June 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] CA0110 analog mic capture flatline
by Ki'Sak 11 Jul '13

11 Jul '13

Good day alsa-devel, I have a "Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG" which I have been unable to capture any content from the analog mic jack. I am a Gentoo Linux user with the 3.8.13 kernel / alsa 1.0.26. This issue has been filed with the wrong devs over at ubuntu's bugtracker and opensuse's forum. Also, I found a report that analog capture was working on this card as recently as September 2009. Thank you for your time, Ki'Sak Note: I have already tried to report this issue, this is a resend without urls due to 2 weeks without a response or it reaching the mailing list.

2 3

[alsa-devel] [PATCH - oss plugin update revised 1/2] oss: Add support of 32-bit and other formats
by alex.d.wiggins＠gmail.com 08 Jul '13

08 Jul '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added support of 32-bit and other audio formats via 3 changes: - Additional audio format definitions at the top of the file (if required); - Additional audio formats in oss_hw_params (to allow as input to pass to OSS); - Additional audio formats in oss_hw_constraint (to allow to report to ALSA). Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d43b44c..d5b5bc4 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -24,6 +24,32 @@ #include <alsa/pcm_external.h> #include <linux/soundcard.h> +/* definitions for additional OSS formats if required */ +#ifndef AFMT_S32_LE +#define AFMT_S32_LE 0x00001000 +#endif +#ifndef AFMT_S32_BE +#define AFMT_S32_BE 0x00002000 +#endif +#ifndef AFMT_S24_LE +#define AFMT_S24_LE 0x00008000 +#endif +#ifndef AFMT_S24_BE +#define AFMT_S24_BE 0x00010000 +#endif +#ifndef AFMT_S24_PACKED +#define AFMT_S24_PACKED 0x00040000 +#endif +#ifndef AFMT_FLOAT +#define AFMT_FLOAT 0x00004000 +#endif +#ifndef AFMT_SPDIF_RAW +#define AFMT_SPDIF_RAW 0x00020000 +#endif +#ifndef AFMT_VORBIS +#define AFMT_VORBIS 0x00000800 +#endif + typedef struct snd_pcm_oss { snd_pcm_ioplug_t io; char *device; @@ -165,6 +191,145 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, case SND_PCM_FORMAT_S16_BE: oss->format = AFMT_S16_BE; break; + /* allow additional data formats (such as 32 bit) as input + to pass to OSS (set in OSS driver) */ + case SND_PCM_FORMAT_S32_LE: + oss->format = AFMT_S32_LE; + break; + case SND_PCM_FORMAT_S32_BE: + oss->format = AFMT_S32_BE; + break; + case SND_PCM_FORMAT_S8: + oss->format = AFMT_S8; + break; + case SND_PCM_FORMAT_U16_LE: + oss->format = AFMT_U16_LE; + break; + case SND_PCM_FORMAT_U16_BE: + oss->format = AFMT_U16_BE; + break; + case SND_PCM_FORMAT_S24_LE: + oss->format = AFMT_S24_LE; + break; + case SND_PCM_FORMAT_S24_BE: + oss->format = AFMT_S24_BE; + break; + case SND_PCM_FORMAT_MU_LAW: + oss->format = AFMT_MU_LAW; + break; + case SND_PCM_FORMAT_A_LAW: + oss->format = AFMT_A_LAW; + break; + case SND_PCM_FORMAT_IMA_ADPCM: + oss->format = AFMT_IMA_ADPCM; + break; + case SND_PCM_FORMAT_MPEG: + oss->format = AFMT_MPEG; + break; + case SND_PCM_FORMAT_S24_3LE: + oss->format = AFMT_S24_PACKED; + break; + case SND_PCM_FORMAT_FLOAT: + oss->format = AFMT_FLOAT; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME: + oss->format = AFMT_SPDIF_RAW; + break; + /* following formats need defining checking native endian: */ + #if 0 + case SND_PCM_FORMAT_S16: + oss->format = AFMT_S16_NE; + break; + case SND_PCM_FORMAT_U16: + oss->format = AFMT_U16_NE; + break; + case SND_PCM_FORMAT_S24: + oss->format = AFMT_S24_NE; + break; + case SND_PCM_FORMAT_S32: + oss->format = AFMT_S32_NE; + #endif + /* following formats found in ALSA but not in OSS: */ + #if 0 + case SND_PCM_FORMAT_U24_LE: + oss->format = AFMT_U24_LE; + break; + case SND_PCM_FORMAT_U24_BE: + oss->format = AFMT_U24_BE; + break; + case SND_PCM_FORMAT_U32_LE: + oss->format = AFMT_U32_LE; + break; + case SND_PCM_FORMAT_U32_BE: + oss->format = AFMT_U32_BE; + break; + case SND_PCM_FORMAT_FLOAT_LE: + oss->format = AFMT_FLOAT_LE; + break; + case