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

[alsa-devel] [PATCH] ASoC: da7213: Update driver to use device_property* FW functions
by Adam Thomson 19 May '17

19 May '17

The driver now supports ACPI based initialisation as well as DT and old pdata methods. However the FW data handling still uses DT specific calls to read firmware data (of_property*) so for ACPI based initialisation the FW data will only be set to default values. This patch updates the FW handling to use device_property* calls instead so that both ACPI and DT are handled as expected. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7213.c | 37 +++++++++++++++++++------------------ 1 file changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 024d83f..c3e1189 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -13,6 +13,8 @@ */ #include <linux/acpi.h> +#include <linux/of_device.h> +#include <linux/property.h> #include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> @@ -1606,12 +1608,12 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, } static struct da7213_platform_data - *da7213_of_to_pdata(struct snd_soc_codec *codec) + *da7213_fw_to_pdata(struct snd_soc_codec *codec) { - struct device_node *np = codec->dev->of_node; + struct device *dev = codec->dev; struct da7213_platform_data *pdata; - const char *of_str; - u32 of_val32; + const char *fw_str; + u32 fw_val32; pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); if (!pdata) { @@ -1619,29 +1621,29 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, return NULL; } - if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0) - pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32); + if (device_property_read_u32(dev, "dlg,micbias1-lvl", &fw_val32) >= 0) + pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, fw_val32); else pdata->micbias1_lvl = DA7213_MICBIAS_2_2V; - if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0) - pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32); + if (device_property_read_u32(dev, "dlg,micbias2-lvl", &fw_val32) >= 0) + pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, fw_val32); else pdata->micbias2_lvl = DA7213_MICBIAS_2_2V; - if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str)) - pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str); + if (!device_property_read_string(dev, "dlg,dmic-data-sel", &fw_str)) + pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, fw_str); else pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL; - if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str)) + if (!device_property_read_string(dev, "dlg,dmic-samplephase", &fw_str)) pdata->dmic_samplephase = - da7213_of_dmic_samplephase(codec, of_str); + da7213_of_dmic_samplephase(codec, fw_str); else pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE; - if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0) - pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32); + if (device_property_read_u32(dev, "dlg,dmic-clkrate", &fw_val32) >= 0) + pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, fw_val32); else pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ; @@ -1713,10 +1715,9 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE); /* Handle DT/Platform data */ - if (codec->dev->of_node) - da7213->pdata = da7213_of_to_pdata(codec); - else - da7213->pdata = dev_get_platdata(codec->dev); + da7213->pdata = dev_get_platdata(codec->dev); + if (!da7213->pdata) + da7213->pdata = da7213_fw_to_pdata(codec); /* Set platform data values */ if (da7213->pdata) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: fix spelling mistake: "missmatch" -> "mismatch"
by Colin King 19 May '17

19 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/generic/audio-graph-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 07e010d38596..e0c26380921e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -107,7 +107,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, int ret; if (rcpu_ep != cpu_ep) { - dev_err(dev, "remote-endpoint missmatch (%s/%s/%s)\n", + dev_err(dev, "remote-endpoint mismatch (%s/%s/%s)\n", cpu_ep->name, codec_ep->name, rcpu_ep->name); ret = -EINVAL; goto dai_link_of_err; -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: simple-card: fix mic jack initialization
by Stefan Agner 19 May '17

19 May '17

Initialize asoc_simple_card_init_mic with the correct struct asoc_simple_jack. Signed-off-by: Stefan Agner <stefan(a)agner.ch> Fixes: 9eac361877b3 ("ASoC: simple-card: add new asoc_simple_jack and use it") --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2c9dedab5184..bc136d2bd7cd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -202,7 +202,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = asoc_simple_card_init_mic(rtd->card, &priv->hp_jack, PREFIX); + ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX); if (ret < 0) return ret; -- 2.13.0

