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

[alsa-devel] [PATCH] ASoC: da7213: Fix incorrect usage of bitwise '&' operator for SRM check
by Adam Thomson 15 May '17

15 May '17

In the SRM lock check section of code the '&' bitwise operator is used as part of checking lock status. Functionally the code works as intended, but the conditional statement is a boolean comparison so should really use '&&' logical operator instead. This commit rectifies this discrepancy. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7213.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 12da558..d114f33 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -771,7 +771,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(50); } - } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + } while ((i < DA7213_SRM_CHECK_RETRIES) && (!srm_lock)); if (!srm_lock) dev_warn(codec->dev, "SRM failed to lock\n"); -- 1.9.1

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[alsa-devel] Applied "ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec" to the asoc tree
by Mark Brown 15 May '17

15 May '17

The patch ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 51827c41c9ce07293b094691673e6ec23dfdc5e8 Mon Sep 17 00:00:00 2001 From: Tomas Vilda <tomas.vilda(a)gmail.com> Date: Sat, 13 May 2017 00:29:37 +0300 Subject: [PATCH] ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec Signed-off-by: Tomas Vilda <tomas.vilda(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8a90ba8cd71..d7d03c92cb8a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1210,7 +1210,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { static struct snd_soc_dai_driver dac31xx_dai_driver[] = { { - .name = "tlv32dac31xx-hifi", + .name = "tlv320dac31xx-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, -- 2.11.0

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[alsa-devel] [PATCH][V2] ASoC: Intel: sst: fix spelling mistake: "allocationf" -> "allocation"
by Colin King 15 May '17

15 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message. Also replace "fail" with "failure". Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 21cac1c8dd4c..e74119113713 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -690,7 +690,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - dev_err(rtd->dev, "dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocation failure\n"); return retval; } } -- 2.11.0

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[alsa-devel] [PATCH] ASoC: nau8824: TDM support
by John Hsu 15 May '17

15 May '17

Support TDM format for NAU88L24. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 12 +++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cca974d..3a309b1 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1125,6 +1125,57 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /** + * nau8824_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: Bitmask representing active TX slots. Ex. + * 0xf for normal 4 channel TDM. + * 0xf0 for shifted 4 channel TDM + * @rx_mask: Bitmask [0:1] representing active DACR RX slots. + * Bitmask [2:3] representing active DACL RX slots. + * 00=CH0,01=CH1,10=CH2,11=CH3. Ex. + * 0xf for DACL/R selecting TDM CH3. + * 0xf0 for DACL/R selecting shifted TDM CH3. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Only support 4 slots TDM. + */ +static int nau8824_set_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 nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int tslot_l = 0, ctrl_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf0) && (rx_mask & 0xf)) || + ((rx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf) && (tx_mask & 0xf0))) + return -EINVAL; + + ctrl_val |= (NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + tslot_l = 4 * slot_width; + ctrl_val |= (tx_mask >> 4); + } else { + ctrl_val |= tx_mask; + } + if (rx_mask & 0xf0) + ctrl_val |= ((rx_mask >> 4) << NAU8824_TDM_DACR_RX_SFT); + else + ctrl_val |= (rx_mask << NAU8824_TDM_DACR_RX_SFT); + + regmap_update_bits(nau8824->regmap, NAU8824_REG_TDM_CTRL, + NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN | + NAU8824_TDM_DACL_RX_MASK | NAU8824_TDM_DACR_RX_MASK | + NAU8824_TDM_TX_MASK, ctrl_val); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_LEFT_TIME_SLOT, + NAU8824_TSLOT_L_MASK, tslot_l); + + return 0; +} + +/** * nau8824_calc_fll_param - Calculate FLL parameters. * @fll_in: external clock provided to codec. * @fs: sampling rate. @@ -1440,6 +1491,7 @@ static struct snd_soc_codec_driver nau8824_codec_driver = { static const struct snd_soc_dai_ops nau8824_dai_ops = { .hw_params = nau8824_hw_params, .set_fmt = nau8824_set_fmt, + .set_tdm_slot = nau8824_set_tdm_slot, }; #define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index 87ac9a3..21eae24 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -258,6 +258,18 @@ #define NAU8824_I2S_MS_SLAVE (0 << NAU8824_I2S_MS_SFT) #define NAU8824_I2S_BLK_DIV_MASK 0x7 +/* PORT0_LEFT_TIME_SLOT (0x1E) */ +#define NAU8824_TSLOT_L_MASK 0x3ff + +/* TDM_CTRL (0x20) */ +#define NAU8824_TDM_MODE (0x1 << 15) +#define NAU8824_TDM_OFFSET_EN (0x1 << 14) +#define NAU8824_TDM_DACL_RX_SFT 6 +#define NAU8824_TDM_DACL_RX_MASK (0x3 << NAU8824_TDM_DACL_RX_SFT) +#define NAU8824_TDM_DACR_RX_SFT 4 +#define NAU8824_TDM_DACR_RX_MASK (0x3 << NAU8824_TDM_DACR_RX_SFT) +#define NAU8824_TDM_TX_MASK 0xf + /* ADC_FILTER_CTRL (0x24) */ #define NAU8824_ADC_SYNC_DOWN_MASK 0x3 #define NAU8824_ADC_SYNC_DOWN_32 0 -- 2.6.4