SND_PCM_FORMAT_FLOAT_BE: + oss->format = AFMT_FLOAT_BE; + break; + case SND_PCM_FORMAT_FLOAT64_LE: + oss->format = AFMT_FLOAT64_LE; + break; + case SND_PCM_FORMAT_FLOAT64_BE: + oss->format = AFMT_FLOAT64_BE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: + oss->format = AFMT_SPDIF_RAW_LE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: + oss->format = AFMT_SPDIF_RAW_BE; + break; + case SND_PCM_FORMAT_GSM: + oss->format = AFMT_GSM; + break; + case SND_PCM_FORMAT_SPECIAL: + oss->format = AFMT_SPECIAL; + break; + case SND_PCM_FORMAT_S24_3BE: + oss->format = AFMT_S24_PACKED_BE; + break; + case SND_PCM_FORMAT_U24_3LE: + oss->format = AFMT_U24_PACKED; + break; + case SND_PCM_FORMAT_U24_3BE: + oss->format = AFMT_U24_PACKED_BE; + break; + case SND_PCM_FORMAT_S20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U24: + oss->format = AFMT_U24_NE; + break; + case SND_PCM_FORMAT_U32: + oss->format = AFMT_U32_NE; + break; + case SND_PCM_FORMAT_FLOAT64: + oss->format = AFMT_FLOAT64; + break; + #endif default: fprintf(stderr, "*** OSS: unsupported format %s\n", snd_pcm_format_name(io->format)); return -EINVAL; @@ -239,7 +404,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) SND_PCM_ACCESS_MMAP_INTERLEAVED }; unsigned int nformats; - unsigned int format[5]; + unsigned int format[20]; unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ @@ -273,6 +438,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) format[nformats++] = SND_PCM_FORMAT_S16_BE; if (tmp & AFMT_MU_LAW) format[nformats++] = SND_PCM_FORMAT_MU_LAW; + /* allow additional data formats (such as 32 bit) + to be reported from OSS to ALSA below */ + if (tmp & AFMT_S32_LE) + format[nformats++] = SND_PCM_FORMAT_S32_LE; + if (tmp & AFMT_S32_BE) + format[nformats++] = SND_PCM_FORMAT_S32_BE; + if (tmp & AFMT_A_LAW) + format[nformats++] = SND_PCM_FORMAT_A_LAW; + if (tmp & AFMT_IMA_ADPCM) + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; + if (tmp & AFMT_S8) + format[nformats++] = SND_PCM_FORMAT_S8; + if (tmp & AFMT_U16_LE) + format[nformats++] = SND_PCM_FORMAT_U16_LE; + if (tmp & AFMT_U16_BE) + format[nformats++] = SND_PCM_FORMAT_U16_BE ; + if (tmp & AFMT_MPEG) + format[nformats++] = SND_PCM_FORMAT_MPEG; + if (tmp & AFMT_FLOAT) + format[nformats++] = SND_PCM_FORMAT_FLOAT; + if (tmp & AFMT_S24_LE) + format[nformats++] = SND_PCM_FORMAT_S24_LE; + if (tmp & AFMT_S24_BE) + format[nformats++] = SND_PCM_FORMAT_S24_BE; + if (tmp & AFMT_SPDIF_RAW) + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; + if (tmp & AFMT_S24_PACKED) + format[nformats++] = SND_PCM_FORMAT_S24_3LE; + /* following formats found in OSS but not in ALSA */ + if (tmp & AFMT_AC3) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + if (tmp & AFMT_VORBIS) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + /* assume signed 16 bit (native CPU endian) if not found above */ if (! nformats) format[nformats++] = SND_PCM_FORMAT_S16; if ((err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, -- 1.8.3.1

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[alsa-devel] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 08 Jul '13

08 Jul '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Version 9 contains some style fixes. Regards, Markus Changes in v9: - Style and compile fixes. - Removed "ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97". This patch is not necessary with the seperated phycore-ac97-dt driver. Changes in v8: - Fix build issues in phycore-ac97. Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: Add phycore-ac97-dt driver ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 451 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97-dt.c | 242 +++++++++++ sound/soc/fsl/phycore-ac97.c | 13 +- 15 files changed, 704 insertions(+), 115 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

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[alsa-devel] [RFC] [patch 1/1] imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found
by Arnaud Patard 04 Jul '13

