October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Delete ads117x.h
by Axel Lin 12 Oct '11

12 Oct '11

This is not required after multi-component patch. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ads117x.h | 13 ------------- 1 files changed, 0 insertions(+), 13 deletions(-) delete mode 100644 sound/soc/codecs/ads117x.h diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h deleted file mode 100644 index 3ce0286..0000000 --- a/sound/soc/codecs/ads117x.h +++ /dev/null @@ -1,13 +0,0 @@ -/* - * ads117x.h -- Driver for ads1174/8 ADC chips - * - * Copyright 2009 ShotSpotter Inc. - * Author: Graeme Gregory <gg(a)slimlogic.co.uk> - * - * 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. - */ -extern struct snd_soc_dai_driver ads117x_dai; -extern struct snd_soc_codec_driver soc_codec_dev_ads117x; -- 1.7.4.1

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[alsa-devel] Driver for daisy-chained AD1938 audio codecs?
by Steve Strobel 12 Oct '11

12 Oct '11

I am considering a Blackfin DSP hardware design which would need eight channels of audio input/output. I haven't been able to find a single audio codec with more than six input channels, so I am looking at daisy-chaining AD1938 codecs as described on page 19 of its datasheet [1]. In the past I worked on a prototype that used two AD1938 chips on separate SPORT interfaces. It worked, but having to deal with two ALSA sound devices made the software relatively complicated (error recovery, synchronization, etc). It would be nice if all of the channels appeared as a single ALSA device. I believe that daisy-chained AD1938 chips would work that way, but I haven't found an ALSA driver for that configuration. Does such a driver already exist? Any suggestions for making one? Thanks for any input. Steve [1] - http://www.analog.com/static/imported-files/data_sheets/AD1938.pdf -- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

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[alsa-devel] [PATCH v2 0/6] ASoC/MFD/Input: twl6040: Support for audio driven vibra
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Hello, Changes since v1: - Patches already taken by Mark has been removed - Patch 4: Return -EBUSY instead of -EPERM Intro mail form v1: The following series adds support for the soc driver to route audio (PCM stream) to the vibra ports of the twl6040. We have ALSA controls for the vibra source selection (PDM, or Input/FF). The route can not be changed from Input/FF to PDM during ongoing effect coming from Input/FF. The Input/FF driver can not execute new effect, while the vibra source is configured to be PDM. TO get things working in legacy mode the omap-mcpdm, and twl6040 soc codec driver's channel count needed to be fixed. To minimize the need for reads through the I2C bus, the MFD driver will cache the vibra control registers. With this series we can route PCM audio to drive the twl6040 vibra output. In order to that we need to play 5 channel audio, and place the vibra sample at the 5th channel. This will change with ABE support (change == simplified). The series has been created on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch If there are no objections, it would be good if this series goes via audio. Regards, Peter --- Peter Ujfalusi (6): Input: twl6040: Simplify vibra regsiter definitions MFD: twl6040: Cache the vibra control registers MFD: twl6040: function to query the vibra status for clients Input: twl6040-vibra: Check the selected path for vibra ASoC: omap-mcpdm: Correct the supported number of channels ASoC: twl6040: Support for vibra output paths drivers/input/misc/twl6040-vibra.c | 19 ++++++--- drivers/mfd/twl6040-core.c | 31 +++++++++++++-- include/linux/mfd/twl6040.h | 24 +++++------ sound/soc/codecs/twl6040.c | 72 ++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.c | 14 ++++--- 5 files changed, 131 insertions(+), 29 deletions(-) -- 1.7.6.1

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[alsa-devel] [PATCH] ASoC: tlv320aic23: convert to soc-cache
by Axel Lin 12 Oct '11

