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

[alsa-devel] Applied "ASoC: max98927: Added max98927_dai_tdm_slot function" to the asoc tree
by Mark Brown 25 Sep '17

25 Sep '17

The patch ASoC: max98927: Added max98927_dai_tdm_slot function 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 d4a8bce81cbd53420988f5db0d096ad04960f189 Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Thu, 14 Sep 2017 17:30:39 -0700 Subject: [PATCH] ASoC: max98927: Added max98927_dai_tdm_slot function Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98927.c | 120 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/max98927.h | 1 + 2 files changed, 105 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 6f7b3ef48e25..a1d39353719d 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -250,6 +250,21 @@ 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) { @@ -275,23 +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; } @@ -391,6 +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 | \ @@ -410,6 +494,7 @@ 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, @@ -419,6 +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 bf7a6f92562a..9ea839735433 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -270,5 +270,6 @@ struct max98927_priv { unsigned int iface; unsigned int master; unsigned int digital_gain; + bool tdm_mode; }; #endif -- 2.14.1

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[alsa-devel] [PATCH 1/2] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 25 Sep '17

25 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 35 ++++++++++++++++++++--------------- sound/soc/codecs/max98927.h | 6 ++++-- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe7..6f7b3ef 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; } 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] [PATCH] ASoC: dapm: Make snd_soc_dapm_add/del_routes use runtime mutex subclass
by Stuart Henderson 25 Sep '17

25 Sep '17

Make snd_soc_dapm_add_routes and snd_soc_dapm_del_routes use the SND_SOC_DAPM_CLASS_RUNTIME dapm subclass to allow dynamic dapm route updates. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..e6b6011 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2884,7 +2884,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, { int i, r, ret = 0; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); for (i = 0; i < num; i++) { r = snd_soc_dapm_add_route(dapm, route); if (r < 0) { @@ -2915,7 +2915,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, { int i; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); for (i = 0; i < num; i++) { snd_soc_dapm_del_route(dapm, route); route++; -- 2.1.4

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[alsa-devel] [PATCH 0/4][RESEND] ASoC: rtd NULL check and unneeded ops check
by Kuninori Morimoto 25 Sep '17

25 Sep '17

Hi Mark These are posted few weeks ago, and nothting happen. Thus, I resend these. I'm posting "Convert Codec/Platform to Component" patch-set too, but I know it is very big patch-set, need more review. OTOH, These are very light patches. Thus, I will repost these 1st. Then, I will repost "Convert ..." patch-set again on top of these if these are accepted. Kuninori Morimoto (4): ASoC: soc-core: remove rtd NULL check on soc_free_pcm_runtime() ASoC: remove unneeded dai->driver check ASoC: remove unneeded dai->driver->ops check ASoC: add null_snd_soc_ops and reduce NULL ops check sound/soc/soc-core.c | 24 ++++++++++++++---------- sound/soc/soc-dapm.c | 8 ++++---- sound/soc/soc-pcm.c | 47 +++++++++++++++++++++++------------------------ 3 files changed, 41 insertions(+), 38 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: rt5663: Change the dev getting function in rt5663_irq
by Oder Chiou 25 Sep '17

25 Sep '17

In the irq handler "rt5663_irq", while the codec is not initialized, rt5663->codec will be null, and it will cause the kernel panic in the debug print enabled. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 767f219..fcd1643 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1827,7 +1827,8 @@ static irqreturn_t rt5663_irq(int irq, void *data) { struct rt5663_priv *rt5663 = data; - dev_dbg(rt5663->codec->dev, "%s IRQ queue work\n", __func__); + dev_dbg(regmap_get_device(rt5663->regmap), "%s IRQ queue work\n", + __func__); queue_delayed_work(system_wq, &rt5663->jack_detect_work, msecs_to_jiffies(250)); -- 2.7.4

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[alsa-devel] [PATCH 0/3] Intel-rt5670 updates
by Pierre-Louis Bossart 25 Sep '17

25 Sep '17

Instead of submitting all my patches at ones, I broke them by platforms/codecs. This series is only for Intel-rt5670 platforms. It fixes a mistake on the Dell Venue 5585 - corrected with the help from Realtek and Dell who kindly provided schematics and data sheet information needed to fix non-working DMIC capture. Since there are still some reports of non-functional audio on some platforms, I added quirks that can be changed from the kernel command line (and later added as DMI-based quirks). The last two patches are minor corrections to fix ACPI HID naming issues and use a more meaningful card name. Pierre-Louis Bossart (3): ASoC: rt5670: refactor DMI quirks and fix Dell Venue settings ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name sound/soc/codecs/rt5670.c | 124 +++++++++++++++++++++++++++----- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++- 2 files changed, 112 insertions(+), 22 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH 00/14] BYT/CHT codec/machine drivers update
by Pierre-Louis Bossart 25 Sep '17

