hv_seq[][2] = { + { + {CS43130_CLASS_H_CTL, 0x1C}, + {CS43130_HP_OUT_CTL_1, 0x10}, + }, + { + {CS43130_CLASS_H_CTL, 0x1E}, + {CS43130_HP_OUT_CTL_1, 0x20}, + }, + { + {CS43130_CLASS_H_CTL, 0x1E}, + {CS43130_HP_OUT_CTL_1, 0x30}, + }, +}; + +static int cs43130_set_hv(struct regmap *regmap, u16 hpload_dc, + const u16 *dc_threshold) +{ + int i; + + for (i = 0; i < CS43130_DC_THRESHOLD; i++) { + if (hpload_dc <= dc_threshold[i]) + break; + } + + regmap_multi_reg_write(regmap, hv_seq[i], ARRAY_SIZE(hv_seq[i])); + + return 0; +} + +static void cs43130_imp_meas(struct work_struct *wk) +{ + unsigned int reg, seq_size; + int i, ret, ac_idx; + struct cs43130_private *cs43130; + struct snd_soc_codec *codec; + struct reg_sequences *hpload_seq; + + cs43130 = container_of(wk, struct cs43130_private, work); + codec = cs43130->codec; + + if (!cs43130->mclk) + return; + + cs43130->hpload_done = false; + + mutex_lock(&cs43130->clk_mutex); + if (!cs43130->clk_req) { + /* clk not in use */ + cs43130_set_pll(codec, 0, 0, cs43130->mclk, CS43130_MCLK_22M); + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + } + + cs43130->clk_req++; + mutex_unlock(&cs43130->clk_mutex); + + regmap_read(cs43130->regmap, CS43130_INT_STATUS_4, &reg); + + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + hpload_seq = hpload_seq1; + seq_size = ARRAY_SIZE(hpload_seq1); + break; + case CS43131_CHIP_ID: + hpload_seq = hpload_seq2; + seq_size = ARRAY_SIZE(hpload_seq2); + } + + i = 0; + ac_idx = 0; + while (i < seq_size) { + ret = cs43130_hpload_proc(cs43130, hpload_seq[i].seq, + hpload_seq[i].size, + hpload_seq[i].msk, ac_idx); + if (ret < 0) + goto exit; + + cs43130_update_hpload(hpload_seq[i].msk, ac_idx, cs43130); + + if (cs43130->ac_meas && + hpload_seq[i].msk == CS43130_HPLOAD_AC_INT && + ac_idx < CS43130_AC_FREQ - 1) { + ac_idx++; + } else { + ac_idx = 0; + i++; + } + } + cs43130->hpload_done = true; + + if (cs43130->hpload_dc[HP_LEFT] >= CS43130_LINEOUT_LOAD) + snd_soc_jack_report(&cs43130->jack, CS43130_JACK_LINEOUT, + CS43130_JACK_MASK); + else + snd_soc_jack_report(&cs43130->jack, CS43130_JACK_HEADPHONE, + CS43130_JACK_MASK); + + dev_dbg(codec->dev, "Set HP output control. DC threshold\n"); + for (i = 0; i < CS43130_DC_THRESHOLD; i++) + dev_dbg(codec->dev, "DC threshold[%d]: %u.\n", i, + cs43130->dc_threshold[i]); + + cs43130_set_hv(cs43130->regmap, cs43130->hpload_dc[HP_LEFT], + cs43130->dc_threshold); + +exit: + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + cs43130_hpload_proc(cs43130, hp_dis_cal_seq, + ARRAY_SIZE(hp_dis_cal_seq), + CS43130_HPLOAD_OFF_INT, ac_idx); + break; + case CS43131_CHIP_ID: + cs43130_hpload_proc(cs43130, hp_dis_cal_seq2, + ARRAY_SIZE(hp_dis_cal_seq2), + CS43130_HPLOAD_OFF_INT, ac_idx); + } + + regmap_multi_reg_write(cs43130->regmap, hp_cln_seq, + ARRAY_SIZE(hp_cln_seq)); + + mutex_lock(&cs43130->clk_mutex); + cs43130->clk_req--; + /* clk not in use */ + if (!cs43130->clk_req) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); + mutex_unlock(&cs43130->clk_mutex); +} + +static irqreturn_t cs43130_irq_thread(int irq, void *data) +{ + struct cs43130_private *cs43130 = (struct cs43130_private *)data; + struct snd_soc_codec *codec = cs43130->codec; + unsigned int stickies[CS43130_NUM_INT]; + unsigned int irq_occurrance = 0; + unsigned int masks[CS43130_NUM_INT]; + int i, j; + + for (i = 0; i < ARRAY_SIZE(stickies); i++) { + regmap_read(cs43130->regmap, CS43130_INT_STATUS_1 + i, + &stickies[i]); + regmap_read(cs43130->regmap, CS43130_INT_MASK_1 + i, + &masks[i]); + } + + for (i = 0; i < ARRAY_SIZE(stickies); i++) { + stickies[i] = stickies[i] & (~masks[i]); + for (j = 0; j < 8; j++) + irq_occurrance += (stickies[i] >> j) & 1; + } + dev_dbg(codec->dev, "number of interrupts occurred (%u)\n", + irq_occurrance); + + if (!irq_occurrance) + return IRQ_NONE; + + if (stickies[0] & CS43130_XTAL_RDY_INT) { + complete(&cs43130->xtal_rdy); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_PLL_RDY_INT) { + complete(&cs43130->pll_rdy); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_NO_DC_INT) { + cs43130->hpload_stat = stickies[3]; + dev_err(codec->dev, + "DC load has not completed before AC load (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_UNPLUG_INT) { + cs43130->hpload_stat = stickies[3]; + dev_err(codec->dev, "HP unplugged during measurement (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_OOR_INT) { + cs43130->hpload_stat = stickies[3]; + dev_err(codec->dev, "HP load out of range (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_AC_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP AC load measurement done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_DC_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP DC load measurement done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_ON_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP load state machine on done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[3] & CS43130_HPLOAD_OFF_INT) { + cs43130->hpload_stat = stickies[3]; + dev_dbg(codec->dev, "HP load state machine off done (%x)\n", + cs43130->hpload_stat); + complete(&cs43130->hpload_evt); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_XTAL_ERR_INT) { + dev_err(codec->dev, "Crystal err: clock is not running\n"); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_HP_UNPLUG_INT) { + dev_dbg(codec->dev, "HP unplugged\n"); + cs43130->hpload_done = false; + snd_soc_jack_report(&cs43130->jack, 0, CS43130_JACK_MASK); + return IRQ_HANDLED; + } + + if (stickies[0] & CS43130_HP_PLUG_INT) { + if (cs43130->dc_meas && !cs43130->hpload_done && + !work_busy(&cs43130->work)) { + dev_dbg(codec->dev, "HP load queue work\n"); + queue_work(cs43130->wq, &cs43130->work); + } + + snd_soc_jack_report(&cs43130->jack, SND_JACK_MECHANICAL, + CS43130_JACK_MASK); + return IRQ_HANDLED; + } + + return IRQ_NONE; +} + +static int cs43130_probe(struct snd_soc_codec *codec) +{ + int ret; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_card *card = codec->component.card; + unsigned int reg; + + cs43130->codec = codec; + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) { + regmap_update_bits(cs43130->regmap, CS43130_CRYSTAL_SET, + CS43130_XTAL_IBIAS_MASK, + cs43130->xtal_ibias); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, 0); + } + + ret = snd_soc_card_jack_new(card, "Headphone", CS43130_JACK_MASK, + &cs43130->jack, NULL, 0); + if (ret < 0) { + dev_err(codec->dev, "Cannot create jack\n"); + return ret; + } + + cs43130->hpload_done = false; + if (cs43130->dc_meas) { + ret = device_create_file(codec->dev, &dev_attr_hpload_dc_l); + if (ret < 0) + return ret; + + ret = device_create_file(codec->dev, &dev_attr_hpload_dc_r); + if (ret < 0) + return ret; + + ret = device_create_file(codec->dev, &dev_attr_hpload_ac_l); + if (ret < 0) + return ret; + + ret = device_create_file(codec->dev, &dev_attr_hpload_ac_r); + if (ret < 0) + return ret; + + cs43130->wq = create_singlethread_workqueue("cs43130_hp"); + INIT_WORK(&cs43130->work, cs43130_imp_meas); + } + + regmap_read(cs43130->regmap, CS43130_INT_STATUS_1, &reg); + regmap_read(cs43130->regmap, CS43130_HP_STATUS, &reg); + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT, 0); + regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT, + CS43130_HP_DETECT_CTRL_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT, + CS43130_HP_DETECT_CTRL_MASK, + CS43130_HP_DETECT_CTRL_MASK); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs43130 = { + .probe = cs43130_probe, + .component_driver = { + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + }, + .set_sysclk = cs43130_codec_set_sysclk, + .set_pll = cs43130_set_pll, +}; + +static const struct regmap_config cs43130_regmap = { + .reg_bits = 24, + .pad_bits = 8, + .val_bits = 8, + + .max_register = CS43130_LASTREG, + .reg_defaults = cs43130_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs43130_reg_defaults), + .readable_reg = cs43130_readable_register, + .precious_reg = cs43130_precious_register, + .volatile_reg = cs43130_volatile_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, /* needed for regcache_sync */ +}; + +static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { + 50, + 120, +}; + +static int cs43130_handle_device_data(struct i2c_client *i2c_client, + struct cs43130_private *cs43130) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + int i; + + if (of_property_read_u32(np, "cirrus,xtal-ibias", &val) < 0) { + /* Crystal is unused. System clock is used for external MCLK */ + cs43130->xtal_ibias = CS43130_XTAL_UNUSED; + return 0; + } + + switch (val) { + case 1: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_7_5UA; + break; + case 2: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_12_5UA; + break; + case 3: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_15UA; + break; + default: + dev_err(&i2c_client->dev, + "Invalid cirrus,xtal-ibias value: %d\n", val); + return -EINVAL; + } + + cs43130->dc_meas = of_property_read_bool(np, "cirrus,dc-measure"); + cs43130->ac_meas = of_property_read_bool(np, "cirrus,ac-measure"); + + if (of_property_read_u16_array(np, "cirrus,ac-freq", cs43130->ac_freq, + CS43130_AC_FREQ) < 0) { + for (i = 0; i < CS43130_AC_FREQ; i++) + cs43130->ac_freq[i] = cs43130_ac_freq[i]; + } + + if (of_property_read_u16_array(np, "cirrus,dc-threshold", + cs43130->dc_threshold, + CS43130_DC_THRESHOLD) < 0) { + for (i = 0; i < CS43130_DC_THRESHOLD; i++) + cs43130->dc_threshold[i] = cs43130_dc_threshold[i]; + } + + return 0; +} + +static int cs43130_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs43130_private *cs43130; + int ret; + unsigned int devid = 0; + unsigned int reg; + int i; + + cs43130 = devm_kzalloc(&client->dev, sizeof(*cs43130), GFP_KERNEL); + if (!cs43130) + return -ENOMEM; + + i2c_set_clientdata(client, cs43130); + + cs43130->regmap = devm_regmap_init_i2c(client, &cs43130_regmap); + if (IS_ERR(cs43130->regmap)) { + ret = PTR_ERR(cs43130->regmap); + return ret; + } + + if (client->dev.of_node) { + ret = cs43130_handle_device_data(client, cs43130); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++) + cs43130->supplies[i].supply = cs43130_supply_names[i]; + + ret = devm_regulator_bulk_get(&client->dev, + ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + ret = regulator_bulk_enable(ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + cs43130->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs43130->reset_gpio)) + return PTR_ERR(cs43130->reset_gpio); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + ret = regmap_read(cs43130->regmap, CS43130_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + switch (devid) { + case CS43130_CHIP_ID: + case CS4399_CHIP_ID: + case CS43131_CHIP_ID: + case CS43198_CHIP_ID: + break; + default: + dev_err(&client->dev, + "CS43130 Device ID %X. Expected ID %X, %X, %X or %X\n", + devid, CS43130_CHIP_ID, CS4399_CHIP_ID, + CS43131_CHIP_ID, CS43198_CHIP_ID); + ret = -ENODEV; + goto err; + } + + cs43130->dev_id = devid; + ret = regmap_read(cs43130->regmap, CS43130_REV_ID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + goto err; + } + + dev_info(&client->dev, + "Cirrus Logic CS43130 (%x), Revision: %02X\n", devid, + reg & 0xFF); + + mutex_init(&cs43130->clk_mutex); + + init_completion(&cs43130->xtal_rdy); + init_completion(&cs43130->pll_rdy); + init_completion(&cs43130->hpload_evt); + + ret = devm_request_threaded_irq(&client->dev, client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(&client->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; + + pm_runtime_set_autosuspend_delay(&client->dev, 100); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_set_active(&client->dev); + pm_runtime_enable(&client->dev); + + switch (cs43130->dev_id) { + case CS43130_CHIP_ID: + case CS43131_CHIP_ID: + memcpy(all_hp_widgets, digital_hp_widgets, + sizeof(digital_hp_widgets)); + memcpy(all_hp_widgets + ARRAY_SIZE(digital_hp_widgets), + analog_hp_widgets, sizeof(analog_hp_widgets)); + memcpy(all_hp_routes, digital_hp_routes, + sizeof(digital_hp_routes)); + memcpy(all_hp_routes + ARRAY_SIZE(digital_hp_routes), + analog_hp_routes, sizeof(analog_hp_routes)); + + soc_codec_dev_cs43130.component_driver.dapm_widgets = + all_hp_widgets; + soc_codec_dev_cs43130.component_driver.num_dapm_widgets = + ARRAY_SIZE(all_hp_widgets); + soc_codec_dev_cs43130.component_driver.dapm_routes = + all_hp_routes; + soc_codec_dev_cs43130.component_driver.num_dapm_routes = + ARRAY_SIZE(all_hp_routes); + break; + case CS43198_CHIP_ID: + case CS4399_CHIP_ID: + soc_codec_dev_cs43130.component_driver.dapm_widgets = + digital_hp_widgets; + soc_codec_dev_cs43130.component_driver.num_dapm_widgets = + ARRAY_SIZE(digital_hp_widgets); + soc_codec_dev_cs43130.component_driver.dapm_routes = + digital_hp_routes; + soc_codec_dev_cs43130.component_driver.num_dapm_routes = + ARRAY_SIZE(digital_hp_routes); + } + + ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_cs43130, + cs43130_dai, ARRAY_SIZE(cs43130_dai)); + if (ret < 0) { + dev_err(&client->dev, + "snd_soc_register_codec failed with ret = %d\n", ret); + goto err; + } + + regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG, + CS43130_ASP_3ST_MASK, 0); + regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG, + CS43130_XSP_3ST_MASK, 0); + + return 0; +err: + return ret; +} + +static int cs43130_i2c_remove(struct i2c_client *client) +{ + struct cs43130_private *cs43130 = i2c_get_clientdata(client); + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, + 1 << CS43130_XTAL_ERR_INT_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT, + CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT); + + if (cs43130->dc_meas) { + cancel_work_sync(&cs43130->work); + flush_workqueue(cs43130->wq); + + device_remove_file(&client->dev, &dev_attr_hpload_dc_l); + device_remove_file(&client->dev, &dev_attr_hpload_dc_r); + device_remove_file(&client->dev, &dev_attr_hpload_ac_l); + device_remove_file(&client->dev, &dev_attr_hpload_ac_r); + } + + if (cs43130->reset_gpio) + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); + + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static int cs43130_runtime_suspend(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, + 1 << CS43130_XTAL_ERR_INT_SHIFT); + + regcache_cache_only(cs43130->regmap, true); + regcache_mark_dirty(cs43130->regmap); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); + + return 0; +} + +static int cs43130_runtime_resume(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(CS43130_NUM_SUPPLIES, cs43130->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs43130->regmap, false); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + ret = regcache_sync(cs43130->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_ERR_INT, 0); + + return 0; +err: + regcache_cache_only(cs43130->regmap, true); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); + + return ret; +} + +static const struct dev_pm_ops cs43130_runtime_pm = { + SET_RUNTIME_PM_OPS(cs43130_runtime_suspend, cs43130_runtime_resume, + NULL) +}; + +static const struct of_device_id cs43130_of_match[] = { + {.compatible = "cirrus,cs43130",}, + {.compatible = "cirrus,cs4399",}, + {.compatible = "cirrus,cs43131",}, + {.compatible = "cirrus,cs43198",}, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs43130_of_match); + +static const struct i2c_device_id cs43130_i2c_id[] = { + {"cs43130", 0}, + {"cs4399", 0}, + {"cs43131", 0}, + {"cs43198", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs43130_i2c_id); + +static struct i2c_driver cs43130_i2c_driver = { + .driver = { + .name = "cs43130", + .of_match_table = cs43130_of_match, + .pm = &cs43130_runtime_pm, + }, + .id_table = cs43130_i2c_id, + .probe = cs43130_i2c_probe, + .remove = cs43130_i2c_remove, +}; + +module_i2c_driver(cs43130_i2c_driver); + +MODULE_AUTHOR("Li Xu <li.xu(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS43130 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h new file mode 100644 index 0000000..7812584 --- /dev/null +++ b/sound/soc/codecs/cs43130.h @@ -0,0 +1,546 @@ +/* + * ALSA SoC CS43130 codec driver + * + * Copyright 2017 Cirrus Logic, Inc. + * + * Author: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS43130_H__ +#define __CS43130_H__ + +/* CS43130 registers addresses */ +/* all reg address is shifted by a byte for control byte to be LSB */ +#define CS43130_FIRSTREG 0x010000 +#define CS43130_LASTREG 0x190000 +#define CS43130_CHIP_ID 0x00043130 +#define CS4399_CHIP_ID 0x00043990 +#define CS43131_CHIP_ID 0x00043131 +#define CS43198_CHIP_ID 0x00043198 +#define CS43130_DEVID_AB 0x010000 /* Device ID A & B [RO] */ +#define CS43130_DEVID_CD 0x010001 /* Device ID C & D [RO] */ +#define CS43130_DEVID_E 0x010002 /* Device ID E [RO] */ +#define CS43130_FAB_ID 0x010003 /* Fab ID [RO] */ +#define CS43130_REV_ID 0x010004 /* Revision ID [RO] */ +#define CS43130_SUBREV_ID 0x010005 /* Subrevision ID */ +#define CS43130_SYS_CLK_CTL_1 0x010006 /* System Clocking Ctl 1 */ +#define CS43130_SP_SRATE 0x01000B /* Serial Port Sample Rate */ +#define CS43130_SP_BITSIZE 0x01000C /* Serial Port Bit Size */ +#define CS43130_PAD_INT_CFG 0x01000D /* Pad Interface Config */ +#define CS43130_DXD1 0x010010 /* DXD1 */ +#define CS43130_DXD7 0x010025 /* DXD7 */ +#define CS43130_DXD19 0x010026 /* DXD19 */ +#define CS43130_DXD17 0x010027 /* DXD17 */ +#define CS43130_DXD18 0x010028 /* DXD18 */ +#define CS43130_DXD12 0x01002C /* DXD12 */ +#define CS43130_DXD8 0x01002E /* DXD8 */ +#define CS43130_PWDN_CTL 0x020000 /* Power Down Ctl */ +#define CS43130_DXD2 0x020019 /* DXD2 */ +#define CS43130_CRYSTAL_SET 0x020052 /* Crystal Setting */ +#define CS43130_PLL_SET_1 0x030001 /* PLL Setting 1 */ +#define CS43130_PLL_SET_2 0x030002 /* PLL Setting 2 */ +#define CS43130_PLL_SET_3 0x030003 /* PLL Setting 3 */ +#define CS43130_PLL_SET_4 0x030004 /* PLL Setting 4 */ +#define CS43130_PLL_SET_5 0x030005 /* PLL Setting 5 */ +#define CS43130_PLL_SET_6 0x030008 /* PLL Setting 6 */ +#define CS43130_PLL_SET_7 0x03000A /* PLL Setting 7 */ +#define CS43130_PLL_SET_8 0x03001B /* PLL Setting 8 */ +#define CS43130_PLL_SET_9 0x040002 /* PLL Setting 9 */ +#define CS43130_PLL_SET_10 0x040003 /* PLL Setting 10 */ +#define CS43130_CLKOUT_CTL 0x040004 /* CLKOUT Ctl */ +#define CS43130_ASP_NUM_1 0x040010 /* ASP Numerator 1 */ +#define CS43130_ASP_NUM_2 0x040011 /* ASP Numerator 2 */ +#define CS43130_ASP_DEN_1 0x040012 /* ASP Denominator 1 */ +#define CS43130_ASP_DEN_2 0x040013 /* ASP Denominator 2 */ +#define CS43130_ASP_LRCK_HI_TIME_1 0x040014 /* ASP LRCK High Time 1 */ +#define CS43130_ASP_LRCK_HI_TIME_2 0x040015 /* ASP LRCK High Time 2 */ +#define CS43130_ASP_LRCK_PERIOD_1 0x040016 /* ASP LRCK Period 1 */ +#define CS43130_ASP_LRCK_PERIOD_2 0x040017 /* ASP LRCK Period 2 */ +#define CS43130_ASP_CLOCK_CONF 0x040018 /* ASP Clock Config */ +#define CS43130_ASP_FRAME_CONF 0x040019 /* ASP Frame Config */ +#define CS43130_XSP_NUM_1 0x040020 /* XSP Numerator 1 */ +#define CS43130_XSP_NUM_2 0x040021 /* XSP Numerator 2 */ +#define CS43130_XSP_DEN_1 0x040022 /* XSP Denominator 1 */ +#define CS43130_XSP_DEN_2 0x040023 /* XSP Denominator 2 */ +#define CS43130_XSP_LRCK_HI_TIME_1 0x040024 /* XSP LRCK High Time 1 */ +#define CS43130_XSP_LRCK_HI_TIME_2 0x040025 /* XSP LRCK High Time 2 */ +#define CS43130_XSP_LRCK_PERIOD_1 0x040026 /* XSP LRCK Period 1 */ +#define CS43130_XSP_LRCK_PERIOD_2 0x040027 /* XSP LRCK Period 2 */ +#define CS43130_XSP_CLOCK_CONF 0x040028 /* XSP Clock Config */ +#define CS43130_XSP_FRAME_CONF 0x040029 /* XSP Frame Config */ +#define CS43130_ASP_CH_1_LOC 0x050000 /* ASP Chan 1 Location */ +#define CS43130_ASP_CH_2_LOC 0x050001 /* ASP Chan 2 Location */ +#define CS43130_ASP_CH_1_SZ_EN 0x05000A /* ASP Chan 1 Size, Enable */ +#define CS43130_ASP_CH_2_SZ_EN 0x05000B /* ASP Chan 2 Size, Enable */ +#define CS43130_XSP_CH_1_LOC 0x060000 /* XSP Chan 1 Location */ +#define CS43130_XSP_CH_2_LOC 0x060001 /* XSP Chan 2 Location */ +#define CS43130_XSP_CH_1_SZ_EN 0x06000A /* XSP Chan 1 Size, Enable */ +#define CS43130_XSP_CH_2_SZ_EN 0x06000B /* XSP Chan 2 Size, Enable */ +#define CS43130_DSD_VOL_B 0x070000 /* DSD Volume B */ +#define CS43130_DSD_VOL_A 0x070001 /* DSD Volume A */ +#define CS43130_DSD_PATH_CTL_1 0x070002 /* DSD Proc Path Sig Ctl 1 */ +#define CS43130_DSD_INT_CFG 0x070003 /* DSD Interface Config */ +#define CS43130_DSD_PATH_CTL_2 0x070004 /* DSD Proc Path Sig Ctl 2 */ +#define CS43130_DSD_PCM_MIX_CTL 0x070005 /* DSD and PCM Mixing Ctl */ +#define CS43130_DSD_PATH_CTL_3 0x070006 /* DSD Proc Path Sig Ctl 3 */ +#define CS43130_HP_OUT_CTL_1 0x080000 /* HP Output Ctl 1 */ +#define CS43130_DXD16 0x080024 /* DXD16 */ +#define CS43130_DXD13 0x080032 /* DXD13 */ +#define CS43130_PCM_FILT_OPT 0x090000 /* PCM Filter Option */ +#define CS43130_PCM_VOL_B 0x090001 /* PCM Volume B */ +#define CS43130_PCM_VOL_A 0x090002 /* PCM Volume A */ +#define CS43130_PCM_PATH_CTL_1 0x090003 /* PCM Path Signal Ctl 1 */ +#define CS43130_PCM_PATH_CTL_2 0x090004 /* PCM Path Signal Ctl 2 */ +#define CS43130_DXD6 0x090097 /* DXD6 */ +#define CS43130_CLASS_H_CTL 0x0B0000 /* Class H Ctl */ +#define CS43130_DXD15 0x0B0005 /* DXD15 */ +#define CS43130_DXD14 0x0B0006 /* DXD14 */ +#define CS43130_DXD3 0x0C0002 /* DXD3 */ +#define CS43130_DXD10 0x0C0003 /* DXD10 */ +#define CS43130_DXD11 0x0C0005 /* DXD11 */ +#define CS43130_DXD9 0x0C0006 /* DXD9 */ +#define CS43130_DXD4 0x0C0009 /* DXD4 */ +#define CS43130_DXD5 0x0C000E /* DXD5 */ +#define CS43130_HP_DETECT 0x0D0000 /* HP Detect */ +#define CS43130_HP_STATUS 0x0D0001 /* HP Status [RO] */ +#define CS43130_HP_LOAD_1 0x0E0000 /* HP Load 1 */ +#define CS43130_HP_MEAS_LOAD_1 0x0E0003 /* HP Load Measurement 1 */ +#define CS43130_HP_MEAS_LOAD_2 0x0E0004 /* HP Load Measurement 2 */ +#define CS43130_HP_DC_STAT_1 0x0E000D /* HP DC Load Status 0 [RO] */ +#define CS43130_HP_DC_STAT_2 0x0E000E /* HP DC Load Status 1 [RO] */ +#define CS43130_HP_AC_STAT_1 0x0E0010 /* HP AC Load Status 0 [RO] */ +#define CS43130_HP_AC_STAT_2 0x0E0011 /* HP AC Load Status 1 [RO] */ +#define CS43130_HP_LOAD_STAT 0x0E001A /* HP Load Status [RO] */ +#define CS43130_INT_STATUS_1 0x0F0000 /* Interrupt Status 1 */ +#define CS43130_INT_STATUS_2 0x0F0001 /* Interrupt Status 2 */ +#define CS43130_INT_STATUS_3 0x0F0002 /* Interrupt Status 3 */ +#define CS43130_INT_STATUS_4 0x0F0003 /* Interrupt Status 4 */ +#define CS43130_INT_STATUS_5 0x0F0004 /* Interrupt Status 5 */ +#define CS43130_INT_MASK_1 0x0F0010 /* Interrupt Mask 1 */ +#define CS43130_INT_MASK_2 0x0F0011 /* Interrupt Mask 2 */ +#define CS43130_INT_MASK_3 0x0F0012 /* Interrupt Mask 3 */ +#define CS43130_INT_MASK_4 0x0F0013 /* Interrupt Mask 4 */ +#define CS43130_INT_MASK_5 0x0F0014 /* Interrupt Mask 5 */ + +#define CS43130_MCLK_SRC_SEL_MASK 0x03 +#define CS43130_MCLK_SRC_SEL_SHIFT 0 +#define CS43130_MCLK_INT_MASK 0x04 +#define CS43130_MCLK_INT_SHIFT 2 +#define CS43130_CH_BITSIZE_MASK 0x03 +#define CS43130_CH_EN_MASK 0x04 +#define CS43130_CH_EN_SHIFT 2 +#define CS43130_ASP_BITSIZE_MASK 0x03 +#define CS43130_XSP_BITSIZE_MASK 0x0C +#define CS43130_XSP_BITSIZE_SHIFT 2 +#define CS43130_SP_BITSIZE_ASP_SHIFT 0 +#define CS43130_HP_DETECT_CTRL_SHIFT 6 +#define CS43130_HP_DETECT_CTRL_MASK (0x03 << CS43130_HP_DETECT_CTRL_SHIFT) +#define CS43130_HP_DETECT_INV_SHIFT 5 +#define CS43130_HP_DETECT_INV_MASK (1 << CS43130_HP_DETECT_INV_SHIFT) + +/* CS43130_INT_MASK_1 */ +#define CS43130_HP_PLUG_INT_SHIFT 6 +#define CS43130_HP_PLUG_INT (1 << CS43130_HP_PLUG_INT_SHIFT) +#define CS43130_HP_UNPLUG_INT_SHIFT 5 +#define CS43130_HP_UNPLUG_INT (1 << CS43130_HP_UNPLUG_INT_SHIFT) +#define CS43130_XTAL_RDY_INT_SHIFT 4 +#define CS43130_XTAL_RDY_INT_MASK 0x10 +#define CS43130_XTAL_RDY_INT (1 << CS43130_XTAL_RDY_INT_SHIFT) +#define CS43130_XTAL_ERR_INT_SHIFT 3 +#define CS43130_XTAL_ERR_INT (1 << CS43130_XTAL_ERR_INT_SHIFT) +#define CS43130_PLL_RDY_INT_MASK 0x04 +#define CS43130_PLL_RDY_INT_SHIFT 2 +#define CS43130_PLL_RDY_INT (1 << CS43130_PLL_RDY_INT_SHIFT) + +/* CS43130_INT_MASK_4 */ +#define CS43130_INT_MASK_ALL 0xFF +#define CS43130_HPLOAD_NO_DC_INT_SHIFT 7 +#define CS43130_HPLOAD_NO_DC_INT (1 << CS43130_HPLOAD_NO_DC_INT_SHIFT) +#define CS43130_HPLOAD_UNPLUG_INT_SHIFT 6 +#define CS43130_HPLOAD_UNPLUG_INT (1 << CS43130_HPLOAD_UNPLUG_INT_SHIFT) +#define CS43130_HPLOAD_OOR_INT_SHIFT 4 +#define CS43130_HPLOAD_OOR_INT (1 << CS43130_HPLOAD_OOR_INT_SHIFT) +#define CS43130_HPLOAD_AC_INT_SHIFT 3 +#define CS43130_HPLOAD_AC_INT (1 << CS43130_HPLOAD_AC_INT_SHIFT) +#define CS43130_HPLOAD_DC_INT_SHIFT 2 +#define CS43130_HPLOAD_DC_INT (1 << CS43130_HPLOAD_DC_INT_SHIFT) +#define CS43130_HPLOAD_OFF_INT_SHIFT 1 +#define CS43130_HPLOAD_OFF_INT (1 << CS43130_HPLOAD_OFF_INT_SHIFT) +#define CS43130_HPLOAD_ON_INT 1 + +/* CS43130_HP_LOAD_1 */ +#define CS43130_HPLOAD_EN_SHIFT 7 +#define CS43130_HPLOAD_EN (1 << CS43130_HPLOAD_EN_SHIFT) +#define CS43130_HPLOAD_CHN_SEL_SHIFT 4 +#define CS43130_HPLOAD_CHN_SEL (1 << CS43130_HPLOAD_CHN_SEL_SHIFT) +#define CS43130_HPLOAD_AC_START_SHIFT 1 +#define CS43130_HPLOAD_AC_START (1 << CS43130_HPLOAD_AC_START_SHIFT) +#define CS43130_HPLOAD_DC_START 1 + +/* Reg CS43130_SP_BITSIZE */ +#define CS43130_SP_BIT_SIZE_8 0x03 +#define CS43130_SP_BIT_SIZE_16 0x02 +#define CS43130_SP_BIT_SIZE_24 0x01 +#define CS43130_SP_BIT_SIZE_32 0x00 + +/* Reg CS43130_SP_CH_SZ_EN */ +#define CS43130_CH_BIT_SIZE_8 0x00 +#define CS43130_CH_BIT_SIZE_16 0x01 +#define CS43130_CH_BIT_SIZE_24 0x02 +#define CS43130_CH_BIT_SIZE_32 0x03 + +/* PLL */ +#define CS43130_PLL_START_MASK 0x01 +#define CS43130_PLL_MODE_MASK 0x02 +#define CS43130_PLL_MODE_SHIFT 1 + +#define CS43130_PLL_REF_PREDIV_MASK 0x3 + +#define CS43130_SP_STP_MASK 0x10 +#define CS43130_SP_STP_SHIFT 4 +#define CS43130_SP_5050_MASK 0x08 +#define CS43130_SP_5050_SHIFT 3 +#define CS43130_SP_FSD_MASK 0x07 + +#define CS43130_SP_MODE_MASK 0x10 +#define CS43130_SP_MODE_SHIFT 4 +#define CS43130_SP_SCPOL_OUT_MASK 0x08 +#define CS43130_SP_SCPOL_OUT_SHIFT 3 +#define CS43130_SP_SCPOL_IN_MASK 0x04 +#define CS43130_SP_SCPOL_IN_SHIFT 2 +#define CS43130_SP_LCPOL_OUT_MASK 0x02 +#define CS43130_SP_LCPOL_OUT_SHIFT 1 +#define CS43130_SP_LCPOL_IN_MASK 0x01 +#define CS43130_SP_LCPOL_IN_SHIFT 0 + +/* Reg CS43130_PWDN_CTL */ +#define CS43130_PDN_XSP_MASK 0x80 +#define CS43130_PDN_XSP_SHIFT 7 +#define CS43130_PDN_ASP_MASK 0x40 +#define CS43130_PDN_ASP_SHIFT 6 +#define CS43130_PDN_DSPIF_MASK 0x20 +#define CS43130_PDN_DSDIF_SHIFT 5 +#define CS43130_PDN_HP_MASK 0x10 +#define CS43130_PDN_HP_SHIFT 4 +#define CS43130_PDN_XTAL_MASK 0x08 +#define CS43130_PDN_XTAL_SHIFT 3 +#define CS43130_PDN_PLL_MASK 0x04 +#define CS43130_PDN_PLL_SHIFT 2 +#define CS43130_PDN_CLKOUT_MASK 0x02 +#define CS43130_PDN_CLKOUT_SHIFT 1 + +/* Reg CS43130_HP_OUT_CTL_1 */ +#define CS43130_HP_IN_EN_SHIFT 3 +#define CS43130_HP_IN_EN_MASK 0x08 + +/* Reg CS43130_PAD_INT_CFG */ +#define CS43130_ASP_3ST_MASK 0x01 +#define CS43130_XSP_3ST_MASK 0x02 + +/* Reg CS43130_PLL_SET_2 */ +#define CS43130_PLL_DIV_DATA_MASK 0x000000FF +#define CS43130_PLL_DIV_FRAC_0_DATA_SHIFT 0 + +/* Reg CS43130_PLL_SET_3 */ +#define CS43130_PLL_DIV_FRAC_1_DATA_SHIFT 8 + +/* Reg CS43130_PLL_SET_4 */ +#define CS43130_PLL_DIV_FRAC_2_DATA_SHIFT 16 + +/* Reg CS43130_SP_DEN_1 */ +#define CS43130_SP_M_LSB_DATA_MASK 0x00FF +#define CS43130_SP_M_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_DEN_2 */ +#define CS43130_SP_M_MSB_DATA_MASK 0xFF00 +#define CS43130_SP_M_MSB_DATA_SHIFT 8 + +/* Reg CS43130_SP_NUM_1 */ +#define CS43130_SP_N_LSB_DATA_MASK 0x00FF +#define CS43130_SP_N_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_NUM_2 */ +#define CS43130_SP_N_MSB_DATA_MASK 0xFF00 +#define CS43130_SP_N_MSB_DATA_SHIFT 8 + +/* Reg CS43130_SP_LRCK_HI_TIME_1 */ +#define CS43130_SP_LCHI_DATA_MASK 0x00FF +#define CS43130_SP_LCHI_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_LRCK_HI_TIME_2 */ +#define CS43130_SP_LCHI_MSB_DATA_SHIFT 8 + +/* Reg CS43130_SP_LRCK_PERIOD_1 */ +#define CS43130_SP_LCPR_DATA_MASK 0x00FF +#define CS43130_SP_LCPR_LSB_DATA_SHIFT 0 + +/* Reg CS43130_SP_LRCK_PERIOD_2 */ +#define CS43130_SP_LCPR_MSB_DATA_SHIFT 8 + +#define CS43130_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define CS43130_DOP_FORMATS (SNDRV_PCM_FMTBIT_DSD_U16_LE | \ + SNDRV_PCM_FMTBIT_DSD_U16_BE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* Reg CS43130_CRYSTAL_SET */ +#define CS43130_XTAL_IBIAS_MASK 0x07 + +/* Reg CS43130_PATH_CTL_1 */ +#define CS43130_MUTE_MASK 0x03 +#define CS43130_MUTE_EN 0x03 + +/* Reg CS43130_DSD_INT_CFG */ +#define CS43130_DSD_MASTER 0x04 + +/* Reg CS43130_DSD_PATH_CTL_2 */ +#define CS43130_DSD_SRC_MASK 0x60 +#define CS43130_DSD_SRC_SHIFT 5 +#define CS43130_DSD_EN_SHIFT 4 +#define CS43130_DSD_SPEED_MASK 0x04 +#define CS43130_DSD_SPEED_SHIFT 2 + +/* Reg CS43130_DSD_PCM_MIX_CTL */ +#define CS43130_MIX_PCM_PREP_SHIFT 1 +#define CS43130_MIX_PCM_PREP_MASK 0x02 + +#define CS43130_MIX_PCM_DSD_SHIFT 0 +#define CS43130_MIX_PCM_DSD_MASK 0x01 + +/* Reg CS43130_HP_MEAS_LOAD */ +#define CS43130_HP_MEAS_LOAD_MASK 0x000000FF +#define CS43130_HP_MEAS_LOAD_1_SHIFT 0 +#define CS43130_HP_MEAS_LOAD_2_SHIFT 8 + +#define CS43130_MCLK_22M 22579200 +#define CS43130_MCLK_24M 24576000 + +#define CS43130_LINEOUT_LOAD 5000 +#define CS43130_JACK_LINEOUT (SND_JACK_MECHANICAL | SND_JACK_LINEOUT) +#define CS43130_JACK_HEADPHONE (SND_JACK_MECHANICAL | \ + SND_JACK_HEADPHONE) +#define CS43130_JACK_MASK (SND_JACK_MECHANICAL | \ + SND_JACK_LINEOUT | \ + SND_JACK_HEADPHONE) + +enum cs43130_dsd_src { + CS43130_DSD_SRC_DSD = 0, + CS43130_DSD_SRC_ASP = 2, + CS43130_DSD_SRC_XSP = 3, +}; + +enum cs43130_asp_rate { + CS43130_ASP_SPRATE_32K = 0, + CS43130_ASP_SPRATE_44_1K, + CS43130_ASP_SPRATE_48K, + CS43130_ASP_SPRATE_88_2K, + CS43130_ASP_SPRATE_96K, + CS43130_ASP_SPRATE_176_4K, + CS43130_ASP_SPRATE_192K, + CS43130_ASP_SPRATE_352_8K, + CS43130_ASP_SPRATE_384K, +}; + +enum cs43130_mclk_src_sel { + CS43130_MCLK_SRC_EXT = 0, + CS43130_MCLK_SRC_PLL, + CS43130_MCLK_SRC_RCO +}; + +enum cs43130_mclk_int_freq { + CS43130_MCLK_24P5 = 0, + CS43130_MCLK_22P5, +}; + +enum cs43130_xtal_ibias { + CS43130_XTAL_UNUSED = -1, + CS43130_XTAL_IBIAS_15UA = 2, + CS43130_XTAL_IBIAS_12_5UA = 4, + CS43130_XTAL_IBIAS_7_5UA = 6, +}; + +enum cs43130_dai_id { + CS43130_ASP_PCM_DAI = 0, + CS43130_ASP_DOP_DAI, + CS43130_XSP_DOP_DAI, + CS43130_XSP_DSD_DAI, + CS43130_DAI_ID_MAX, +}; + +struct cs43130_clk_gen { + unsigned int mclk_int; + int fs; + u16 den; + u16 num; +}; + +/* frm_size = 16 */ +static const struct cs43130_clk_gen cs43130_16_clk_gen[] = { + {22579200, 32000, 441, 10,}, + {22579200, 44100, 32, 1,}, + {22579200, 48000, 147, 5,}, + {22579200, 88200, 16, 1,}, + {22579200, 96000, 147, 10,}, + {22579200, 176400, 8, 1,}, + {22579200, 192000, 147, 20,}, + {22579200, 352800, 4, 1,}, + {22579200, 384000, 147, 40,}, + {24576000, 32000, 48, 1,}, + {24576000, 44100, 5120, 147,}, + {24576000, 48000, 32, 1,}, + {24576000, 88200, 2560, 147,}, + {24576000, 96000, 16, 1,}, + {24576000, 176400, 1280, 147,}, + {24576000, 192000, 8, 1,}, + {24576000, 352800, 640, 147,}, + {24576000, 384000, 4, 1,}, +}; + +/* frm_size = 32 */ +static const struct cs43130_clk_gen cs43130_32_clk_gen[] = { + {22579200, 32000, 441, 20,}, + {22579200, 44100, 16, 1,}, + {22579200, 48000, 147, 10,}, + {22579200, 88200, 8, 1,}, + {22579200, 96000, 147, 20,}, + {22579200, 176400, 4, 1,}, + {22579200, 192000, 147, 40,}, + {22579200, 352800, 2, 1,}, + {22579200, 384000, 147, 80,}, + {24576000, 32000, 24, 1,}, + {24576000, 44100, 2560, 147,}, + {24576000, 48000, 16, 1,}, + {24576000, 88200, 1280, 147,}, + {24576000, 96000, 8, 1,}, + {24576000, 176400, 640, 147,}, + {24576000, 192000, 4, 1,}, + {24576000, 352800, 320, 147,}, + {24576000, 384000, 2, 1,}, +}; + +/* frm_size = 48 */ +static const struct cs43130_clk_gen cs43130_48_clk_gen[] = { + {22579200, 32000, 147, 100,}, + {22579200, 44100, 32, 3,}, + {22579200, 48000, 49, 5,}, + {22579200, 88200, 16, 3,}, + {22579200, 96000, 49, 10,}, + {22579200, 176400, 8, 3,}, + {22579200, 192000, 49, 20,}, + {22579200, 352800, 4, 3,}, + {22579200, 384000, 49, 40,}, + {24576000, 32000, 16, 1,}, + {24576000, 44100, 5120, 441,}, + {24576000, 48000, 32, 3,}, + {24576000, 88200, 2560, 441,}, + {24576000, 96000, 16, 3,}, + {24576000, 176400, 1280, 441,}, + {24576000, 192000, 8, 3,}, + {24576000, 352800, 640, 441,}, + {24576000, 384000, 4, 3,}, +}; + +/* frm_size = 64 */ +static const struct cs43130_clk_gen cs43130_64_clk_gen[] = { + {22579200, 32000, 441, 40,}, + {22579200, 44100, 8, 1,}, + {22579200, 48000, 147, 20,}, + {22579200, 88200, 4, 1,}, + {22579200, 96000, 147, 40,}, + {22579200, 176400, 2, 1,}, + {22579200, 192000, 147, 80,}, + {22579200, 352800, 1, 1,}, + {24576000, 32000, 12, 1,}, + {24576000, 44100, 1280, 147,}, + {24576000, 48000, 8, 1,}, + {24576000, 88200, 640, 147,}, + {24576000, 96000, 4, 1,}, + {24576000, 176400, 320, 147,}, + {24576000, 192000, 2, 1,}, + {24576000, 352800, 160, 147,}, + {24576000, 384000, 1, 1,}, +}; + +struct cs43130_bitwidth_map { + unsigned int bitwidth; + u8 sp_bit; + u8 ch_bit; +}; + +struct cs43130_rate_map { + int fs; + int val; +}; + +#define HP_LEFT 0 +#define HP_RIGHT 1 +#define CS43130_AC_FREQ 10 +#define CS43130_DC_THRESHOLD 2 + +#define CS43130_NUM_SUPPLIES 5 +static const char *const cs43130_supply_names[CS43130_NUM_SUPPLIES] = { + "VA", + "VP", + "VCP", + "VD", + "VL", +}; + +#define CS43130_NUM_INT 5 /* number of interrupt status reg */ + +struct cs43130_dai { + unsigned int sclk; + unsigned int dai_format; + unsigned int dai_mode; +}; + +struct cs43130_private { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct regulator_bulk_data supplies[CS43130_NUM_SUPPLIES]; + struct gpio_desc *reset_gpio; + unsigned int dev_id; /* codec device ID */ + int xtal_ibias; + + /* shared by both DAIs */ + struct mutex clk_mutex; + int clk_req; + bool pll_bypass; + struct completion xtal_rdy; + struct completion pll_rdy; + unsigned int mclk; + unsigned int mclk_int; + int mclk_int_src; + + /* DAI specific */ + struct cs43130_dai dais[CS43130_DAI_ID_MAX]; + + /* HP load specific */ + bool dc_meas; + bool ac_meas; + bool hpload_done; + struct completion hpload_evt; + unsigned int hpload_stat; + u16 hpload_dc[2]; + u16 dc_threshold[CS43130_DC_THRESHOLD]; + u16 ac_freq[CS43130_AC_FREQ]; + u16 hpload_ac[CS43130_AC_FREQ][2]; + struct workqueue_struct *wq; + struct work_struct work; + struct snd_soc_jack jack; +}; + +#endif /* __CS43130_H__ */ -- 1.9.1