12 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, This patch converts tlv320aic23 to use soc-cache APIs. I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/tlv320aic23.c | 169 +++++++++++++--------------------------- 1 files changed, 54 insertions(+), 115 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3a4bb2..c9d85db 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -47,63 +47,6 @@ static const u16 tlv320aic23_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ }; -/* - * read tlv320aic23 register cache - */ -static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec - *codec, unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return -1; - return cache[reg]; -} - -/* - * write tlv320aic23 register cache - */ -static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u16 value) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return; - cache[reg] = value; -} - -/* - * write to the tlv320aic23 register space - */ -static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - - u8 data[2]; - - /* TLV320AIC23 has 7 bit address and 9 bits of data - * so we need to switch one data bit into reg and rest - * of data into val - */ - - if (reg > 9 && reg != 15) { - printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); - return -1; - } - - data[0] = (reg << 1) | (value >> 8 & 0x01); - data[1] = value & 0xff; - - tlv320aic23_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - - printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, - value, reg); - - return -EIO; -} - static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -139,8 +82,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, */ val = (val >= 4) ? 4 : (3 - val); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); - tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); + snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } @@ -151,7 +94,7 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val; - val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; @@ -232,7 +175,6 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { enum snd_soc_control_type control_type; - void *control_data; int mclk; int requested_adc; int requested_dac; @@ -344,7 +286,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int src = snd_soc_read(codec, TLV320AIC23_SRATE); int sr = (src >> 2) & 0x0f; int val = (mclk / bosr_usb_divisor_table[src & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; @@ -368,7 +310,7 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + snd_soc_write(codec, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; @@ -407,9 +349,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - iface_reg = - tlv320aic23_read_reg_cache(codec, - TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -423,7 +364,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg |= (0x03 << 2); break; } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -435,7 +376,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; /* set active */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); return 0; } @@ -450,7 +391,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; @@ -463,14 +404,14 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; u16 reg; - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + reg = snd_soc_read(codec, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; - tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + snd_soc_write(codec, TLV320AIC23_DIGT, reg); return 0; } @@ -481,8 +422,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 iface_reg; - iface_reg = - tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -516,7 +456,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -532,26 +472,26 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ TLV320AIC23_DAC_OFF); - tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + snd_soc_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ - TLV320AIC23_CLK_OFF); + snd_soc_write(codec, TLV320AIC23_PWR, + reg | TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->dapm.bias_level = level; @@ -598,13 +538,7 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec, static int tlv320aic23_resume(struct snd_soc_codec *codec) { - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } + snd_soc_cache_sync(codec); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -613,46 +547,54 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); - int reg; + int ret; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec->control_data = aic23->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; - codec->hw_read = NULL; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* Reset codec */ - tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + snd_soc_write(codec, TLV320AIC23_RESET, 0); + + /* Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + codec->cache_only = 1; + snd_soc_write(codec, 0x0A, 0); + snd_soc_write(codec, 0x0B, 0); + snd_soc_write(codec, 0x0C, 0); + snd_soc_write(codec, 0x0D, 0); + snd_soc_write(codec, 0x0E, 0); + codec->cache_only = 0; /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); - tlv320aic23_write(codec, TLV320AIC23_LINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - (TLV320AIC23_LRS_ENABLED)); + snd_soc_update_bits(codec, TLV320AIC23_LINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); - tlv320aic23_write(codec, TLV320AIC23_RINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - TLV320AIC23_LRS_ENABLED); + snd_soc_update_bits(codec, TLV320AIC23_RINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); - tlv320aic23_write(codec, TLV320AIC23_ANLG, - (reg) & (~TLV320AIC23_BYPASS_ON) & - (~TLV320AIC23_MICM_MUTED)); + snd_soc_update_bits(codec, TLV320AIC23_ANLG, + TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, + 0); /* Default output volume */ - tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); @@ -674,8 +616,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, - .read = tlv320aic23_read_reg_cache, - .write = tlv320aic23_write, .set_bias_level = tlv320aic23_set_bias_level, .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), @@ -702,7 +642,6 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, aic23); - aic23->control_data = i2c; aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

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[alsa-devel] Realtek ALC5630 codec driver
by 卞磊 12 Oct '11

12 Oct '11

Hi ALL, I can't find the Realtek ALC5630 codec driver in current kernel. Where can i get it? Thanks. BRs, Lei.Bian

