February 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] arm64: dts: renesas: r8a77965: add SSIU support for sound
by twischer＠de.adit-jv.com 25 Feb '19

25 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> rsnd driver supports SSIU now, let's use it. Then, BUSIF DMA settings on rcar_sound,ssi (= rxu, txu) are no longer needed. Applies commit 8d14bfa074db ("arm64: dts: renesas: r8a7796: add SSIU support for sound") for r8a77965. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 211 ++++++++++++++++++++++++++++++ 1 file changed, 211 insertions(+) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index 4b3730f..cc77c2e 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1585,6 +1585,217 @@ }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + ssiu02: ssiu-2 { + dmas = <&audma0 0x37>, <&audma1 0x38>; + dma-names = "rx", "tx"; + }; + ssiu03: ssiu-3 { + dmas = <&audma0 0x47>, <&audma1 0x48>; + dma-names = "rx", "tx"; + }; + ssiu04: ssiu-4 { + dmas = <&audma0 0x3F>, <&audma1 0x40>; + dma-names = "rx", "tx"; + }; + ssiu05: ssiu-5 { + dmas = <&audma0 0x43>, <&audma1 0x44>; + dma-names = "rx", "tx"; + }; + ssiu06: ssiu-6 { + dmas = <&audma0 0x4F>, <&audma1 0x50>; + dma-names = "rx", "tx"; + }; + ssiu07: ssiu-7 { + dmas = <&audma0 0x53>, <&audma1 0x54>; + dma-names = "rx", "tx"; + }; + ssiu10: ssiu-8 { + dmas = <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx"; + }; + ssiu11: ssiu-9 { + dmas = <&audma0 0x4B>, <&audma1 0x4C>; + dma-names = "rx", "tx"; + }; + ssiu12: ssiu-10 { + dmas = <&audma0 0x57>, <&audma1 0x58>; + dma-names = "rx", "tx"; + }; + ssiu13: ssiu-11 { + dmas = <&audma0 0x59>, <&audma1 0x5A>; + dma-names = "rx", "tx"; + }; + ssiu14: ssiu-12 { + dmas = <&audma0 0x5F>, <&audma1 0x60>; + dma-names = "rx", "tx"; + }; + ssiu15: ssiu-13 { + dmas = <&audma0 0xC3>, <&audma1 0xC4>; + dma-names = "rx", "tx"; + }; + ssiu16: ssiu-14 { + dmas = <&audma0 0xC7>, <&audma1 0xC8>; + dma-names = "rx", "tx"; + }; + ssiu17: ssiu-15 { + dmas = <&audma0 0xCB>, <&audma1 0xCC>; + dma-names = "rx", "tx"; + }; + ssiu20: ssiu-16 { + dmas = <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx"; + }; + ssiu21: ssiu-17 { + dmas = <&audma0 0x67>, <&audma1 0x68>; + dma-names = "rx", "tx"; + }; + ssiu22: ssiu-18 { + dmas = <&audma0 0x6B>, <&audma1 0x6C>; + dma-names = "rx", "tx"; + }; + ssiu23: ssiu-19 { + dmas = <&audma0 0x6D>, <&audma1 0x6E>; + dma-names = "rx", "tx"; + }; + ssiu24: ssiu-20 { + dmas = <&audma0 0xCF>, <&audma1 0xCE>; + dma-names = "rx", "tx"; + }; + ssiu25: ssiu-21 { + dmas = <&audma0 0xEB>, <&audma1 0xEC>; + dma-names = "rx", "tx"; + }; + ssiu26: ssiu-22 { + dmas = <&audma0 0xED>, <&audma1 0xEE>; + dma-names = "rx", "tx"; + }; + ssiu27: ssiu-23 { + dmas = <&audma0 0xEF>, <&audma1 0xF0>; + dma-names = "rx", "tx"; + }; + ssiu30: ssiu-24 { + dmas = <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx"; + }; + ssiu31: ssiu-25 { + dmas = <&audma0 0x21>, <&audma1 0x22>; + dma-names = "rx", "tx"; + }; + ssiu32: ssiu-26 { + dmas = <&audma0 0x23>, <&audma1 0x24>; + dma-names = "rx", "tx"; + }; + ssiu33: ssiu-27 { + dmas = <&audma0 0x25>, <&audma1 0x26>; + dma-names = "rx", "tx"; + }; + ssiu34: ssiu-28 { + dmas = <&audma0 0x27>, <&audma1 0x28>; + dma-names = "rx", "tx"; + }; + ssiu35: ssiu-29 { + dmas = <&audma0 0x29>, <&audma1 0x2A>; + dma-names = "rx", "tx"; + }; + ssiu36: ssiu-30 { + dmas = <&audma0 0x2B>, <&audma1 0x2C>; + dma-names = "rx", "tx"; + }; + ssiu37: ssiu-31 { + dmas = <&audma0 0x2D>, <&audma1 0x2E>; + dma-names = "rx", "tx"; + }; + ssiu40: ssiu-32 { + dmas = <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx"; + }; + ssiu41: ssiu-33 { + dmas = <&audma0 0x17>, <&audma1 0x18>; + dma-names = "rx", "tx"; + }; + ssiu42: ssiu-34 { + dmas = <&audma0 0x19>, <&audma1 0x1A>; + dma-names = "rx", "tx"; + }; + ssiu43: ssiu-35 { + dmas = <&audma0 0x1B>, <&audma1 0x1C>; + dma-names = "rx", "tx"; + }; + ssiu44: ssiu-36 { + dmas = <&audma0 0x1D>, <&audma1 0x1E>; + dma-names = "rx", "tx"; + }; + ssiu45: ssiu-37 { + dmas = <&audma0 0x1F>, <&audma1 0x20>; + dma-names = "rx", "tx"; + }; + ssiu46: ssiu-38 { + dmas = <&audma0 0x31>, <&audma1 0x32>; + dma-names = "rx", "tx"; + }; + ssiu47: ssiu-39 { + dmas = <&audma0 0x33>, <&audma1 0x34>; + dma-names = "rx", "tx"; + }; + ssiu50: ssiu-40 { + dmas = <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx"; + }; + ssiu60: ssiu-41 { + dmas = <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx"; + }; + ssiu70: ssiu-42 { + dmas = <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx"; + }; + ssiu80: ssiu-43 { + dmas = <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx"; + }; + ssiu90: ssiu-44 { + dmas = <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx"; + }; + ssiu91: ssiu-45 { + dmas = <&audma0 0x7F>, <&audma1 0x80>; + dma-names = "rx", "tx"; + }; + ssiu92: ssiu-46 { + dmas = <&audma0 0x81>, <&audma1 0x82>; + dma-names = "rx", "tx"; + }; + ssiu93: ssiu-47 { + dmas = <&audma0 0x83>, <&audma1 0x84>; + dma-names = "rx", "tx"; + }; + ssiu94: ssiu-48 { + dmas = <&audma0 0xA3>, <&audma1 0xA4>; + dma-names = "rx", "tx"; + }; + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 25 Feb '19