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[alsa-devel] [PATCH] ASoC: rt5514: Add more width and channels support in the TDM mode
by Oder Chiou 15 May '17

15 May '17

This patch adds more width and channels support in the TDM mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 23 +++++++++++++++++++++-- sound/soc/codecs/rt5514.h | 6 ++++++ 2 files changed, 27 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b..ff97360 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -906,9 +906,23 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; - if (slots == 4) + switch (slots) { + case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; + break; + + case 6: + val |= RT5514_TDMSLOT_SEL_RX_6CH | RT5514_TDMSLOT_SEL_TX_6CH; + break; + + case 8: + val |= RT5514_TDMSLOT_SEL_RX_8CH | RT5514_TDMSLOT_SEL_TX_8CH; + break; + case 2: + default: + break; + } switch (slot_width) { case 20: @@ -919,6 +933,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24; break; + case 25: + val |= RT5514_TDM_MODE2; + break; + case 32: val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32; break; @@ -930,7 +948,8 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE | RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK | - RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val); + RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | + RT5514_TDM_MODE2, val); return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 5d343fb..02bc212 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -117,6 +117,8 @@ #define RT5514_POW_ADCFEDL_BIT 0 /* RT5514_I2S_CTRL1 (0x2010) */ +#define RT5514_TDM_MODE2 (0x1 << 30) +#define RT5514_TDM_MODE2_SFT 30 #define RT5514_TDM_MODE (0x1 << 28) #define RT5514_TDM_MODE_SFT 28 #define RT5514_I2S_LR_MASK (0x1 << 26) @@ -136,6 +138,8 @@ #define RT5514_TDMSLOT_SEL_RX_MASK (0x3 << 10) #define RT5514_TDMSLOT_SEL_RX_SFT 10 #define RT5514_TDMSLOT_SEL_RX_4CH (0x1 << 10) +#define RT5514_TDMSLOT_SEL_RX_6CH (0x2 << 10) +#define RT5514_TDMSLOT_SEL_RX_8CH (0x3 << 10) #define RT5514_CH_LEN_RX_MASK (0x3 << 8) #define RT5514_CH_LEN_RX_SFT 8 #define RT5514_CH_LEN_RX_16 (0x0 << 8) @@ -145,6 +149,8 @@ #define RT5514_TDMSLOT_SEL_TX_MASK (0x3 << 6) #define RT5514_TDMSLOT_SEL_TX_SFT 6 #define RT5514_TDMSLOT_SEL_TX_4CH (0x1 << 6) +#define RT5514_TDMSLOT_SEL_TX_6CH (0x2 << 6) +#define RT5514_TDMSLOT_SEL_TX_8CH (0x3 << 6) #define RT5514_CH_LEN_TX_MASK (0x3 << 4) #define RT5514_CH_LEN_TX_SFT 4 #define RT5514_CH_LEN_TX_16 (0x0 << 4) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt5665: add ADC STO2 ASRC support
by Bard Liao 15 May '17

15 May '17

"ADC Stereo2 Filter" is with ASRC supported. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd2230..ece77b8 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2684,6 +2684,8 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { RT5665_DAC_MONO_R_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5665_ASRC_1, RT5665_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5665_ASRC_1, + RT5665_ADC_STO2_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono L ASRC", 1, RT5665_ASRC_1, RT5665_ADC_MONO_L_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono R ASRC", 1, RT5665_ASRC_1, @@ -3227,6 +3229,7 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc}, {"ADC Mono Left Filter", NULL, "ADC Mono L ASRC", is_using_asrc}, {"ADC Mono Right Filter", NULL, "ADC Mono R ASRC", is_using_asrc}, {"DAC Mono Left Filter", NULL, "DAC Mono L ASRC", is_using_asrc}, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: tidyup Sampling rate convert example on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "renesas,rsrc-card" is exchanged to "simple-scu-card". Let's update Document Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316..3332910 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- 1.9.1