3 2

[alsa-devel] [PATCH] ASoC: max98927: Adding support for TDM mode and Envelop tracking control
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes * Added code to support TDM mode. * Added "EnvTrack Switch" and "EnvTrack Headroom" controls. * Added six more registers into volatile register list. * Added missing \n to end of dev_err messages. * Removed an obsolete variable from 'max98927_priv'. sound/soc/codecs/max98927.c | 173 ++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98927.h | 17 +++-- 2 files changed, 152 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b5ee294..9b864e2 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); @@ -159,7 +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; } @@ -175,7 +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; } @@ -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, @@ -311,7 +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; } @@ -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, @@ -495,6 +586,12 @@ static bool max98927_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + case MAX98927_R004C_MEAS_ADC_CH0_READ: + case MAX98927_R004D_MEAS_ADC_CH1_READ: + case MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS: + case MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R01FF_REV_ID: return true; default: return false; @@ -523,6 +620,18 @@ static SOC_ENUM_SINGLE_DECL(max98927_current_limit, MAX98927_R0042_BOOST_CTRL1, 1, max98927_current_limit_text); +static const char * const max98927_env_track_headroom_text[] = { + "0.000V", "0.125V", "0.250V", "0.375V", "0.500V", "0.625V", + "0.750V", "0.875V", "1.000V", "1.125V", "1.250V", "1.375V", + "1.500V", "1.625V", "1.750V", "1.875V", "2.000V", "2.125V", + "2.250V", "2.375V", "2.500V", "2.625V", "2.750V", "2.875V", + "3.000V", "3.125V", "3.250V", "3.375V", "3.500V" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_env_track_headroom, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0, + max98927_env_track_headroom_text); + static const struct snd_kcontrol_new max98927_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, 0, 6, 0, @@ -540,6 +649,9 @@ static const struct snd_kcontrol_new max98927_snd_controls[] = { MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), SOC_ENUM("Current Limit", max98927_current_limit), + SOC_SINGLE("EnvTrack Switch", MAX98927_R0086_ENV_TRACK_CTRL, + MAX98927_ENV_TRACKER_EN_SHIFT, 1, 0), + SOC_ENUM("EnvTrack Headroom", max98927_env_track_headroom), }; static const struct snd_soc_dapm_route max98927_audio_map[] = { @@ -635,13 +747,10 @@ static int max98927_probe(struct snd_soc_codec *codec) /* Envelope Tracking configuration */ regmap_write(max98927->regmap, MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, - 0x08); + 0x0A); regmap_write(max98927->regmap, MAX98927_R0086_ENV_TRACK_CTRL, 0x01); - regmap_write(max98927->regmap, - MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, - 0x10); /* voltage, current slot configuration */ regmap_write(max98927->regmap, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..43d3c74 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) @@ -194,7 +196,7 @@ /* MAX98927_R0024_PCM_SR_SETUP2 */ #define MAX98927_PCM_SR_SET2_SR_MASK (0xF << 4) #define MAX98927_PCM_SR_SET2_SR_SHIFT (4) -#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xf << 0) +#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xF << 0) /* MAX98927_R0025_PCM_TO_SPK_MONOMIX_A */ #define MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK (0x3 << 6) @@ -207,7 +209,7 @@ #define MAX98927_AMP_VOL_SEL (0x1 << 7) #define MAX98927_AMP_VOL_SEL_WIDTH (1) #define MAX98927_AMP_VOL_SEL_SHIFT (7) -#define MAX98927_AMP_VOL_MASK (0x7f << 0) +#define MAX98927_AMP_VOL_MASK (0x7F << 0) #define MAX98927_AMP_VOL_WIDTH (7) #define MAX98927_AMP_VOL_SHIFT (0) @@ -238,7 +240,7 @@ #define MAX98927_MEAS_I_EN (0x1 << 1) /* MAX98927_R0040_BOOST_CTRL0 */ -#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1f << 0) +#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1F << 0) #define MAX98927_BOOST_CTRL0_PVDD_MASK (0x1 << 7) #define MAX98927_BOOST_CTRL0_PVDD_EN_SHIFT (7) @@ -248,6 +250,10 @@ #define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) #define MAX98927_BROWNOUT_DSP_SHIFT (2) +/* MAX98927_R0086_ENV_TRACK_CTRL */ +#define MAX98927_ENV_TRACKER_EN (0x1 << 0) +#define MAX98927_ENV_TRACKER_EN_SHIFT (0) + /* MAX98927_R0100_SOFT_RESET */ #define MAX98927_SOFT_RESET (0x1 << 0) @@ -257,7 +263,6 @@ 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; -- 2.7.4

2 2

[alsa-devel] [PATCH 5/5] staging: max98927: Removed an obsolete variable
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index 3069a09..3551e7d 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -263,7 +263,6 @@ 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; -- 2.7.4

1 0

[alsa-devel] [PATCH 3/5] staging: max98927: Updated volatile register list
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index cdee3a3..e787a03 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -586,6 +586,12 @@ static bool max98927_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + case MAX98927_R004C_MEAS_ADC_CH0_READ: + case MAX98927_R004D_MEAS_ADC_CH1_READ: + case MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS: + case MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R01FF_REV_ID: return true; default: return false; -- 2.7.4