25 Feb '19

From: Shengjiu Wang <shengjiu.wang(a)freescale.com> There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the begginning with the normal enable flow. This is hardware issue, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 32 ++++++++++++++++++++++---------- 1 file changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..23bd0ad4ac31 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -54,6 +54,8 @@ struct fsl_esai { u32 fifo_depth; u32 slot_width; u32 slots; + u32 tx_mask; + u32 rx_mask; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -361,21 +363,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); - esai_priv->slot_width = slot_width; esai_priv->slots = slots; + esai_priv->tx_mask = tx_mask; + esai_priv->rx_mask = rx_mask; return 0; } @@ -596,6 +590,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 mask; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -611,12 +606,23 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); + mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, 0); /* Disable and reset FIFO */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), @@ -906,6 +912,12 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + /* Clear the TSMA, TSMB, RSMA, RSMB */ + regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { -- 1.9.1

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[alsa-devel] simple-audio-card,mclk-fs
by Daniel Baluta 25 Feb '19

25 Feb '19

Hello Morimoto-san, Do you have any plans on adding support for multiple mclk-fs values in simple-card? Usually one constant mclk-fs value is not enough because usually codecs have different value depending on rate. For example, for wm8524 [1] one can use mclk-fs = 256 for rates between 8kHz and 96kHz but it needs to change mclk-fs = 128 for 176.4hHz and 192kHz. I am thinking of something like this: --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -44,6 +44,8 @@ compatible = "simple-audio-card"; simple-audio-card,name = "wm8524-audio"; simple-audio-card,format = "i2s"; + simple-audio-card,lrclk-fs = <8000>, <32000>, <44100> <48000>, <96000>, <176400>, <192000>; + simple-audio-card,mclk-fs = <256>, <256>, <256>, <256>, <256>, <128>, <128>; Let me know if this would be a good direction and I will send a patch. thanks, Daniel. [1] https://www.cirrus.com/products/wm8524/