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[alsa-devel] [PATCH][RESEND] ASoC: rsnd: add missing clocks/clock-names on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 3332910..13cb21d5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -368,6 +368,10 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI +- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. +- clock-names : List of necessary clock names. + "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", + "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout -- 1.9.1

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[alsa-devel] [PATCH v2][RESEND] ASoC: rsnd: control kctrl items acceptance anytime/runtime
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SRC/DVC/CTU adds kctrl for each device, and SRC can adjust its sampling rate during playback, thus, this feature should be enabled only *during* playback. This patch controls it more clearly Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 17 +++++++++++++++++ sound/soc/sh/rcar/ctu.c | 6 ++++++ sound/soc/sh/rcar/dvc.c | 5 +++++ sound/soc/sh/rcar/rsnd.h | 16 ++++++++++------ sound/soc/sh/rcar/src.c | 2 ++ 5 files changed, 40 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1744015..d3c9078 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -982,6 +982,9 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); int i, change = 0; + if (!cfg->accept(cfg->io)) + return 0; + for (i = 0; i < cfg->size; i++) { if (cfg->texts) { change |= (uc->value.enumerated.item[i] != cfg->val[i]); @@ -998,6 +1001,18 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, return change; } +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) +{ + return 1; +} + +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + return !!runtime; +} + struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; @@ -1016,6 +1031,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -1050,6 +1066,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->texts = texts; cfg->max = max; cfg->size = size; + cfg->accept = accept; cfg->update = update; cfg->card = card; cfg->kctrl = kctrl; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 9dcc1f9..4ba8f2f 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -279,12 +279,14 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* CTU Pass */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + rsnd_kctrl_accept_anytime, NULL, &ctu->pass, RSND_MAX_CHANNELS, 0xC); /* ROW0 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv0, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -293,6 +295,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW1 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv1, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -301,6 +304,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW2 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv2, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -309,6 +313,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW3 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv3, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -317,6 +322,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* Reset */ ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_kctrl_accept_anytime, rsnd_ctu_value_reset, &ctu->reset, 1); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 463de83..75af6e7 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -257,6 +257,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Playback Volume" : "DVC In Capture Volume", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->volume, slots, 0x00800000 - 1); @@ -267,6 +268,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Mute Switch" : "DVC In Mute Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->mute, slots, 1); @@ -277,6 +279,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_s(mod, io, rtd, is_play ? "DVC Out Ramp Switch" : "DVC In Ramp Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->ren, 1); if (ret < 0) @@ -285,6 +288,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rup, dvc_ramp_rate); @@ -294,6 +298,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rdown, dvc_ramp_rate); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index dbf4163..f68aecb 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -594,6 +594,7 @@ struct rsnd_kctrl_cfg { unsigned int size; u32 *val; const char * const *texts; + int (*accept)(struct rsnd_dai_stream *io); void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod); struct rsnd_dai_stream *io; struct snd_card *card; @@ -611,12 +612,15 @@ struct rsnd_kctrl_cfg_s { u32 val; }; +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -624,16 +628,16 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int size, u32 max); -#define rsnd_kctrl_new_m(mod, io, rtd, name, update, cfg, size, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_m(cfg), \ +#define rsnd_kctrl_new_m(mod, io, rtd, name, accept, update, cfg, size, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_m(cfg), \ NULL, size, max) -#define rsnd_kctrl_new_s(mod, io, rtd, name, update, cfg, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_s(mod, io, rtd, name, accept, update, cfg, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ NULL, 1, max) -#define rsnd_kctrl_new_e(mod, io, rtd, name, update, cfg, texts) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_e(mod, io, rtd, name, accept, update, cfg, texts) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ texts, 1, ARRAY_SIZE(texts)) /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e4bde0c..e395fcb 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -489,6 +489,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate Switch" : "SRC In Rate Switch", + rsnd_kctrl_accept_anytime, rsnd_src_set_convert_rate, &src->sen, 1); if (ret < 0) @@ -498,6 +499,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", + rsnd_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); -- 1.9.1

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[alsa-devel] linux-4.12-rc1/sound/soc/codecs/cs35l35.c: false comparison ?
by David Binderman 14 May '17

14 May '17

hello there, [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==1' is always false. [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==3' is always false. Source code is errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; if (classh->classh_wk_fet_disable == 0x00 && (errata_chk == 0x01 || errata_chk == 0x03)) { but ./sound/soc/codecs/cs35l35.h:#define CS35L35_SP_RATE_MASK 0xC0 Regards David Binderman

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