04 Jul '13

If the ssi or codec drivers are not loaded (for instance, because spi or i2c bus drivers are not loaded), returning -EINVAL will for people to unload and then reload the module to get sound working. Returning E_PROBE_DEFER will mitigate this. Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: linux/sound/soc/fsl/imx-sgtl5000.c =================================================================== --- linux.orig/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:33:01.000000000 +0200 +++ linux/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:35:14.000000000 +0200 @@ -113,13 +113,13 @@ static int imx_sgtl5000_probe(struct pla ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + return -EPROBE_DEFER; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);

3 4

[alsa-devel] [PATCH] Add -lm to test programs to fix build failure.
by Jordi Mallach 04 Jul '13

04 Jul '13

Add -lm to test programs that use sin(), to fix a build failure with recent binutils. Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- test/Makefile.am | 2 ++ 1 file changed, 2 insertions(+) diff --git a/test/Makefile.am b/test/Makefile.am index 8705402..c1896f8 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -7,8 +7,10 @@ check_PROGRAMS=control pcm pcm_min latency seq \ control_LDADD=../src/libasound.la pcm_LDADD=../src/libasound.la +pcm_LDFLAGS= -lm pcm_min_LDADD=../src/libasound.la latency_LDADD=../src/libasound.la +latency_LDFLAGS= -lm seq_LDADD=../src/libasound.la playmidi1_LDADD=../src/libasound.la timer_LDADD=../src/libasound.la -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ASoC: Add PCM1681 codec driver.
by Marek Belisko 01 Jul '13

01 Jul '13

PCM1681 can be controlled via I2C, SPI or in bootstrap mode. This code add support only for I2C mode. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 328 ++++++++++++++++++++ 4 files changed, 349 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 sound/soc/codecs/pcm1681.c diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 0000000..4df1718 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fb..e2daf82 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd806..4a068d2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 0000000..6122fd1 --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,328 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko <marek.belisko(a)streamunlimited.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, ret; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + /* enable choosen frequency */ + if (val != -1) + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + + /* enable/disable deemphasis functionality */ + ret = regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, priv->deemph); + + return ret; +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int ret, val = 0; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + + ret = regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + + return 0; +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("PCM1681 Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("PCM1681 De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults), + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static int pcm1681_probe(struct snd_soc_codec *codec) +{ + return 0; +} + +static int pcm1681_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .probe = pcm1681_probe, + .remove = pcm1681_remove, + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko <marek.belisko(a)streamunlimited.com>"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

4 4

[alsa-devel] [PATCH 1/2] ASoC: adau1701: more direct regmap usage
by Daniel Mack 30 Jun '13

30 Jun '13

Replace calls to snd_soc_update_bits() with regmap_update_bits(). Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/adau1701.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ac8b2c4..ce56065 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -334,7 +334,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; } - snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, mask, val); return 0; } @@ -362,7 +362,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_SERICTL, + regmap_update_bits(adau1701->regmap, ADAU1701_SERICTL, ADAU1701_SERICTL_MODE_MASK, val); return 0; @@ -403,7 +403,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val); format = params_format(params); @@ -489,6 +489,7 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: @@ -497,11 +498,13 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Enable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, 0x00); break; case SND_SOC_BIAS_OFF: /* Disable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, mask); break; } @@ -513,6 +516,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; unsigned int mask = ADAU1701_DSPCTRL_DAM; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int val; if (mute) @@ -520,7 +524,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) else val = mask; - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, mask, val); return 0; } @@ -542,7 +546,8 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + regmap_update_bits(adau1701->regmap, ADAU1701_OSCIPOW, + ADAU1701_OSCIPOW_OPD, val); adau1701->sysclk = freq; return 0; -- 1.8.1.4

3 4

[alsa-devel] [PATCH - oss plugin update revised 2/2] oss: Add user variables to set period and buffer config
by alex.d.wiggins＠gmail.com 30 Jun '13