25 Sep '17

Finally took the time to cleanup known fixes for codec and machine drivers. I tested on Baytrail/Cherrytrail and w/ RT5640, 45 and 72. This should be the last batch of fixes before we move on to the integration of SOF (Sound Open Firmware) support in the mainline. This batch contains two patches already submitted by Thierry Escande on alsa-devel but never merged. Fang, Yang A (1): ASoC: ts3a227e: add acpi table Pierre-Louis Bossart (12): ASoC: Intel: boards: use devm_clk_get() unconditionally ASoC: intel: cht_bsw_max98090: add support for clock framework ASoC: Intel: cht_bsw_max98090: Fix I2S config + unused code ASoC: Intel: atom: add support for Baytrail Chromebooks ASoC: Intel: bytcr-rt5651: fix capture routes ASoC: Intel: bytcr_rt5651: add MCLK support ASoC: Intel: bytcr_rt5651: filter codec name ASoC: codecs: rt5645: add quirks for Asus T100HA ASoC: Intel: bytcr_rt5640: simplify MCLK quirk tests ASoC: codecs: rt5670: add jack detection quirk for Dell Venue 5585 ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name Thierry Escande (1): ASoc: cht_bsw_max98090_ti: Fix jack initialization sound/soc/codecs/rt5645.c | 20 ++ sound/soc/codecs/rt5670.c | 16 ++ sound/soc/codecs/ts3a227e.c | 10 + sound/soc/intel/atom/sst/sst_acpi.c | 8 + sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 278 ++++++++++++++++++++++++--- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 157 ++++++++++++--- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 36 ++-- 9 files changed, 460 insertions(+), 87 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix swapped order of function arguments dir and pin_index
by Colin King 25 Sep '17

25 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> The call to slk_tplg_fill_res_tkn is passing dir and pin_index in the wrong order, they need to be in pin_index, dir order to match the function slk_tplg_fill_res_tkn correctly. Detected by CoveritScan, CID#1454992 ("Arguments in wrong order") Fixes: f6fa56e22559 ("ASoC: Intel: Skylake: Parse and update module config structure") Signed-off-by: Colin Ian King <colin.king(a)canonical.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..27bcb62568fb 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2382,7 +2382,7 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U32_MAX_MCPS: case SKL_TKN_U32_OBS: case SKL_TKN_U32_IBS: - ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, dir, pin_index); + ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, pin_index, dir); if (ret < 0) return ret; -- 2.14.1

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[alsa-devel] [PATCH] ASoC: wm8998: Correct handling of input muxes
by Charles Keepax 25 Sep '17

