diff options
| author | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2011-05-30 10:54:18 +0800 |
|---|---|---|
| committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2011-05-30 10:54:18 +0800 |
| commit | d21685ec258f803d3badae5eae821383a34815a9 (patch) | |
| tree | 7ab60a2a5d557a4f345b01a79ca2f877c06d9b92 | |
| parent | 74ab24af4fe165de5af01d0507250dd099f096b0 (diff) | |
| parent | ea02c63d57d7ec099f66ddb2942b4022e865cd5f (diff) | |
Merge branch 'for-2.6.40' into for-2.6.41
45 files changed, 1304 insertions, 252 deletions
diff --git a/MAINTAINERS b/MAINTAINERS index 9f926c0229db..75318bfb1650 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5840,7 +5840,7 @@ F: include/sound/ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) -M: Liam Girdwood <lrg@slimlogic.co.uk> +M: Liam Girdwood <lrg@ti.com> M: Mark Brown <broonie@opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git L: alsa-devel@alsa-project.org (moderated for non-subscribers) @@ -6093,7 +6093,7 @@ F: drivers/mmc/host/tifm_sd.c F: include/linux/tifm.h TI TWL4030 SERIES SOC CODEC DRIVER -M: Peter Ujfalusi <peter.ujfalusi@nokia.com> +M: Peter Ujfalusi <peter.ujfalusi@ti.com> L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/twl4030* @@ -6736,7 +6736,7 @@ F: drivers/scsi/vmw_pvscsi.c F: drivers/scsi/vmw_pvscsi.h VOLTAGE AND CURRENT REGULATOR FRAMEWORK -M: Liam Girdwood <lrg@slimlogic.co.uk> +M: Liam Girdwood <lrg@ti.com> M: Mark Brown <broonie@opensource.wolfsonmicro.com> W: http://opensource.wolfsonmicro.com/node/15 W: http://www.slimlogic.co.uk/?p=48 diff --git a/include/sound/ak4641.h b/include/sound/ak4641.h new file mode 100644 index 000000000000..96d1991c811d --- /dev/null +++ b/include/sound/ak4641.h @@ -0,0 +1,26 @@ +/* + * AK4641 ALSA SoC Codec driver + * + * Copyright 2009 Philipp Zabel + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __AK4641_H +#define __AK4641_H + +/** + * struct ak4641_platform_data - platform specific AK4641 configuration + * @gpio_power: GPIO to control external power to AK4641 + * @gpio_npdn: GPIO connected to AK4641 nPDN pin + * + * Both GPIO parameters are optional. + */ +struct ak4641_platform_data { + int gpio_power; + int gpio_npdn; +}; + +#endif /* __AK4641_H */ diff --git a/include/sound/soc.h b/include/sound/soc.h index b27c7a2d3bb0..f1de3e0c75bc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -702,6 +702,8 @@ struct snd_soc_aux_dev { /* SoC card */ struct snd_soc_card { const char *name; + const char *long_name; + const char *driver_name; struct device *dev; struct snd_card *snd_card; struct module *owner; diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h index 6c6649656798..0b94192a8cdf 100644 --- a/include/sound/tlv320dac33-plat.h +++ b/include/sound/tlv320dac33-plat.h @@ -1,7 +1,7 @@ /* * Platform header for Texas Instruments TLV320DAC33 codec driver * - * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> * * Copyright: (C) 2009 Nokia Corporation * diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h index e29fde6b5cbe..89beccb57edd 100644 --- a/include/sound/tpa6130a2-plat.h +++ b/include/sound/tpa6130a2-plat.h @@ -3,7 +3,7 @@ * * Copyright (C) Nokia Corporation * - * Written by Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 28afbbf69ce0..95572d290c27 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -146,7 +146,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) "at91sam9g20ek_wm8731 " ": at91sam9g20ek_wm8731_init() called\n"); - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_MCLK, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2a6971891d31..98175a096df2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C @@ -139,6 +140,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4641 + tristate + config SND_SOC_AK4642 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4cb2f42dbffa..fd8558406ef0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o @@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c new file mode 100644 index 000000000000..ed96f247c2da --- /dev/null +++ b/sound/soc/codecs/ak4641.c @@ -0,0 +1,664 @@ +/* + * ak4641.c -- AK4641 ALSA Soc Audio driver + * + * Copyright (C) 2008 Harald Welte <laforge@gnufiish.org> + * Copyright (C) 2011 Dmitry Artamonow <mad_soft@inbox.ru> + * + * Based on ak4535.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/ak4641.h> + +#include "ak4641.h" + +/* codec private data */ +struct ak4641_priv { + struct snd_soc_codec *codec; + unsigned int sysclk; + int deemph; + int playback_fs; +}; + +/* + * ak4641 register cache + */ +static const u8 ak4641_reg[AK4641_CACHEREGNUM] = { + 0x00, 0x80, 0x00, 0x80, + 0x02, 0x00, 0x11, 0x05, + 0x00, 0x00, 0x36, 0x10, + 0x00, 0x00, 0x57, 0x00, + 0x88, 0x88, 0x08, 0x08 +}; + +static const int deemph_settings[] = {44100, 0, 48000, 32000}; + +static int ak4641_set_deemph(struct snd_soc_codec *codec) +{ + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + int i, best = 0; + + for (i = 0 ; i < ARRAY_SIZE(deemph_settings); i++) { + /* if deemphasis is on, select the nearest available rate */ + if (ak4641->deemph && deemph_settings[i] != 0 && + abs(deemph_settings[i] - ak4641->playback_fs) < + abs(deemph_settings[best] - ak4641->playback_fs)) + best = i; + + if (!ak4641->deemph && deemph_settings[i] == 0) + best = i; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", best); + + return snd_soc_update_bits(codec, AK4641_DAC, 0x3, best); +} + +static int