30 Jun '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added 3 user variables to control the OSS period and buffer config: - period_size (integer): if not set then queries OSS for period size (by using SNDCTL_DSP_GETBLKSIZE), if positive then sets the OSS period size (SND_PCM_IOPLUG_HW_PERIOD_BYTES), if negative then allows any power of 2 period size. - periods (integer): sets the number of OSS periods (SND_PCM_IOPLUG_HW_PERIODS); - buffer_size (integer): sets the OSS buffer size (SND_PCM_IOPLUG_HW_BUFFER_BYTES). Note that ALSA performs sanity checks to prevent conflicts. Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d5b5bc4..a92c3e9 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -60,6 +60,12 @@ typedef struct snd_pcm_oss { unsigned int period_shift; unsigned int periods; unsigned int frame_bytes; + /* user variables set using ALSA config files */ + /* included here as this snd_pcm_oss_t struct + is stored in io->private_data */ + long int user_period_size; + long int user_periods; + long int user_buffer_size; } snd_pcm_oss_t; static snd_pcm_sframes_t oss_write(snd_pcm_ioplug_t *io, @@ -408,6 +414,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ + /* minimum: 2^8 = 256; maximum: 2^23 = 8388608 */ static const unsigned int bytes_list[] = { 1U<<8, 1U<<9, 1U<<10, 1U<<11, 1U<<12, 1U<<13, 1U<<14, 1U<<15, 1U<<16, 1U<<17, 1U<<18, 1U<<19, 1U<<20, 1U<<21, 1U<<22, 1U<<23 @@ -500,20 +507,97 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) if (err < 0) return err; - /* period size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; - /* periods */ - err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); - if (err < 0) - return err; - /* buffer size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; + /* period and buffer sizes: default behaviour below is to + query period size from OSS */ + + /* consider user set paramaters + (period_size; periods; buffer_size) */ + + /* note: ALSA performs sanity checks + to prevent conflicting parameters */ + + /* period_size */ + if (oss->user_period_size < 0) { + /* period_size set to negative value: + allow any power of 2 period size */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } else { + if (oss->user_period_size > 0) { + /* period_size set to positive value: + use user set period size */ + const unsigned int user_period_size + = oss->user_period_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, 1, + &user_period_size); + } else { + /* query OSS for period size (called fragment size) */ + unsigned int fragsize; + if (ioctl(oss->fd, SNDCTL_DSP_GETBLKSIZE, &fragsize) + < 0) + fprintf(stderr, +"*** alsa-plugins-oss: error querying SNDCTL_DSP_GETBLKSIZE.\n"); + + /* note: second query of SNDCTL_DSP_GETBLKSIZE + can give different result + (i.e. 4096 instead of 2048) */ + /* in oss_audio_core.c getblksize function, + 4096 is the default return value: + so second time probably not reliable */ + + /* period size (in power of two) */ + unsigned int x = fragsize; + /* check if fragsize is a power of two */ + if ((x != 0) && ((x & (~x + 1)) == x)) + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + 1, &fragsize); + else + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } + } + + /* periods (number of periods per buffer) */ + if (oss->user_periods > 0) { + /* force number of periods to user set periods */ + const unsigned int user_periods = oss->user_periods; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIODS, 1, &user_periods); + if (err < 0) + return err; + } else { + /* allow any number of periods between 2 and 1024 inclusive + (ALSA can choose) */ + err = snd_pcm_ioplug_set_param_minmax(io, + SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); + if (err < 0) + return err; + } + + /* buffer_size */ + if (oss->user_buffer_size > 0) { + /* force buffer size to user set buffer_size */ + const unsigned int user_buffer_size = oss->user_buffer_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, 1, &user_buffer_size); + if (err < 0) + return err; + } else { + /* buffer size (in power of two) */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } return 0; } @@ -558,6 +642,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) const char *device = "/dev/dsp"; int err; snd_pcm_oss_t *oss; + /* default values for user variables*/ + long int user_period_size = 0; + long int user_periods = 0; + long int user_buffer_size = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -573,6 +661,30 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) } continue; } + + /* user variables */ + if (strcmp(id, "period_size") == 0) { + if (snd_config_get_integer(n, &user_period_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "periods") == 0) { + if (snd_config_get_integer(n, &user_periods) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "buffer_size") == 0) { + if (snd_config_get_integer(n, &user_buffer_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + SNDERR("Unknown field %s", id); return -EINVAL; } @@ -597,6 +709,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) goto error; } + /* save user variables from ALSA config files */ + oss->user_period_size = user_period_size; + oss->user_periods = user_periods; + oss->user_buffer_size = user_buffer_size; + oss->io.version = SND_PCM_IOPLUG_VERSION; oss->io.name = "ALSA <-> OSS PCM I/O Plugin"; oss->io.poll_fd = oss->fd; -- 1.8.3.1

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Re: [alsa-devel] [PATCH 0/2] ASoc: ac97: fix multi-platform build failures
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 11:17:47AM -0700, Kevin Hilman wrote: > Hi Mark, > > When building next-20130628, I noticed a few build errors for > cm_x300_defconfig that were introduced by the new multi-platform AC'97 > support. Applied both, thanks.

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: tegra20-ac97: Remove duplicate error message
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 09:53:24PM +0200, Thierry Reding wrote: > devm_ioremap_resource() already outputs an error message when any of the > operations it performs fails, so the duplicate in the caller can be > removed. Applied both, thanks.

1 0