25 Sep '17

Currently, wm8998 has two input mux controls on IN1 and attempts to switch these together when the A position is configured to be in digital mode. This is because the digital mode requires pins from both the L and R channels. However, this doesn't work as intended because whilst the registers on the chip are changed the corresponding DAPM representation is only updated for the mux actually being changed by the user. The DAPM graph being out of sync with the hardware can cause some odd issues with incorrect things being powered etc. To avoid this issue and simplify the code somewhat, simply let the user set the muxes as they desire. If they set an invalid configuration they might not get audio from the DMIC but most of the chip requires you to set a valid audio route to get audio. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm8998.c | 84 +++++++++++------------------------------------ 1 file changed, 19 insertions(+), 65 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index c59caaa75ba0..2d211dbe7422 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -101,7 +101,7 @@ static int wm8998_asrc_ev(struct snd_soc_dapm_widget *w, return 0; } -static int wm8998_in1mux_put(struct snd_kcontrol *kcontrol, +static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); @@ -109,84 +109,38 @@ static int wm8998_in1mux_put(struct snd_kcontrol *kcontrol, struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = wm8998->core.arizona; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux, inmode; - unsigned int mode_val, src_val; + unsigned int mode_reg, mode_index; + unsigned int mux, inmode, src_val, mode_val; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) return -EINVAL; - /* L and R registers have same shift and mask */ - inmode = arizona->pdata.inmode[2 * mux]; - src_val = mux << ARIZONA_IN1L_SRC_SHIFT; - if (inmode & ARIZONA_INMODE_SE) - src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT; - - switch (arizona->pdata.inmode[0]) { - case ARIZONA_INMODE_DMIC: - if (mux) - mode_val = 0; /* B always analogue */ - else - mode_val = 1 << ARIZONA_IN1_MODE_SHIFT; - - snd_soc_update_bits(codec, ARIZONA_IN1L_CONTROL, - ARIZONA_IN1_MODE_MASK, mode_val); - - /* IN1A is digital so L and R must change together */ - /* src_val setting same for both registers */ - snd_soc_update_bits(codec, - ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, src_val); - snd_soc_update_bits(codec, - ARIZONA_ADC_DIGITAL_VOLUME_1R, - ARIZONA_IN1R_SRC_MASK | - ARIZONA_IN1R_SRC_SE_MASK, src_val); + switch (e->reg) { + case ARIZONA_ADC_DIGITAL_VOLUME_2L: + mode_reg = ARIZONA_IN2L_CONTROL; + mode_index = 1 + (2 * mux); break; default: - /* both analogue */ - snd_soc_update_bits(codec, - e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + mode_reg = ARIZONA_IN1L_CONTROL; + mode_index = (2 * mux); break; } - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); -} - -static int wm8998_in2mux_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); - struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec); - struct arizona *arizona = wm8998->core.arizona; - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux, inmode, src_val, mode_val; - - mux = ucontrol->value.enumerated.item[0]; - if (mux > 1) - return -EINVAL; - - inmode = arizona->pdata.inmode[1 + (2 * mux)]; + inmode = arizona->pdata.inmode[mode_index]; if (inmode & ARIZONA_INMODE_DMIC) - mode_val = 1 << ARIZONA_IN2_MODE_SHIFT; + mode_val = 1 << ARIZONA_IN1_MODE_SHIFT; else mode_val = 0; - src_val = mux << ARIZONA_IN2L_SRC_SHIFT; + src_val = mux << ARIZONA_IN1L_SRC_SHIFT; if (inmode & ARIZONA_INMODE_SE) - src_val |= 1 << ARIZONA_IN2L_SRC_SE_SHIFT; + src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT; - snd_soc_update_bits(codec, ARIZONA_IN2L_CONTROL, - ARIZONA_IN2_MODE_MASK, mode_val); + snd_soc_update_bits(codec, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_update_bits(codec, ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_IN2L_SRC_MASK | ARIZONA_IN2L_SRC_SE_MASK, + snd_soc_update_bits(codec, e->reg, + ARIZONA_IN1L_SRC_MASK | ARIZONA_IN1L_SRC_SE_MASK, src_val); return snd_soc_dapm_mux_update_power(dapm, kcontrol, @@ -216,14 +170,14 @@ static SOC_ENUM_SINGLE_DECL(wm8998_in2mux_enum, static const struct snd_kcontrol_new wm8998_in1mux[2] = { SOC_DAPM_ENUM_EXT("IN1L Mux", wm8998_in1muxl_enum, - snd_soc_dapm_get_enum_double, wm8998_in1mux_put), + snd_soc_dapm_get_enum_double, wm8998_inmux_put), SOC_DAPM_ENUM_EXT("IN1R Mux", wm8998_in1muxr_enum, - snd_soc_dapm_get_enum_double, wm8998_in1mux_put), + snd_soc_dapm_get_enum_double, wm8998_inmux_put), }; static const struct snd_kcontrol_new wm8998_in2mux = SOC_DAPM_ENUM_EXT("IN2 Mux", wm8998_in2mux_enum, - snd_soc_dapm_get_enum_double, wm8998_in2mux_put); + snd_soc_dapm_get_enum_double, wm8998_inmux_put); static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); -- 2.11.0

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[alsa-devel] [PATCH] Documentation: DT bindings: Add chip compatible string for audio codec.
by Shreeya Patel 25 Sep '17

25 Sep '17

This patch adds chip compatible string to remove the warnings generated by checkpatch that -DT compatible string "toshiba,apb-dummy-codec" appears un-documented. Signed-off-by: Shreeya Patel <shreeya.patel23498(a)gmail.com> --- Documentation/devicetree/bindings/sound/sirf-audio-codec.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt index 062f5ec..a61560b 100644 --- a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt +++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt @@ -3,6 +3,7 @@ SiRF internal audio CODEC Required properties: - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec" + "toshiba,apb-dummy-codec" - reg : the register address of the device. @@ -11,7 +12,7 @@ Required properties: Example: audiocodec: audiocodec@b0040000 { - compatible = "sirf,atlas6-audio-codec"; + compatible = "sirf,atlas6-audio-codec", "toshiba,apb-dummy-codec"; reg = <0xb0040000 0x10000>; clocks = <&clks 27>; }; -- 2.7.4

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