1 0

[alsa-devel] [PATCH 2/5] staging: max98927: Added controls for Envelope tracking
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 20 ++++++++++++++++---- sound/soc/codecs/max98927.h | 4 ++++ 2 files changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 99d6e41..cdee3a3 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -614,6 +614,18 @@ static SOC_ENUM_SINGLE_DECL(max98927_current_limit, MAX98927_R0042_BOOST_CTRL1, 1, max98927_current_limit_text); +static const char * const max98927_env_track_headroom_text[] = { + "0.000V", "0.125V", "0.250V", "0.375V", "0.500V", "0.625V", + "0.750V", "0.875V", "1.000V", "1.125V", "1.250V", "1.375V", + "1.500V", "1.625V", "1.750V", "1.875V", "2.000V", "2.125V", + "2.250V", "2.375V", "2.500V", "2.625V", "2.750V", "2.875V", + "3.000V", "3.125V", "3.250V", "3.375V", "3.500V" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_env_track_headroom, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0, + max98927_env_track_headroom_text); + static const struct snd_kcontrol_new max98927_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, 0, 6, 0, @@ -631,6 +643,9 @@ static const struct snd_kcontrol_new max98927_snd_controls[] = { MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), SOC_ENUM("Current Limit", max98927_current_limit), + SOC_SINGLE("EnvTrack Switch", MAX98927_R0086_ENV_TRACK_CTRL, + MAX98927_ENV_TRACKER_EN_SHIFT, 1, 0), + SOC_ENUM("EnvTrack Headroom", max98927_env_track_headroom), }; static const struct snd_soc_dapm_route max98927_audio_map[] = { @@ -726,13 +741,10 @@ static int max98927_probe(struct snd_soc_codec *codec) /* Envelope Tracking configuration */ regmap_write(max98927->regmap, MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, - 0x08); + 0x0A); regmap_write(max98927->regmap, MAX98927_R0086_ENV_TRACK_CTRL, 0x01); - regmap_write(max98927->regmap, - MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, - 0x10); /* voltage, current slot configuration */ regmap_write(max98927->regmap, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index bf7a6f92..3069a09 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -250,6 +250,10 @@ #define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) #define MAX98927_BROWNOUT_DSP_SHIFT (2) +/* MAX98927_R0086_ENV_TRACK_CTRL */ +#define MAX98927_ENV_TRACKER_EN (0x1 << 0) +#define MAX98927_ENV_TRACKER_EN_SHIFT (0) + /* MAX98927_R0100_SOFT_RESET */ #define MAX98927_SOFT_RESET (0x1 << 0) -- 2.7.4

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[alsa-devel] [PATCH 1/5] staging: max98927: Added TDM support
by Ryan Lee 19 May '17

19 May '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

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[alsa-devel] [bug report] ASoC: add audio-graph-card support
by Dan Carpenter 19 May '17

19 May '17

Hello Kuninori Morimoto, The patch 2692c1c63c29: "ASoC: add audio-graph-card support" from Apr 20, 2017, leads to the following static checker warning: sound/soc/generic/audio-graph-card.c:275 asoc_graph_card_probe() info: return a literal instead of 'ret' sound/soc/generic/audio-graph-card.c 268 goto err; 269 } 270 271 snd_soc_card_set_drvdata(card, priv); 272 273 ret = devm_snd_soc_register_card(dev, card); 274 if (ret >= 0) 275 return ret; This code is confusing. I'm pretty sure that devm_snd_soc_register_card() returns zero on success or negative error codes but why are we checking >= 0 here? It's better to do error handling than success handling. People often get a bit creative with the last condition in a function... :/ In other words, it's cleaner like this: ret = devm_snd_soc_register_card(dev, card); if (ret) goto err; return 0; err: asoc_simple_card_clean_reference(card); return ret; 276 err: 277 asoc_simple_card_clean_reference(card); 278 279 return ret; 280 } regards, dan carpenter

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Re: [alsa-devel] WM5102 - Help to make baytrail machine driver work
by Paulo Sergio 18 May '17