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[alsa-devel] Driver for daisy-chained AD1938 audio codecs?
by Steve Strobel 11 Oct '11

11 Oct '11

I am considering a Blackfin DSP hardware design which would need eight channels of audio input/output. I haven't been able to find a single audio codec with more than six input channels, so I am looking at daisy-chaining AD1938 codecs as described on page 19 of its datasheet [1]. In the past I worked on a prototype that used two AD1938 chips on separate SPORT interfaces. It worked, but having to deal with two ALSA sound devices made the software relatively complicated (error recovery, synchronization, etc). It would be nice if all of the channels appeared as a single ALSA device. I believe that daisy-chained AD1938 chips would work that way, but I haven't found an ALSA driver for that configuration. Does such a driver already exist? Any suggestions for making one? Thanks for any input. Steve [1] - http://www.analog.com/static/imported-files/data_sheets/AD1938.pdf -- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

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[alsa-devel] [PATCH] ASoC: Implement WM5100 accessory detection support
by Mark Brown 11 Oct '11

11 Oct '11

The WM5100 includes an advanced, low power, accessory detect subsystem capable of detecting both accessory presence and button presses while the device is in an ultra low power mode. Implement initial support for this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 162 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm5100.h | 2 + 2 files changed, 164 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 80f3fee..9ce3e57 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -26,6 +26,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/jack.h> #include <sound/initval.h> #include <sound/tlv.h> #include <sound/wm5100.h> @@ -68,6 +69,11 @@ struct wm5100_priv { bool out_ena[2]; + struct snd_soc_jack *jack; + bool jack_detecting; + bool jack_mic; + int jack_mode; + struct wm5100_fll fll[2]; struct wm5100_pdata pdata; @@ -2114,6 +2120,159 @@ static int wm5100_dig_vu[] = { WM5100_DAC_DIGITAL_VOLUME_6R, }; +static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; + + BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); + + gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); + snd_soc_update_bits(codec, WM5100_ACCESSORY_DETECT_MODE_1, + WM5100_ACCDET_BIAS_SRC_MASK | + WM5100_ACCDET_SRC, + (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | + mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + + wm5100->jack_mode = the_mode; + + dev_dbg(codec->dev, "Set microphone polarity to %d\n", + wm5100->jack_mode); +} + +static void wm5100_micd_irq(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int val; + + val = snd_soc_read(codec, WM5100_MIC_DETECT_3); + + dev_dbg(codec->dev, "Microphone event: %x\n", val); + + if (!(val & WM5100_ACCDET_VALID)) { + dev_warn(codec->dev, "Microphone detection state invalid\n"); + return; + } + + /* No accessory, reset everything and report removal */ + if (!(val & WM5100_ACCDET_STS)) { + dev_dbg(codec->dev, "Jack removal detected\n"); + wm5100->jack_mic = false; + wm5100->jack_detecting = true; + snd_soc_jack_report(wm5100->jack, 0, + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + WM5100_ACCDET_RATE_MASK); + return; + } + + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. + */ + if (val & 0x400) { + if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm5100->jack_mic = true; + snd_soc_jack_report(wm5100->jack, + SND_JACK_HEADSET, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 5 << WM5100_ACCDET_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm5100->jack, 0, SND_JACK_BTN_0); + } + + return; + } + + /* If we detected a lower impedence during initial startup + * then we probably have the wrong polarity, flip it. Don't + * do this for the lowest impedences to speed up detection of + * plain headphones. + */ + if (wm5100->jack_detecting && (val & 0x3f8)) { + wm5100_set_detect_mode(codec, !wm5100->jack_mode); + + return; + } + + /* Don't distinguish between buttons, just report any low + * impedence as BTN_0. + */ + if (val & 0x3fc) { + if (wm5100->jack_mic) { + dev_dbg(codec->dev, "Mic button detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + } else if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, + SND_JACK_HEADPHONE); + + /* Increase the detection rate a bit for + * responsiveness. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); + } + } +} + +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + + if (jack) { + wm5100->jack = jack; + wm5100->jack_detecting = true; + + wm5100_set_detect_mode(codec, 0); + + /* Slowest detection rate, gives debounce for initial + * detection */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_BIAS_STARTTIME_MASK | + WM5100_ACCDET_RATE_MASK, + (7 << WM5100_ACCDET_BIAS_STARTTIME_SHIFT) | + WM5100_ACCDET_RATE_MASK); + + /* We need the charge pump to power MICBIAS */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_sync(&codec->dapm); + + /* We start off just enabling microphone detection - even a + * plain headphone will trigger detection. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, WM5100_ACCDET_ENA); + + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_ACCDET_EINT, 0); + } else { + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT); + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, 0); + wm5100->jack = NULL; + } + + return 0; +} + static irqreturn_t wm5100_irq(int irq, void *data) { struct snd_soc_codec *codec = data; @@ -2145,6 +2304,9 @@ static irqreturn_t wm5100_irq(int irq, void *data) complete(&wm5100->fll[1].lock); } + if (irq_val & WM5100_ACCDET_EINT) + wm5100_micd_irq(codec); + irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); if (irq_val < 0) { dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index 345b3cf..fa32b12 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -16,6 +16,8 @@ #include <sound/soc.h> +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + #define WM5100_CLK_AIF1 1 #define WM5100_CLK_AIF2 2 #define WM5100_CLK_AIF3 3 -- 1.7.6.3