3 4

[alsa-devel] [PATCH v2 2/2] ASoC: cs43130: Add devicetree bindings for CS43130
by Li Xu 01 Sep '17

01 Sep '17

Add devicetree bindings documentation file for Cirrus Logic CS43130 codec. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs43130.txt | 67 ++++++++++++++++++++++ 1 file changed, 67 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs43130.txt diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 0000000..8b1dd5a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,67 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130", "cirrus,cs4399", "cirrus,cs43131", + "cirrus,cs43198" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + + - cirrus,dc-measure: + Boolean, define to enable headphone DC impedance measurement. + + - cirrus,ac-measure: + Boolean, define to enable headphone AC impedance measurement. + DC impedance must also be enabled for AC impedance measurement. + + - cirrus,dc-threshold: + Define 2 DC impedance thresholds in ohms for HP output control. + Default values are 50 and 120 Ohms. + + - cirrus,ac-freq: + Define the frequencies at which to measure HP AC impedance. + Only used if "cirrus,dc-measure" is defined. + Exactly 10 frequencies must be defined. + If this properties is undefined, by default, + following frequencies are used: + <24 43 93 200 431 928 2000 4309 9283 20000> + The above frequencies are logarithmically equally spaced. + Log base is 10. + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 0>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + cirrus,dc-measure; + cirrus,ac-measure; + cirrus,dc-threshold = /bits/ 16 <20 100>; + cirrus,ac-freq = /bits/ 16 <24 43 93 200 431 928 2000 4309 9283 20000>; +}; -- 1.9.1

3 3

[alsa-devel] Applied "ASoC: add missing compile rule for sirf-audio-codec" to the asoc tree
by Mark Brown 01 Sep '17

01 Sep '17

The patch ASoC: add missing compile rule for sirf-audio-codec has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2dc3b7b06bace6d8986a5fae6569ef30276cdd73 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 1 Sep 2017 04:33:27 +0000 Subject: [PATCH] ASoC: add missing compile rule for sirf-audio-codec It is still using old driver style, this patch also fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 10 ++++++---- 2 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b4d576dab48a..8ed3c9256a74 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -373,6 +373,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 6bfd25c289d1..01d7690a1520 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -432,10 +432,12 @@ static int sirf_audio_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = { .probe = sirf_audio_codec_probe, .remove = sirf_audio_codec_remove, - .dapm_widgets = sirf_audio_codec_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets), - .dapm_routes = sirf_audio_codec_map, - .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map), + .component_driver = { + .dapm_widgets = sirf_audio_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets), + .dapm_routes = sirf_audio_codec_map, + .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map), + }, .idle_bias_off = true, }; -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: add missing compile rule for sirf-audio-codec
by Kuninori Morimoto 01 Sep '17

01 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is still using old driver style, this patch also fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 10 ++++++---- 2 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d93318e..5a6f841 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -376,6 +376,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 50bc222..7ae8c18 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -432,10 +432,12 @@ static int sirf_audio_codec_remove(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = { .probe = sirf_audio_codec_probe, .remove = sirf_audio_codec_remove, - .dapm_widgets = sirf_audio_codec_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets), - .dapm_routes = sirf_audio_codec_map, - .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map), + .component_driver = { + .dapm_widgets = sirf_audio_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets), + .dapm_routes = sirf_audio_codec_map, + .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map), + }, .idle_bias_off = true, }; -- 1.9.1

2 1

Re: [alsa-devel] ALC700 in the Intel chipset
by Pshou 01 Sep '17

01 Sep '17

Hi alsa-devel ML: Sorry , this file is right. Best regards pshou -----Original Message----- From: Pshou Sent: Friday, September 01, 2017 4:55 PM To: 'Takashi Iwai'; 'alsa-devel(a)alsa-project.org' Cc: CHLin; Kailang; Singh, Guneshwor O Subject: RE: ALC700 in the Intel chipset Hi alsa-devel ML: Could you merge this 000-alc700-intel.patch file to your ALSA driver? I changed "Intel debug" to "Intel Reference board". Best regards Pshou -----Original Message----- From: Takashi Iwai [mailto:tiwai@suse.de] Sent: Friday, September 01, 2017 4:13 PM To: Pshou Cc: CHLin; Kailang; Singh, Guneshwor O Subject: Re: ALC700 in the Intel chipset On Fri, 01 Sep 2017 10:06:13 +0200, Pshou wrote: > > Hi Takashi: > > This is 0000-alc700-intel.patch file. Thanks, it looks good now. One thing to be changed is the name "Intel debug". Maybe "Intel Reference board" or such would fit better. In anyway, could you submit the patch to alsa-devel ML? Once after it's submitted, I'll merge it. thanks, Takashi > > Best regards > pshou > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Tuesday, August 22, 2017 5:04 PM > To: CHLin > Cc: Pshou; Kailang; Singh, Guneshwor O > Subject: Re: ALC700 in the Intel chipset > > On Tue, 22 Aug 2017 10:05:18 +0200, > CHLin wrote: > > > > Hi Takashi > > > > ALC 700 needs this patch for jack detection function. > > Because ALC700's jack detect function defaults is disable. > > So alc700 needs pathc to enable jack detection function. > > OK, then could you update the patch description for giving more such information, prepare the proper patch in a form described in Documentation/SubmittingPatches with signed-off-by tags? > This is mandatory for merging to upstream. > > > thanks, > > Takashi > > > Thanks. > > > > ======================= > > Realtek Semiconductor Corp. > > 林家賢 CH_Lin > > System Design Dept II PCBG > > Tel: 886-3-578-0211 ext:13162 > > ======================= > > > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Monday, August 21, 2017 6:23 PM > > To: Pshou > > Cc: Kailang; Singh, Guneshwor O; CHLin > > Subject: Re: ALC700 in the Intel chipset > > > > On Mon, 21 Aug 2017 10:05:25 +0200, > > Pshou wrote: > > > > > > Hi Tiwai: > > > > > > Nice to meet you. > > > > > > This is the case of intel. > > > Fixing Jack detection on ALC700 > > > > Could you give more information about this patch? > > For which platform and for what problem? At best, please add more information in the patch changelog itself. > > > > > > thanks, > > > > Takashi > > > > ------Please consider the environment before printing this e-mail.

1 0

[alsa-devel] [PATCH] ALSA: ac97c: Fix an error handling path in 'atmel_ac97c_probe()'
by Christophe JAILLET 01 Sep '17

01 Sep '17

If 'clk_prepare_enable()' fails, we must release some resources before returning. Add a new label in the existing error handling path and 'goto' there. Fixes: 260ea95cc027 ("ASoC: atmel: ac97c: Handle return value of clk_prepare_enable.") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/atmel/ac97c.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 30c64ab210d9..5ffefac2fa8f 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -785,7 +785,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) } retval = clk_prepare_enable(pclk); if (retval) - return retval; + goto err_prepare_enable; retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -881,6 +881,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) snd_card_free(card); err_snd_card_new: clk_disable_unprepare(pclk); +err_prepare_enable: clk_put(pclk); return retval; } -- 2.11.0

6 19

[alsa-devel] [PATCH v1 0/4] *** add XRUN injections suppport in alsabat ***
by Zhang Keqiao 31 Aug '17

31 Aug '17

The patch set add a new feature in alsabat, now alsabat can test XRUNs for both playback and capture by writing to proc file '/proc/sound/card*/pcm*/sub*/xrun_injection'. Zhang Keqiao (4): alsabat: add one option for XRUN injections alsabat: parse the card and PCM ID for XRUN injections path alsabat: add XRUN injections function alsabat: XRUN injections test criteria bat/analyze.c | 22 ++++++-- bat/bat.c | 171 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++- bat/common.h | 13 +++++ 3 files changed, 202 insertions(+), 4 deletions(-) -- 2.9.3

2 7

[alsa-devel] [PATCH] ALSA: ctxfi: Remove null check before kfree
by Himanshu Jha 31 Aug '17

31 Aug '17

kfree on NULL pointer is a no-op and therefore checking is redundant. Signed-off-by: Himanshu Jha <himanshujha199640(a)gmail.com> --- sound/pci/ctxfi/ctresource.c | 6 ++---- sound/pci/ctxfi/ctsrc.c | 6 ++---- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index c5124c3..80c4d84 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -258,10 +258,8 @@ int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, int rsc_mgr_uninit(struct rsc_mgr *mgr) { - if (NULL != mgr->rscs) { - kfree(mgr->rscs); - mgr->rscs = NULL; - } + kfree(mgr->rscs); + mgr->rscs = NULL; if ((NULL != mgr->hw) && (NULL != mgr->ctrl_blk)) { switch (mgr->type) { diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index a5a72df..bb4c9c3 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -702,10 +702,8 @@ static int srcimp_rsc_init(struct srcimp *srcimp, static int srcimp_rsc_uninit(struct srcimp *srcimp) { - if (NULL != srcimp->imappers) { - kfree(srcimp->imappers); - srcimp->imappers = NULL; - } + kfree(srcimp->imappers); + srcimp->imappers = NULL; srcimp->ops = NULL; srcimp->mgr = NULL; rsc_uninit(&srcimp->rsc); -- 2.7.4

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.13
by Mark Brown 31 Aug '17

31 Aug '17

The following changes since commit 5771a8c08880cdca3bfb4a3fc6d309d6bba20877: Linux v4.13-rc1 (2017-07-15 15:22:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.13-rc7 for you to fetch changes up to 05997302b7aa56a7f4a7b9d49d0661865c17874d: Merge remote-tracking branch 'asoc/fix/rt5670' into asoc-fixes (2017-08-31 12:49:46 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.13 A couple of fixes, one for a regression in simple-card introduced during the merge window that was only reported this week and another for a regression in registration of ACPI GPIOs. ---------------------------------------------------------------- Lucas Stach (1): ASoC: simple_card_utils: fix fallback when "label" property isn't present Mark Brown (1): Merge remote-tracking branch 'asoc/fix/rt5670' into asoc-fixes Takashi Iwai (1): ASoC: rt5670: Fix GPIO headset detection regression sound/soc/codecs/rt5670.c | 2 +- sound/soc/generic/simple-card-utils.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++++++++++ 3 files changed, 12 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH 1/7] staging: max98927: Added TDM support
by Ryan Lee 31 Aug '17

31 Aug '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 141 ++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98927.h | 6 +- 2 files changed, 120 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b5ee294..99d6e41 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +236,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + } return 0; } @@ -245,6 +252,21 @@ static const int rate_table[] = { 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,19 +292,17 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: + if ((max98927->iface == SND_SOC_DAIFMT_DSP_A) || + (max98927->iface == SND_SOC_DAIFMT_DSP_B)) + return 0; + + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "BCLK %d not supported\n", blr_clk_ratio); return -EINVAL; } + regmap_update_bits(max98927->regmap, MAX98927_R0022_PCM_CLK_SETUP, MAX98927_PCM_CLK_SETUP_BSEL_MASK, @@ -386,6 +406,76 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +495,7 @@ static const struct snd_soc_dai_ops max98927_dai_ops = { .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..bf7a6f92 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 2.7.4