2 1

[alsa-devel] [PATCH] ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
by twischer＠de.adit-jv.com 25 Feb '19

25 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif mode/adinr/dalign address is registered by: (in busif4 case) RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_ADINR,0x504, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80) But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif mode/adinr/dalign register address ( SSI9-[4/5/6/7]_BUSIF_[MODE/ADINR/DALIGN] ) are out of this rule. This patch registers ssi9 4/5/6/7 mode/adinr/dalign register as single register, and access these these registers in case of SSI9 BUSIF 4/5/6/7. Fixes: commit 8c9d75033340 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/soc/sh/rcar/gen.c | 24 ++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 27 +++++++++++++++++++++++++++ sound/soc/sh/rcar/ssiu.c | 36 ++++++++++++++++-------------------- 3 files changed, 67 insertions(+), 20 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 7cda601..af19010 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -255,6 +255,30 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), + RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c), + RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484), + RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488), + RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0), + RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4), + RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8), + RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0), + RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4), + RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8), + RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0), + RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4), + RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8), + RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80), + RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84), + RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88), + RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0), + RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4), + RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8), + RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0), + RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4), + RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8), + RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0), + RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4), + RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8), }; static const struct rsnd_regmap_field_conf conf_scu[] = { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 605e4b9..90625c5 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -191,6 +191,30 @@ enum rsnd_reg { SSI_SYS_STATUS7, HDMI0_SEL, HDMI1_SEL, + SSI9_BUSIF0_MODE, + SSI9_BUSIF1_MODE, + SSI9_BUSIF2_MODE, + SSI9_BUSIF3_MODE, + SSI9_BUSIF4_MODE, + SSI9_BUSIF5_MODE, + SSI9_BUSIF6_MODE, + SSI9_BUSIF7_MODE, + SSI9_BUSIF0_ADINR, + SSI9_BUSIF1_ADINR, + SSI9_BUSIF2_ADINR, + SSI9_BUSIF3_ADINR, + SSI9_BUSIF4_ADINR, + SSI9_BUSIF5_ADINR, + SSI9_BUSIF6_ADINR, + SSI9_BUSIF7_ADINR, + SSI9_BUSIF0_DALIGN, + SSI9_BUSIF1_DALIGN, + SSI9_BUSIF2_DALIGN, + SSI9_BUSIF3_DALIGN, + SSI9_BUSIF4_DALIGN, + SSI9_BUSIF5_DALIGN, + SSI9_BUSIF6_DALIGN, + SSI9_BUSIF7_DALIGN, /* SSI */ SSICR, @@ -209,6 +233,9 @@ enum rsnd_reg { #define SSI_BUSIF_MODE(i) (SSI_BUSIF0_MODE + (i)) #define SSI_BUSIF_ADINR(i) (SSI_BUSIF0_ADINR + (i)) #define SSI_BUSIF_DALIGN(i) (SSI_BUSIF0_DALIGN + (i)) +#define SSI9_BUSIF_MODE(i) (SSI9_BUSIF0_MODE + (i)) +#define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) +#define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index c74991d..c036654 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -181,29 +181,25 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (rsnd_ssi_use_busif(io)) { int id = rsnd_mod_id(mod); int busif = rsnd_mod_id_sub(mod); + int chnl = rsnd_io_is_play(io) ? + rsnd_runtime_channel_after_ctu(io) : + rsnd_runtime_channel_original(io); - /* - * FIXME - * - * We can't support SSI9-4/5/6/7, because its address is - * out of calculation rule - */ if ((id == 9) && (busif >= 4)) { - struct device *dev = rsnd_priv_to_dev(priv); - - dev_err(dev, "This driver doesn't support SSI%d-%d, so far", - id, busif); + rsnd_mod_write(mod, SSI9_BUSIF_ADINR(busif), + rsnd_get_adinr_bit(mod, io) | chnl); + rsnd_mod_write(mod, SSI9_BUSIF_MODE(busif), + rsnd_get_busif_shift(io, mod) | 1); + rsnd_mod_write(mod, SSI9_BUSIF_DALIGN(busif), + rsnd_get_dalign(mod, io)); + } else { + rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), + rsnd_get_adinr_bit(mod, io) | chnl); + rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), + rsnd_get_busif_shift(io, mod) | 1); + rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), + rsnd_get_dalign(mod, io)); } - - rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), - rsnd_get_adinr_bit(mod, io) | - (rsnd_io_is_play(io) ? - rsnd_runtime_channel_after_ctu(io) : - rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), - rsnd_get_busif_shift(io, mod) | 1); - rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), - rsnd_get_dalign(mod, io)); } if (has_hdmi0 || has_hdmi1) { -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsl_ssi: remove unsupported formats in i2s master mode
by Sven Van Asbroeck 23 Feb '19