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[alsa-devel] How to use "hw" or "plughw"?
by Vidblain Amaro 11 Oct '11

11 Oct '11

Hi everbody I have a beginner in this area, i have to develop an audio application to implement filter algorithms to reduce the noisy of an audio I have readed the alsa api, and i decided to partition into 3 threads mi application, one for read samples from an audio file, the second for filter de samples, and the third for playback the audio filtered using snd_pcm_writei, but i am confused about the use "hw" and "plughw" when open de audio device. Some suggestion -- Vidblain "Lo verdadero es demasiado sencillo, pero siempre se llega a ello por lo más complicado."

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[alsa-devel] [PATCH] ASoC: Add missing default for WM5100 Clocking 1
by Mark Brown 11 Oct '11

11 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100-tables.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 960617b..e9ce81a 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -794,6 +794,7 @@ u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1] = { [0x0030] = 0x0000, /* R48 - PWM Drive 1 */ [0x0031] = 0x0100, /* R49 - PWM Drive 2 */ [0x0032] = 0x0100, /* R50 - PWM Drive 3 */ + [0x0100] = 0x0002, /* R256 - Clocking 1 */ [0x0101] = 0x0000, /* R257 - Clocking 3 */ [0x0102] = 0x0011, /* R258 - Clocking 4 */ [0x0103] = 0x0011, /* R259 - Clocking 5 */ -- 1.7.6.3

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[alsa-devel] question for the snd_pcm_mmap_writen.
by Anders Gnistrup 11 Oct '11

11 Oct '11

Hi First post to the mail list. I have been reading through the Alsa API and tutorials and it all seems fairly simple. But I still have some questions for the sample format possible to send using the mmap method. First a short description of my problem. I have a embedded device with a 8 channel audio chip. I only have to use 5 of these channels My goal is not to use any cpu cycles on the 3 unused channels I have decided to use the non-interleaved method (application specific, streams comes from different sources in a correct hardware specific format). For this, the function snd_pcm_mmap_writen could be used. When sending data to the device the format should/could be done using a 5 seperate buffers. snd_pcm_sframes_t snd_pcm_mmap_writen (snd_pcm_t * pcm, void ** bufs, snd_pcm_uframes_t size) where *bufs[0] equal to first channel *bufs[1] equals to next etc. Q1) Should/must the **bufs be continuations -> bufs[0][size]==bufs[1][0]??? Q2) Is is possible the set bufs[2] == NULL, to signal that the channel is unused? I think I know the answers. Yes, it should be continues, No data can be set to NULL. Q3) I have tried to find some information about the SND_PCM_ACCESS_MMAP_COMPLEX. What is this??? My best gees is that it is some entirely hardware specific access format? Regards Anders Gnistrup

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