2 9

[alsa-devel] [PATCH 4/5] staging: max98927: Added missing \n to end of dev_err messages
by Ryan Lee 31 Aug '17

31 Aug '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index e787a03..9b864e2 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -160,7 +160,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) mode = MAX98927_PCM_MASTER_MODE_MASTER; break; default: - dev_err(codec->dev, "DAI clock mode unsupported"); + dev_err(codec->dev, "DAI clock mode unsupported\n"); return -EINVAL; } @@ -176,7 +176,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) invert = MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE; break; default: - dev_err(codec->dev, "DAI invert mode unsupported"); + dev_err(codec->dev, "DAI invert mode unsupported\n"); return -EINVAL; } @@ -331,7 +331,7 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; break; default: - dev_err(codec->dev, "format unsupported %d", + dev_err(codec->dev, "format unsupported %d\n", params_format(params)); goto err; } -- 2.7.4

2 1

[alsa-devel] [PATCH 1/9] ASoC: max98927: Added TDM support
by Ryan Lee 31 Aug '17

31 Aug '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Added code to support TDM mode sound/soc/codecs/max98927.c | 141 ++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98927.h | 6 +- 2 files changed, 120 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b0380b5..951cc05 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +236,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + } return 0; } @@ -245,6 +252,21 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,19 +292,17 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: + if ((max98927->iface == SND_SOC_DAIFMT_DSP_A) || + (max98927->iface == SND_SOC_DAIFMT_DSP_B)) + return 0; + + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "BCLK %d not supported\n", blr_clk_ratio); return -EINVAL; } + regmap_update_bits(max98927->regmap, MAX98927_R0022_PCM_CLK_SETUP, MAX98927_PCM_CLK_SETUP_BSEL_MASK, @@ -386,6 +406,76 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +495,7 @@ static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..bf7a6f92 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 1.9.1

2 12

[alsa-devel] [PATCH v5 0/8] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 31 Aug '17

31 Aug '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 I don't own a A33 device which uses the i2s block for the audio codec so if someone could test against that it would be much appreciated. I'm also wondering if there is a preferred way of setting the lrclk size in the dts?..currently it is set to the sample width but for example the pcm5102a wants it to be 32 bits whatever the sample rate. Thanks in advance, CK --- v5 changes compared to v4 are: - fixed bad logic in fmt0 set LRCK period define. - fixed PTR_ERR checks in init of regmap fields - added reviewed-by to commit messages - removed delivered patches v4 changes compared to v3 are: - moved clkdiv variant adjustment out of function - used PTR_ERR_OR_ZERO for checks - tidy up of extra lines and lines over 80 chars. - reduced names of polarity, wss and sr reg fields. - added reviewed-by to commit messages - added comments for functionality that hasn't been implemented yet. - removed delivered patches v3 changes compared to v2 are: - initial changes to prepare driver for newer SoCs has been broken down into smaller patches - reduce use of regmap fields to where just needed. - clkdiv expansion will be delivered later. - defines for H3 variant segregated. - fixed regmap config issue with SUN8I_I2S_FIFO_TX_REG. v2 changes compared to v1 are: - massive refactoring to remove duplicate code making use of regmap_fields. - extending the clock divisors. - removed code that should be delivered when we support 20/24bits --- Marcus Cooper (8): ASoC: sun4i-i2s: Add regmap fields for channels ASoC: sun4i-i2s: Add regfields for word size select and sample resolution ASoC: sun4i-i2s: bclk and lrclk polarity tidyup ASoC: sun4i-i2s: Add mclk enable regmap field ASoC: sun4i-i2s: Add regmap field to set DAI format ASoC: sun4i-i2s: Check for slave select bit ASoC: sun4i-i2s: Update global enable with bitmask ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 406 ++++++++++++++++++--- 2 files changed, 357 insertions(+), 51 deletions(-) -- 2.14.1

3 14

[alsa-devel] [PATCH 0/6] Put missing KERN_* prefix
by Takashi Iwai 31 Aug '17

31 Aug '17

Hi, it's a trivial fix patch set for missing KERN_* prefix to some debug codes found in various drivers. Takashi === Takashi Iwai (6): ALSA: opl3: Put missing KERN_CONT prefix ALSA: vx: Put missing KERN_CONT prefix ALSA: asihpi: Put missing KERN_CONT prefix ALSA: usb-audio: Put missing KERN_CONT prefix ALSA: usx2y: Put missing KERN_CONT prefix ALSA: sh: Put missing KERN_* prefix sound/drivers/opl3/opl3_midi.c | 4 ++-- sound/drivers/vx/vx_core.c | 4 ++-- sound/pci/asihpi/hpidebug.c | 4 ++-- sound/sh/sh_dac_audio.c | 2 +- sound/usb/midi.c | 4 ++-- sound/usb/usx2y/usb_stream.c | 4 ++-- 6 files changed, 11 insertions(+), 11 deletions(-) -- 2.14.1

1 6

[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: Support for OCMV configuration
by Peter Ujfalusi 31 Aug '17

31 Aug '17

In aic3x class of devices Output Common-Mode Voltage can be configured for better analog performance. The OCMV value depends on the Analog and digital domain power supply voltage configuration. The default OCMV of 1.35V gives best performance when AVDD is around 2.7V and DVDD is 1.525V, but for higher AVDD/DVDD higher OCMV setting is recommended. The patch gives an automatic way of guessing the best OCMV which can be overwritten by a DT parameter if needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Change since v1: - Correction for the lower supported voltage range (when OCMV would be set to 1.35V) Regards, Peter .../devicetree/bindings/sound/tlv320aic3x.txt | 5 +++ sound/soc/codecs/tlv320aic3x.c | 45 ++++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.h | 8 ++++ 3 files changed, 58 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 47a213c411ce..ba5b45c483f5 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -26,6 +26,11 @@ Optional properties: 3 - MICBIAS output is connected to AVDD, If this node is not mentioned or if the value is incorrect, then MicBias is powered down. +- ai3x-ocmv - Output Common-Mode Voltage selection: + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V - AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the device as covered in Documentation/devicetree/bindings/regulator/regulator.txt diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 405f4602888a..06f92571eba4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -93,6 +93,8 @@ struct aic3x_priv { /* Selects the micbias voltage */ enum aic3x_micbias_voltage micbias_vg; + /* Output Common-Mode Voltage */ + u8 ocmv; }; static const struct reg_default aic3x_reg[] = { @@ -1572,6 +1574,10 @@ static int aic3x_init(struct snd_soc_codec *codec) break; } + /* Output common-mode voltage = 1.5 V */ + snd_soc_update_bits(codec, HPOUT_SC, HPOUT_SC_OCMV_MASK, + aic3x->ocmv << HPOUT_SC_OCMV_SHIFT); + return 0; } @@ -1699,6 +1705,43 @@ static const struct snd_soc_codec_driver soc_codec_dev_aic3x = { }, }; +static void aic3x_configure_ocmv(struct i2c_client *client) +{ + struct device_node *np = client->dev.of_node; + struct aic3x_priv *aic3x = i2c_get_clientdata(client); + u32 value; + int dvdd, avdd; + + if (np && !of_property_read_u32(np, "ai3x-ocmv", &value)) { + /* OCMV setting is forced by DT */ + if (value <= 3) { + aic3x->ocmv = value; + return; + } + } + + dvdd = regulator_get_voltage(aic3x->supplies[1].consumer); + avdd = regulator_get_voltage(aic3x->supplies[2].consumer); + + if (avdd > 3600000 || dvdd > 1950000) { + dev_warn(&client->dev, + "Too high supply voltage(s) AVDD: %d, DVDD: %d\n", + avdd, dvdd); + } else if (avdd == 3600000 && dvdd == 1950000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_8V; + } else if (avdd > 3300000 && dvdd > 1800000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_65V; + } else if (avdd > 3000000 && dvdd > 1650000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_5V; + } else if (avdd >= 2700000 && dvdd >= 1525000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_35V; + } else { + dev_warn(&client->dev, + "Invalid supply voltage(s) AVDD: %d, DVDD: %d\n", + avdd, dvdd); + } +} + /* * AIC3X 2 wire address can be up to 4 devices with device addresses * 0x18, 0x19, 0x1A, 0x1B @@ -1816,6 +1859,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_gpio; } + aic3x_configure_ocmv(i2c); + if (aic3x->model == AIC3X_MODEL_3007) { ret = regmap_register_patch(aic3x->regmap, aic3007_class_d, ARRAY_SIZE(aic3007_class_d)); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 89fa692df206..34c35196aa0d 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -243,6 +243,14 @@ #define MICBIAS_LEVEL_SHIFT (6) #define MICBIAS_LEVEL_MASK (3 << 6) +/* HPOUT_SC */ +#define HPOUT_SC_OCMV_MASK (3 << 6) +#define HPOUT_SC_OCMV_SHIFT (6) +#define HPOUT_SC_OCMV_1_35V 0 +#define HPOUT_SC_OCMV_1_5V 1 +#define HPOUT_SC_OCMV_1_65V 2 +#define HPOUT_SC_OCMV_1_8V 3 + /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) -- 2.14.1 Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH v3] ASoC: rt5645: Add jack detection workaround for MINIX Z83-4 based devices
by Ian W MORRISON 31 Aug '17

31 Aug '17

The MINIX NEO Z83-4 and MINIX NEO Z83-4 Pro devices requires jd_mode=3 to make the jack detection work. Using a BIOS DMI product of "Z83-4" will match both devices of 'NEO Z83-4' and 'Z83-4 Pro'. Signed-off-by: Ian W Morrison <ianwmorrison(a)gmail.com> --- v3: Tabs are now tabs and not spaces v2: Applied and tested again sound -b for-next and linux-next -b master v1: Applied and tested again linux -b master --- sound/soc/codecs/rt5645.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index cd82302..6a7778a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3655,6 +3655,21 @@ static int rt5645_resume(struct snd_soc_codec *codec) { } }; +static struct rt5645_platform_data minix_z83_4_platform_data = { + .jd_mode = 3, +}; + +static struct dmi_system_id dmi_platform_minix_z83_4[] = { + { + .ident = "MINIX Z83-4", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"), + DMI_MATCH(DMI_PRODUCT_NAME, "Z83-4"), + }, + }, + { } +}; + static bool rt5645_check_dp(struct device *dev) { if (device_property_present(dev, "realtek,in2-differential") || @@ -3709,6 +3724,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, rt5645->pdata = gpd_win_platform_data; else if (dmi_check_system(dmi_platform_asus_t100ha)) rt5645->pdata = general_platform_data2; + else if (dmi_check_system(dmi_platform_minix_z83_4)) + rt5645->pdata = minix_z83_4_platform_data; if (quirk != -1) { rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Support for OCMV configuration
by Peter Ujfalusi 31 Aug '17

31 Aug '17

In aic3x class of devices Output Common-Mode Voltage can be configured for better analog performance. The OCMV value depends on the Analog and digital domain power supply voltage configuration. The default OCMV of 1.35V gives best performance when AVDD is around 2.7V and DVDD is 1.525V, but for higher AVDD/DVDD higher OCMV setting is recommended. The patch gives an automatic way of guessing the best OCMV which can be overwritten by a DT parameter if needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- .../devicetree/bindings/sound/tlv320aic3x.txt | 5 +++ sound/soc/codecs/tlv320aic3x.c | 45 ++++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.h | 8 ++++ 3 files changed, 58 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 47a213c411ce..ba5b45c483f5 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -26,6 +26,11 @@ Optional properties: 3 - MICBIAS output is connected to AVDD, If this node is not mentioned or if the value is incorrect, then MicBias is powered down. +- ai3x-ocmv - Output Common-Mode Voltage selection: + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V - AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the device as covered in Documentation/devicetree/bindings/regulator/regulator.txt diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 405f4602888a..4f24d7552968 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -93,6 +93,8 @@ struct aic3x_priv { /* Selects the micbias voltage */ enum aic3x_micbias_voltage micbias_vg; + /* Output Common-Mode Voltage */ + u8 ocmv; }; static const struct reg_default aic3x_reg[] = { @@ -1572,6 +1574,10 @@ static int aic3x_init(struct snd_soc_codec *codec) break; } + /* Output common-mode voltage = 1.5 V */ + snd_soc_update_bits(codec, HPOUT_SC, HPOUT_SC_OCMV_MASK, + aic3x->ocmv << HPOUT_SC_OCMV_SHIFT); + return 0; } @@ -1699,6 +1705,43 @@ static const struct snd_soc_codec_driver soc_codec_dev_aic3x = { }, }; +static void aic3x_configure_ocmv(struct i2c_client *client) +{ + struct device_node *np = client->dev.of_node; + struct aic3x_priv *aic3x = i2c_get_clientdata(client); + u32 value; + int dvdd, avdd; + + if (np && !of_property_read_u32(np, "ai3x-ocmv", &value)) { + /* OCMV setting is forced by DT */ + if (value <= 3) { + aic3x->ocmv = value; + return; + } + } + + dvdd = regulator_get_voltage(aic3x->supplies[1].consumer); + avdd = regulator_get_voltage(aic3x->supplies[2].consumer); + + if (avdd > 3600000 || dvdd > 1950000) { + dev_warn(&client->dev, + "Too high supply voltage(s) AVDD: %d, DVDD: %d\n", + avdd, dvdd); + } else if (avdd == 3600000 && dvdd == 1950000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_8V; + } else if (avdd > 3300000 && dvdd > 1800000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_65V; + } else if (avdd > 3000000 && dvdd > 1650000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_5V; + } else if (avdd > 2700000 && dvdd > 1525000) { + aic3x->ocmv = HPOUT_SC_OCMV_1_35V; + } else { + dev_warn(&client->dev, + "Invalid supply voltage(s) AVDD: %d, DVDD: %d\n", + avdd, dvdd); + } +} + /* * AIC3X 2 wire address can be up to 4 devices with device addresses * 0x18, 0x19, 0x1A, 0x1B @@ -1816,6 +1859,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_gpio; } + aic3x_configure_ocmv(i2c); + if (aic3x->model == AIC3X_MODEL_3007) { ret = regmap_register_patch(aic3x->regmap, aic3007_class_d, ARRAY_SIZE(aic3007_class_d)); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 89fa692df206..34c35196aa0d 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -243,6 +243,14 @@ #define MICBIAS_LEVEL_SHIFT (6) #define MICBIAS_LEVEL_MASK (3 << 6) +/* HPOUT_SC */ +#define HPOUT_SC_OCMV_MASK (3 << 6) +#define HPOUT_SC_OCMV_SHIFT (6) +#define HPOUT_SC_OCMV_1_35V 0 +#define HPOUT_SC_OCMV_1_5V 1 +#define HPOUT_SC_OCMV_1_65V 2 +#define HPOUT_SC_OCMV_1_8V 3 + /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) -- 2.14.1 Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 1

[alsa-devel] [PATCH v2] ASoC: rt5645: Add jack detection workaround for MINIX Z83-4 based devices
by Ian W MORRISON 31 Aug '17

31 Aug '17

The MINIX NEO Z83-4 and MINIX NEO Z83-4 Pro devices requires jd_mode=3 to make the jack detection work. Using a BIOS DMI product of "Z83-4" will match both devices of 'NEO Z83-4' and 'Z83-4 Pro'. Signed-off-by: Ian W Morrison <ianwmorrison(a)gmail.com> --- v2: Applied and tested again sound -b for-next and linux-next -b master v1: Applied and tested again linux -b master --- sound/soc/codecs/rt5645.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index cd82302..6a7778a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3655,6 +3655,21 @@ static int rt5645_resume(struct snd_soc_codec *codec) { } }; +static struct rt5645_platform_data minix_z83_4_platform_data = { + .jd_mode = 3, +}; + +static struct dmi_system_id dmi_platform_minix_z83_4[] = { + { + .ident = "MINIX Z83-4", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"), + DMI_MATCH(DMI_PRODUCT_NAME, "Z83-4"), + }, + }, + { } +}; + static bool rt5645_check_dp(struct device *dev) { if (device_property_present(dev, "realtek,in2-differential") || @@ -3709,6 +3724,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, rt5645->pdata = gpd_win_platform_data; else if (dmi_check_system(dmi_platform_asus_t100ha)) rt5645->pdata = general_platform_data2; + else if (dmi_check_system(dmi_platform_minix_z83_4)) + rt5645->pdata = minix_z83_4_platform_data; if (quirk != -1) { rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); -- 1.9.1

2 3

[alsa-devel] [RESENT PATCH] ASoC: hdmi-codec: Use different name for playback streams
by Jeffy Chen 31 Aug '17

31 Aug '17

Currently the hdmi i2s playback stream and hdmi spdif playback stream are using the same name. So when they are enabled at the same time, kernel will print this warning: [ 2.201835] hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: Failed to create Playback debugfs file Assign different names to them to avoid that. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/codecs/hdmi-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 509ab51..a2af440 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -696,7 +696,7 @@ static struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, .playback = { - .stream_name = "Playback", + .stream_name = "I2S Playback", .channels_min = 2, .channels_max = 8, .rates = HDMI_RATES, @@ -711,7 +711,7 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, .playback = { - .stream_name = "Playback", + .stream_name = "SPDIF Playback", .channels_min = 2, .channels_max = 2, .rates = HDMI_RATES, -- 2.1.4

2 1

[alsa-devel] USB dB data
by Jack Bates 30 Aug '17

30 Aug '17

I confirmed with Wireshark that my USB sound card responds to GET_MIN with 0x0000 and GET_MAX with 0x6300 .. The audio device class definition [1] says the units are 1/256 dB (section 5.2.2.2.3: Mixer Control) ... So by my calculation, the min and max are 0 and 99 dB (0x6300 / 256 == 99) ... Why then does alsactl print: > range '0 - 99' > dbmin 0 > dbmax 38 (/usr/sbin/aslactl -f - store) I originally asked about this on the pulseaudio-discuss list [2] because it uses "dB data" [3]. The "0 - 99" makes sense -- it lines up with my calculation ... Where does the 38 come from? (I expect it's obvious, I just haven't found the explanation?) Background: I just got a Bose SoundLink Revolve ... When I connect it with USB, the volume is silent from 0 to 99% and deafening at 100% (using PulseAudio). Directly adjusting the volume with alsamixer works properly: It increases smoothly from 0 to 100% -- changes that are reflected in the PulseAudio volume as fine adjustments in between 99 and 100% (64587 to 65536 PulseAudio volume). From what I've read there's a common problem where especially USB devices report the wrong dB data (don't follow the spec [4]) -- this has previously been handled by patching the USB audio driver [5] ... However I haven't figured out how my device deviates from the spec (0x0000 to 0x6300 or 0 to 99 dB seem plausible) -- my understanding is incomplete: I haven't figured out where the value 38 comes from ... [1] http://www.usb.org/developers/docs/devclass_docs/audio10.pdf [2] https://lists.freedesktop.org/archives/pulseaudio-discuss/2017-August/02860… [3] http://0pointer.de/blog/projects/decibel-data [4] https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/637066 [5] https://bugs.freedesktop.org/show_bug.cgi?id=93193#c14

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[alsa-devel] Unable to play MP3 using TinyCompress and bytcr-rt5640
by Lambdadroid 30 Aug '17

30 Aug '17

Hi, I'm attempting to use TinyCompress to play a MP3 file on a Baytrail tablet (using bytcr-rt5640 on Linux 4.13-rc7) with ALSA's Compress-Offload API. After finding the right quirks and setting up the UCM files normal audio output is working, but I'm unable to play a MP3 file using the `cplay` tool included with TinyCompress. Note that `cplay` appears to be broken as of commit 50c2b03 "cplay: check the values of fread". fread returns the number of items that were read but the commit assumes it returns the number of bytes. I've simply changed the check to `read != 1` to fix it. When I play a MP3 file after fixing that I get a cryptic SST firmware error in dmesg and there is nothing playing on the speakers: [ 29.080173] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 39.575366] intel_sst_acpi 80860F28:00: FW sent async error msg: [ 39.578059] (null)00040009 00000000 [ 39.578068] (null)00040009 00000000 [ 53.872092] intel_sst_acpi 80860F28:00: FW sent async error msg: [ 53.874999] (null)0002000b 00000000 [ 53.875006] (null)0002000b 00000000 I've tried various different MP3 files with different bit rates but this happens for all of them. Is this an indication that it's not supported on my device or is something else wrong? Other related dmesg messages (but these look all fine to me): [ 14.085226] intel_sst_acpi 80860F28:00: Detected Baytrail-CR platform [ 14.085422] intel_sst_acpi 80860F28:00: LPE base: 0x51000000 size:0x200000 [ 14.085425] intel_sst_acpi 80860F28:00: IRAM base: 0x510c0000 [ 14.085475] intel_sst_acpi 80860F28:00: DRAM base: 0x51100000 [ 14.085484] intel_sst_acpi 80860F28:00: SHIM base: 0x51140000 [ 14.085492] intel_sst_acpi 80860F28:00: Mailbox base: 0x51144000 [ 14.085499] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [ 14.085730] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [ 14.976784] bytcr_rt5640 bytcr_rt5640: Overriding quirk 0x640002 => 0x510000 [ 14.976788] bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled [ 14.976791] bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled [ 14.976793] bytcr_rt5640 bytcr_rt5640: quirk SSP0_AIF1 enabled [ 14.976795] bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled [ 14.980650] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 14.980710] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [ 14.980740] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [ 14.981233] bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp0-port mapping ok The full dmesg and alsa-info.txt is quite big so I've uploaded it to gist. I hope that's okay. https://gist.github.com/lambdadroid/d46675e54720292ed7718c74247753b5 Thanks in advance! Lambdadroid

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[alsa-devel] [PATCH 1/2] ALSA: Get rid of card power_lock
by Takashi Iwai 30 Aug '17

30 Aug '17

Currently we're taking power_lock at each card component for assuring the power-up sequence, but it doesn't help anything in the implementation at the moment: it just serializes unnecessarily the callers, but it doesn't protect about the power state change itself. It used to have some usefulness in the early days where we managed the PM manually. But now the suspend/resume core procedure is beyond our hands, and power_lock lost its meaning. This patch drops the power_lock from allover the places. There shouldn't be any issues by this change, as it's no helper regarding the power state change. Rather we'll get better performance by removing the serialization; which is the only slight concern of any behavior change, but it can't be a showstopper, after all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 13 ---------- sound/core/control.c | 58 ++++++++++++++++++++++++--------------------- sound/core/control_compat.c | 34 ++++++++++++-------------- sound/core/init.c | 5 ---- sound/core/pcm_native.c | 30 ++++++----------------- sound/soc/soc-core.c | 2 -- 6 files changed, 53 insertions(+), 89 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 357f36b5ee80..4104a9d1001f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -136,7 +136,6 @@ struct snd_card { #ifdef CONFIG_PM unsigned int power_state; /* power state */ - struct mutex power_lock; /* power lock */ wait_queue_head_t power_sleep; #endif @@ -149,16 +148,6 @@ struct snd_card { #define dev_to_snd_card(p) container_of(p, struct snd_card, card_dev) #ifdef CONFIG_PM -static inline void snd_power_lock(struct snd_card *card) -{ - mutex_lock(&card->power_lock); -} - -static inline void snd_power_unlock(struct snd_card *card) -{ - mutex_unlock(&card->power_lock); -} - static inline unsigned int snd_power_get_state(struct snd_card *card) { return card->power_state; @@ -175,8 +164,6 @@ int snd_power_wait(struct snd_card *card, unsigned int power_state); #else /* ! CONFIG_PM */ -#define snd_power_lock(card) do { (void)(card); } while (0) -#define snd_power_unlock(card) do { (void)(card); } while (0) static inline int snd_power_wait(struct snd_card *card, unsigned int state) { return 0; } #define snd_power_get_state(card) ({ (void)(card); SNDRV_CTL_POWER_D0; }) #define snd_power_change_state(card, state) do { (void)(card); } while (0) diff --git a/sound/core/control.c b/sound/core/control.c index 51d4b4ad3e1d..56b3e2d49c82 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -864,14 +864,14 @@ static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl, if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; - snd_power_lock(ctl->card); result = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0); - if (result >= 0) - result = snd_ctl_elem_info(ctl, &info); - snd_power_unlock(ctl->card); - if (result >= 0) - if (copy_to_user(_info, &info, sizeof(info))) - return -EFAULT; + if (result < 0) + return result; + result = snd_ctl_elem_info(ctl, &info); + if (result < 0) + return result; + if (copy_to_user(_info, &info, sizeof(info))) + return -EFAULT; return result; } @@ -905,17 +905,19 @@ static int snd_ctl_elem_read_user(struct snd_card *card, if (IS_ERR(control)) return PTR_ERR(control); - snd_power_lock(card); result = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (result >= 0) { - down_read(&card->controls_rwsem); - result = snd_ctl_elem_read(card, control); - up_read(&card->controls_rwsem); - } - snd_power_unlock(card); - if (result >= 0) - if (copy_to_user(_control, control, sizeof(*control))) - result = -EFAULT; + if (result < 0) + goto error; + + down_read(&card->controls_rwsem); + result = snd_ctl_elem_read(card, control); + up_read(&card->controls_rwsem); + if (result < 0) + goto error; + + if (copy_to_user(_control, control, sizeof(*control))) + result = -EFAULT; + error: kfree(control); return result; } @@ -964,17 +966,19 @@ static int snd_ctl_elem_write_user(struct snd_ctl_file *file, return PTR_ERR(control); card = file->card; - snd_power_lock(card); result = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (result >= 0) { - down_write(&card->controls_rwsem); - result = snd_ctl_elem_write(card, file, control); - up_write(&card->controls_rwsem); - } - snd_power_unlock(card); - if (result >= 0) - if (copy_to_user(_control, control, sizeof(*control))) - result = -EFAULT; + if (result < 0) + goto error; + + down_write(&card->controls_rwsem); + result = snd_ctl_elem_write(card, file, control); + up_write(&card->controls_rwsem); + if (result < 0) + goto error; + + if (copy_to_user(_control, control, sizeof(*control))) + result = -EFAULT; + error: kfree(control); return result; } diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 1fa70766ffab..a848836a5de0 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -111,12 +111,10 @@ static int snd_ctl_elem_info_compat(struct snd_ctl_file *ctl, if (get_user(data->value.enumerated.item, &data32->value.enumerated.item)) goto error; - snd_power_lock(ctl->card); err = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0); - if (err >= 0) - err = snd_ctl_elem_info(ctl, data); - snd_power_unlock(ctl->card); - + if (err < 0) + goto error; + err = snd_ctl_elem_info(ctl, data); if (err < 0) goto error; /* restore info to 32bit */ @@ -315,14 +313,13 @@ static int ctl_elem_read_user(struct snd_card *card, if (err < 0) goto error; - snd_power_lock(card); err = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (err >= 0) - err = snd_ctl_elem_read(card, data); - snd_power_unlock(card); - if (err >= 0) - err = copy_ctl_value_to_user(userdata, valuep, data, - type, count); + if (err < 0) + goto error; + err = snd_ctl_elem_read(card, data); + if (err < 0) + goto error; + err = copy_ctl_value_to_user(userdata, valuep, data, type, count); error: kfree(data); return err; @@ -344,14 +341,13 @@ static int ctl_elem_write_user(struct snd_ctl_file *file, if (err < 0) goto error; - snd_power_lock(card); err = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (err >= 0) - err = snd_ctl_elem_write(card, file, data); - snd_power_unlock(card); - if (err >= 0) - err = copy_ctl_value_to_user(userdata, valuep, data, - type, count); + if (err < 0) + goto error; + err = snd_ctl_elem_write(card, file, data); + if (err < 0) + goto error; + err = copy_ctl_value_to_user(userdata, valuep, data, type, count); error: kfree(data); return err; diff --git a/sound/core/init.c b/sound/core/init.c index 6e219dc23f96..32ebe2f6bc59 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -253,7 +253,6 @@ int snd_card_new(struct device *parent, int idx, const char *xid, spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); #ifdef CONFIG_PM - mutex_init(&card->power_lock); init_waitqueue_head(&card->power_sleep); #endif @@ -978,8 +977,6 @@ EXPORT_SYMBOL(snd_card_file_remove); * Waits until the power-state is changed. * * Return: Zero if successful, or a negative error code. - * - * Note: the power lock must be active before call. */ int snd_power_wait(struct snd_card *card, unsigned int power_state) { @@ -999,9 +996,7 @@ int snd_power_wait(struct snd_card *card, unsigned int power_state) if (snd_power_get_state(card) == power_state) break; set_current_state(TASK_UNINTERRUPTIBLE); - snd_power_unlock(card); schedule_timeout(30 * HZ); - snd_power_lock(card); } remove_wait_queue(&card->power_sleep, &wait); return result; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cf0433f80067..621142ea9ec6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1830,7 +1830,6 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, add_wait_queue(&to_check->sleep, &wait); snd_pcm_stream_unlock_irq(substream); up_read(&snd_pcm_link_rwsem); - snd_power_unlock(card); if (runtime->no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; else { @@ -1842,7 +1841,6 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, tout = msecs_to_jiffies(tout * 1000); } tout = schedule_timeout_interruptible(tout); - snd_power_lock(card); down_read(&snd_pcm_link_rwsem); snd_pcm_stream_lock_irq(substream); remove_wait_queue(&to_check->sleep, &wait); @@ -2764,11 +2762,16 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) return 0; } -static int snd_pcm_common_ioctl(struct file *file, +static int snd_pcm_common_ioctl1(struct file *file, struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) { struct snd_pcm_file *pcm_file = file->private_data; + int res; + + res = snd_power_wait(substream->pcm->card, SNDRV_CTL_POWER_D0); + if (res < 0) + return res; switch (cmd) { case SNDRV_PCM_IOCTL_PVERSION: @@ -2846,21 +2849,6 @@ static int snd_pcm_common_ioctl(struct file *file, return -ENOTTY; } -static int snd_pcm_common_ioctl1(struct file *file, - struct snd_pcm_substream *substream, - unsigned int cmd, void __user *arg) -{ - struct snd_card *card = substream->pcm->card; - int res; - - snd_power_lock(card); - res = snd_power_wait(card, SNDRV_CTL_POWER_D0); - if (res >= 0) - res = snd_pcm_common_ioctl(file, substream, cmd, arg); - snd_power_unlock(card); - return res; -} - static int snd_pcm_playback_ioctl1(struct file *file, struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) @@ -3064,7 +3052,6 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, { snd_pcm_uframes_t *frames = arg; snd_pcm_sframes_t result; - int err; switch (cmd) { case SNDRV_PCM_IOCTL_FORWARD: @@ -3084,10 +3071,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, case SNDRV_PCM_IOCTL_START: return snd_pcm_start_lock_irq(substream); case SNDRV_PCM_IOCTL_DRAIN: - snd_power_lock(substream->pcm->card); - err = snd_pcm_drain(substream, NULL); - snd_power_unlock(substream->pcm->card); - return err; + return snd_pcm_drain(substream, NULL); case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); case SNDRV_PCM_IOCTL_DELAY: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 13c875e2392a..62c11e26ce5c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -653,9 +653,7 @@ int snd_soc_suspend(struct device *dev) /* Due to the resume being scheduled into a workqueue we could * suspend before that's finished - wait for it to complete. */ - snd_power_lock(card->snd_card); snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0); - snd_power_unlock(card->snd_card); /* we're going to block userspace touching us until resume completes */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); -- 2.14.1