18 May '17

Adding alsa list + intel devs (Sorry for that ) Inline answers for Charles Keepax questions bellow: 2017-05-08 5:24 GMT-03:00 Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>: > On Sun, May 07, 2017 at 10:42:31PM -0300, Paulo Sergio wrote: >> Hi, >> >> >> I'm trying to make wm5102 work on Lenovo Yoga 2 1051F (Baytrail) again >> and would like some tips. >> >> With a revised machine driver code [1] ( based on bytcr_rt5640 and >> some ports from Lenovo kernel [2]) I can initialize >> and setting routes with alsa mixer [3], I still don't have sound. >> >> However now if I connect 'HPOUT1' to 'Tone Generator 1' I can hear the 1k tone. >> Question: When I do this association, it is using my route? (in my >> case CODEC OUT -> SSP0 TX -> AIF1RX) >> Or this internal tone generator has a direct link to HPOUTs? >> >> Here's some lines from my dmesg output while attemptig to play >> something (complete dmesg - see [4]): >> >> [ 211.801176] arizona spi-WM510204:00: FLL1: Fref=25000000 Fout=49152000 >> [ 211.801184] arizona spi-WM510204:00: FLL1: Fvco=98304000Hz >> [ 211.801190] arizona spi-WM510204:00: FLL1: GCD=4000 >> [ 211.801195] arizona spi-WM510204:00: FLL1: N=7 THETA=a8d LAMBDA=c35 >> [ 211.801200] arizona spi-WM510204:00: FLL1: FRATIO=0(0) OUTDIV=2 REFCLK_DIV=1 >> [ 211.801204] arizona spi-WM510204:00: FLL1: GAIN=4 >> [ 211.801210] arizona spi-WM510204:00: 173 <= a8d >> [ 211.801227] arizona spi-WM510204:00: 174 <= c35 >> [ 211.801234] arizona spi-WM510204:00: 176 <= 40 >> [ 211.801241] arizona spi-WM510204:00: 179 <= 10 >> [ 211.813301] arizona spi-WM510204:00: 172 <= 8007 >> [ 211.813353] arizona spi-WM510204:00: 171 <= 1 >> [ 211.813366] arizona spi-WM510204:00: FLL1: Waiting for FLL lock... >> [ 211.814098] arizona spi-WM510204:00: d23 => 100 >> [ 211.814641] arizona spi-WM510204:00: d23 => 101 >> [ 211.814651] arizona spi-WM510204:00: FLL1: FLL locked (1 polls) >> [ 211.814660] arizona spi-WM510204:00: SYSCLK set to 49152000Hz >> [ 211.814668] arizona spi-WM510204:00: SYSCLK set to 49152000Hz >> [ 211.814675] arizona spi-WM510204:00: 101 <= 300 >> [ 211.814882] wm5102-codec wm5102-codec: AIF1: BCLK 1536000Hz LRCLK 48000Hz >> [ 211.814892] arizona spi-WM510204:00: 80 <= 3 >> [ 211.815191] arizona spi-WM510204:00: 4dc <= 0 >> [ 211.815339] arizona spi-WM510204:00: 4dd <= 0 >> [ 211.815460] arizona spi-WM510204:00: 80 <= 0 >> [ 211.815662] sst-mfld-platform sst-mfld-platform: Enter: enable=1 >> port_name=ssp0-port >> [ 211.816833] wm5102-codec wm5102-codec: AIF1: BCLK 1536000Hz LRCLK 48000Hz >> [ 211.816845] arizona spi-WM510204:00: 80 <= 3 >> [ 211.817030] arizona spi-WM510204:00: 4dc <= 0 >> [ 211.817287] arizona spi-WM510204:00: 4dd <= 0 >> >> Thanks in advance! >> >> >> [1] - https://github.com/pstglia/linux/tree/lenovo_yoga2_returns >> [2] - https://github.com/lenovo-yt2-dev/android_kernel_lenovo_baytrail/blob/cm-12… >> [3] - https://drive.google.com/file/d/0BxO6THtB865fQnQyZkx6c05Nbjg/view?usp=shari… >> [4] - https://drive.google.com/file/d/0BxO6THtB865fRy0xV3F0elRrUHM/view?usp=shari… > > OK good news you are getting sound from the Tone generator, that > likely implies things are setup sensibly on the CODEC side. My > suspicion would be that data is not making it over the I2S. > Probalby worth copying the Intel guys on the email, as I don't > know know much about that side of things. Copied alsa list and Intel guys (hit reply instead of reply all - my apologies) > > How are you setting up the I2S? As in which side is master, what > format are you using? Also do you see any over/underflow error, > does the system complain with "Error playing sample". The only part where I2S is explicity used on machine driver is this setting on backend DAI (SSP0-Codec) dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, And after setting the routes using routes mentioned on [3], I have no errors, but no sound Note: Based on Lenovo code, I concluded this device uses ssp0, so I'm using a fw file "fw_sst_0f28_ssp0.bin" posted another thread. Could use the hacks used on kernel 4.11, but as I'm still testing with kernel 4.4, using this fw would make things easier x86_64:/data # alsa_aplay -v TNT.mp3 Playing raw data 'TNT.mp3' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 262144000 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 Slave: Hardware PCM card 0 'baytrailcraudio' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 appl_ptr : 0 hw_ptr : 0 Regards, Pstglia

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