23 Feb '19

Quote from the ssi manual: "The word length is fixed to 32 in I2S Master mode and the WL bits determine the number of bits that will contain valid data (out of the 32 transmitted/received bits in each channel)." (61.8.1.4, page 5138, IMX6DQRM Rev. 3, 07/2015) In other words, in i2s master mode, we always get 32 physical bits/channel, no matter what. However, the supported list of formats for this ssi contains S8, S16_XX, S18_3XX, and S20_3XX, which have 8, 16, 24 and 24 physical bits/channel respectively. If one of these is selected in i2s master mode, the format generated by the ssi will be incorrect. Fix by constraining the frame bits in i2s master mode to 32 * channels. This will filter out those formats that the ssi cannot support. Tested on imx6 with tda998x and sgtl5k codecs. Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0a648229e643..3d375f60ecf4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -627,6 +627,28 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi *ssi) regmap_write(regs, REG_SSI_SOR, SSI_SOR_WAIT(3)); } +static int fsl_ssi_hw_rule_i2s_master(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *it = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_FRAME_BITS); + struct fsl_ssi *ssi = rule->private; + struct snd_interval t; + + if (!fsl_ssi_is_i2s_master(ssi)) + return 0; + + /* + * In i2s master mode, the ssi always generates 32 physical + * bits/channel. Filter out formats that don't have 32 physical + * bits/channel, they are unsupported. + */ + memset(&t, 0, sizeof(t)); + t.min = t.max = params_channels(params) * 32; + + return snd_interval_refine(it, &t); +} + static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -648,6 +670,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2); + snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + fsl_ssi_hw_rule_i2s_master, ssi, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); + return 0; } -- 2.17.1

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[alsa-devel] [PATCH 0/2] ASoC: adau1977: Add support for micbias in DT
by Bogdan Togorean 22 Feb '19

22 Feb '19

Currently micbias voltage is set based on platform_data. If driver is instantiated using DT only default voltage for micbias is possible. This patch adds the possibility to specify an optional DT property to configure the output voltage for micbias. Bogdan Togorean (2): ASoC: adau1977: Add MICBIAS example in DT bindings ASoC: adau1977: Add support for setting MICBIAS via DT .../devicetree/bindings/sound/adi,adau1977.txt | 7 +++++++ sound/soc/codecs/adau1977.c | 12 +++++++----- 2 files changed, 14 insertions(+), 5 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] printk: Remove no longer used LOG_PREFIX.
by Tetsuo Handa 22 Feb '19