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[alsa-devel] [asoc:topic/component 7/8] include/sound/soc.h:811:8: error: duplicate member 'set_pll'
by kbuild test robot 30 Aug '17

30 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/component head: 880f02224c624fa2d6e4f8b2d37410fdc98b35fe commit: 47bdbaa2c9a6dd7b112fb403b05dd9293b35a41c [7/8] ASoC: add Component level set_pll config: xtensa-allmodconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 47bdbaa2c9a6dd7b112fb403b05dd9293b35a41c # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): In file included from sound/soc/adi/axi-i2s.c:20:0: include/sound/soc.h:809:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ >> include/sound/soc.h:811:8: error: duplicate member 'set_pll' int (*set_pll)(struct snd_soc_component *component, int pll_id, ^ include/sound/soc.h:891:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ include/sound/soc.h:893:8: error: duplicate member 'set_pll' int (*set_pll)(struct snd_soc_component *component, int pll_id, ^ vim +/set_pll +811 include/sound/soc.h 780 781 /* component interface */ 782 struct snd_soc_component_driver { 783 const char *name; 784 785 /* Default control and setup, added after probe() is run */ 786 const struct snd_kcontrol_new *controls; 787 unsigned int num_controls; 788 const struct snd_soc_dapm_widget *dapm_widgets; 789 unsigned int num_dapm_widgets; 790 const struct snd_soc_dapm_route *dapm_routes; 791 unsigned int num_dapm_routes; 792 793 int (*probe)(struct snd_soc_component *); 794 void (*remove)(struct snd_soc_component *); 795 int (*suspend)(struct snd_soc_component *); 796 int (*resume)(struct snd_soc_component *); 797 798 /* pcm creation and destruction */ 799 int (*pcm_new)(struct snd_soc_pcm_runtime *); 800 void (*pcm_free)(struct snd_pcm *); 801 802 /* component wide operations */ 803 int (*set_sysclk)(struct snd_soc_component *component, 804 int clk_id, int source, unsigned int freq, int dir); 805 int (*set_pll)(struct snd_soc_component *component, int pll_id, 806 int source, unsigned int freq_in, unsigned int freq_out); 807 808 /* component wide operations */ > 809 int (*set_sysclk)(struct snd_soc_component *component, 810 int clk_id, int source, unsigned int freq, int dir); > 811 int (*set_pll)(struct snd_soc_component *component, int pll_id, 812 int source, unsigned int freq_in, unsigned int freq_out); 813 814 /* DT */ 815 int (*of_xlate_dai_name)(struct snd_soc_component *component, 816 struct of_phandle_args *args, 817 const char **dai_name); 818 int (*of_xlate_dai_id)(struct snd_soc_component *comment, 819 struct device_node *endpoint); 820 void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, 821 int subseq); 822 int (*stream_event)(struct snd_soc_component *, int event); 823 824 /* probe ordering - for components with runtime dependencies */ 825 int probe_order; 826 int remove_order; 827 }; 828 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/component 6/8] include/sound/soc.h:809:8: error: duplicate member 'set_sysclk'
by kbuild test robot 30 Aug '17

30 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/component head: 880f02224c624fa2d6e4f8b2d37410fdc98b35fe commit: 9cfeb53411be7c08d7dd19bc4bf0599a6d211390 [6/8] ASoC: add Component level set_sysclk config: xtensa-allmodconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 9cfeb53411be7c08d7dd19bc4bf0599a6d211390 # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): In file included from sound//soc/adi/axi-i2s.c:20:0: >> include/sound/soc.h:809:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ include/sound/soc.h:889:8: error: duplicate member 'set_sysclk' int (*set_sysclk)(struct snd_soc_component *component, ^ vim +/set_sysclk +809 include/sound/soc.h 780 781 /* component interface */ 782 struct snd_soc_component_driver { 783 const char *name; 784 785 /* Default control and setup, added after probe() is run */ 786 const struct snd_kcontrol_new *controls; 787 unsigned int num_controls; 788 const struct snd_soc_dapm_widget *dapm_widgets; 789 unsigned int num_dapm_widgets; 790 const struct snd_soc_dapm_route *dapm_routes; 791 unsigned int num_dapm_routes; 792 793 int (*probe)(struct snd_soc_component *); 794 void (*remove)(struct snd_soc_component *); 795 int (*suspend)(struct snd_soc_component *); 796 int (*resume)(struct snd_soc_component *); 797 798 /* pcm creation and destruction */ 799 int (*pcm_new)(struct snd_soc_pcm_runtime *); 800 void (*pcm_free)(struct snd_pcm *); 801 802 /* component wide operations */ 803 int (*set_sysclk)(struct snd_soc_component *component, 804 int clk_id, int source, unsigned int freq, int dir); 805 int (*set_pll)(struct snd_soc_component *component, int pll_id, 806 int source, unsigned int freq_in, unsigned int freq_out); 807 808 /* component wide operations */ > 809 int (*set_sysclk)(struct snd_soc_component *component, 810 int clk_id, int source, unsigned int freq, int dir); 811 812 /* DT */ 813 int (*of_xlate_dai_name)(struct snd_soc_component *component, 814 struct of_phandle_args *args, 815 const char **dai_name); 816 int (*of_xlate_dai_id)(struct snd_soc_component *comment, 817 struct device_node *endpoint); 818 void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, 819 int subseq); 820 int (*stream_event)(struct snd_soc_component *, int event); 821 822 /* probe ordering - for components with runtime dependencies */ 823 int probe_order; 824 int remove_order; 825 }; 826 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: simple_card_utils: fix fallback when "label" property isn't present
by Lucas Stach 30 Aug '17

30 Aug '17

snd_soc_of_parse_card_name() doesn't return an error if the requested property isn't present, but silently fails to fill the card name. This can not be changed, as it is a backwards compatibility measure itself. We can not rely on the return value of this function alone, but must check if the card name has been filled sucessfully when deciding to skip the fallback path, which is in place for existing users. Fixes: dedfaa1eedc7 (ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name) Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- This may be applied instead of the revert I just sent. After examining the function more closely, the fix seems easy enough to go into 4.13 final. --- sound/soc/generic/simple-card-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 7d7ab4aee42e..d72f7d58102f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -132,7 +132,7 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, /* Parse the card name from DT */ ret = snd_soc_of_parse_card_name(card, "label"); - if (ret < 0) { + if (ret < 0 || !card->name) { char prop[128]; snprintf(prop, sizeof(prop), "%sname", prefix); -- 2.11.0

3 2

[alsa-devel] Applied "ASoC: add Component level set_sysclk" to the asoc tree
by Mark Brown 30 Aug '17

30 Aug '17

The patch ASoC: add Component level set_sysclk has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9cfeb53411be7c08d7dd19bc4bf0599a6d211390 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 24 Aug 2017 00:57:35 +0000 Subject: [PATCH] ASoC: add Component level set_sysclk In current ALSA SoC, Codec only has set_sysclk feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 29 +++++++++++++++++++++++++++++ sound/soc/soc-core.c | 45 ++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 69 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2bcb38045dc1..e7c476febf82 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -799,6 +799,16 @@ struct snd_soc_component_driver { int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); + /* component wide operations */ + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); + + /* component wide operations */ + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, @@ -871,6 +881,14 @@ struct snd_soc_component { int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); + + int (*set_sysclk)(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + /* machine specific init */ int (*init)(struct snd_soc_component *component); @@ -1465,6 +1483,17 @@ void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +/* component wide operations */ +int snd_soc_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); +int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out); + +/* component wide operations */ +int snd_soc_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir); + #ifdef CONFIG_REGMAP void snd_soc_component_init_regmap(struct snd_soc_component *component, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b8f96a41b594..046ab9189b36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2590,11 +2590,9 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, { if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); - else if (dai->codec && dai->codec->driver->set_sysclk) - return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, - freq, dir); - else - return -ENOTSUPP; + + return snd_soc_component_set_sysclk(dai->component, clk_id, 0, + freq, dir); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); @@ -2619,6 +2617,32 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); +/** + * snd_soc_component_set_sysclk - configure COMPONENT system or master clock. + * @component: COMPONENT + * @clk_id: DAI specific clock ID + * @source: Source for the clock + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, + int source, unsigned int freq, int dir) +{ + /* will be removed */ + if (component->set_sysclk) + return component->set_sysclk(component, clk_id, source, + freq, dir); + + if (component->driver->set_sysclk) + return component->driver->set_sysclk(component, clk_id, source, + freq, dir); + + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_component_set_sysclk); + /** * snd_soc_dai_set_clkdiv - configure DAI clock dividers. * @dai: DAI @@ -3176,6 +3200,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->resume = component->driver->resume; component->pcm_new = component->driver->pcm_new; component->pcm_free = component->driver->pcm_free; + component->set_sysclk = component->driver->set_sysclk; dapm = &component->dapm; dapm->dev = dev; @@ -3582,6 +3607,14 @@ static int snd_soc_codec_drv_read(struct snd_soc_component *component, return 0; } +static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir); +} + static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { @@ -3633,6 +3666,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; + if (codec_drv->set_sysclk) + codec->component.set_sysclk = snd_soc_codec_set_sysclk_; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; dapm = snd_soc_codec_get_dapm(codec); -- 2.14.1

1 0

[alsa-devel] Applied "ASoC: add Component level set_pll" to the asoc tree
by Mark Brown 30 Aug '17

30 Aug '17

The patch ASoC: add Component level set_pll has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 47bdbaa2c9a6dd7b112fb403b05dd9293b35a41c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 24 Aug 2017 00:57:51 +0000 Subject: [PATCH] ASoC: add Component level set_pll In current ALSA SoC, Codec only has set_pll feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 7 +++++++ sound/soc/soc-core.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 49 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e7c476febf82..02a510183ab4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -808,6 +808,8 @@ struct snd_soc_component_driver { /* component wide operations */ int (*set_sysclk)(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, @@ -888,6 +890,8 @@ struct snd_soc_component { int (*set_sysclk)(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out); /* machine specific init */ int (*init)(struct snd_soc_component *component); @@ -1493,6 +1497,9 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, /* component wide operations */ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir); +int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out); #ifdef CONFIG_REGMAP diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 046ab9189b36..299d23e832ea 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2679,11 +2679,9 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, if (dai->driver && dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); - else if (dai->codec && dai->codec->driver->set_pll) - return dai->codec->driver->set_pll(dai->codec, pll_id, source, - freq_in, freq_out); - else - return -EINVAL; + + return snd_soc_component_set_pll(dai->component, pll_id, source, + freq_in, freq_out); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); @@ -2708,6 +2706,33 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); +/* + * snd_soc_component_set_pll - configure component PLL. + * @component: COMPONENT + * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + /* will be removed */ + if (component->set_pll) + return component->set_pll(component, pll_id, source, + freq_in, freq_out); + + if (component->driver->set_pll) + return component->driver->set_pll(component, pll_id, source, + freq_in, freq_out); + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_component_set_pll); + /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI @@ -3201,6 +3226,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->pcm_new = component->driver->pcm_new; component->pcm_free = component->driver->pcm_free; component->set_sysclk = component->driver->set_sysclk; + component->set_pll = component->driver->set_pll; dapm = &component->dapm; dapm->dev = dev; @@ -3615,6 +3641,15 @@ static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component, return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir); } +static int snd_soc_codec_set_pll_(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out); +} + static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { @@ -3668,6 +3703,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.read = snd_soc_codec_drv_read; if (codec_drv->set_sysclk) codec->component.set_sysclk = snd_soc_codec_set_sysclk_; + if (codec_drv->set_pll) + codec->component.set_pll = snd_soc_codec_set_pll_; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; dapm = snd_soc_codec_get_dapm(codec); -- 2.14.1

1 0

[alsa-devel] Applied "ASoC: add Component level set_jack" to the asoc tree
by Mark Brown 30 Aug '17

30 Aug '17

The patch ASoC: add Component level set_jack has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 880f02224c624fa2d6e4f8b2d37410fdc98b35fe Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 24 Aug 2017 00:58:07 +0000 Subject: [PATCH] ASoC: add Component level set_jack In current ALSA SoC, Codec only has set_jack feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 4 ++++ sound/soc/soc-core.c | 11 +++++++++++ sound/soc/soc-jack.c | 22 ++++++++++++++++++++++ 3 files changed, 37 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 02a510183ab4..18676ab4f87c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -810,6 +810,8 @@ struct snd_soc_component_driver { int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); + int (*set_jack)(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, @@ -892,6 +894,8 @@ struct snd_soc_component { int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); + int (*set_jack)(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); /* machine specific init */ int (*init)(struct snd_soc_component *component); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 299d23e832ea..78b8886a26d6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3227,6 +3227,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->pcm_free = component->driver->pcm_free; component->set_sysclk = component->driver->set_sysclk; component->set_pll = component->driver->set_pll; + component->set_jack = component->driver->set_jack; dapm = &component->dapm; dapm->dev = dev; @@ -3650,6 +3651,14 @@ static int snd_soc_codec_set_pll_(struct snd_soc_component *component, return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out); } +static int snd_soc_codec_set_jack_(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return snd_soc_codec_set_jack(codec, jack, data); +} + static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { @@ -3705,6 +3714,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.set_sysclk = snd_soc_codec_set_sysclk_; if (codec_drv->set_pll) codec->component.set_pll = snd_soc_codec_set_pll_; + if (codec_drv->set_jack) + codec->component.set_jack = snd_soc_codec_set_jack_; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; dapm = snd_soc_codec_get_dapm(codec); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7daf21fee355..2f9f496ab14f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -40,6 +40,28 @@ int snd_soc_codec_set_jack(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); +/** + * snd_soc_component_set_jack - configure component jack. + * @component: COMPONENTs + * @jack: structure to use for the jack + * @data: can be used if codec driver need extra data for configuring jack + * + * Configures and enables jack detection function. + */ +int snd_soc_component_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + /* will be removed */ + if (component->set_jack) + return component->set_jack(component, jack, data); + + if (component->driver->set_jack) + return component->driver->set_jack(component, jack, data); + + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); + /** * snd_soc_card_jack_new - Create a new jack * @card: ASoC card -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: rt274: add acpi id
by Bard Liao 30 Aug '17

30 Aug '17

Add ACPI id for Intel platform. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt274.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index fb683ff..0be69c1 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1102,6 +1102,7 @@ MODULE_DEVICE_TABLE(i2c, rt274_i2c_id); static const struct acpi_device_id rt274_acpi_match[] = { { "10EC0274", 0 }, + { "INT34C2", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, rt274_acpi_match); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Add delay for jack plug in
by Oder Chiou 30 Aug '17

30 Aug '17

Codec initialize takes some time when 3.5mm jack plugged in. Add a delay to report jack plugged event to user space to avoid pop noise. Signed-off-by: Hsinyu Chao <hychao(a)chromium.org> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index c98a8a7..ab9e0eb 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1705,6 +1705,9 @@ static void rt5663_jack_detect_work(struct work_struct *work) default: dev_err(codec->dev, "Unknown CODEC Version\n"); } + + /* Delay the jack insert report to avoid pop noise */ + msleep(30); } else { /* jack is already in, report button event */ report = SND_JACK_HEADSET; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rockchip: constify snd_soc_ops structures
by Jeffy Chen 30 Aug '17

30 Aug '17

Add const to dp/dmic snd_soc_ops. Fixes: 626d84db64d7 (ASoC: rockchip: Add support for DMIC codec) Fixes: 3313faf1053e (ASoC: rockchip: Add support for DP codec) Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 5ab25962cabd..0513fe480353 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -301,11 +301,11 @@ static const struct snd_soc_ops rockchip_sound_da7219_ops = { .hw_params = rockchip_sound_da7219_hw_params, }; -static struct snd_soc_ops rockchip_sound_cdndp_ops = { +static const struct snd_soc_ops rockchip_sound_cdndp_ops = { .hw_params = rockchip_sound_cdndp_hw_params, }; -static struct snd_soc_ops rockchip_sound_dmic_ops = { +static const struct snd_soc_ops rockchip_sound_dmic_ops = { .hw_params = rockchip_sound_dmic_hw_params, }; -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: rsnd: Drop unit-addresses without reg properties
by Geert Uytterhoeven 30 Aug '17

30 Aug '17

Nodes without reg properties must not have unit addresses: Warning (unit_address_vs_reg): Node .../rcar_sound,dvc/dvc@0 has a unit name, but no reg property Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Despite the example, there don't seem to be any DTSes with multiple "simple-audio-card,cpu" nodes? --- .../devicetree/bindings/sound/renesas,rsnd.txt | 68 +++++++++++----------- 1 file changed, 34 insertions(+), 34 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 7246bb268bf9bab6..a1536fdc60e62c49 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -199,10 +199,10 @@ Ex) sound { compatible = "simple-scu-audio-card"; ... - simple-audio-card,cpu@0 { + simple-audio-card,cpu-0 { sound-dai = <&rcar_sound 0>; }; - simple-audio-card,cpu@1 { + simple-audio-card,cpu-1 { sound-dai = <&rcar_sound 1>; }; simple-audio-card,codec { @@ -441,79 +441,79 @@ rcar_sound: sound@ec500000 { "clk_a", "clk_b", "clk_c", "clk_i"; rcar_sound,dvc { - dvc0: dvc@0 { + dvc0: dvc-0 { dmas = <&audma0 0xbc>; dma-names = "tx"; }; - dvc1: dvc@1 { + dvc1: dvc-1 { dmas = <&audma0 0xbe>; dma-names = "tx"; }; }; rcar_sound,mix { - mix0: mix@0 { }; - mix1: mix@1 { }; + mix0: mix-0 { }; + mix1: mix-1 { }; }; rcar_sound,ctu { - ctu00: ctu@0 { }; - ctu01: ctu@1 { }; - ctu02: ctu@2 { }; - ctu03: ctu@3 { }; - ctu10: ctu@4 { }; - ctu11: ctu@5 { }; - ctu12: ctu@6 { }; - ctu13: ctu@7 { }; + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; }; rcar_sound,src { - src0: src@0 { + src0: src-0 { interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x85>, <&audma1 0x9a>; dma-names = "rx", "tx"; }; - src1: src@1 { + src1: src-1 { interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x87>, <&audma1 0x9c>; dma-names = "rx", "tx"; }; - src2: src@2 { + src2: src-2 { interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x89>, <&audma1 0x9e>; dma-names = "rx", "tx"; }; - src3: src@3 { + src3: src-3 { interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x8b>, <&audma1 0xa0>; dma-names = "rx", "tx"; }; - src4: src@4 { + src4: src-4 { interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x8d>, <&audma1 0xb0>; dma-names = "rx", "tx"; }; - src5: src@5 { + src5: src-5 { interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x8f>, <&audma1 0xb2>; dma-names = "rx", "tx"; }; - src6: src@6 { + src6: src-6 { interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x91>, <&audma1 0xb4>; dma-names = "rx", "tx"; }; - src7: src@7 { + src7: src-7 { interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x93>, <&audma1 0xb6>; dma-names = "rx", "tx"; }; - src8: src@8 { + src8: src-8 { interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x95>, <&audma1 0xb8>; dma-names = "rx", "tx"; }; - src9: src@9 { + src9: src-9 { interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x97>, <&audma1 0xba>; dma-names = "rx", "tx"; @@ -521,52 +521,52 @@ rcar_sound: sound@ec500000 { }; rcar_sound,ssi { - ssi0: ssi@0 { + ssi0: ssi-0 { interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi1: ssi@1 { + ssi1: ssi-1 { interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi2: ssi@2 { + ssi2: ssi-2 { interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi3: ssi@3 { + ssi3: ssi-3 { interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi4: ssi@4 { + ssi4: ssi-4 { interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi5: ssi@5 { + ssi5: ssi-5 { interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi6: ssi@6 { + ssi6: ssi-6 { interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi7: ssi@7 { + ssi7: ssi-7 { interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi8: ssi@8 { + ssi8: ssi-8 { interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; dma-names = "rx", "tx", "rxu", "txu"; }; - ssi9: ssi@9 { + ssi9: ssi-9 { interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; dma-names = "rx", "tx", "rxu", "txu"; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Add jack detection workaround for MINIX Z83-4 based devices
by Ian W MORRISON 30 Aug '17

30 Aug '17

The MINIX NEO Z83-4 and MINIX NEO Z83-4 Pro devices requires jd_mode=3 to make the jack detection work. Using a BIOS DMI product of "Z83-4" will match both devices of 'NEO Z83-4' and 'Z83-4 Pro'. Signed-off-by: Ian W Morrison <ianwmorrison(a)gmail.com> --- sound/soc/codecs/rt5645.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ec5816..7d889b6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3637,6 +3637,21 @@ static int rt5645_resume(struct snd_soc_codec *codec) {} }; +static struct rt5645_platform_data minix_z83_4_platform_data = { + .jd_mode = 3, +}; + +static struct dmi_system_id dmi_platform_minix_z83_4[] = { + { + .ident = "MINIX Z83-4", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"), + DMI_MATCH(DMI_PRODUCT_NAME, "Z83-4"), + }, + }, + { } +}; + static bool rt5645_check_dp(struct device *dev) { if (device_property_present(dev, "realtek,in2-differential") || @@ -3689,6 +3704,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, rt5645->pdata = general_platform_data; else if (dmi_check_system(dmi_platform_gpd_win)) rt5645->pdata = gpd_win_platform_data; + else if (dmi_check_system(dmi_platform_minix_z83_4)) + rt5645->pdata = minix_z83_4_platform_data; if (quirk != -1) { rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); -- 1.9.1

2 1

[alsa-devel] ALSA hang on 4.13.0-rc7
by Markus Trippelsdorf 30 Aug '17

30 Aug '17

On my system ALSA sometimes hangs: sysrq: SysRq : Show Blocked State task PC stack pid father output:ALSA def D 0 171 1 0x00000000 Call Trace: ? __schedule+0x17c/0x720 ? schedule_preempt_disabled+0x2d/0x80 ? __mutex_lock_slowpath+0x141/0x420 ? snd_pcm_common_ioctl1+0x2f/0x1400 ? snd_pcm_common_ioctl1+0x2f/0x1400 ? snd_card_file_remove+0x76/0x120 ? snd_pcm_playback_ioctl+0x1c7/0x560 ? dput+0xb6/0x1e0 ? SyS_ioctl+0xa7/0x860 ? task_work_run+0x70/0xa0 ? entry_SYSCALL_64_fastpath+0x13/0x94 Only a reboot will fix the issue. -- Markus

3 12

[alsa-devel] memory leak in mix/pcm_upmix.c mix/pcm_downmix.c!!!
by boozer asm 30 Aug '17

30 Aug '17

eg. in function:SND_PCM_PLUGIN_DEFINE_FUNC(upmix) 487 mix = calloc(1, sizeof(*mix)); 488 if (mix == NULL) 489 return -ENOMEM; mix is allocated here. but at close function, it is not freed, and no __destructor function available to free it. and in alsa-lib/src/pcm/pcm_extplug.c static int snd_pcm_extplug_close(snd_pcm_t *pcm) { extplug_priv_t *ext = pcm->private_data; snd_pcm_close(ext->plug.gen.slave); clear_ext_params(ext); if (ext->data->callback->close) { ext->data->callback->close(ext->data); } free(ext); return 0; } this function does NOT free ext->data. and 366 static int upmix_close(snd_pcm_extplug_t *ext) 367 { 368 snd_pcm_upmix_t *mix = (snd_pcm_upmix_t *)ext; 369 free(mix->delayline[0]); 370 free(mix->delayline[1]); 371 return 0; 372 } and this function does NOT free mix itself. so I think there is memory leak.

