September 2017 - Alsa-devel - mailman.alsa-project.org

[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

2 1

[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

2 1

[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

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[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

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[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

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[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

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[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

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[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

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[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

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