ak4641_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + ak4641->deemph = deemph; + + return ak4641_set_deemph(codec); +} + +static int ak4641_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = ak4641->deemph; + return 0; +}; + +static const char *ak4641_mono_out[] = {"(L + R)/2", "Hi-Z"}; +static const char *ak4641_hp_out[] = {"Stereo", "Mono"}; +static const char *ak4641_mic_select[] = {"Internal", "External"}; +static const char *ak4641_mic_or_dac[] = {"Microphone", "Voice DAC"}; + + +static const DECLARE_TLV_DB_SCALE(mono_gain_tlv, -1700, 2300, 0); +static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(master_tlv, -12750, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_stereo_sidetone_tlv, -2700, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_mono_sidetone_tlv, -400, 400, 0); +static const DECLARE_TLV_DB_SCALE(capture_tlv, -800, 50, 0); +static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0); +static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0); + + +static const struct soc_enum ak4641_mono_out_enum = + SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out); +static const struct soc_enum ak4641_hp_out_enum = + SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out); +static const struct soc_enum ak4641_mic_select_enum = + SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select); +static const struct soc_enum ak4641_mic_or_dac_enum = + SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac); + +static const struct snd_kcontrol_new ak4641_snd_controls[] = { + SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum), + SOC_SINGLE_TLV("Mono 1 Gain Volume", AK4641_SIG1, 7, 1, 1, + mono_gain_tlv), + SOC_ENUM("Headphone Output", ak4641_hp_out_enum), + SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0, + ak4641_get_deemph, ak4641_put_deemph), + + SOC_SINGLE_TLV("Mic Boost Volume", AK4641_MIC, 0, 1, 0, mic_boost_tlv), + + SOC_SINGLE("ALC Operation Time", AK4641_TIMER, 0, 3, 0), + SOC_SINGLE("ALC Recovery Time", AK4641_TIMER, 2, 3, 0), + SOC_SINGLE("ALC ZC Time", AK4641_TIMER, 4, 3, 0), + + SOC_SINGLE("ALC 1 Switch", AK4641_ALC1, 5, 1, 0), + + SOC_SINGLE_TLV("ALC Volume", AK4641_ALC2, 0, 71, 0, alc_tlv), + SOC_SINGLE("Left Out Enable Switch", AK4641_SIG2, 1, 1, 0), + SOC_SINGLE("Right Out Enable Switch", AK4641_SIG2, 0, 1, 0), + + SOC_SINGLE_TLV("Capture Volume", AK4641_PGA, 0, 71, 0, capture_tlv), + + SOC_DOUBLE_R_TLV("Master Playback Volume", AK4641_LATT, + AK4641_RATT, 0, 255, 1, master_tlv), + + SOC_SINGLE_TLV("AUX In Volume", AK4641_VOL, 0, 15, 0, aux_in_tlv), + + SOC_SINGLE("Equalizer Switch", AK4641_DAC, 2, 1, 0), + SOC_SINGLE_TLV("EQ1 100 Hz Volume", AK4641_EQLO, 0, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 250 Hz Volume", AK4641_EQLO, 4, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 1 kHz Volume", AK4641_EQMID, 0, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 3.5 kHz Volume", AK4641_EQMID, 4, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 10 kHz Volume", AK4641_EQHI, 0, 15, 1, eq_tlv), +}; + +/* Mono 1 Mixer */ +static const struct snd_kcontrol_new ak4641_mono1_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("Mic Mono Sidetone Volume", AK4641_VOL, 7, 1, 0, + mic_mono_sidetone_tlv), + SOC_DAPM_SINGLE("Mic Mono Sidetone Switch", AK4641_SIG1, 4, 1, 0), + SOC_DAPM_SINGLE("Mono Playback Switch", AK4641_SIG1, 5, 1, 0), +}; + +/* Stereo Mixer */ +static const struct snd_kcontrol_new ak4641_stereo_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("Mic Sidetone Volume", AK4641_VOL, 4, 7, 0, + mic_stereo_sidetone_tlv), + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4641_SIG2, 4, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", AK4641_SIG2, 7, 1, 0), + SOC_DAPM_SINGLE("Aux Bypass Switch", AK4641_SIG2, 5, 1, 0), +}; + +/* Input Mixer */ +static const struct snd_kcontrol_new ak4641_input_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Capture Switch", AK4641_MIC, 2, 1, 0), + SOC_DAPM_SINGLE("Aux Capture Switch", AK4641_MIC, 5, 1, 0), +}; + +/* Mic mux */ +static const struct snd_kcontrol_new ak4641_mic_mux_control = + SOC_DAPM_ENUM("Mic Select", ak4641_mic_select_enum); + +/* Input mux */ +static const struct snd_kcontrol_new ak4641_input_mux_control = + SOC_DAPM_ENUM("Input Select", ak4641_mic_or_dac_enum); + +/* mono 2 switch */ +static const struct snd_kcontrol_new ak4641_mono2_control = + SOC_DAPM_SINGLE("Switch", AK4641_SIG1, 0, 1, 0); + +/* ak4641 dapm widgets */ +static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, + &ak4641_stereo_mixer_controls[0], + ARRAY_SIZE(ak4641_stereo_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, + &ak4641_mono1_mixer_controls[0], + ARRAY_SIZE(ak4641_mono1_mixer_controls)), + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, + &ak4641_input_mixer_controls[0], + ARRAY_SIZE(ak4641_input_mixer_controls)), + SND_SOC_DAPM_MUX("Mic Mux", SND_SOC_NOPM, 0, 0, + &ak4641_mic_mux_control), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4641_input_mux_control), + SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, + &ak4641_mono2_control), + + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("MOUT1"), + SND_SOC_DAPM_OUTPUT("MOUT2"), + SND_SOC_DAPM_OUTPUT("MICOUT"), + + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AK4641_PM1, 0, 0), + SND_SOC_DAPM_PGA("Mic", AK4641_PM1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX In", AK4641_PM1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", AK4641_PM1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line Out", AK4641_PM1, 4, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AK4641_PM2, 0, 0), + SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0), + + SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0), + SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), + + SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0), + SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0), + + SND_SOC_DAPM_INPUT("MICIN"), + SND_SOC_DAPM_INPUT("MICEXT"), + SND_SOC_DAPM_INPUT("AUX"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const struct snd_soc_dapm_route ak4641_audio_map[] = { + /* Stereo Mixer */ + {"Stereo Mixer", "Playback Switch", "DAC"}, + {"Stereo Mixer", "Mic Sidetone Switch", "Input Mux"}, + {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, + + /* Mono 1 Mixer */ + {"Mono1 Mixer", "Mic Mono Sidetone Switch", "Input Mux"}, + {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, + + /* Mic */ + {"Mic", NULL, "AIN"}, + {"Mic Mux", "Internal", "Mic Int Bias"}, + {"Mic Mux", "External", "Mic Ext Bias"}, + {"Mic Int Bias", NULL, "MICIN"}, + {"Mic Ext Bias", NULL, "MICEXT"}, + {"MICOUT", NULL, "Mic Mux"}, + + /* Input Mux */ + {"Input Mux", "Microphone", "Mic"}, + {"Input Mux", "Voice DAC", "Voice DAC"}, + + /* Line Out */ + {"LOUT", NULL, "Line Out"}, + {"ROUT", NULL, "Line Out"}, + {"Line Out", NULL, "Stereo Mixer"}, + + /* Mono 1 Out */ + {"MOUT1", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono1 Mixer"}, + + /* Mono 2 Out */ + {"MOUT2", NULL, "Mono 2 Enable"}, + {"Mono 2 Enable", "Switch", "Mono Out 2"}, + {"Mono Out 2", NULL, "Stereo Mixer"}, + + {"Voice ADC", NULL, "Mono 2 Enable"}, + + /* Aux In */ + {"AUX In", NULL, "AUX"}, + + /* ADC */ + {"ADC", NULL, "Input Mixer"}, + {"Input Mixer", "Mic Capture Switch", "Mic"}, + {"Input Mixer", "Aux Capture Switch", "AUX In"}, +}; + +static int ak4641_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + + ak4641->sysclk = freq; + return 0; +} + +static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + int rate = params_rate(params), fs = 256; + u8 mode2; + + if (rate) + fs = ak4641->sysclk / rate; + else + return -EINVAL; + + /* set fs */ + switch (fs) { + case 1024: + mode2 = (0x2 << 5); + break; + case 512: + mode2 = (0x1 << 5); + break; + case 256: + mode2 = (0x0 << 5); + break; + default: + dev_err(codec->dev, "Error: unsupported fs=%d\n", fs); + return -EINVAL; + } + + snd_soc_update_bits(codec, AK4641_MODE2, (0x3 << 5), mode2); + + /* Update de-emphasis filter for the new rate */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ak4641->playback_fs = rate; + ak4641_set_deemph(codec); + }; + + return 0; +} + +static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 btif; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + btif = (0x3 << 5); + break; + case SND_SOC_DAIFMT_LEFT_J: + btif = (0x2 << 5); + break; + case SND_SOC_DAIFMT_DSP_A: /* MSB after FRM */ + btif = (0x0 << 5); + break; + case SND_SOC_DAIFMT_DSP_B: /* MSB during FRM */ + btif = (0x1 << 5); + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); +} + +static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode1 = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode1 = 0x02; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode1 = 0x01; + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, AK4641_MODE1, mode1); +} + +static int ak4641_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_update_bits(codec, AK4641_DAC, 0x20, mute ? 0x20 : 0); +} + +static int ak4641_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ak4641_platform_data *pdata = codec->dev->platform_data; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + /* unmute */ + snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0); + break; + case SND_SOC_BIAS_PREPARE: + /* mute */ + snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 1); + mdelay(1); + if (pdata && gpio_is_valid(pdata->gpio_npdn)) + gpio_set_value(pdata->gpio_npdn, 1); + mdelay(1); + + ret = snd_soc_cache_sync(codec); + if (ret) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0x80); + snd_soc_update_bits(codec, AK4641_PM2, 0x80, 0); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0); + if (pdata && gpio_is_valid(pdata->gpio_npdn)) + gpio_set_value(pdata->gpio_npdn, 0); + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + codec->cache_sync = 1; + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define AK4641_RATES (SNDRV_PCM_RATE_8000_48000) +#define AK4641_RATES_BT (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000) +#define AK4641_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops ak4641_i2s_dai_ops = { + .hw_params = ak4641_i2s_hw_params, + .set_fmt = ak4641_i2s_set_dai_fmt, + .digital_mute = ak4641_mute, + .set_sysclk = ak4641_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops ak4641_pcm_dai_ops = { + .hw_params = NULL, /* rates are controlled by BT chip */ + .set_fmt = ak4641_pcm_set_dai_fmt, + .digital_mute = ak4641_mute, + .set_sysclk = ak4641_set_dai_sysclk, +}; + +struct snd_soc_dai_driver ak4641_dai[] = { +{ + .name = "ak4641-hifi", + .id = 1, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4641_RATES, + .formats = AK4641_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4641_RATES, + .formats = AK4641_FORMATS, + }, + .ops = &ak4641_i2s_dai_ops, + .symmetric_rates = 1, +}, +{ + .name = "ak4641-voice", + .id = 1, + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = AK4641_RATES_BT, + .formats = AK4641_FORMATS, + }, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 1, + .rates = AK4641_RATES_BT, + .formats = AK4641_FORMATS, + }, + .ops = &ak4641_pcm_dai_ops, + .symmetric_rates = 1, +}, +}; + +static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak4641_resume(struct snd_soc_codec *codec) +{ + ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int ak4641_probe(struct snd_soc_codec *codec) +{ + struct ak4641_platform_data *pdata = codec->dev->platform_data; + int ret; + + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + ret = gpio_request_one(pdata->gpio_power, + GPIOF_OUT_INIT_LOW, "ak4641 power"); + if (ret) + goto err_out; + } + if (gpio_is_valid(pdata->gpio_npdn)) { + ret = gpio_request_one(pdata->gpio_npdn, + GPIOF_OUT_INIT_LOW, "ak4641 npdn"); + if (ret) + goto err_gpio; + + udelay(1); /* > 150 ns */ + gpio_set_value(pdata->gpio_npdn, 1); + } + } + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err_register; + } + + /* power on device */ + ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; + +err_register: + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } +err_gpio: + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_free(pdata->gpio_power); +err_out: + return ret; +} + +static int ak4641_remove(struct snd_soc_codec *codec) +{ + struct ak4641_platform_data *pdata = codec->dev->platform_data; + + ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_power); + } + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } + return 0; +} + + +static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { + .probe = ak4641_probe, + .remove = ak4641_remove, + .suspend = ak4641_suspend, + .resume = ak4641_resume, + .controls = ak4641_snd_controls, + .num_controls = ARRAY_SIZE(ak4641_snd_controls), + .dapm_widgets = ak4641_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4641_dapm_widgets), + .dapm_routes = ak4641_audio_map, + .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map), + .set_bias_level = ak4641_set_bias_level, + .reg_cache_size = ARRAY_SIZE(ak4641_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = ak4641_reg, + .reg_cache_step = 1, +}; + + +static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4641_priv *ak4641; + int ret; + + ak4641 = kzalloc(sizeof(struct ak4641_priv), GFP_KERNEL); + if (!ak4641) + return -ENOMEM; + + i2c_set_clientdata(i2c, ak4641); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641, + ak4641_dai, ARRAY_SIZE(ak4641_dai)); + if (ret < 0) + kfree(ak4641); + + return ret; +} + +static int __devexit ak4641_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + kfree(i2c_get_clientdata(i2c)); + return 0; +} + +static const struct i2c_device_id ak4641_i2c_id[] = { + { "ak4641", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id); + +static struct i2c_driver ak4641_i2c_driver = { + .driver = { + .name = "ak4641", + .owner = THIS_MODULE, + }, + .probe = ak4641_i2c_probe, + .remove = __devexit_p(ak4641_i2c_remove), + .id_table = ak4641_i2c_id, +}; + +static int __init ak4641_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&ak4641_i2c_driver); + if (ret != 0) + pr_err("Failed to register AK4641 I2C driver: %d\n", ret); + + return ret; +} +module_init(ak4641_modinit); + +static void __exit ak4641_exit(void) +{ + i2c_del_driver(&ak4641_i2c_driver); +} +module_exit(ak4641_exit); + +MODULE_DESCRIPTION("SoC AK4641 driver"); +MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4641.h b/sound/soc/codecs/ak4641.h new file mode 100644 index 000000000000..4a263248efea --- /dev/null +++ b/sound/soc/codecs/ak4641.h @@ -0,0 +1,47 @@ +/* + * ak4641.h -- AK4641 SoC Audio driver + * + * Copyright 2008 Harald Welte <laforge@gnufiish.org> + * + * Based on ak4535.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4641_H +#define _AK4641_H + +/* AK4641 register space */ + +#define AK4641_PM1 0x00 +#define AK4641_PM2 0x01 +#define AK4641_SIG1 0x02 +#define AK4641_SIG2 0x03 +#define AK4641_MODE1 0x04 +#define AK4641_MODE2 0x05 +#define AK4641_DAC 0x06 +#define AK4641_MIC 0x07 +#define AK4641_TIMER 0x08 +#define AK4641_ALC1 0x09 +#define AK4641_ALC2 0x0a +#define AK4641_PGA 0x0b +#define AK4641_LATT 0x0c +#define AK4641_RATT 0x0d +#define AK4641_VOL 0x0e +#define AK4641_STATUS 0x0f +#define AK4641_EQLO 0x10 +#define AK4641_EQMID 0x11 +#define AK4641_EQHI 0x12 +#define AK4641_BTIF 0x13 + +#define AK4641_CACHEREGNUM 0x14 + + + +#define AK4641_DAI_HIFI 0 +#define AK4641_DAI_VOICE 1 + + +#endif diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 57e9dac88d38..f9a87737ec16 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -39,7 +39,31 @@ static struct snd_soc_dai_driver dmic_dai = { }, }; -static struct snd_soc_codec_driver soc_dmic = {}; +static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_INPUT("DMic"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DMIC AIF", NULL, "DMic"}, +}; + +static int dmic_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, + ARRAY_SIZE(dmic_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_widgets(dapm); + + return 0; +} + +static struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_probe, +}; static int __devinit dmic_dev_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 93255ff48b4f..ac65a2d36408 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum = ARRAY_SIZE(max98088_exmode_texts), max98088_exmode_texts, max98088_exmode_values); -static const struct snd_kcontrol_new max98088_exmode_controls = - SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum); static const char *max98088_ex_thresh[] = { /* volts PP */ "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; @@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + SOC_ENUM("EX Limiter Mode", max98088_exmode_enum), SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum), @@ -808,10 +807,10 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { /* Left speaker mixer switch */ static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1, 0), @@ -836,10 +835,10 @@ static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = { /* Left headphone mixer switch */ static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1, 0), @@ -864,10 +863,10 @@ static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = { /* Left earpiece/receiver mixer switch */ static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1, 0), @@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = { SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0), - SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0, - &max98088_exmode_controls), - SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), SND_SOC_DAPM_OUTPUT("SPKL"), @@ -1568,6 +1564,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int reg; + + if (mute) + reg = M98088_DAI_MUTE; + else + reg = 0; + + snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + +static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int reg; + + if (mute) + reg = M98088_DAI_MUTE; + else + reg = 0; + + snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + static void max98088_sync_cache(struct snd_soc_codec *codec) { u16 *reg_cache = codec->reg_cache; @@ -1629,12 +1655,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops = { .set_sysclk = max98088_dai_set_sysclk, .set_fmt = max98088_dai1_set_fmt, .hw_params = max98088_dai1_hw_params, + .digital_mute = max98088_dai1_digital_mute, }; static struct snd_soc_dai_ops max98088_dai2_ops = { .set_sysclk = max98088_dai_set_sysclk, .set_fmt = max98088_dai2_set_fmt, .hw_params = max98088_dai2_hw_params, + .digital_mute = max98088_dai2_digital_mute, }; static struct snd_soc_dai_driver max98088_dai[] = { diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index 56554c797fef..be89a4f4aab8 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -133,6 +133,19 @@ #define M98088_REC_LINEMODE (1<<7) #define M98088_REC_LINEMODE_MASK (1<<7) +/* M98088_REG_2D_MIX_SPK_CNTL */ + #define M98088_MIX_SPKR_GAIN_MASK (3<<2) + #define M98088_MIX_SPKR_GAIN_SHIFT 2 + #define M98088_MIX_SPKL_GAIN_MASK (3<<0) + #define M98088_MIX_SPKL_GAIN_SHIFT 0 + +/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */ + #define M98088_DAI_MUTE (1<<7) + #define M98088_DAI_MUTE_MASK (1<<7) + #define M98088_DAI_VOICE_GAIN_MASK (3<<4) + #define M98088_DAI_ATTENUATION_MASK (0xF<<0) + #define M98088_DAI_ATTENUATION_SHIFT 0 + /* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */ #define M98088_MICPRE_MASK (3<<5) #define M98088_MICPRE_SHIFT 5 diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index fe19677bf4b5..872a5fa4bf1f 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1870,16 +1870,14 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, BUG_ON(channel > 1); - cdata = &max98095->dai[channel]; + if (!pdata || !max98095->eq_textcnt) + return 0; if (sel >= pdata->eq_cfgcnt) return -EINVAL; + cdata = &max98095->dai[channel]; cdata->eq_sel = sel; - - if (!pdata || !max98095->eq_textcnt) - return 0; - fs = cdata->rate; /* Find the selected configuration with nearest sample rate */ @@ -2018,16 +2016,14 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, BUG_ON(channel > 1); - cdata = &max98095->dai[channel]; + if (!pdata || !max98095->bq_textcnt) + return 0; if (sel >= pdata->bq_cfgcnt) return -EINVAL; + cdata = &max98095->dai[channel]; cdata->bq_sel = sel; - - if (!pdata || !max98095->bq_textcnt) - return 0; - fs = cdata->rate; /* Find the selected configuration with nearest sample rate */ diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 4c32b54913ad..6a1a7e705cd7 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -21,7 +21,7 @@ #include <sound/pcm.h> #include <sound/initval.h> -MODULE_LICENSE("GPL"); +#define DRV_NAME "spdif-dit" #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE @@ -56,7 +56,7 @@ static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { - .name = "spdif-dit", + .name = DRV_NAME, .owner = THIS_MODULE, }, }; @@ -74,3 +74,7 @@ static void __exit dit_exit(void) module_init(dit_modinit); module_exit(dit_exit); +MODULE_AUTHOR("Steve Chen <schen@mvista.com>"); +MODULE_DESCRIPTION("SPDIF dummy codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 70099c9d63c7..84f4ad568556 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -137,7 +137,7 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), -SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0), +SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6c43c13f0430..c3d96fc8c267 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 90c361ef598f..faa5e9fb1471 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1,7 +1,7 @@ /* * ALSA SoC Texas Instruments TLV320DAC33 codec driver * - * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> * * Copyright: (C) 2009 Nokia Corporation * @@ -1658,5 +1658,5 @@ module_exit(dac33_module_exit); MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); -MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@nokia.com>"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h index 7c318b5da437..ed69670747bf 100644 --- a/sound/soc/codecs/tlv320dac33.h +++ b/sound/soc/codecs/tlv320dac33.h @@ -1,7 +1,7 @@ /* * ALSA SoC Texas Instruments TLV320DAC33 codec driver * - * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> * * Copyright: (C) 2009 Nokia Corporation * diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 1f1ac8110bef..239e0c461068 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -3,7 +3,7 @@ * * Copyright (C) Nokia Corporation * - * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -495,7 +495,7 @@ static void __exit tpa6130a2_exit(void) i2c_del_driver(&tpa6130a2_i2c_driver); } -MODULE_AUTHOR("Peter Ujfalusi"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>"); MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h index 5df49c8756b2..417444020ba6 100644 --- a/sound/soc/codecs/tpa6130a2.h +++ b/sound/soc/codecs/tpa6130a2.h @@ -3,7 +3,7 @@ * * Copyright (C) Nokia Corporation * - * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 255901c4460d..4c336636d4f5 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -960,9 +960,9 @@ static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0); /* * AFMGAIN volume control: - * from 18 to 24 dB in 6 dB steps + * from -18 to 24 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 1800, 600, 0); +static DECLARE_TLV_DB_SCALE(afm_amp_tlv, -1800, 600, 0); /* * HSGAIN volume control: @@ -1049,7 +1049,7 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* AFM gains */ SOC_DOUBLE_TLV("Aux FM Volume", - TWL6040_REG_LINEGAIN, 0, 4, 0xF, 0, afm_amp_tlv), + TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 14d0716bf009..bcc208967917 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -22,7 +22,7 @@ SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC", "wm1250-ev1 Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_INPUT("WM1250 Input"), -SND_SOC_DAPM_INPUT("WM1250 Output"), +SND_SOC_DAPM_OUTPUT("WM1250 Output"), }; static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 6dec7cee2cb4..2dc964b55e4f 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -198,7 +198,7 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, { struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec); - return wm8731->sysclk_type == WM8731_SYSCLK_MCLK; + return wm8731->sysclk_type == WM8731_SYSCLK_XTAL; } static const struct snd_soc_dapm_route wm8731_intercon[] = { diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 957cd66177d6..43e3d760766f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index ccc9bd832794..a0b1a7278284 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -19,7 +19,6 @@ #include <linux/gcd.h> #include <linux/gpio.h> #include <linux/i2c.h> -#include <linux/delay.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/workqueue.h> diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 74983ee2b87f..0293763debe5 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -99,7 +99,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, len = fw->size - len; while (len) { if (len < 12) { - dev_err(codec->dev, "%s short data block of %d\n", + dev_err(codec->dev, "%s short data block of %zd\n", name, len); goto err; } @@ -107,7 +107,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, memcpy(&data32, data + 4, sizeof(data32)); block_len = be32_to_cpu(data32); if (block_len + 8 > len) { - dev_err(codec->dev, "%d byte block longer than file\n", + dev_err(codec->dev, "%zd byte block longer than file\n", block_len); goto err; } @@ -141,7 +141,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, case WM_FW_BLOCK_I: case WM_FW_BLOCK_A: case WM_FW_BLOCK_C: - dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name, + dev_dbg(codec->dev, "%s: %zd bytes of %x@%x\n", name, block_len, (data32 >> 24) & 0xff, data32 & 0xffffff); @@ -362,6 +362,10 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) path, wm8994->dsp_active, start, pwr_reg, reg); if (start && ena) { + /* If the DSP is already running then noop */ + if (reg & WM8958_DSP2_ENA) + return; + /* If either AIFnCLK is not yet enabled postpone */ if (!(snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA_MASK) && @@ -508,6 +512,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->mbc_ena[mbc] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -628,6 +635,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->vss_ena[vss] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -689,6 +699,16 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (hpf < 3) { + if (wm8994->hpf1_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } else { + if (wm8994->hpf2_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -782,6 +802,9 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->enh_eq_ena[eq] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 056aef904347..9e5ff789b805 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, static int class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b6d47e771519..970a95c5360b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -877,7 +877,8 @@ static const char *hp_mux_text[] = { static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; @@ -1004,7 +1005,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; @@ -2416,8 +2418,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; + switch (control->type) { + case WM8994: + snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); + break; + case WM8958: + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, 0); + break; + } + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -2435,6 +2448,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8994_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; unsigned int val, mask; @@ -2473,6 +2487,19 @@ static int wm8994_resume(struct snd_soc_codec *codec) i + 1, ret); } + switch (control->type) { + case WM8994: + if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) + snd_soc_update_bits(codec, WM8994_MICBIAS, + WM8994_MICD_ENA, WM8994_MICD_ENA); + break; + case WM8958: + if (wm8994->jack_cb) + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, WM8958_MICD_ENA); + break; + } + return 0; } #else diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 67eaaecbb42e..5ad873fda814 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -305,11 +305,11 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source, static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec; int ret; - w = snd_kcontrol_chip(kcontrol); codec = w->codec; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); wm8995_update_class_w(codec); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4005e9af5d61..9e370d14ad88 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -215,23 +215,23 @@ static const struct snd_kcontrol_new analogue_snd_controls[] = { SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), -SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), +SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), -SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), +SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), -SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), +SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), -SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), +SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0, inmix_sw_tlv), @@ -787,17 +787,17 @@ static const struct snd_soc_dapm_route analogue_routes[] = { static const struct snd_soc_dapm_route lineout1_diff_routes[] = { { "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" }, { "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" }, - { "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" }, + { "LINEOUT1 Mixer", "Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" }, { "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" }, }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { - { "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" }, - { "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" }, + { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, + { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" }, + { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, { "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" }, @@ -806,17 +806,17 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { static const struct snd_soc_dapm_route lineout2_diff_routes[] = { { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, - { "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" }, + { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, { "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" }, }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { - { "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" }, - { "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" }, + { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, + { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" }, + { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, { "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" }, @@ -836,17 +836,21 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, WM8993_IN2_VU, WM8993_IN2_VU); + snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT, + WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT, WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME, - WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC); + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC, + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC); snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC); snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME, - WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC); + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU, + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU); snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4ddc6d3b6678..8566238db2a5 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -909,6 +909,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_size = pdata->sram_size_playback; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + mem->start); @@ -925,6 +926,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_size = pdata->sram_size_capture; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + mem->start); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2175f09e57b6..07b772357244 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula <jhnikula@gmail.com> - * Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x() || cpu_is_omap44xx()) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (cpu_is_omap343x()) { + if (cpu_is_omap34xx()) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (omap_mcbsp_get_dma_op_mode(bus_id) == diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 37dc7211ed3f..9a7dedd6f5a9 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula <jhnikula@gmail.com> - * Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 8caeb8d305c3..e6a6b991d05f 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula <jhnikula@gmail.com> - * Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, @@ -195,7 +196,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); - else + else if (!substream->runtime->no_period_wakeup) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); if (!