2 6

[alsa-devel] [PATCH] alsabat: fix one uninitialized warning issue
by Zhang Keqiao 30 Aug '17

30 Aug '17

From: Zhang Keqiao <keqiaox.k.zhang(a)linux.intel.com> Fix a variable uninitialized issue, adding the initialized assignment to fix it. Signed-off-by: Zhang Keqiao <keqiaox.k.zhang(a)linux.intel.com> --- bat/analyze.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/bat/analyze.c b/bat/analyze.c index 2b8ca61..9808b73 100644 --- a/bat/analyze.c +++ b/bat/analyze.c @@ -34,7 +34,7 @@ static void check_amplitude(struct bat *bat, float *buf) int i, percent; /* calculate average value */ - for (i = 0, sum = 0.0; i < bat->frames; i++) + for (i = 0, sum = 0.0, average = 0.0; i < bat->frames; i++) sum += buf[i]; average = sum / bat->frames; -- 2.9.3

2 1

[alsa-devel] [PATCH] Revert "ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name"
by Lucas Stach 30 Aug '17

30 Aug '17

This commit breaks existing systems, as snd_soc_of_parse_card_name() doesn't return an error if the DT name property is missing, which would be required to hit the fallback path in the offending commit at all. Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- This should be fixed properly, but given the timing with the 4.13 release being really close, I opted for just reverting the offending commit. --- sound/soc/generic/simple-card-utils.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 7d7ab4aee42e..e6f472360cfc 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -125,21 +125,15 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_set_dailink_name); int asoc_simple_card_parse_card_name(struct snd_soc_card *card, char *prefix) { + char prop[128]; int ret; - if (!prefix) - prefix = ""; + snprintf(prop, sizeof(prop), "%sname", prefix); /* Parse the card name from DT */ - ret = snd_soc_of_parse_card_name(card, "label"); - if (ret < 0) { - char prop[128]; - - snprintf(prop, sizeof(prop), "%sname", prefix); - ret = snd_soc_of_parse_card_name(card, prop); - if (ret < 0) - return ret; - } + ret = snd_soc_of_parse_card_name(card, prop); + if (ret < 0) + return ret; if (!card->name && card->dai_link) card->name = card->dai_link->name; -- 2.11.0

2 2

[alsa-devel] [PATCH] ASoC: tas2552: Fix fraction overflow in PLL calculation
by Oskar Schirmer 29 Aug '17

29 Aug '17

Setting the PLL involves the calculation of a fixed point ratio with 4 decimal digits fraction, referred to as "J.D". The fraction "D" is stored separately from the integer part "J" and is limited to 0..9999. The current algorithm uses integer registers to calculate the fraction part, but failed to compensate for rounding errors, resulting in values larger than 9999 for the fraction part occasionally, e.g. for 44.1kHz audio rate and pll_clkin = 3763400 it would set J to 11 and D to 10002, which will at best result in wrong pitch. The critical part is the "pll_clkin / 10000", which would be ok with real numbers, but using integer arithmetic the rounding decreases the divisor, thus increasing the final quotient. The issue is solved by linear interpolation over the reciprocal function between the two adjacent points with integer divisor, i.e. pll_clkin / 10000 and pll_clkin / 10000 + 1, and doing all rounding to the lower result. As a side effect to the bug fix, the approximation to the desired frequency is much better, for the above mentioned example we get 11.9993, while the true ratio is 11.9993623. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/codecs/tas2552.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 8840f72..1011616 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -192,7 +192,7 @@ static int tas2552_setup_pll(struct snd_soc_codec *codec, * pll_clk = (.5 * pll_clkin * J.D) / 2^p * Need to fill in J and D here based on incoming freq */ - unsigned int d; + unsigned int d, q, t; u8 j; u8 pll_sel = (tas2552->pll_clk_id << 3) & TAS2552_PLL_SRC_MASK; u8 p = snd_soc_read(codec, TAS2552_PLL_CTRL_1); @@ -200,9 +200,12 @@ static int tas2552_setup_pll(struct snd_soc_codec *codec, p = (p >> 7); recalc: - j = (pll_clk * 2 * (1 << p)) / pll_clkin; - d = (pll_clk * 2 * (1 << p)) % pll_clkin; - d /= (pll_clkin / 10000); + t = (pll_clk * 2) << p; + j = t / pll_clkin; + d = t % pll_clkin; + t = pll_clkin / 10000; + q = d / (t + 1); + d = q + ((9999 - pll_clkin % 10000) * (d / t - q)) / 10000; if (d && (pll_clkin < 512000 || pll_clkin > 9200000)) { if (tas2552->pll_clk_id == TAS2552_PLL_CLKIN_BCLK) { -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: Intel: kbl: Machine driver modifications to support MST Audio
by Naveen M 29 Aug '17

29 Aug '17

Patch modifies dapm routes to enable HDMI Audio and invokes machine pin widget for hdmi port to enable/disable pin, when monitor is connected/disconnected based on the pin status. Naveen M (2): ASoC: Intel: kbl: Add MST route change to kbl machine drivers ASoC: Intel: kbl: Add jack port initialize in kbl machine drivers sound/soc/intel/boards/kbl_rt5663_max98927.c | 15 +++++++++------ sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 14 ++++++++------ 2 files changed, 17 insertions(+), 12 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH v3] ASoC: fsl_dma: remove dma_object path member
by Rob Herring 29 Aug '17

29 Aug '17

dma_object.path is unused, so rather than fix it to work with DT full_name changes, just remove it. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org --- v3: - Update subject: s/sound/ASoC/ v2: - Move printf specifier change to correct patch sound/soc/fsl/fsl_dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index c17359648915..0c11f434a374 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -63,7 +63,6 @@ struct dma_object { struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; - char path[1]; }; /* @@ -903,13 +902,12 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) return ret; } - dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + dma = kzalloc(sizeof(*dma), GFP_KERNEL); if (!dma) { of_node_put(ssi_np); return -ENOMEM; } - strcpy(dma->path, np->full_name); dma->dai.ops = &fsl_dma_ops; dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: rt5514: Guard Hotword Model bytes loading
by Hsin-Yu Chao 29 Aug '17

29 Aug '17

If the rt5514-spi driver is not enabled in kernel, hotword model will not be loaded when "DSP Voice Wake Up" is set to turn on DSP mode, and an error is logged instead. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 7482df8ff737..250581082e9d 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -299,7 +299,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); struct snd_soc_codec *codec = rt5514->codec; const struct firmware *fw = NULL; - int ret = 0; if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) return 0; @@ -337,6 +336,9 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, } if (rt5514->model_buf && rt5514->model_len) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) + int ret; + ret = rt5514_spi_burst_write(0x4ff80000, rt5514->model_buf, ((rt5514->model_len / 8) + 1) * 8); @@ -345,13 +347,22 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, "Model load failed %d\n", ret); return ret; } +#else + dev_err(codec->dev, + "No SPI driver for loading firmware\n"); +#endif } else { request_firmware(&fw, RT5514_FIRMWARE3, codec->dev); if (fw) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_write(0x4ff80000, fw->data, ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, + "No SPI driver to load fw\n"); +#endif release_firmware(fw); fw = NULL; } -- 2.12.2

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: kbl: Add map for Maxim IV Feedback
by Naveen M 29 Aug '17

29 Aug '17

MAX98927 provides IV feedback on the capture widget. Here we are connecting the capture widget to SSP0_RX and SSP0_RX to the algorithm running on host. v2: rebased on newer kernel (d46b182) Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index c144214..8d2a1c0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -107,6 +107,11 @@ enum { { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "dmic01_hifi", NULL, "DMIC01 Rx" }, { "DMIC01 Rx", NULL, "DMIC AIF" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index c2ef538..8d19da9 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -119,6 +119,11 @@ enum { { "codec1_in", NULL, "ssp0 Rx" }, { "ssp0 Rx", NULL, "AIF1 Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, -- 1.9.1

3 2

[alsa-devel] [RESENT PATCH v7 0/7] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 29 Aug '17

29 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v7: Rebase on the newest for-next Changes in v6: Add dmic wakeup delay(not used for now). Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (7): ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 255 +++++++++++++++------ 4 files changed, 186 insertions(+), 80 deletions(-) -- 2.11.0

3 8

[alsa-devel] [PATCH v3 0/9] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 29 Aug '17

29 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (9): ASoC: rt5514: Switch to snd_soc_register_codec ASoC: rockchip: Remove obsolete dmic-delay ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 9 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/codecs/rt5514-spi.c | 19 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 263 ++++++++++++++------- 5 files changed, 196 insertions(+), 104 deletions(-) -- 2.11.0

4 15

[alsa-devel] [RESENT PATCH v6 0/8] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 29 Aug '17

29 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v6: Add dmic wakeup delay(not used for now). Changes in v5: Keep the dmic-delay property, but correct it with the name in dt-binding. -- Suggested-by Mark Brown <broonie(a)kernel.org> Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (8): ASoC: rockchip: Correct 'dmic-delay' property name ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 270 ++++++++++++++------- 4 files changed, 193 insertions(+), 88 deletions(-) -- 2.11.0

2 8

[alsa-devel] [PATCH v2 00/10] Remove unnecessary function
by Peng Donglin 29 Aug '17

29 Aug '17

When calling the function devm_snd_soc_register_card to register a sound card,the address of snd_soc_card will be assigned to dev->driver_data automatically,so there is no extra need to call platform_set_drvdata. Peng Donglin (10): ASoC: s3c24xx_uda134x: Remove unnecessary function call ASoC: mediatek: Remove unnecessary function call ASoC: Intel: Remove unnecessary function call ASoC: atmel: Remove unnecessary function call ASoC: rockchip: Remove unnecessary function call ASoC: mxs-sgtl5000: Remove unnecessary function call ASoC: qcom: Remove useless function call ASoC: sun4i-codec: Remove unnecessary function call ASoC: tegra: Remove unnecessary function call ASoC: ux500: Remove unnecessary function call sound/soc/atmel/atmel-classd.c | 1 - sound/soc/atmel/atmel-pdmic.c | 1 - sound/soc/intel/boards/bytcht_da7213.c | 6 ++---- sound/soc/intel/boards/bytcht_es8316.c | 6 ++---- sound/soc/intel/boards/bytcht_nocodec.c | 6 ++---- sound/soc/intel/boards/bytcr_rt5640.c | 7 ++----- sound/soc/intel/boards/bytcr_rt5651.c | 7 ++----- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 ++---- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 ++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 ++---- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 7 ++----- sound/soc/qcom/apq8016_sbc.c | 1 - sound/soc/qcom/storm.c | 1 - sound/soc/rockchip/rk3288_hdmi_analog.c | 6 +----- sound/soc/samsung/s3c24xx_uda134x.c | 1 - sound/soc/sunxi/sun4i-codec.c | 1 - sound/soc/tegra/tegra_alc5632.c | 1 - sound/soc/tegra/tegra_max98090.c | 1 - sound/soc/tegra/tegra_rt5640.c | 1 - sound/soc/tegra/tegra_rt5677.c | 1 - sound/soc/tegra/tegra_sgtl5000.c | 1 - sound/soc/tegra/tegra_wm8753.c | 1 - sound/soc/tegra/tegra_wm8903.c | 1 - sound/soc/tegra/tegra_wm9712.c | 1 - sound/soc/tegra/trimslice.c | 1 - sound/soc/ux500/mop500.c | 1 - 29 files changed, 19 insertions(+), 63 deletions(-) -- 2.14.1

3 14

[alsa-devel] [PATCH v3] sound: aoa: Convert to using %pOF instead of full_name
by Rob Herring 29 Aug '17

29 Aug '17

Now that we have a custom printf format specifier, convert users of full_name to use %pOF instead. This is preparation to remove storing of the full path string for each node. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org --- v3: Split aoa to separate patch sound/aoa/codecs/tas.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 733b6365dad6..15c05755d270 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -905,8 +905,8 @@ static int tas_i2c_probe(struct i2c_client *client, goto fail; } printk(KERN_DEBUG - "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - (unsigned int)client->addr, node->full_name); + "snd-aoa-codec-tas: tas found, addr 0x%02x on %pOF\n", + (unsigned int)client->addr, node); return 0; fail: mutex_destroy(&tas->mtx); -- 2.11.0

2 2

[alsa-devel] [PATCH v4] ASoC: Intel: kbl: Add map for Maxim IV Feedback
by Naveen M 28 Aug '17

28 Aug '17

MAX98927 provides IV feedback on the capture widget. Here we are connecting the capture widget to SSP0_RX and SSP0_RX to the algorithm running on host. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- v2: rebased on newer kernel (d46b182) v3: moved version changelog info after s-o-b lines v4: rebased (9149916), moved ver changelog info to correct position sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +++++ sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +++++ 2 files changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index c144214..8d2a1c0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -107,6 +107,11 @@ enum { { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "dmic01_hifi", NULL, "DMIC01 Rx" }, { "DMIC01 Rx", NULL, "DMIC AIF" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 690b54f..509aeb9 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -120,6 +120,11 @@ enum { { "codec1_in", NULL, "ssp0 Rx" }, { "ssp0 Rx", NULL, "AIF1 Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, -- 1.9.1

2 1

[alsa-devel] Can ALSA MIDI use timestamps for precise timing?
by Roger Dannenberg 28 Aug '17

28 Aug '17

Hi, I wrote PortMIDI long ago and now that it's going into Audacity and we're having timing problems in Linux, I'm revisiting the code and wondering if ALSA has a fundamental design problem. I'm hoping I'm missing something, but here's the problem: PortMIDI wants to send MIDI messages at precise times according to arbitrary clocks. In Audacity, the important clock to synchronize to is the audio sample clock. To synchronize MIDI, one can in principle use timestamps and let the device driver, which presumably runs with very low latency, dispatch MIDI from a queue to the device or destination. The question is then what timestamp to use. ALSA has some options. There's relative timing, but if you send with relative time and you get unscheduled just as you are executing the write call, you'll be late. There's absolute timing, but the docs say that's relative to when you start the queue, but when exactly is that? The problem becomes precise timing of queue creation doesn't it? And there's no timestamp provided for when to start. If "absolute" mean exactly the system time, that would be great, but I think that's not the case. Is this a fundamental design error, or am I missing something? Thanks. -Roger Dannenberg

2 1

[alsa-devel] Scarlett 6i6 gain switches patch
by Jens Verwiebe 28 Aug '17

28 Aug '17

Hi folks As an owner of an Focusrite Scarlett 6i6 (gen 1)i was bothered that i could not switch rear inputs 3/4 gain. After investigating the code i found the specs of that interface not used right. I wrote a patch and tested this well so far. Afaik the same could be used also for the 18i18 and 18i20, but i left those untouched for i could not test myself. Pls review and take over in case its accepted. Cheers ... Jens The patch: --- /sound/usb/mixer_scarlett_nv.c 2017-04-08 09:53:53.000000000 +0200 +++ /sound/usb/mixer_scarlett.c 2017-08-26 03:15:15.734930629 +0200 @@ -152,6 +152,7 @@ SCARLETT_OUTPUTS, SCARLETT_SWITCH_IMPEDANCE, SCARLETT_SWITCH_PAD, + SCARLETT_SWITCH_GAIN, }; enum { @@ -202,6 +203,15 @@ } }; +static const struct scarlett_mixer_elem_enum_info opt_gain = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Lo", "Hi" + } +}; + static const struct scarlett_mixer_elem_enum_info opt_impedance = { .start = 0, .len = 2, @@ -664,8 +674,8 @@ { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, }, .matrix_mux_init = { @@ -895,6 +905,15 @@ if (err < 0) return err; break; + case SCARLETT_SWITCH_GAIN: + sprintf(mx, "Input %d Gain Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, + 0x08, ctl->num, USB_MIXER_S16, 1, mx, + &opt_gain, &elem); + if (err < 0) + return err; + break; } } -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

2 1

[alsa-devel] [PATCH] ASoC: rt5514: expose Hotword Model control
by Hsin-Yu Chao 28 Aug '17

28 Aug '17

This change exposes mixer control 'Hotword Model' for switching hotword model in runtime. This new function requires updated firmware to load hotword model at address 0x4ff80000. Note that if the rt5514-spi driver is not enabled in kernel, hotword model will not be loaded when "DSP Voice Wake Up" is set to turn on DSP mode, and an error is logged instead. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 63 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 3 +++ 2 files changed, 66 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a2722177470e..250581082e9d 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -335,6 +335,39 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } + if (rt5514->model_buf && rt5514->model_len) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) + int ret; + + ret = rt5514_spi_burst_write(0x4ff80000, + rt5514->model_buf, + ((rt5514->model_len / 8) + 1) * 8); + if (ret) { + dev_err(codec->dev, + "Model load failed %d\n", ret); + return ret; + } +#else + dev_err(codec->dev, + "No SPI driver for loading firmware\n"); +#endif + } else { + request_firmware(&fw, RT5514_FIRMWARE3, + codec->dev); + if (fw) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ff80000, + fw->data, + ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, + "No SPI driver to load fw\n"); +#endif + release_firmware(fw); + fw = NULL; + } + } + /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -349,6 +382,34 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } +static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, + const unsigned int __user *bytes, unsigned int size) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + int ret = 0; + + if (rt5514->model_buf || rt5514->model_len < size) { + if (rt5514->model_buf) + devm_kfree(codec->dev, rt5514->model_buf); + rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); + if (!rt5514->model_buf) { + ret = -ENOMEM; + goto done; + } + } + + /* Skips the TLV header. */ + bytes += 2; + + if (copy_from_user(rt5514->model_buf, bytes, size)) + ret = -EFAULT; +done: + rt5514->model_len = (ret ? 0 : size); + return ret; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -360,6 +421,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), + SND_SOC_BYTES_TLV("Hotword Model", 0x8504, + NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212a86d9..07475a9918d2 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -236,6 +236,7 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" +#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -262,6 +263,8 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; + u8 *model_buf; + unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.12.2

1 1

[alsa-devel] [PATCH v2] ASoC: rt5514: expose Hotword Model control
by Hsin-Yu Chao 28 Aug '17

28 Aug '17

This change exposes mixer control 'Hotword Model' for switching hotword model in runtime. This new function requires updated firmware to load hotword model at address 0x4ff80000. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 63 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 3 +++ 2 files changed, 66 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a2722177470e..250581082e9d 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -335,6 +335,39 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } + if (rt5514->model_buf && rt5514->model_len) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) + int ret; + + ret = rt5514_spi_burst_write(0x4ff80000, + rt5514->model_buf, + ((rt5514->model_len / 8) + 1) * 8); + if (ret) { + dev_err(codec->dev, + "Model load failed %d\n", ret); + return ret; + } +#else + dev_err(codec->dev, + "No SPI driver for loading firmware\n"); +#endif + } else { + request_firmware(&fw, RT5514_FIRMWARE3, + codec->dev); + if (fw) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ff80000, + fw->data, + ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, + "No SPI driver to load fw\n"); +#endif + release_firmware(fw); + fw = NULL; + } + } + /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -349,6 +382,34 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } +static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, + const unsigned int __user *bytes, unsigned int size) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + int ret = 0; + + if (rt5514->model_buf || rt5514->model_len < size) { + if (rt5514->model_buf) + devm_kfree(codec->dev, rt5514->model_buf); + rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); + if (!rt5514->model_buf) { + ret = -ENOMEM; + goto done; + } + } + + /* Skips the TLV header. */ + bytes += 2; + + if (copy_from_user(rt5514->model_buf, bytes, size)) + ret = -EFAULT; +done: + rt5514->model_len = (ret ? 0 : size); + return ret; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -360,6 +421,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), + SND_SOC_BYTES_TLV("Hotword Model", 0x8504, + NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212a86d9..07475a9918d2 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -236,6 +236,7 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" +#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -262,6 +263,8 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; + u8 *model_buf; + unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.12.2

2 2

[alsa-devel] [PATCH 0/4] ASoC: add Component level set_xxx
by Kuninori Morimoto 28 Aug '17

28 Aug '17

Hi Mark, Lars-Peter These add Component level set_xxx feature which are supported on Codec side. After this, we can replace - snd_soc_codec_set_xxx(...); + snd_soc_component_set_xxx(...); Kuninori Morimoto (4): ASoC: add Component level set_sysclk ASoC: add Component level set_pll ASoC: add Component level set_jack ASoC: add Component level set_bias_level include/sound/soc.h | 26 ++++++++++++ sound/soc/soc-core.c | 113 ++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/soc-jack.c | 22 ++++++++++ 3 files changed, 151 insertions(+), 10 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

2 8

[alsa-devel] [PATCH v2] ASoC: Add a sanity check before using dai driver name
by Jeffy Chen 28 Aug '17

28 Aug '17

The dai driver's name is allowed to be NULL. So add a sanity check for that. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> Reported-by: Donglin Peng <dolinux.peng(a)gmail.com> --- Changes in v2: Keep the oringinal check style. sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fc1bb2da3e2e..c38c8f844546 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1029,7 +1029,8 @@ struct snd_soc_dai *snd_soc_find_dai( continue; list_for_each_entry(dai, &component->dai_list, list) { if (dlc->dai_name && strcmp(dai->name, dlc->dai_name) - && strcmp(dai->driver->name, dlc->dai_name)) + && (!dai->driver->name + || !strcmp(dai->driver->name, dlc->dai_name)) continue; return dai; -- 2.11.0

5 11

[alsa-devel] [PATCH v3] ASoC: Add a sanity check before using dai driver name
by Jeffy Chen 27 Aug '17

27 Aug '17

The dai driver's name is allowed to be NULL. So add a sanity check for that. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> Reported-by: Donglin Peng <dolinux.peng(a)gmail.com> --- Changes in v3: Fix typo Changes in v2: Keep the oringinal check style. sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fc1bb2da3e2e..8827c2225ba3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1029,7 +1029,8 @@ struct snd_soc_dai *snd_soc_find_dai( continue; list_for_each_entry(dai, &component->dai_list, list) { if (dlc->dai_name && strcmp(dai->name, dlc->dai_name) - && strcmp(dai->driver->name, dlc->dai_name)) + && (!dai->driver->name + || strcmp(dai->driver->name, dlc->dai_name))) continue; return dai; -- 2.11.0

2 1

[alsa-devel] [PATCH 0/7] ASoC: Intel: Introduce tokens for CNL
by Guneshwor Singh 27 Aug '17

27 Aug '17

This series introduces topology changes for CNL platform. Cannonlake DSP supports various formats, rates and channels for each module. To represent the buffer size, cycles per second and required memory page per format/channel/rate per module, common module data are abstracted out and are now part of the topology manifest. This is necessary for pipelines which involve algo modules as they required floating point calculations. With the addition of manifest tokens, buffer size, formats and page required by dsp modules can be pre-calculated and populate in the conf manifest instead of computing them in the driver. This change is backward compatible with the existing model. Jeeja KP (1): ASoC: Intel: Skylake: Update module id in pin connections Ramesh Babu (1): ASoC: Intel: Skylake: Parse and update module config structure Shreyas NC (5): ASoC: Intel: Skylake: Parse multiple manifest data blocks ASoC: Intel: uapi: Add new tokens for module common data ASoC: Intel: Skylake: Commonize parsing of format tokens ASoC: Intel: Skylake: Add driver structures to be filled from topology manifest ASoC: Intel: Skylake: Populate module data from topology manifest include/uapi/sound/snd_sst_tokens.h | 96 ++++- sound/soc/intel/skylake/skl-messages.c | 59 ++- sound/soc/intel/skylake/skl-pcm.c | 44 +- sound/soc/intel/skylake/skl-topology.c | 718 ++++++++++++++++++++++++++++----- sound/soc/intel/skylake/skl-topology.h | 79 ++++ sound/soc/intel/skylake/skl.h | 2 + 6 files changed, 871 insertions(+), 127 deletions(-) -- 2.14.1

2 10

[alsa-devel] [PATCH v2 0/7] ASoC: Intel: Introduce tokens for CNL
by Guneshwor Singh 27 Aug '17

27 Aug '17

This series introduces topology changes for CNL platform. Cannonlake DSP supports various formats, rates and channels for each module. To represent the buffer size, cycles per second and required memory page per format/channel/rate per module, common module data are abstracted out and are now part of the topology manifest. This is necessary for pipelines which involve algo modules as they required floating point calculations. With the addition of manifest tokens, buffer size, formats and page required by dsp modules can be pre-calculated and populate in the conf manifest instead of computing them in the driver. This change is backward compatible with the existing model. Changes in v2: - Removed token SKL_TKN_MM_U8_MAX_INST_COUNT and its structure member which is never used. Jeeja KP (1): ASoC: Intel: Skylake: Update module id in pin connections Ramesh Babu (1): ASoC: Intel: Skylake: Parse and update module config structure Shreyas NC (5): ASoC: Intel: Skylake: Parse multiple manifest data blocks ASoC: Intel: uapi: Add new tokens for module common data ASoC: Intel: Skylake: Commonize parsing of format tokens ASoC: Intel: Skylake: Add driver structures to be filled from topology manifest ASoC: Intel: Skylake: Populate module data from topology manifest include/uapi/sound/snd_sst_tokens.h | 92 ++++- sound/soc/intel/skylake/skl-messages.c | 59 ++- sound/soc/intel/skylake/skl-pcm.c | 44 +- sound/soc/intel/skylake/skl-topology.c | 713 ++++++++++++++++++++++++++++----- sound/soc/intel/skylake/skl-topology.h | 78 ++++ sound/soc/intel/skylake/skl.h | 2 + 6 files changed, 861 insertions(+), 127 deletions(-) -- 2.14.1

3 10

[alsa-devel] Applied "ASoC: Intel: Skylake: Populate module data from topology manifest" to the asoc tree
by Mark Brown 27 Aug '17