22 Feb '19

When commit 5becfb1df5ac8e49 ("kmsg: merge continuation records while printing") introduced LOG_PREFIX, we used KERN_DEFAULT etc. as a flag for setting LOG_PREFIX in order to tell whether to call cont_add() (i.e. whether to append the message to "struct cont"). But since commit 4bcc595ccd80decb ("printk: reinstate KERN_CONT for printing continuation lines") inverted the behavior (i.e. don't append the message to "struct cont" unless KERN_CONT is specified) and commit 5aa068ea4082b39e ("printk: remove games with previous record flags") removed the last LOG_PREFIX check, setting LOG_PREFIX via KERN_DEFAULT etc. is no longer meaningful. Therefore, we can remove LOG_PREFIX and make KERN_DEFAULT empty string. Signed-off-by: Tetsuo Handa <penguin-kernel(a)I-love.SAKURA.ne.jp> --- include/linux/kern_levels.h | 2 +- include/linux/printk.h | 1 - kernel/printk/printk.c | 6 +----- 3 files changed, 2 insertions(+), 7 deletions(-) diff --git a/include/linux/kern_levels.h b/include/linux/kern_levels.h index d237fe8..bf2389c 100644 --- a/include/linux/kern_levels.h +++ b/include/linux/kern_levels.h @@ -14,7 +14,7 @@ #define KERN_INFO KERN_SOH "6" /* informational */ #define KERN_DEBUG KERN_SOH "7" /* debug-level messages */ -#define KERN_DEFAULT KERN_SOH "d" /* the default kernel loglevel */ +#define KERN_DEFAULT "" /* the default kernel loglevel */ /* * Annotation for a "continued" line of log printout (only done after a diff --git a/include/linux/printk.h b/include/linux/printk.h index 77740a5..9fe6c9e 100644 --- a/include/linux/printk.h +++ b/include/linux/printk.h @@ -18,7 +18,6 @@ static inline int printk_get_level(const char *buffer) if (buffer[0] == KERN_SOH_ASCII && buffer[1]) { switch (buffer[1]) { case '0' ... '7': - case 'd': /* KERN_DEFAULT */ case 'c': /* KERN_CONT */ return buffer[1]; } diff --git a/kernel/printk/printk.c b/kernel/printk/printk.c index d3d1703..3e4a735 100644 --- a/kernel/printk/printk.c +++ b/kernel/printk/printk.c @@ -344,7 +344,6 @@ enum con_msg_format_flags { enum log_flags { LOG_NEWLINE = 2, /* text ended with a newline */ - LOG_PREFIX = 4, /* text started with a prefix */ LOG_CONT = 8, /* text is a fragment of a continuation line */ }; @@ -1867,9 +1866,6 @@ int vprintk_store(int facility, int level, case '0' ... '7': if (level == LOGLEVEL_DEFAULT) level = kern_level - '0'; - /* fallthrough */ - case 'd': /* KERN_DEFAULT */ - lflags |= LOG_PREFIX; break; case 'c': /* KERN_CONT */ lflags |= LOG_CONT; @@ -1884,7 +1880,7 @@ int vprintk_store(int facility, int level, level = default_message_loglevel; if (dict) - lflags |= LOG_PREFIX|LOG_NEWLINE; + lflags |= LOG_NEWLINE; return log_output(facility, level, lflags, dict, dictlen, text, text_len); -- 1.8.3.1

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[alsa-devel] [PATCH 1/3] ALSA: hda/tegra: property for card name
by Sameer Pujar 22 Feb '19

22 Feb '19

An optional property "nvidia,model" is introduced for hda to pass custom name for the sound card. The suffix "-hda" in the name passed is useful to distinguish between multiple cards available for a platform. When the property is not specified, default name("tegra-hda") mentioned in hda driver is used. This property can be added in platform specific file of the board and card name can relate to the board in use. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Jonathan Hunter <jonathanh(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt index 44d2745..21cd310 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -13,6 +13,10 @@ Required properties: See ../reset/reset.txt for details. - reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x +Optional properties: +- nvidia,model : The user-visible name of this sound complex. Since the property + is optional, legacy boards can use default name provided in hda driver. + Example: hda@70030000 { @@ -27,4 +31,5 @@ hda@70030000 { <&tegra_car 128>, /* hda2hdmi */ <&tegra_car 111>; /* hda2codec_2x */ reset-names = "hda", "hda2hdmi", "hda2codec_2x"; + nvidia,model = "jetson-tk1-hda"; }; -- 2.7.4

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[alsa-devel] doc pages don't work
by Jarek 22 Feb '19

22 Feb '19

Hi, https://www.alsa-project.org/alsa-doc/alsa-lib/ return 404 Jarek

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[alsa-devel] Playback samples at a specific time on Intel HDA
by Rajagopal Aravindan 21 Feb '19

21 Feb '19

Hello, I am trying to accomplish synchronized audio playback using ALSA & gPTP, on an Intel SkyLake platform have Intel HDA. I do understand that, starting with SkyLake platform, there is support for querying link time. But, how do I instruct to play the samples at a given (gPTP) time ? Is this possible ? Can someone please point me to code or doc, relevant to my goal ? In essence, I would like to play the samples at the stroke of a given (gPTP) time. Also, the ALSA page says ... "The link time can be directly measured if supported in hardware by sample counters or wallclocks (e.g. with HDAudio 24MHz or PTP clock for networked solutions)" What is the connection between PTP clock and sound card here, are there sound cards that can latch on to a PTP clock on a NIC or what ? Thanks in advance. Regards, Rajagopal

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