27 Aug '17

The patch ASoC: Intel: Skylake: Populate module data from topology manifest has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From db6ed55de6da1125ff349bfe4ef12d18c366e893 Mon Sep 17 00:00:00 2001 From: Shreyas NC <shreyas.nc(a)intel.com> Date: Wed, 23 Aug 2017 19:33:52 +0530 Subject: [PATCH] ASoC: Intel: Skylake: Populate module data from topology manifest All the module common data will now be populated in the topology manifest. This includes the resource and interface list supported by the module. With this, driver need not compute the resources required by each dsp module for a particular pcm parameter since it comes as a part of the topology manifest. So, add functions to parse the manifest tokens to populate the module config data structure. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> Acked-By: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 320 ++++++++++++++++++++++++++++++++- 1 file changed, 316 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index df808fbf14b5..f39883067f63 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2038,6 +2038,104 @@ static void skl_tplg_fill_pin_dynamic_val( } /* + * Resource table in the manifest has pin specific resources + * like pin and pin buffer size + */ +static int skl_tplg_manifest_pin_res_tkn(struct device *dev, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + struct skl_module_res *res, int pin_idx, int dir) +{ + struct skl_module_pin_resources *m_pin; + + switch (dir) { + case SKL_DIR_IN: + m_pin = &res->input[pin_idx]; + break; + + case SKL_DIR_OUT: + m_pin = &res->output[pin_idx]; + break; + + default: + dev_err(dev, "Invalid pin direction: %d\n", dir); + return -EINVAL; + } + + switch (tkn_elem->token) { + case SKL_TKN_MM_U32_RES_PIN_ID: + m_pin->pin_index = tkn_elem->value; + break; + + case SKL_TKN_MM_U32_PIN_BUF: + m_pin->buf_size = tkn_elem->value; + break; + + default: + dev_err(dev, "Invalid token: %d\n", tkn_elem->token); + return -EINVAL; + } + + return 0; +} + +/* + * Fill module specific resources from the manifest's resource + * table like CPS, DMA size, mem_pages. + */ +static int skl_tplg_fill_res_tkn(struct device *dev, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + struct skl_module_res *res, + int pin_idx, int dir) +{ + int ret, tkn_count = 0; + + if (!res) + return -EINVAL; + + switch (tkn_elem->token) { + case SKL_TKN_MM_U32_CPS: + res->cps = tkn_elem->value; + break; + + case SKL_TKN_MM_U32_DMA_SIZE: + res->dma_buffer_size = tkn_elem->value; + break; + + case SKL_TKN_MM_U32_CPC: + res->cpc = tkn_elem->value; + break; + + case SKL_TKN_U32_MEM_PAGES: + res->is_pages = tkn_elem->value; + break; + + case SKL_TKN_U32_OBS: + res->obs = tkn_elem->value; + break; + + case SKL_TKN_U32_IBS: + res->ibs = tkn_elem->value; + break; + + case SKL_TKN_MM_U32_RES_PIN_ID: + case SKL_TKN_MM_U32_PIN_BUF: + ret = skl_tplg_manifest_pin_res_tkn(dev, tkn_elem, res, + pin_idx, dir); + if (ret < 0) + return ret; + break; + + default: + dev_err(dev, "Not a res type token: %d", tkn_elem->token); + return -EINVAL; + + } + tkn_count++; + + return tkn_count; +} + +/* * Parse tokens to fill up the module private data */ static int skl_tplg_get_token(struct device *dev, @@ -2625,13 +2723,13 @@ static int skl_tplg_fill_str_mfest_tkn(struct device *dev, str_elem->string, ARRAY_SIZE(skl->skl_sst->lib_info[ref_count].name)); ref_count++; - tkn_count++; break; default: dev_err(dev, "Not a string token %d\n", str_elem->token); break; } + tkn_count++; return tkn_count; } @@ -2657,26 +2755,236 @@ static int skl_tplg_get_str_tkn(struct device *dev, return tkn_count; } +static int skl_tplg_manifest_fill_fmt(struct device *dev, + struct skl_module_iface *fmt, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + u32 dir, int fmt_idx) +{ + struct skl_module_pin_fmt *dst_fmt; + struct skl_module_fmt *mod_fmt; + int ret; + + if (!fmt) + return -EINVAL; + + switch (dir) { + case SKL_DIR_IN: + dst_fmt = &fmt->inputs[fmt_idx]; + break; + + case SKL_DIR_OUT: + dst_fmt = &fmt->outputs[fmt_idx]; + break; + + default: + dev_err(dev, "Invalid direction: %d\n", dir); + return -EINVAL; + } + + mod_fmt = &dst_fmt->fmt; + + switch (tkn_elem->token) { + case SKL_TKN_MM_U32_INTF_PIN_ID: + dst_fmt->id = tkn_elem->value; + break; + + default: + ret = skl_tplg_fill_fmt(dev, mod_fmt, tkn_elem->token, + tkn_elem->value); + if (ret < 0) + return ret; + break; + } + + return 0; +} + +static int skl_tplg_fill_mod_info(struct device *dev, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + struct skl_module *mod) +{ + + if (!mod) + return -EINVAL; + + switch (tkn_elem->token) { + case SKL_TKN_U8_IN_PIN_TYPE: + mod->input_pin_type = tkn_elem->value; + break; + + case SKL_TKN_U8_OUT_PIN_TYPE: + mod->output_pin_type = tkn_elem->value; + break; + + case SKL_TKN_U8_IN_QUEUE_COUNT: + mod->max_input_pins = tkn_elem->value; + break; + + case SKL_TKN_U8_OUT_QUEUE_COUNT: + mod->max_output_pins = tkn_elem->value; + break; + + case SKL_TKN_MM_U8_NUM_RES: + mod->nr_resources = tkn_elem->value; + break; + + case SKL_TKN_MM_U8_NUM_INTF: + mod->nr_interfaces = tkn_elem->value; + break; + + default: + dev_err(dev, "Invalid mod info token %d", tkn_elem->token); + return -EINVAL; + } + + return 0; +} + + static int skl_tplg_get_int_tkn(struct device *dev, struct snd_soc_tplg_vendor_value_elem *tkn_elem, struct skl *skl) { - int tkn_count = 0; + int tkn_count = 0, ret; + static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx; + struct skl_module_res *res = NULL; + struct skl_module_iface *fmt = NULL; + struct skl_module *mod = NULL; + int i; + + if (skl->modules) { + mod = skl->modules[mod_idx]; + res = &mod->resources[res_val_idx]; + fmt = &mod->formats[intf_val_idx]; + } switch (tkn_elem->token) { case SKL_TKN_U32_LIB_COUNT: skl->skl_sst->lib_count = tkn_elem->value; - tkn_count++; + break; + + case SKL_TKN_U8_NUM_MOD: + skl->nr_modules = tkn_elem->value; + skl->modules = devm_kcalloc(dev, skl->nr_modules, + sizeof(*skl->modules), GFP_KERNEL); + if (!skl->modules) + return -ENOMEM; + + for (i = 0; i < skl->nr_modules; i++) { + skl->modules[i] = devm_kzalloc(dev, + sizeof(struct skl_module), GFP_KERNEL); + if (!skl->modules[i]) + return -ENOMEM; + } + break; + + case SKL_TKN_MM_U8_MOD_IDX: + mod_idx = tkn_elem->value; + break; + + case SKL_TKN_U8_IN_PIN_TYPE: + case SKL_TKN_U8_OUT_PIN_TYPE: + case SKL_TKN_U8_IN_QUEUE_COUNT: + case SKL_TKN_U8_OUT_QUEUE_COUNT: + case SKL_TKN_MM_U8_NUM_RES: + case SKL_TKN_MM_U8_NUM_INTF: + ret = skl_tplg_fill_mod_info(dev, tkn_elem, mod); + if (ret < 0) + return ret; + break; + + case SKL_TKN_U32_DIR_PIN_COUNT: + dir = tkn_elem->value & SKL_IN_DIR_BIT_MASK; + pin_idx = (tkn_elem->value & SKL_PIN_COUNT_MASK) >> 4; + break; + + case SKL_TKN_MM_U32_RES_ID: + if (!res) + return -EINVAL; + + res->id = tkn_elem->value; + res_val_idx = tkn_elem->value; + break; + + case SKL_TKN_MM_U32_FMT_ID: + if (!fmt) + return -EINVAL; + + fmt->fmt_idx = tkn_elem->value; + intf_val_idx = tkn_elem->value; + break; + + case SKL_TKN_MM_U32_CPS: + case SKL_TKN_MM_U32_DMA_SIZE: + case SKL_TKN_MM_U32_CPC: + case SKL_TKN_U32_MEM_PAGES: + case SKL_TKN_U32_OBS: + case SKL_TKN_U32_IBS: + case SKL_TKN_MM_U32_RES_PIN_ID: + case SKL_TKN_MM_U32_PIN_BUF: + ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, pin_idx, dir); + if (ret < 0) + return ret; + + break; + + case SKL_TKN_MM_U32_NUM_IN_FMT: + if (!fmt) + return -EINVAL; + + res->nr_input_pins = tkn_elem->value; + break; + + case SKL_TKN_MM_U32_NUM_OUT_FMT: + if (!fmt) + return -EINVAL; + + res->nr_output_pins = tkn_elem->value; + break; + + case SKL_TKN_U32_FMT_CH: + case SKL_TKN_U32_FMT_FREQ: + case SKL_TKN_U32_FMT_BIT_DEPTH: + case SKL_TKN_U32_FMT_SAMPLE_SIZE: + case SKL_TKN_U32_FMT_CH_CONFIG: + case SKL_TKN_U32_FMT_INTERLEAVE: + case SKL_TKN_U32_FMT_SAMPLE_TYPE: + case SKL_TKN_U32_FMT_CH_MAP: + case SKL_TKN_MM_U32_INTF_PIN_ID: + ret = skl_tplg_manifest_fill_fmt(dev, fmt, tkn_elem, + dir, pin_idx); + if (ret < 0) + return ret; break; default: dev_err(dev, "Not a manifest token %d\n", tkn_elem->token); return -EINVAL; } + tkn_count++; return tkn_count; } +static int skl_tplg_get_manifest_uuid(struct device *dev, + struct skl *skl, + struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) +{ + static int ref_count; + struct skl_module *mod; + + if (uuid_tkn->token == SKL_TKN_UUID) { + mod = skl->modules[ref_count]; + memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid)); + ref_count++; + } else { + dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); + return -EINVAL; + } + + return 0; +} + /* * Fill the manifest structure by parsing the tokens based on the * type. @@ -2709,7 +3017,11 @@ static int skl_tplg_get_manifest_tkn(struct device *dev, continue; case SND_SOC_TPLG_TUPLE_TYPE_UUID: - dev_warn(dev, "no uuid tokens for skl tplf manifest\n"); + ret = skl_tplg_get_manifest_uuid(dev, skl, array->uuid); + if (ret < 0) + return ret; + + tuple_size += sizeof(*array->uuid); continue; default: -- 2.13.2

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: Parse and update module config structure" to the asoc tree
by Mark Brown 27 Aug '17

27 Aug '17

The patch ASoC: Intel: Skylake: Parse and update module config structure has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f6fa56e22559ade7287ca7c1b7218a08f516f379 Mon Sep 17 00:00:00 2001 From: Ramesh Babu <ramesh.babu(a)intel.com> Date: Wed, 23 Aug 2017 19:33:53 +0530 Subject: [PATCH] ASoC: Intel: Skylake: Parse and update module config structure A dsp path and the modules in the path can support various pcm configurations. The list of supported pcm configurations from topology manifest would be stored and later selected runtime based on the hw pcm params. For legacy, module data is filled in the 0th index of resource and interface table. To accommodate both models, change the relevant structures and populate them by parsing newly defined tokens. This change is backward compatible with the existing model where driver computes the resources required by each dsp module. Signed-off-by: Ramesh Babu <ramesh.babu(a)intel.com> Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> Acked-By: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 59 ++++-- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl-topology.c | 342 +++++++++++++++++++++++++-------- sound/soc/intel/skylake/skl-topology.h | 25 +++ 4 files changed, 334 insertions(+), 95 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 697d07bc1b5a..ea83e8bfb136 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -443,9 +443,12 @@ static void skl_set_base_module_format(struct skl_sst *ctx, struct skl_module_cfg *mconfig, struct skl_base_cfg *base_cfg) { - struct skl_module_fmt *format = &mconfig->in_fmt[0]; + struct skl_module *module = mconfig->module; + struct skl_module_res *res = &module->resources[mconfig->res_idx]; + struct skl_module_iface *fmt = &module->formats[mconfig->fmt_idx]; + struct skl_module_fmt *format = &fmt->inputs[0].fmt; - base_cfg->audio_fmt.number_of_channels = (u8)format->channels; + base_cfg->audio_fmt.number_of_channels = format->channels; base_cfg->audio_fmt.s_freq = format->s_freq; base_cfg->audio_fmt.bit_depth = format->bit_depth; @@ -460,10 +463,10 @@ static void skl_set_base_module_format(struct skl_sst *ctx, base_cfg->audio_fmt.interleaving = format->interleaving_style; - base_cfg->cps = mconfig->mcps; - base_cfg->ibs = mconfig->ibs; - base_cfg->obs = mconfig->obs; - base_cfg->is_pages = mconfig->mem_pages; + base_cfg->cps = res->cps; + base_cfg->ibs = res->ibs; + base_cfg->obs = res->obs; + base_cfg->is_pages = res->is_pages; } /* @@ -551,6 +554,9 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, struct skl_cpr_cfg *cpr_mconfig) { u32 dma_io_buf; + struct skl_module_res *res; + int res_idx = mconfig->res_idx; + struct skl *skl = get_skl_ctx(ctx->dev); cpr_mconfig->gtw_cfg.node_id = skl_get_node_id(ctx, mconfig); @@ -559,19 +565,27 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, return; } + if (skl->nr_modules) { + res = &mconfig->module->resources[mconfig->res_idx]; + cpr_mconfig->gtw_cfg.dma_buffer_size = res->dma_buffer_size; + goto skip_buf_size_calc; + } else { + res = &mconfig->module->resources[res_idx]; + } + switch (mconfig->hw_conn_type) { case SKL_CONN_SOURCE: if (mconfig->dev_type == SKL_DEVICE_HDAHOST) - dma_io_buf = mconfig->ibs; + dma_io_buf = res->ibs; else - dma_io_buf = mconfig->obs; + dma_io_buf = res->obs; break; case SKL_CONN_SINK: if (mconfig->dev_type == SKL_DEVICE_HDAHOST) - dma_io_buf = mconfig->obs; + dma_io_buf = res->obs; else - dma_io_buf = mconfig->ibs; + dma_io_buf = res->ibs; break; default: @@ -586,11 +600,12 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, /* fallback to 2ms default value */ if (!cpr_mconfig->gtw_cfg.dma_buffer_size) { if (mconfig->hw_conn_type == SKL_CONN_SOURCE) - cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs; + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * res->obs; else - cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs; + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * res->ibs; } +skip_buf_size_calc: cpr_mconfig->cpr_feature_mask = 0; cpr_mconfig->gtw_cfg.config_length = 0; @@ -638,7 +653,9 @@ static void skl_setup_out_format(struct skl_sst *ctx, struct skl_module_cfg *mconfig, struct skl_audio_data_format *out_fmt) { - struct skl_module_fmt *format = &mconfig->out_fmt[0]; + struct skl_module *module = mconfig->module; + struct skl_module_iface *fmt = &module->formats[mconfig->fmt_idx]; + struct skl_module_fmt *format = &fmt->outputs[0].fmt; out_fmt->number_of_channels = (u8)format->channels; out_fmt->s_freq = format->s_freq; @@ -663,7 +680,9 @@ static void skl_set_src_format(struct skl_sst *ctx, struct skl_module_cfg *mconfig, struct skl_src_module_cfg *src_mconfig) { - struct skl_module_fmt *fmt = &mconfig->out_fmt[0]; + struct skl_module *module = mconfig->module; + struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx]; + struct skl_module_fmt *fmt = &iface->outputs[0].fmt; skl_set_base_module_format(ctx, mconfig, (struct skl_base_cfg *)src_mconfig); @@ -680,7 +699,9 @@ static void skl_set_updown_mixer_format(struct skl_sst *ctx, struct skl_module_cfg *mconfig, struct skl_up_down_mixer_cfg *mixer_mconfig) { - struct skl_module_fmt *fmt = &mconfig->out_fmt[0]; + struct skl_module *module = mconfig->module; + struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx]; + struct skl_module_fmt *fmt = &iface->outputs[0].fmt; int i = 0; skl_set_base_module_format(ctx, mconfig, @@ -1013,8 +1034,8 @@ int skl_unbind_modules(struct skl_sst *ctx, struct skl_ipc_bind_unbind_msg msg; struct skl_module_inst_id src_id = src_mcfg->id; struct skl_module_inst_id dst_id = dst_mcfg->id; - int in_max = dst_mcfg->max_in_queue; - int out_max = src_mcfg->max_out_queue; + int in_max = dst_mcfg->module->max_input_pins; + int out_max = src_mcfg->module->max_output_pins; int src_index, dst_index, src_pin_state, dst_pin_state; skl_dump_bind_info(ctx, src_mcfg, dst_mcfg); @@ -1075,8 +1096,8 @@ int skl_bind_modules(struct skl_sst *ctx, { int ret; struct skl_ipc_bind_unbind_msg msg; - int in_max = dst_mcfg->max_in_queue; - int out_max = src_mcfg->max_out_queue; + int in_max = dst_mcfg->module->max_input_pins; + int out_max = src_mcfg->module->max_output_pins; int src_index, dst_index; skl_dump_bind_info(ctx, src_mcfg, dst_mcfg); diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index f6c9adb6faa2..a756882b9029 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1233,7 +1233,8 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) list_for_each_entry(module, &ctx->uuid_list, list) { if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { mconfig->id.module_id = module->id; - mconfig->is_loadable = module->is_loadable; + if (mconfig->module) + mconfig->module->loadable = module->is_loadable; return 0; } } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index f39883067f63..22f768ca3c73 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -49,6 +49,9 @@ static const int mic_quatro_list[][SKL_CH_QUATRO] = { {0, 1, 2, 3}, }; +#define CHECK_HW_PARAMS(ch, freq, bps, prm_ch, prm_freq, prm_bps) \ + ((ch == prm_ch) && (bps == prm_bps) && (freq == prm_freq)) + void skl_tplg_d0i3_get(struct skl *skl, enum d0i3_capability caps) { struct skl_d0i3_data *d0i3 = &skl->skl_sst->d0i3; @@ -153,8 +156,10 @@ static bool skl_is_pipe_mcps_avail(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; + u8 res_idx = mconfig->res_idx; + struct skl_module_res *res = &mconfig->module->resources[res_idx]; - if (skl->resource.mcps + mconfig->mcps > skl->resource.max_mcps) { + if (skl->resource.mcps + res->cps > skl->resource.max_mcps) { dev_err(ctx->dev, "%s: module_id %d instance %d\n", __func__, mconfig->id.module_id, mconfig->id.instance_id); @@ -170,7 +175,10 @@ static bool skl_is_pipe_mcps_avail(struct skl *skl, static void skl_tplg_alloc_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig) { - skl->resource.mcps += mconfig->mcps; + u8 res_idx = mconfig->res_idx; + struct skl_module_res *res = &mconfig->module->resources[res_idx]; + + skl->resource.mcps += res->cps; } /* @@ -179,7 +187,11 @@ static void skl_tplg_alloc_pipe_mcps(struct skl *skl, static void skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig) { - skl->resource.mcps -= mconfig->mcps; + u8 res_idx = mconfig->res_idx; + struct skl_module_res *res = &mconfig->module->resources[res_idx]; + + res = &mconfig->module->resources[res_idx]; + skl->resource.mcps -= res->cps; } /* @@ -195,17 +207,21 @@ skl_tplg_free_pipe_mem(struct skl *skl, struct skl_module_cfg *mconfig) static void skl_dump_mconfig(struct skl_sst *ctx, struct skl_module_cfg *mcfg) { + struct skl_module_iface *iface = &mcfg->module->formats[0]; + dev_dbg(ctx->dev, "Dumping config\n"); dev_dbg(ctx->dev, "Input Format:\n"); - dev_dbg(ctx->dev, "channels = %d\n", mcfg->in_fmt[0].channels); - dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->in_fmt[0].s_freq); - dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->in_fmt[0].ch_cfg); - dev_dbg(ctx->dev, "valid bit depth = %d\n", mcfg->in_fmt[0].valid_bit_depth); + dev_dbg(ctx->dev, "channels = %d\n", iface->inputs[0].fmt.channels); + dev_dbg(ctx->dev, "s_freq = %d\n", iface->inputs[0].fmt.s_freq); + dev_dbg(ctx->dev, "ch_cfg = %d\n", iface->inputs[0].fmt.ch_cfg); + dev_dbg(ctx->dev, "valid bit depth = %d\n", + iface->inputs[0].fmt.valid_bit_depth); dev_dbg(ctx->dev, "Output Format:\n"); - dev_dbg(ctx->dev, "channels = %d\n", mcfg->out_fmt[0].channels); - dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->out_fmt[0].s_freq); - dev_dbg(ctx->dev, "valid bit depth = %d\n", mcfg->out_fmt[0].valid_bit_depth); - dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg); + dev_dbg(ctx->dev, "channels = %d\n", iface->outputs[0].fmt.channels); + dev_dbg(ctx->dev, "s_freq = %d\n", iface->outputs[0].fmt.s_freq); + dev_dbg(ctx->dev, "valid bit depth = %d\n", + iface->outputs[0].fmt.valid_bit_depth); + dev_dbg(ctx->dev, "ch_cfg = %d\n", iface->outputs[0].fmt.ch_cfg); } static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs) @@ -273,8 +289,8 @@ static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg, struct skl_module_fmt *in_fmt, *out_fmt; /* Fixups will be applied to pin 0 only */ - in_fmt = &m_cfg->in_fmt[0]; - out_fmt = &m_cfg->out_fmt[0]; + in_fmt = &m_cfg->module->formats[0].inputs[0].fmt; + out_fmt = &m_cfg->module->formats[0].outputs[0].fmt; if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (is_fe) { @@ -312,21 +328,23 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, { int multiplier = 1; struct skl_module_fmt *in_fmt, *out_fmt; + struct skl_module_res *res; /* Since fixups is applied to pin 0 only, ibs, obs needs * change for pin 0 only */ - in_fmt = &mcfg->in_fmt[0]; - out_fmt = &mcfg->out_fmt[0]; + res = &mcfg->module->resources[0]; + in_fmt = &mcfg->module->formats[0].inputs[0].fmt; + out_fmt = &mcfg->module->formats[0].outputs[0].fmt; if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) multiplier = 5; - mcfg->ibs = DIV_ROUND_UP(in_fmt->s_freq, 1000) * + res->ibs = DIV_ROUND_UP(in_fmt->s_freq, 1000) * in_fmt->channels * (in_fmt->bit_depth >> 3) * multiplier; - mcfg->obs = DIV_ROUND_UP(out_fmt->s_freq, 1000) * + res->obs = DIV_ROUND_UP(out_fmt->s_freq, 1000) * out_fmt->channels * (out_fmt->bit_depth >> 3) * multiplier; } @@ -365,6 +383,8 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, struct nhlt_specific_cfg *cfg; struct skl *skl = get_skl_ctx(ctx->dev); u8 dev_type = skl_tplg_be_dev_type(m_cfg->dev_type); + int fmt_idx = m_cfg->fmt_idx; + struct skl_module_iface *m_iface = &m_cfg->module->formats[fmt_idx]; /* check if we already have blob */ if (m_cfg->formats_config.caps_size > 0) @@ -375,23 +395,23 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, case SKL_DEVICE_DMIC: link_type = NHLT_LINK_DMIC; dir = SNDRV_PCM_STREAM_CAPTURE; - s_freq = m_cfg->in_fmt[0].s_freq; - s_fmt = m_cfg->in_fmt[0].bit_depth; - ch = m_cfg->in_fmt[0].channels; + s_freq = m_iface->inputs[0].fmt.s_freq; + s_fmt = m_iface->inputs[0].fmt.bit_depth; + ch = m_iface->inputs[0].fmt.channels; break; case SKL_DEVICE_I2S: link_type = NHLT_LINK_SSP; if (m_cfg->hw_conn_type == SKL_CONN_SOURCE) { dir = SNDRV_PCM_STREAM_PLAYBACK; - s_freq = m_cfg->out_fmt[0].s_freq; - s_fmt = m_cfg->out_fmt[0].bit_depth; - ch = m_cfg->out_fmt[0].channels; + s_freq = m_iface->outputs[0].fmt.s_freq; + s_fmt = m_iface->outputs[0].fmt.bit_depth; + ch = m_iface->outputs[0].fmt.channels; } else { dir = SNDRV_PCM_STREAM_CAPTURE; - s_freq = m_cfg->in_fmt[0].s_freq; - s_fmt = m_cfg->in_fmt[0].bit_depth; - ch = m_cfg->in_fmt[0].channels; + s_freq = m_iface->inputs[0].fmt.s_freq; + s_fmt = m_iface->inputs[0].fmt.bit_depth; + ch = m_iface->inputs[0].fmt.channels; } break; @@ -549,6 +569,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) struct snd_soc_dapm_widget *w; struct skl_module_cfg *mconfig; struct skl_sst *ctx = skl->skl_sst; + u8 cfg_idx; int ret = 0; list_for_each_entry(w_module, &pipe->w_list, node) { @@ -564,11 +585,15 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) return -EIO; } + cfg_idx = mconfig->pipe->cur_config_idx; + mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx; + mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx; + /* check resource available */ if (!skl_is_pipe_mcps_avail(skl, mconfig)) return -ENOMEM; - if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) { + if (mconfig->module->loadable && ctx->dsp->fw_ops.load_mod) { ret = ctx->dsp->fw_ops.load_mod(ctx->dsp, mconfig->id.module_id, mconfig->guid); if (ret < 0) @@ -633,7 +658,7 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, mconfig = w_module->w->priv; uuid_mod = (uuid_le *)mconfig->guid; - if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod && + if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod && mconfig->m_state > SKL_MODULE_UNINIT) { ret = ctx->dsp->fw_ops.unload_mod(ctx->dsp, mconfig->id.module_id); @@ -655,6 +680,65 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, } /* + * Here, we select pipe format based on the pipe type and pipe + * direction to determine the current config index for the pipeline. + * The config index is then used to select proper module resources. + * Intermediate pipes currently have a fixed format hence we select the + * 0th configuratation by default for such pipes. + */ +static int +skl_tplg_get_pipe_config(struct skl *skl, struct skl_module_cfg *mconfig) +{ + struct skl_sst *ctx = skl->skl_sst; + struct skl_pipe *pipe = mconfig->pipe; + struct skl_pipe_params *params = pipe->p_params; + struct skl_path_config *pconfig = &pipe->configs[0]; + struct skl_pipe_fmt *fmt = NULL; + bool in_fmt = false; + int i; + + if (pipe->nr_cfgs == 0) { + pipe->cur_config_idx = 0; + return 0; + } + + if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE) { + dev_dbg(ctx->dev, "No conn_type detected, take 0th config\n"); + pipe->cur_config_idx = 0; + pipe->memory_pages = pconfig->mem_pages; + + return 0; + } + + if ((pipe->conn_type == SKL_PIPE_CONN_TYPE_FE && + pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) || + (pipe->conn_type == SKL_PIPE_CONN_TYPE_BE && + pipe->direction == SNDRV_PCM_STREAM_CAPTURE)) + in_fmt = true; + + for (i = 0; i < pipe->nr_cfgs; i++) { + pconfig = &pipe->configs[i]; + if (in_fmt) + fmt = &pconfig->in_fmt; + else + fmt = &pconfig->out_fmt; + + if (CHECK_HW_PARAMS(params->ch, params->s_freq, params->s_fmt, + fmt->channels, fmt->freq, fmt->bps)) { + pipe->cur_config_idx = i; + pipe->memory_pages = pconfig->mem_pages; + dev_dbg(ctx->dev, "Using pipe config: %d\n", i); + + return 0; + } + } + + dev_err(ctx->dev, "Invalid pipe config: %d %d %d for pipe: %d\n", + params->ch, params->s_freq, params->s_fmt, pipe->ppl_id); + return -EINVAL; +} + +/* * Mixer module represents a pipeline. So in the Pre-PMU event of mixer we * need create the pipeline. So we do following: * - check the resources @@ -673,6 +757,10 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, struct skl_sst *ctx = skl->skl_sst; struct skl_module_deferred_bind *modules; + ret = skl_tplg_get_pipe_config(skl, mconfig); + if (ret < 0) + return ret; + /* check resource available */ if (!skl_is_pipe_mcps_avail(skl, mconfig)) return -EBUSY; @@ -776,12 +864,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, * check all out/in pins are in bind state. * if so set the module param */ - for (i = 0; i < mcfg->max_out_queue; i++) { + for (i = 0; i < mcfg->module->max_output_pins; i++) { if (mcfg->m_out_pin[i].pin_state != SKL_PIN_BIND_DONE) return 0; } - for (i = 0; i < mcfg->max_in_queue; i++) { + for (i = 0; i < mcfg->module->max_input_pins; i++) { if (mcfg->m_in_pin[i].pin_state != SKL_PIN_BIND_DONE) return 0; } @@ -832,7 +920,7 @@ static int skl_tplg_module_add_deferred_bind(struct skl *skl, int i; /* only supported for module with static pin connection */ - for (i = 0; i < dst->max_in_queue; i++) { + for (i = 0; i < dst->module->max_input_pins; i++) { struct skl_module_pin *pin = &dst->m_in_pin[i]; if (pin->is_dynamic) @@ -1092,7 +1180,7 @@ static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w, if (ret) return ret; - for (i = 0; i < sink_mconfig->max_in_queue; i++) { + for (i = 0; i < sink_mconfig->module->max_input_pins; i++) { if (sink_mconfig->m_in_pin[i].pin_state == SKL_PIN_BIND_DONE) { src_mconfig = sink_mconfig->m_in_pin[i].tgt_mcfg; if (!src_mconfig) @@ -1203,7 +1291,7 @@ static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, if (ret) return ret; - for (i = 0; i < src_mconfig->max_out_queue; i++) { + for (i = 0; i < src_mconfig->module->max_output_pins; i++) { if (src_mconfig->m_out_pin[i].pin_state == SKL_PIN_BIND_DONE) { sink_mconfig = src_mconfig->m_out_pin[i].tgt_mcfg; if (!sink_mconfig) @@ -1497,14 +1585,22 @@ int skl_tplg_update_pipe_params(struct device *dev, struct skl_module_cfg *mconfig, struct skl_pipe_params *params) { + struct skl_module_res *res = &mconfig->module->resources[0]; + struct skl *skl = get_skl_ctx(dev); struct skl_module_fmt *format = NULL; + u8 cfg_idx = mconfig->pipe->cur_config_idx; skl_tplg_fill_dma_id(mconfig, params); + mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx; + mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx; + + if (skl->nr_modules) + return 0; if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) - format = &mconfig->in_fmt[0]; + format = &mconfig->module->formats[0].inputs[0].fmt; else - format = &mconfig->out_fmt[0]; + format = &mconfig->module->formats[0].outputs[0].fmt; /* set the hw_params */ format->s_freq = params->s_freq; @@ -1532,11 +1628,11 @@ int skl_tplg_update_pipe_params(struct device *dev, } if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { - mconfig->ibs = (format->s_freq / 1000) * + res->ibs = (format->s_freq / 1000) * (format->channels) * (format->bit_depth >> 3); } else { - mconfig->obs = (format->s_freq / 1000) * + res->obs = (format->s_freq / 1000) * (format->channels) * (format->bit_depth >> 3); } @@ -1810,6 +1906,54 @@ static const struct snd_soc_tplg_kcontrol_ops skl_tplg_kcontrol_ops[] = { }, }; +static int skl_tplg_fill_pipe_cfg(struct device *dev, + struct skl_pipe *pipe, u32 tkn, + u32 tkn_val, int conf_idx, int dir) +{ + struct skl_pipe_fmt *fmt; + struct skl_path_config *config; + + switch (dir) { + case SKL_DIR_IN: + fmt = &pipe->configs[conf_idx].in_fmt; + break; + + case SKL_DIR_OUT: + fmt = &pipe->configs[conf_idx].out_fmt; + break; + + default: + dev_err(dev, "Invalid direction: %d\n", dir); + return -EINVAL; + } + + config = &pipe->configs[conf_idx]; + + switch (tkn) { + case SKL_TKN_U32_CFG_FREQ: + fmt->freq = tkn_val; + break; + + case SKL_TKN_U8_CFG_CHAN: + fmt->channels = tkn_val; + break; + + case SKL_TKN_U8_CFG_BPS: + fmt->bps = tkn_val; + break; + + case SKL_TKN_U32_PATH_MEM_PGS: + config->mem_pages = tkn_val; + break; + + default: + dev_err(dev, "Invalid token config: %d\n", tkn); + return -EINVAL; + } + + return 0; +} + static int skl_tplg_fill_pipe_tkn(struct device *dev, struct skl_pipe *pipe, u32 tkn, u32 tkn_val) @@ -1832,6 +1976,14 @@ static int skl_tplg_fill_pipe_tkn(struct device *dev, pipe->lp_mode = tkn_val; break; + case SKL_TKN_U32_PIPE_DIRECTION: + pipe->direction = tkn_val; + break; + + case SKL_TKN_U32_NUM_CONFIGS: + pipe->nr_cfgs = tkn_val; + break; + default: dev_err(dev, "Token not handled %d\n", tkn); return -EINVAL; @@ -1993,18 +2145,21 @@ static int skl_tplg_fill_fmt(struct device *dev, } static int skl_tplg_widget_fill_fmt(struct device *dev, - struct skl_module_cfg *mconfig, + struct skl_module_iface *fmt, u32 tkn, u32 val, u32 dir, int fmt_idx) { struct skl_module_fmt *dst_fmt; + if (!fmt) + return -EINVAL; + switch (dir) { case SKL_DIR_IN: - dst_fmt = &mconfig->in_fmt[fmt_idx]; + dst_fmt = &fmt->inputs[fmt_idx].fmt; break; case SKL_DIR_OUT: - dst_fmt = &mconfig->out_fmt[fmt_idx]; + dst_fmt = &fmt->outputs[fmt_idx].fmt; break; default: @@ -2117,6 +2272,10 @@ static int skl_tplg_fill_res_tkn(struct device *dev, res->ibs = tkn_elem->value; break; + case SKL_TKN_U32_MAX_MCPS: + res->cps = tkn_elem->value; + break; + case SKL_TKN_MM_U32_RES_PIN_ID: case SKL_TKN_MM_U32_PIN_BUF: ret = skl_tplg_manifest_pin_res_tkn(dev, tkn_elem, res, @@ -2145,49 +2304,54 @@ static int skl_tplg_get_token(struct device *dev, int tkn_count = 0; int ret; static int is_pipe_exists; - static int pin_index, dir; + static int pin_index, dir, conf_idx; + struct skl_module_iface *iface = NULL; + struct skl_module_res *res = NULL; + int res_idx = mconfig->res_idx; + int fmt_idx = mconfig->fmt_idx; + + /* + * If the manifest structure contains no modules, fill all + * the module data to 0th index. + * res_idx and fmt_idx are default set to 0. + */ + if (skl->nr_modules == 0) { + res = &mconfig->module->resources[res_idx]; + iface = &mconfig->module->formats[fmt_idx]; + } if (tkn_elem->token > SKL_TKN_MAX) return -EINVAL; switch (tkn_elem->token) { case SKL_TKN_U8_IN_QUEUE_COUNT: - mconfig->max_in_queue = tkn_elem->value; - mconfig->m_in_pin = devm_kzalloc(dev, mconfig->max_in_queue * - sizeof(*mconfig->m_in_pin), - GFP_KERNEL); - if (!mconfig->m_in_pin) - return -ENOMEM; - + mconfig->module->max_input_pins = tkn_elem->value; break; case SKL_TKN_U8_OUT_QUEUE_COUNT: - mconfig->max_out_queue = tkn_elem->value; - mconfig->m_out_pin = devm_kzalloc(dev, mconfig->max_out_queue * - sizeof(*mconfig->m_out_pin), - GFP_KERNEL); - - if (!mconfig->m_out_pin) - return -ENOMEM; - + mconfig->module->max_output_pins = tkn_elem->value; break; case SKL_TKN_U8_DYN_IN_PIN: if (!mconfig->m_in_pin) + mconfig->m_in_pin = devm_kzalloc(dev, MAX_IN_QUEUE * + sizeof(*mconfig->m_in_pin), GFP_KERNEL); + if (!mconfig->m_in_pin) return -ENOMEM; - skl_tplg_fill_pin_dynamic_val(mconfig->m_in_pin, - mconfig->max_in_queue, tkn_elem->value); - + skl_tplg_fill_pin_dynamic_val(mconfig->m_in_pin, MAX_IN_QUEUE, + tkn_elem->value); break; case SKL_TKN_U8_DYN_OUT_PIN: if (!mconfig->m_out_pin) + mconfig->m_out_pin = devm_kzalloc(dev, MAX_IN_QUEUE * + sizeof(*mconfig->m_in_pin), GFP_KERNEL); + if (!mconfig->m_out_pin) return -ENOMEM; - skl_tplg_fill_pin_dynamic_val(mconfig->m_out_pin, - mconfig->max_out_queue, tkn_elem->value); - + skl_tplg_fill_pin_dynamic_val(mconfig->m_out_pin, MAX_OUT_QUEUE, + tkn_elem->value); break; case SKL_TKN_U8_TIME_SLOT: @@ -2215,19 +2379,13 @@ static int skl_tplg_get_token(struct device *dev, break; case SKL_TKN_U32_MEM_PAGES: - mconfig->mem_pages = tkn_elem->value; - break; - case SKL_TKN_U32_MAX_MCPS: - mconfig->mcps = tkn_elem->value; - break; - case SKL_TKN_U32_OBS: - mconfig->obs = tkn_elem->value; - break; - case SKL_TKN_U32_IBS: - mconfig->ibs = tkn_elem->value; + ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, dir, pin_index); + if (ret < 0) + return ret; + break; case SKL_TKN_U32_VBUS_ID: @@ -2260,10 +2418,16 @@ static int skl_tplg_get_token(struct device *dev, break; + case SKL_TKN_U32_PIPE_CONFIG_ID: + conf_idx = tkn_elem->value; + break; + case SKL_TKN_U32_PIPE_CONN_TYPE: case SKL_TKN_U32_PIPE_PRIORITY: case SKL_TKN_U32_PIPE_MEM_PGS: case SKL_TKN_U32_PMODE: + case SKL_TKN_U32_PIPE_DIRECTION: + case SKL_TKN_U32_NUM_CONFIGS: if (is_pipe_exists) { ret = skl_tplg_fill_pipe_tkn(dev, mconfig->pipe, tkn_elem->token, tkn_elem->value); @@ -2273,6 +2437,27 @@ static int skl_tplg_get_token(struct device *dev, break; + case SKL_TKN_U32_PATH_MEM_PGS: + case SKL_TKN_U32_CFG_FREQ: + case SKL_TKN_U8_CFG_CHAN: + case SKL_TKN_U8_CFG_BPS: + if (mconfig->pipe->nr_cfgs) { + ret = skl_tplg_fill_pipe_cfg(dev, mconfig->pipe, + tkn_elem->token, tkn_elem->value, + conf_idx, dir); + if (ret < 0) + return ret; + } + break; + + case SKL_TKN_CFG_MOD_RES_ID: + mconfig->mod_cfg[conf_idx].res_idx = tkn_elem->value; + break; + + case SKL_TKN_CFG_MOD_FMT_ID: + mconfig->mod_cfg[conf_idx].fmt_idx = tkn_elem->value; + break; + /* * SKL_TKN_U32_DIR_PIN_COUNT token has the value for both * direction and the pin count. The first four bits represent @@ -2293,7 +2478,7 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U32_FMT_INTERLEAVE: case SKL_TKN_U32_FMT_SAMPLE_TYPE: case SKL_TKN_U32_FMT_CH_MAP: - ret = skl_tplg_widget_fill_fmt(dev, mconfig, tkn_elem->token, + ret = skl_tplg_widget_fill_fmt(dev, iface, tkn_elem->token, tkn_elem->value, dir, pin_index); if (ret < 0) @@ -2518,11 +2703,11 @@ static void skl_clear_pin_config(struct snd_soc_platform *platform, strlen(platform->component.name))) { mconfig = w->priv; pipe = mconfig->pipe; - for (i = 0; i < mconfig->max_in_queue; i++) { + for (i = 0; i < mconfig->module->max_input_pins; i++) { mconfig->m_in_pin[i].in_use = false; mconfig->m_in_pin[i].pin_state = SKL_PIN_UNBIND; } - for (i = 0; i < mconfig->max_out_queue; i++) { + for (i = 0; i < mconfig->module->max_output_pins; i++) { mconfig->m_out_pin[i].in_use = false; mconfig->m_out_pin[i].pin_state = SKL_PIN_UNBIND; } @@ -2581,6 +2766,13 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, if (!mconfig) return -ENOMEM; + if (skl->nr_modules == 0) { + mconfig->module = devm_kzalloc(bus->dev, + sizeof(*mconfig->module), GFP_KERNEL); + if (!mconfig->module) + return -ENOMEM; + } + w->priv = mconfig; /* diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 93b2ba8c40f7..90903a15fa1d 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -273,6 +273,23 @@ struct skl_pipe_params { unsigned int link_bps; }; +struct skl_pipe_fmt { + u32 freq; + u8 channels; + u8 bps; +}; + +struct skl_pipe_mcfg { + u8 res_idx; + u8 fmt_idx; +}; + +struct skl_path_config { + u8 mem_pages; + struct skl_pipe_fmt in_fmt; + struct skl_pipe_fmt out_fmt; +}; + struct skl_pipe { u8 ppl_id; u8 pipe_priority; @@ -281,6 +298,10 @@ struct skl_pipe { u8 lp_mode; struct skl_pipe_params *p_params; enum skl_pipe_state state; + u8 direction; + u8 cur_config_idx; + u8 nr_cfgs; + struct skl_path_config configs[SKL_MAX_PATH_CONFIGS]; struct list_head w_list; bool passthru; }; @@ -347,6 +368,9 @@ struct skl_module { struct skl_module_cfg { u8 guid[16]; struct skl_module_inst_id id; + struct skl_module *module; + int res_idx; + int fmt_idx; u8 domain; bool homogenous_inputs; bool homogenous_outputs; @@ -381,6 +405,7 @@ struct skl_module_cfg { enum skl_module_state m_state; struct skl_pipe *pipe; struct skl_specific_cfg formats_config; + struct skl_pipe_mcfg mod_cfg[SKL_MAX_MODULES_IN_PIPE]; }; struct skl_algo_data { -- 2.13.2

1 0

[alsa-devel] [PATCH] ASoC: use snd_soc_component_get_dapm()
by Kuninori Morimoto 27 Aug '17

27 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we have snd_soc_component_get_dapm(), and as soc.h say below, let's use it. /* Don't use these, use snd_soc_component_get_dapm() */ struct snd_soc_dapm_context dapm; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 03875ea..01b2d26 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3222,7 +3222,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->pcm_new = component->driver->pcm_new; component->pcm_free = component->driver->pcm_free; - dapm = &component->dapm; + dapm = snd_soc_component_get_dapm(component); dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: nau8825: correct typo of semaphore comment
by John Hsu 27 Aug '17

27 Aug '17

There are a lot of typo about semaphore in the comment. Correct it from semaphone to semaphore. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 000aa79..714ce17 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -260,11 +260,11 @@ static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) if (timeout) { ret = down_timeout(&nau8825->xtalk_sem, timeout); if (ret < 0) - dev_warn(nau8825->dev, "Acquire semaphone timeout\n"); + dev_warn(nau8825->dev, "Acquire semaphore timeout\n"); } else { ret = down_interruptible(&nau8825->xtalk_sem); if (ret < 0) - dev_warn(nau8825->dev, "Acquire semaphone fail\n"); + dev_warn(nau8825->dev, "Acquire semaphore fail\n"); } return ret; @@ -1299,7 +1299,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, NAU8825_I2S_DL_MASK, val_len); - /* Release the semaphone. */ + /* Release the semaphore. */ nau8825_sema_release(nau8825); return 0; @@ -1361,7 +1361,7 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, NAU8825_I2S_MS_MASK, ctrl2_val); - /* Release the semaphone. */ + /* Release the semaphore. */ nau8825_sema_release(nau8825); return 0; @@ -2140,7 +2140,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_MCLK: - /* Acquire the semaphone to synchronize the playback and + /* Acquire the semaphore to synchronize the playback and * interrupt handler. In order to avoid the playback inter- * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. @@ -2150,7 +2150,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, /* MCLK not changed by clock tree */ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0); - /* Release the semaphone. */ + /* Release the semaphore. */ nau8825_sema_release(nau8825); ret = nau8825_mclk_prepare(nau8825, freq); @@ -2188,7 +2188,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_FLL_MCLK: - /* Acquire the semaphone to synchronize the playback and + /* Acquire the semaphore to synchronize the playback and * interrupt handler. In order to avoid the playback inter- * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. @@ -2201,7 +2201,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_FLL3, NAU8825_FLL_CLK_SRC_MASK | NAU8825_GAIN_ERR_MASK, NAU8825_FLL_CLK_SRC_MCLK | 0); - /* Release the semaphone. */ + /* Release the semaphore. */ nau8825_sema_release(nau8825); ret = nau8825_mclk_prepare(nau8825, freq); @@ -2210,7 +2210,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_FLL_BLK: - /* Acquire the semaphone to synchronize the playback and + /* Acquire the semaphore to synchronize the playback and * interrupt handler. In order to avoid the playback inter- * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. @@ -2226,7 +2226,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_FLL_CLK_SRC_MASK | NAU8825_GAIN_ERR_MASK, NAU8825_FLL_CLK_SRC_BLK | (0xf << NAU8825_GAIN_ERR_SFT)); - /* Release the semaphone. */ + /* Release the semaphore. */ nau8825_sema_release(nau8825); if (nau8825->mclk_freq) { @@ -2236,7 +2236,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_FLL_FS: - /* Acquire the semaphone to synchronize the playback and + /* Acquire the semaphore to synchronize the playback and * interrupt handler. In order to avoid the playback inter- * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. @@ -2252,7 +2252,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_FLL_CLK_SRC_MASK | NAU8825_GAIN_ERR_MASK, NAU8825_FLL_CLK_SRC_FS | (0xf << NAU8825_GAIN_ERR_SFT)); - /* Release the semaphone. */ + /* Release the semaphore. */ nau8825_sema_release(nau8825); if (nau8825->mclk_freq) { @@ -2563,7 +2563,7 @@ static int nau8825_i2c_probe(struct i2c_client *i2c, return PTR_ERR(nau8825->regmap); nau8825->dev = dev; nau8825->irq = i2c->irq; - /* Initiate parameters, semaphone and work queue which are needed in + /* Initiate parameters, semaphore and work queue which are needed in * cross talk suppression measurment function. */ nau8825->xtalk_state = NAU8825_XTALK_DONE; -- 2.6.4

2 1

[alsa-devel] [PATCH 0/3 v2] ASoC: remove duplicated definitions
by Kuninori Morimoto 27 Aug '17

27 Aug '17

Hi Mark These are v2 of "ASoC: remove duplicated definitions" patches. These are based on mark/topic/core branch Kuninori Morimoto (3): ASoC: remove duplicate definition of controls/num_controls ASoC: remove duplicate definition of dapm_widgets/num_dapm_widgets ASoC: remove duplicate definition of dapm_routes/num_dapm_routes include/sound/soc.h | 6 ------ sound/soc/soc-core.c | 28 ++++++++++++---------------- 2 files changed, 12 insertions(+), 22 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH] ASoC: davinci-mcasp: check memory allocation failure
by Christophe JAILLET 27 Aug '17

27 Aug '17

Check memory allocation failures and return -ENOMEM in such cases, as already done above for another memory allocation. This avoids NULL pointers dereference. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/davinci/davinci-mcasp.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d486908d4e0e..f395bbc7c354 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1851,6 +1851,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev, sizeof(u32) * mcasp->num_serializer, GFP_KERNEL); + if (!mcasp->context.xrsr_regs) { + ret = -ENOMEM; + goto err; + } #endif mcasp->serial_dir = pdata->serial_dir; mcasp->version = pdata->version; -- 2.11.0

2 1

Re: [alsa-devel] Alsa-devel Digest, Vol 126, Issue 233
by cuhl＠eit.uni-kl.de 27 Aug '17

27 Aug '17

Hello Clemens, hello Alsa Developers, thank you for your interest and answer to my question. Why is the behaviour hardware dependent? As long as the I2S' are synced at the inputs of the SoC, are the relative differences / phases preserved during processing and is this ensured? I would like to use a device with multiple I2S-inputs and outputs, as far as available and supported by the ALSA drivers. Is this driver dependent? Are there drivers which take special effort to preserve the relative position of channels? Kind regards > Date: Wed, 23 Aug 2017 15:41:36 +0200 > From: Clemens Ladisch <clemens(a)ladisch.de> > To: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Fixed latency between multiple channels > Message-ID: <b6c4c2f2-a01c-fc06-e1bb-73e14b3cf56c(a)ladisch.de> > Content-Type: text/plain; charset=us-ascii > > cuhl(a)eit.uni-kl.de wrote: >> I've read, that even with a realtime kernel the input-output latency >> varies with time - especially if USB is inbetween. > > Yes, USB cam have a small but unknown delay between playback and capture. > >> I expect to require a fixed latency between multiple ( I2S ) channels. > >> So every channel's signal shall be exactly determined to its >> relative position with respect to the other channels, e. g. it shall be >> known inside the system which events occured simultanously. >> >> Likewise, the processed or generated signals shall be transferred to the >> outputs in a way, that they occur there with intended and modifiable >> relative latency to each other. > > This is essentially a hardware problem. > > Can't you just use a single device with many channels? > > > Regards, > Clemens >

2 1

[alsa-devel] [PATCH 00/23] alsa-utils: rewrite aplay
by Takashi Sakamoto 26 Aug '17

26 Aug '17

Hi, I think all of you in this mailing list might have used the command 'aplay', at least once. This patchset is my attempt to rewrite it because current shape of code is not enough good. Especially, ALSA PCM interface have got extensions since aplay was firstly developed. One of my motivation is to refine the tool to test such extensions. Below diagram illustrates new internal design of this program: <-> container <-> (file) (device) <-> (alsa-lib) <-> xfer <-> aligner <-> container <-> (file) <-> container <-> (file) For description of each modules, please read patch comment. This patchset is early preview for developers to see the above design, thus some features of original aplay are not still implemented. Especially, some options have no actual effect even if they're parsed. Presently, I see below command lines run as I expected: $ ./aplay -C (-v) (-M)(-N) -D [hw:0,0|default] record.wav $ ./aplay -P (-v) (-M)(-N) -D [hw:0,0|default] playback.wav $ ./aplay --list-devices $ ./aplay --list-pcm $ ./aplay --version Here, I work with Intel HDA (hw:0,0) and pulse PCM plugin (default). It's my pleasure if received some comments from you. Takashi Sakamoto (23): aplay: add an abstraction of container to parse/build audio-specific data format aplay: add an implementation of container for Microsoft/IBM RIFF/Wave format aplay: add an implementation of container for Sparc AU format aplay: add an implementation of container for Creative Tech. voice format aplay: add an implementation of container for raw format aplay: aligner: add an abstraction to align buffers with different data aplay: add an implementation of aligner for single target aplay: add an implementation of aligner for multiple target aplay: add an abstruction of waiter for I/O event notification aplay: add an implementation of waiter for poll(2) aplay: add an implementation of waiter for epoll(7) aplay: options: add a parser for command-line options aplay: add an abstraction for transferring of PCM frames aplay: add an implementation for transferring by ALSA PCM APIs aplay: add implementation of I/O aplay: add implementations to scheduling aplay: add a sub-command to print list of PCMs/devices aplay: add a sub-command to transfer data frames aplay: obsolete main routine and introduce sub-command style aplay: add an implementation for volume unit meter aplay: add a parser for channel map API aplay: add a handler for key events aplay: add a feature to generate PID file aplay/Makefile.am | 48 +- aplay/aligner-multiple.c | 337 ++++ aplay/aligner-single.c | 256 +++ aplay/aligner.c | 122 ++ aplay/aligner.h | 88 ++ aplay/aplay.c | 3425 ----------------------------------------- aplay/container-au.c | 203 +++ aplay/container-riff-wave.c | 549 +++++++ aplay/container-voc.c | 736 +++++++++ aplay/container.c | 375 +++++ aplay/container.h | 127 ++ aplay/formats.h | 134 -- aplay/key-event.c | 120 ++ aplay/key-event.h | 21 + aplay/main.c | 178 +++ aplay/main.h | 32 + aplay/options.c | 674 ++++++++ aplay/options.h | 71 + aplay/subcmd-list.c | 233 +++ aplay/subcmd-transfer.c | 431 ++++++ aplay/vumeter.c | 299 ++++ aplay/vumeter.h | 50 + aplay/waiter-epoll.c | 76 + aplay/waiter-poll.c | 66 + aplay/waiter.c | 61 + aplay/waiter.h | 59 + aplay/xfer-alsa-io-mmap.c | 135 ++ aplay/xfer-alsa-io-rw.c | 370 +++++ aplay/xfer-alsa-sched-irq.c | 18 + aplay/xfer-alsa-sched-timer.c | 49 + aplay/xfer-alsa.c | 610 ++++++++ aplay/xfer-alsa.h | 67 + aplay/xfer.c | 124 ++ aplay/xfer.h | 74 + configure.ac | 2 +- 35 files changed, 6655 insertions(+), 3565 deletions(-) create mode 100644 aplay/aligner-multiple.c create mode 100644 aplay/aligner-single.c create mode 100644 aplay/aligner.c create mode 100644 aplay/aligner.h delete mode 100644 aplay/aplay.c create mode 100644 aplay/container-au.c create mode 100644 aplay/container-riff-wave.c create mode 100644 aplay/container-voc.c create mode 100644 aplay/container.c create mode 100644 aplay/container.h delete mode 100644 aplay/formats.h create mode 100644 aplay/key-event.c create mode 100644 aplay/key-event.h create mode 100644 aplay/main.c create mode 100644 aplay/main.h create mode 100644 aplay/options.c create mode 100644 aplay/options.h create mode 100644 aplay/subcmd-list.c create mode 100644 aplay/subcmd-transfer.c create mode 100644 aplay/vumeter.c create mode 100644 aplay/vumeter.h create mode 100644 aplay/waiter-epoll.c create mode 100644 aplay/waiter-poll.c create mode 100644 aplay/waiter.c create mode 100644 aplay/waiter.h create mode 100644 aplay/xfer-alsa-io-mmap.c create mode 100644 aplay/xfer-alsa-io-rw.c create mode 100644 aplay/xfer-alsa-sched-irq.c create mode 100644 aplay/xfer-alsa-sched-timer.c create mode 100644 aplay/xfer-alsa.c create mode 100644 aplay/xfer-alsa.h create mode 100644 aplay/xfer.c create mode 100644 aplay/xfer.h -- 2.11.0

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[alsa-devel] [patch 4/4] ASoC: dapm: Remove the redundant check
by anish kumar 26 Aug '17

26 Aug '17

Remove this redundant check, the check for creating the control in the case of codec to codec dai link is moved to snd_soc_dapm_new_pcm function. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5d792dc..3ecaad0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1032,10 +1032,6 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; - /* create control for links with > 1 config */ - if (w->num_params <= 1) - return 0; - /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, -- 2.10.1 (Apple Git-78)

1 0

[alsa-devel] [patch 3/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 26 Aug '17

26 Aug '17

Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 116 +++++++++++++++++++++++++++++---------------------- 1 file changed, 67 insertions(+), 49 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1687c8a..5d792dc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,12 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3792,16 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; + const char **w_param_text; + unsigned long private_value; + int count; w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); + sizeof(char *), GFP_KERNEL); if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + return NULL; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,24 +3818,14 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), @@ -3852,22 +3833,67 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } kcontrol_dai_link[0].private_value = private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; goto outfree_private_value; } + return kcontrol_news; + +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + if (w_param_text[count] != NULL) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + return NULL; +} +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto outfree_link_name; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,16 +3925,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - if (w_param_text[count]) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.10.1 (Apple Git-78)

1 0

[alsa-devel] [PATCH] ASoC: rt5677: Reintroduce I2C device IDs
by Tom Rini 25 Aug '17

25 Aug '17

Not all devices with ACPI and this combination of sound devices will have the required information provided via ACPI. Reintroduce the I2C device ID to restore sound functionality on on the Chromebook 'Samus' model. Fixes: a36afb0ab648 ("ASoC: rt5677: Introduce proper table for ACPI enumeration") Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org CC: linux-kernel(a)vger.kernel.org Cc: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Linus Torvalds <torvalds(a)linux-foundation.org> Signed-off-by: Tom Rini <trini(a)konsulko.com> --- This is a regression from v4.12 on my laptop (a Chromebook 'Samus' that's not running ChromeOS). My fault for getting out of the habit of trying -rc1 when it comes out and not spotting this sooner. I'm not 100% sure if this fix is correct for all cases as I'm only able to test my hardware here, and this does fix my laptop. --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 36e530a36c82..6f629278d982 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5021,6 +5021,7 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) static const struct i2c_device_id rt5677_i2c_id[] = { { "rt5677", RT5677 }, { "rt5676", RT5676 }, + { "RT5677CE:00", RT5677 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); -- 1.9.1

5 35

[alsa-devel] [PATCH v1] ASoC: rt5677: Convert I2C driver to ->probe_new()
by Andy Shevchenko 25 Aug '17

25 Aug '17

There is no platform code that uses i2c module table. Remove it altogether and adjust ->probe() to be ->probe_new(). Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/codecs/rt5677.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c735590c5a25..6448b7a00203 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5009,11 +5009,6 @@ static const struct regmap_config rt5677_regmap = { .num_ranges = ARRAY_SIZE(rt5677_ranges), }; -static const struct i2c_device_id rt5677_i2c_id[] = { - { } -}; -MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); - static const struct of_device_id rt5677_of_match[] = { { .compatible = "realtek,rt5677", RT5677 }, { } @@ -5131,8 +5126,7 @@ static void rt5677_free_irq(struct i2c_client *i2c) regmap_del_irq_chip(i2c->irq, rt5677->irq_data); } -static int rt5677_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int rt5677_i2c_probe(struct i2c_client *i2c) { struct rt5677_priv *rt5677; int ret; @@ -5279,9 +5273,8 @@ static struct i2c_driver rt5677_i2c_driver = { .of_match_table = rt5677_of_match, .acpi_match_table = ACPI_PTR(rt5677_acpi_match), }, - .probe = rt5677_i2c_probe, + .probe_new = rt5677_i2c_probe, .remove = rt5677_i2c_remove, - .id_table = rt5677_i2c_id, }; module_i2c_driver(rt5677_i2c_driver); -- 2.14.1

3 11

[alsa-devel] Developing a ti-pcm1863 (adc) codec driver
by Oleksandr Müller 25 Aug '17

25 Aug '17

Hello Mr. Brown, I have a soundcard which doesnt have a linux driver. Quick information it has a pcm1863 adc and a pcm5142 dac on it. I began with the sample dt-overlay and edited it to get the playback working. No problem at all using the pcm512x.c codec driver. After that I began with the development of the codec driver, i2c driver and a dt-overlay with 2 sound links as I need the record and playback functionality. Right now the i2c driver seems to work since I get the hardware into the used state "UU" when executing the i2cdetect command and dmesg says "mapping okay". My codec driver has the codec driver and dai definitions and some more functions (here I am stuck). I know that I need the hw_params and some set functions to have the minimal functionality. 1. Question would be towards the dapm_widgets and dapm_routes definitions. When defining the dapm_adc widget do I put there the Capture stream or do I seperate it with "NULL" and then adding the dapm_routes definitions? Which one is correct since I have seen both possibilities (which ad-/disadventages do I have?) 2. Question in my layout plan I need to set the ADC to VIN4 l/r because only this one is connected to the microphone port. I have put it into the sound controls but how would I make this switch from the command line (arecord) or in the players (like vlc) 3. It seems that I have a bug in the code since I cant execute arecord right now it says file doesnt exist. Furthermore I would really be interested in gaining more knowledge about the accesses and routines from e.g. the vlc player using alsa (?) and then the accesses to the hardware? Also the whole registering process in which order are which files executed to get into the state where the os knows my soundcard. Since in a stracktrace I dont see this high level functions executed. I have found some diagramms on a chinese blog which look not bad but it would be a lot better to find it out myself with the os if possible. Here is an example of this kind of diagramm: http://img.blog.csdn.net/20160127104235066?watermark/2/text/aHR0cDovL2Jsb2c… This is knowledge is right now optional but would be very helpful for the understanding. I will attach the codec driver, codec driver-header, i2c driver, dt-overlay and a strack trace (ltrace just hangs up) so it would be possible to find out what is causing the problem right now. Please note that this is my first driver development and the code is work in progress :) Thanks a lot for Your time and Your help! If an information is missing please ask for it and if possible I will try to provide it :) Sanj3k

2 3

[alsa-devel] Question about sound device release timing
by Kuninori Morimoto 25 Aug '17

25 Aug '17

Hi Takashi-san, Mark I noticed snd_soc_dapm_stream_event() is called strange timing if I unbind sound device during playing > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/xxx/unbind ... > Unable to handle kernel paging request at virtual address dead000000000220 > Mem abort info: (snip) > PC is at soc_dapm_dai_stream_event.isra.14+0x20/0xd0 > LR is at snd_soc_dapm_stream_event+0x74/0xa8 (snip) > [<ffff000008715610>] soc_dapm_dai_stream_event.isra.14+0x20/0xd0 > [<ffff00000871989c>] snd_soc_dapm_stream_event+0x74/0xa8 > [<ffff00000871b23c>] close_delayed_work+0x3c/0x50 > [<ffff0000080bbd6c>] process_one_work+0x1ac/0x318 > [<ffff0000080bbf20>] worker_thread+0x48/0x420 > [<ffff0000080c201c>] kthread+0xfc/0x128 > [<ffff0000080842f0>] ret_from_fork+0x10/0x18 > Code: b40005c9 a9bf7bfd 91048120 910003fd (f9409121) > ---[ end trace 6739a0ea1013e0b2 ]--- This snd_soc_dapm_stream_event() was called from close_delayed_work() and it was set on soc_pcm_close() static int soc_pcm_close(struct snd_pcm_substream *substream) { ... if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (snd_soc_runtime_ignore_pmdown_time(rtd)) { ... } else { /* start delayed pop wq here for playback streams */ rtd->pop_wait = 1; => queue_delayed_work(system_power_efficient_wq, &rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } ... } But, this soc_pcm_close() is called *after* flush_delayed_work() of soc_cleanup_card_resources(). It seems this close function seems called from snd_card_free() and this rtd will be freed on soc_remove_pcm_runtimes(). static int soc_cleanup_card_resources(struct snd_soc_card *card) { ... /* make sure any delayed work runs */ list_for_each_entry(rtd, &card->rtd_list, list) { => flush_delayed_work(&rtd->delayed_work); } /* free the ALSA card at first; this syncs with pending operations */ => snd_card_free(card->snd_card); ... => soc_remove_pcm_runtimes(card); ... } Because of this, delayed called close_delayed_work() will call soc_dapm_dai_stream_event() and it access to already freed rtd, and kernel will die. I don't know why, but it seems this close is wokring on other thread (?), but I'm not sure. Q1. does this close function run on other thread ? Q2. I can avoid this delayed close by setting rtd->pmdown_time = 0; before snd_card_free(). but is this correct approach ? I can avoid this delayed calling close_delayed_work() by local patch, but then, re-bind doesn't work correctly. /* * custom kernel for avoiding close_delayed_work() */ > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/rcar_sound/unbind /* unbind codec and card drivers here */ > /* * try re-bind */ /* re-bind codec and card drivers here */ > echo xxxx > /sys/bus/platform/drivers/rcar_sound/bind rcar_sound ec500000.sound: probed asoc-simple-card sound: ak4613-hifi <-> ec500000.sound mapping ok > > aplay xxx.wav Playing WAVE '/home/Calm_16bit_48k.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable aplay: set_params:1204: Broken configuration for this PCM: no configurations available Do you know why or have hint to solve these issue ? Best regards --- Kuninori Morimoto

2 4

[alsa-devel] [lkp-robot] [ALSA] c8da9be4a7: WARNING:at_fs/proc/generic.c:#proc_register
by kernel test robot 25 Aug '17

25 Aug '17

FYI, we noticed the following commit: commit: c8da9be4a75f1f63024a16b976c5be47405c13dc ("ALSA: pcm: Adjust nine function calls together with a variable assignment") https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git for-next in testcase: trinity with following parameters: runtime: 300s test-description: Trinity is a linux system call fuzz tester. test-url: http://codemonkey.org.uk/projects/trinity/ on test machine: qemu-system-i386 -enable-kvm -m 256M caused below changes (please refer to attached dmesg/kmsg for entire log/backtrace): +------------------------------------------------+------------+------------+ | | 97d15a141f | c8da9be4a7 | +------------------------------------------------+------------+------------+ | boot_successes | 4 | 0 | | boot_failures | 2 | 6 | | IP-Config:Auto-configuration_of_network_failed | 2 | 2 | | WARNING:at_fs/proc/generic.c:#proc_register | 0 | 6 | | EIP:proc_register | 0 | 6 | +------------------------------------------------+------------+------------+ [ 8.559972] WARNING: CPU: 0 PID: 1 at fs/proc/generic.c:328 proc_register+0xe4/0x130 [ 8.559972] WARNING: CPU: 0 PID: 1 at fs/proc/generic.c:328 proc_register+0xe4/0x130 [ 8.561652] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.13.0-rc3-00171-gc8da9be #1 [ 8.561652] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.13.0-rc3-00171-gc8da9be #1 [ 8.562958] task: cf644000 task.stack: cf646000 [ 8.562958] task: cf644000 task.stack: cf646000 [ 8.563748] EIP: proc_register+0xe4/0x130 [ 8.563748] EIP: proc_register+0xe4/0x130 [ 8.564423] EFLAGS: 00210282 CPU: 0 [ 8.564423] EFLAGS: 00210282 CPU: 0 [ 8.565001] EAX: 0000002e EBX: ced3a038 ECX: 00000000 EDX: 00000000 [ 8.565001] EAX: 0000002e EBX: ced3a038 ECX: 00000000 EDX: 00000000 [ 8.566049] ESI: ced3dab0 EDI: cf724d24 EBP: cf647d08 ESP: cf647ce8 [ 8.566049] ESI: ced3dab0 EDI: cf724d24 EBP: cf647d08 ESP: cf647ce8 [ 8.567102] DS: 007b ES: 007b FS: 00d8 GS: 0000 SS: 0068 [ 8.567102] DS: 007b ES: 007b FS: 00d8 GS: 0000 SS: 0068 [ 8.568042] CR0: 80050033 CR2: 00000000 CR3: 021d5000 CR4: 000006b0 [ 8.568042] CR0: 80050033 CR2: 00000000 CR3: 021d5000 CR4: 000006b0 [ 8.569131] Call Trace: [ 8.569131] Call Trace: [ 8.569561] proc_create_data+0x81/0xf0 [ 8.569561] proc_create_data+0x81/0xf0 [ 8.570264] snd_info_register+0x6d/0xe0 [ 8.570264] snd_info_register+0x6d/0xe0 [ 8.570941] snd_pcm_new_stream+0x21b/0x4a0 [ 8.570941] snd_pcm_new_stream+0x21b/0x4a0 [ 8.571694] ? strlcpy+0x33/0x50 [ 8.571694] ? strlcpy+0x33/0x50 [ 8.572252] _snd_pcm_new+0xc6/0x150 [ 8.572252] _snd_pcm_new+0xc6/0x150 [ 8.572874] snd_pcm_new+0x18/0x20 [ 8.572874] snd_pcm_new+0x18/0x20 [ 8.573463] snd_dummy_probe+0xec/0x400 [ 8.573463] snd_dummy_probe+0xec/0x400 [ 8.574103] platform_drv_probe+0x1e/0x60 [ 8.574103] platform_drv_probe+0x1e/0x60 [ 8.574785] really_probe+0x1c2/0x260 [ 8.574785] really_probe+0x1c2/0x260 [ 8.575424] __device_attach_driver+0x72/0xb0 [ 8.575424] __device_attach_driver+0x72/0xb0 [ 8.576183] ? __driver_attach+0xb0/0xb0 [ 8.576183] ? __driver_attach+0xb0/0xb0 [ 8.576844] bus_for_each_drv+0x4f/0x80 [ 8.576844] bus_for_each_drv+0x4f/0x80 [ 8.577486] __device_attach+0x9e/0xd0 [ 8.577486] __device_attach+0x9e/0xd0 [ 8.578113] ? __driver_attach+0xb0/0xb0 [ 8.578113] ? __driver_attach+0xb0/0xb0 [ 8.578772] device_initial_probe+0x12/0x20 [ 8.578772] device_initial_probe+0x12/0x20 [ 8.579486] bus_probe_device+0x77/0x80 [ 8.579486] bus_probe_device+0x77/0x80 [ 8.580201] device_add+0x2fc/0x530 [ 8.580201] device_add+0x2fc/0x530 [ 8.580797] ? dev_set_name+0x19/0x20 [ 8.580797] ? dev_set_name+0x19/0x20 [ 8.581416] platform_device_add+0x124/0x240 [ 8.581416] platform_device_add+0x124/0x240 [ 8.582156] platform_device_register_full+0xe2/0x120 [ 8.582156] platform_device_register_full+0xe2/0x120 [ 8.583002] alsa_card_dummy_init+0x97/0xf4 [ 8.583002] alsa_card_dummy_init+0x97/0xf4 [ 8.583784] ? alsa_seq_midi_event_init+0xc/0xc [ 8.583784] ? alsa_seq_midi_event_init+0xc/0xc [ 8.584604] do_one_initcall+0x80/0x151 [ 8.584604] do_one_initcall+0x80/0x151 [ 8.585270] ? parse_args+0x130/0x2a0 [ 8.585270] ? parse_args+0x130/0x2a0 [ 8.585890] ? kernel_init_freeable+0xe5/0x178 [ 8.585890] ? kernel_init_freeable+0xe5/0x178 [ 8.586630] kernel_init_freeable+0x105/0x178 [ 8.586630] kernel_init_freeable+0x105/0x178 [ 8.587655] ? rest_init+0xf0/0xf0 [ 8.587655] ? rest_init+0xf0/0xf0 [ 8.588230] kernel_init+0xd/0xf0 [ 8.588230] kernel_init+0xd/0xf0 [ 8.588778] ret_from_fork+0x19/0x24 [ 8.588778] ret_from_fork+0x19/0x24 [ 8.589384] Code: af 8d b6 00 00 00 00 6a 01 31 c9 ba 01 00 00 00 b8 90 4e f7 c1 83 c7 7c e8 ba 5a ef ff ff 75 f0 57 68 ac f0 d4 c1 e8 6c 2f e9 ff <0f> ff 6a 01 31 c9 ba 01 00 00 00 b8 78 4e f7 c1 e8 97 5a ef ff [ 8.589384] Code: af 8d b6 00 00 00 00 6a 01 31 c9 ba 01 00 00 00 b8 90 4e f7 c1 83 c7 7c e8 ba 5a ef ff ff 75 f0 57 68 ac f0 d4 c1 e8 6c 2f e9 ff <0f> ff 6a 01 31 c9 ba 01 00 00 00 b8 78 4e f7 c1 e8 97 5a ef ff [ 8.592689] ---[ end trace 8498fc54cd971cf8 ]--- To reproduce: git clone https://github.com/01org/lkp-tests.git cd lkp-tests bin/lkp qemu -k <bzImage> job-script # job-script is attached in this email Thanks, Xiaolong

1 0

[alsa-devel] [PATCH v3] ASoC: Intel: kbl: Add map for Maxim IV Feedback
by Naveen M 25 Aug '17

25 Aug '17

MAX98927 provides IV feedback on the capture widget. Here we are connecting the capture widget to SSP0_RX and SSP0_RX to the algorithm running on host. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> v2: rebased on newer kernel (d46b182) v3: moved version changelog info after s-o-b lines --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +++++ sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +++++ 2 files changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index c144214..8d2a1c0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -107,6 +107,11 @@ enum { { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "dmic01_hifi", NULL, "DMIC01 Rx" }, { "DMIC01 Rx", NULL, "DMIC AIF" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index c2ef538..8d19da9 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -119,6 +119,11 @@ enum { { "codec1_in", NULL, "ssp0 Rx" }, { "ssp0 Rx", NULL, "AIF1 Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix uninitialized return
by Subhransu S. Prusty 25 Aug '17

25 Aug '17

On failure to get dsp_ops, dsp_init returns error without assigning ret. ret is assigned in code path which will never be executed. Fix it. Fixes: f77d443c4c29 ("ASoC: Intel: Skylake: Fix to free resources for dsp_init failure" Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index f0f11f597b21..697d07bc1b5a 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -278,8 +278,8 @@ int skl_init_dsp(struct skl *skl) ops = skl_get_dsp_ops(skl->pci->device); if (!ops) { - goto unmap_mmio; ret = -EIO; + goto unmap_mmio; } loader_ops = ops->loader_ops(); -- 1.9.1

3 2

[alsa-devel] proc_dir_entry 'asound/pcm' already registered
by Fabio Estevam 24 Aug '17

24 Aug '17

Hi, I am running linux-next 20170824 on a imx6q-sabresd and getting the warnings below. Any ideas as to why 'asound/pcm' is getting registered twice? Thanks [ 3.842972] fsl-ssi-dai 202c000.ssi: No cache defaults, reading back from HW [ 3.859270] input: WM8962 Beep Generator as /devices/soc0/soc/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input1 [ 3.921494] ------------[ cut here ]------------ [ 3.926181] WARNING: CPU: 1 PID: 1 at fs/proc/generic.c:330 proc_register+0xc8/0x13c [ 3.934020] proc_dir_entry 'asound/pcm' already registered [ 3.939609] Modules linked in: [ 3.942712] CPU: 1 PID: 1 Comm: swapper/0 Not tainted 4.13.0-rc6-next-20170824 #49 [ 3.950301] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 3.956849] Backtrace: [ 3.959340] [<c010ca24>] (dump_backtrace) from [<c010ccf4>] (show_stack+0x18/0x1c) [ 3.966933] r7:00000000 r6:60000013 r5:00000000 r4:c0e79b80 [ 3.972620] [<c010ccdc>] (show_stack) from [<c09ca560>] (dump_stack+0xb4/0xe8) [ 3.979869] [<c09ca4ac>] (dump_stack) from [<c0125654>] (__warn+0xd8/0x104) [ 3.986856] r9:c029892c r8:0000014a r7:00000009 r6:c0c06374 r5:00000000 r4:ec069b48 [ 3.994623] [<c012557c>] (__warn) from [<c01256bc>] (warn_slowpath_fmt+0x3c/0x44) [ 4.002129] r9:ec338f2c r8:00000000 r7:ec338f00 r6:00000001 r5:ec887b40 r4:c0c06404 [ 4.009898] [<c0125684>] (warn_slowpath_fmt) from [<c029892c>] (proc_register+0xc8/0x13c) [ 4.018096] r3:ec338f7d r2:c0c06404 [ 4.021691] r4:ec12c874 [ 4.024254] [<c0298864>] (proc_register) from [<c0298c64>] (proc_create_data+0x80/0xc4) [ 4.032281] r10:ec8a9e00 r9:00000000 r8:00000000 r7:ec138000 r6:ec887a80 r5:c0a900f0 [ 4.040130] r4:ec887b40 [ 4.042695] [<c0298be4>] (proc_create_data) from [<c075456c>] (snd_info_register+0x6c/0xd4) [ 4.051066] r5:ec338f00 r4:ec887a80 [ 4.054670] [<c0754500>] (snd_info_register) from [<c07590ac>] (snd_pcm_new_stream+0x15c/0x454) [ 4.063388] r5:ec145800 r4:00000000 [ 4.066987] [<c0758f50>] (snd_pcm_new_stream) from [<c0759a54>] (_snd_pcm_new+0xb4/0x13c) [ 4.075188] r10:00000000 r9:00000000 r8:00000001 r7:ec138000 r6:ec069ca4 r5:ec069ca8 [ 4.083036] r4:ec145800 [ 4.085593] [<c07599a0>] (_snd_pcm_new) from [<c0759afc>] (snd_pcm_new+0x20/0x28) [ 4.093100] r10:ec069ca8 r9:00000001 r8:ec132c80 r7:00000000 r6:ec8a9a0c r5:00000001 [ 4.100948] r4:00000001 r3:00000001 [ 4.104550] [<c0759adc>] (snd_pcm_new) from [<c0777c0c>] (soc_new_pcm+0x12c/0x45c) [ 4.112139] r4:ec144800 [ 4.114700] [<c0777ae0>] (soc_new_pcm) from [<c076b658>] (snd_soc_register_card+0xc40/0xe38) [ 4.123162] r10:ec144800 r9:ec8cd810 r8:00000002 r7:ec8cd898 r6:ec8cd9d4 r5:ec8cd978 [ 4.131011] r4:c0c8dcb8 [ 4.133570] [<c076aa18>] (snd_soc_register_card) from [<c0778474>] (devm_snd_soc_register_card+0x38/0x78) [ 4.143161] r10:ec224400 r9:ec8cd810 r8:c0aaf308 r7:ec284c10 r6:ec8cd898 r5:ec89b150 [ 4.151009] r4:ec8cd898 [ 4.153575] [<c077843c>] (devm_snd_soc_register_card) from [<c07913a4>] (imx_wm8962_probe+0x244/0x37c) [ 4.162905] r7:ef7d9650 r6:ec284c10 r5:ec284c00 r4:ec8cd898 [ 4.168595] [<c0791160>] (imx_wm8962_probe) from [<c0546780>] (platform_drv_probe+0x54/0xb8) [ 4.177056] r10:00000000 r9:00000000 r8:c0e70228 r7:fffffdfb r6:c0e70228 r5:ec284c10 [ 4.184904] r4:ec284c10 [ 4.187470] [<c054672c>] (platform_drv_probe) from [<c0544bf4>] (driver_probe_device+0x274/0x2fc) [ 4.196365] r7:c15e6c10 r6:00000000 r5:c15e6c0c r4:ec284c10 [ 4.202051] [<c0544980>] (driver_probe_device) from [<c0544d44>] (__driver_attach+0xc8/0xcc) [ 4.210513] r10:c0d00618 r9:c0d6089c r8:00000000 r7:00000000 r6:ec284c44 r5:c0e70228 [ 4.218361] r4:ec284c10 r3:00000000 [ 4.221964] [<c0544c7c>] (__driver_attach) from [<c0542ebc>] (bus_for_each_dev+0x74/0xa8) [ 4.230164] r7:00000000 r6:c0544c7c r5:c0e70228 r4:00000000 [ 4.235852] [<c0542e48>] (bus_for_each_dev) from [<c0544530>] (driver_attach+0x20/0x28) [ 4.243876] r6:c0e3b288 r5:ec132300 r4:c0e70228 [ 4.248520] [<c0544510>] (driver_attach) from [<c054402c>] (bus_add_driver+0x190/0x218) [ 4.256549] [<c0543e9c>] (bus_add_driver) from [<c05456e4>] (driver_register+0x80/0x100) [ 4.264662] r7:c0cf5ec4 r6:c0d60894 r5:c0d4e44c r4:c0e70228 [ 4.270347] [<c0545664>] (driver_register) from [<c05466d4>] (__platform_driver_register+0x38/0x4c) [ 4.279412] r5:c0d4e44c r4:ffffe000 [ 4.283020] [<c054669c>] (__platform_driver_register) from [<c0d4e464>] (imx_wm8962_driver_init+0x18/0x20) [ 4.292702] [<c0d4e44c>] (imx_wm8962_driver_init) from [<c0101c54>] (do_one_initcall+0x44/0x17c) [ 4.301517] [<c0101c10>] (do_one_initcall) from [<c0d00e7c>] (kernel_init_freeable+0x118/0x1d0) [ 4.310238] r8:000000f1 r7:c0cf5ec4 r6:c0d60894 r5:c0e7d000 r4:c0d710e0 [ 4.316967] [<c0d00d64>] (kernel_init_freeable) from [<c09de898>] (kernel_init+0x10/0x118) [ 4.325254] r10:00000000 r9:00000000 r8:00000000 r7:00000000 r6:00000000 r5:c09de888 [ 4.333102] r4:00000000 [ 4.335665] [<c09de888>] (kernel_init) from [<c01080f8>] (ret_from_fork+0x14/0x3c) [ 4.343255] r5:c09de888 r4:00000000 [ 4.346962] ---[ end trace 3ea29ef261f4780c ]--- [ 4.351955] ------------[ cut here ]------------ [ 4.356611] WARNING: CPU: 1 PID: 1 at fs/proc/generic.c:330 proc_register+0xc8/0x13c [ 4.364434] proc_dir_entry 'asound/pcm' already registered [ 4.369982] Modules linked in: [ 4.373083] CPU: 1 PID: 1 Comm: swapper/0 Tainted: G W 4.13.0-rc6-next-20170824 #49 [ 4.381887] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 4.388433] Backtrace: [ 4.390918] [<c010ca24>] (dump_backtrace) from [<c010ccf4>] (show_stack+0x18/0x1c) [ 4.398512] r7:00000000 r6:60000013 r5:00000000 r4:c0e79b80 [ 4.404196] [<c010ccdc>] (show_stack) from [<c09ca560>] (dump_stack+0xb4/0xe8) [ 4.411443] [<c09ca4ac>] (dump_stack) from [<c0125654>] (__warn+0xd8/0x104) [ 4.418430] r9:c029892c r8:0000014a r7:00000009 r6:c0c06374 r5:00000000 r4:ec069b48 [ 4.426196] [<c012557c>] (__warn) from [<c01256bc>] (warn_slowpath_fmt+0x3c/0x44) [ 4.433703] r9:ec338f2c r8:00000000 r7:ec338f00 r6:00000001 r5:ec88c480 r4:c0c06404 [ 4.441470] [<c0125684>] (warn_slowpath_fmt) from [<c029892c>] (proc_register+0xc8/0x13c) [ 4.449668] r3:ec338f7d r2:c0c06404 [ 4.453263] r4:ec12c874 [ 4.455822] [<c0298864>] (proc_register) from [<c0298c64>] (proc_create_data+0x80/0xc4) [ 4.463850] r10:ec8c8000 r9:00000001 r8:00000000 r7:ec138000 r6:ec88c3c0 r5:c0a900f0 [ 4.471698] r4:ec88c480 [ 4.474261] [<c0298be4>] (proc_create_data) from [<c075456c>] (snd_info_register+0x6c/0xd4) [ 4.482632] r5:ec338f00 r4:ec88c3c0 [ 4.486234] [<c0754500>] (snd_info_register) from [<c07590ac>] (snd_pcm_new_stream+0x15c/0x454) [ 4.494953] r5:ec145800 r4:00000000 [ 4.498552] [<c0758f50>] (snd_pcm_new_stream) from [<c0759a6c>] (_snd_pcm_new+0xcc/0x13c) [ 4.506753] r10:00000000 r9:00000000 r8:00000001 r7:ec138000 r6:ec069ca4 r5:00000000 [ 4.514601] r4:ec145800 [ 4.517159] [<c07599a0>] (_snd_pcm_new) from [<c0759afc>] (snd_pcm_new+0x20/0x28) [ 4.524665] r10:ec069ca8 r9:00000001 r8:ec132c80 r7:00000000 r6:ec8a9a0c r5:00000001 [ 4.532513] r4:00000001 r3:00000001 [ 4.536112] [<c0759adc>] (snd_pcm_new) from [<c0777c0c>] (soc_new_pcm+0x12c/0x45c) [ 4.543700] r4:ec144800 [ 4.546260] [<c0777ae0>] (soc_new_pcm) from [<c076b658>] (snd_soc_register_card+0xc40/0xe38) [ 4.554721] r10:ec144800 r9:ec8cd810 r8:00000002 r7:ec8cd898 r6:ec8cd9d4 r5:ec8cd978 [ 4.562569] r4:c0c8dcb8 [ 4.565129] [<c076aa18>] (snd_soc_register_card) from [<c0778474>] (devm_snd_soc_register_card+0x38/0x78) [ 4.574719] r10:ec224400 r9:ec8cd810 r8:c0aaf308 r7:ec284c10 r6:ec8cd898 r5:ec89b150 [ 4.582567] r4:ec8cd898 [ 4.585132] [<c077843c>] (devm_snd_soc_register_card) from [<c07913a4>] (imx_wm8962_probe+0x244/0x37c) [ 4.594460] r7:ef7d9650 r6:ec284c10 r5:ec284c00 r4:ec8cd898 [ 4.600146] [<c0791160>] (imx_wm8962_probe) from [<c0546780>] (platform_drv_probe+0x54/0xb8) [ 4.608607] r10:00000000 r9:00000000 r8:c0e70228 r7:fffffdfb r6:c0e70228 r5:ec284c10 [ 4.616455] r4:ec284c10 [ 4.619019] [<c054672c>] (platform_drv_probe) from [<c0544bf4>] (driver_probe_device+0x274/0x2fc) [ 4.627913] r7:c15e6c10 r6:00000000 r5:c15e6c0c r4:ec284c10 [ 4.633600] [<c0544980>] (driver_probe_device) from [<c0544d44>] (__driver_attach+0xc8/0xcc) [ 4.642062] r10:c0d00618 r9:c0d6089c r8:00000000 r7:00000000 r6:ec284c44 r5:c0e70228 [ 4.649910] r4:ec284c10 r3:00000000 [ 4.653512] [<c0544c7c>] (__driver_attach) from [<c0542ebc>] (bus_for_each_dev+0x74/0xa8) [ 4.661711] r7:00000000 r6:c0544c7c r5:c0e70228 r4:00000000 [ 4.667397] [<c0542e48>] (bus_for_each_dev) from [<c0544530>] (driver_attach+0x20/0x28) [ 4.675421] r6:c0e3b288 r5:ec132300 r4:c0e70228 [ 4.680064] [<c0544510>] (driver_attach) from [<c054402c>] (bus_add_driver+0x190/0x218) [ 4.688093] [<c0543e9c>] (bus_add_driver) from [<c05456e4>] (driver_register+0x80/0x100) [ 4.696206] r7:c0cf5ec4 r6:c0d60894 r5:c0d4e44c r4:c0e70228 [ 4.701889] [<c0545664>] (driver_register) from [<c05466d4>] (__platform_driver_register+0x38/0x4c) [ 4.710954] r5:c0d4e44c r4:ffffe000 [ 4.714558] [<c054669c>] (__platform_driver_register) from [<c0d4e464>] (imx_wm8962_driver_init+0x18/0x20) [ 4.724239] [<c0d4e44c>] (imx_wm8962_driver_init) from [<c0101c54>] (do_one_initcall+0x44/0x17c) [ 4.733053] [<c0101c10>] (do_one_initcall) from [<c0d00e7c>] (kernel_init_freeable+0x118/0x1d0) [ 4.741774] r8:000000f1 r7:c0cf5ec4 r6:c0d60894 r5:c0e7d000 r4:c0d710e0 [ 4.748502] [<c0d00d64>] (kernel_init_freeable) from [<c09de898>] (kernel_init+0x10/0x118) [ 4.756789] r10:00000000 r9:00000000 r8:00000000 r7:00000000 r6:00000000 r5:c09de888 [ 4.764637] r4:00000000 [ 4.767197] [<c09de888>] (kernel_init) from [<c01080f8>] (ret_from_fork+0x14/0x3c) [ 4.774787] r5:c09de888 r4:00000000 [ 4.778454] ---[ end trace 3ea29ef261f4780d ]---

2 2