(cpu_class_is_omap1())) { diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index fea0515331fb..a0ed1dbb52d6 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula <jhnikula@gmail.com> - * Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index d0986220eff9..0aae998b6540 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 - 2009 Nokia Corporation * - * Contact: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com> * Eduardo Valentin <eduardo.valentin@nokia.com> * Jarkko Nikula <jhnikula@gmail.com> * diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 580f48571303..33ebc46b45b5 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -155,6 +155,15 @@ config SND_SOC_RAUMFELD help Say Y if you want to add support for SoC audio on Raumfeld devices +config SND_PXA2XX_SOC_HX4700 + tristate "SoC Audio support for HP iPAQ hx4700" + depends on SND_PXA2XX_SOC && MACH_H4700 + select SND_PXA2XX_SOC_I2S + select SND_SOC_AK4641 + help + Say Y if you want to add support for SoC audio on the + HP iPAQ hx4700. + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 07660165ec8d..af357623be9d 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o snd-soc-saarb-objs := saarb.o snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o +snd-soc-hx4700-objs := hx4700.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-z2-objs := z2.o @@ -37,6 +38,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c new file mode 100644 index 000000000000..65c124831a00 --- /dev/null +++ b/sound/soc/pxa/hx4700.c @@ -0,0 +1,255 @@ +/* + * SoC audio for HP iPAQ hx4700 + * + * Copyright (c) 2009 Philipp Zabel + * + * 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. + * + */ + +#include <linux/module.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hx4700.h> +#include <asm/mach-types.h> +#include "pxa2xx-i2s.h" + +#include "../codecs/ak4641.h" + +static struct snd_soc_jack hs_jack; + +/* Headphones jack detection DAPM pin */ +static struct snd_soc_jack_pin hs_jack_pin[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + /* disable speaker when hp jack is inserted */ + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +/* Headphones jack detection GPIO */ +static struct snd_soc_jack_gpio hs_jack_gpio = { + .gpio = GPIO75_HX4700_EARPHONE_nDET, + .invert = true, + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, +}; + +/* + * iPAQ hx4700 uses I2S for capture and playback. + */ +static int hx4700_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the I2S system clock as output */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* inform codec driver about clock freq * + * (PXA I2S always uses divider 256) */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params), + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops hx4700_ops = { + .hw_params = hx4700_hw_params, +}; + +static int hx4700_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(GPIO107_HX4700_SPK_nSD, !!SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int hx4700_hp_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(GPIO92_HX4700_HP_DRIVER, !!SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* hx4700 machine dapm widgets */ +static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power), + SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power), + SND_SOC_DAPM_MIC("Built-in Microphone", NULL), +}; + +/* hx4700 machine audio_map */ +static const struct snd_soc_dapm_route hx4700_audio_map[] = { + + /* Headphone connected to LOUT, ROUT */ + {"Headphone Jack", NULL, "LOUT"}, + {"Headphone Jack", NULL, "ROUT"}, + + /* Speaker connected to MOUT2 */ + {"Speaker", NULL, "MOUT2"}, + + /* Microphone connected to MICIN */ + {"MICIN", NULL, "Built-in Microphone"}, + {"AIN", NULL, "MICOUT"}, +}; + +/* + * Logic for a ak4641 as connected on a HP iPAQ hx4700 + */ +static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int err; + + /* NC codec pins */ + /* FIXME: is anything connected here? */ + snd_soc_dapm_nc_pin(dapm, "MOUT1"); + snd_soc_dapm_nc_pin(dapm, "MICEXT"); + snd_soc_dapm_nc_pin(dapm, "AUX"); + + /* Jack detection API stuff */ + err = snd_soc_jack_new(codec, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin), + hs_jack_pin); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio); + + return err; +} + +/* hx4700 digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link hx4700_dai = { + .name = "ak4641", + .stream_name = "AK4641", + .cpu_dai_name = "pxa2xx-i2s", + .codec_dai_name = "ak4641-hifi", + .platform_name = "pxa-pcm-audio", + .codec_name = "ak4641.0-0012", + .init = hx4700_ak4641_init, + .ops = &hx4700_ops, +}; + +/* hx4700 audio machine driver */ +static struct snd_soc_card snd_soc_card_hx4700 = { + .name = "iPAQ hx4700", + .dai_link = &hx4700_dai, + .num_links = 1, + .dapm_widgets = hx4700_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(hx4700_dapm_widgets), + .dapm_routes = hx4700_audio_map, + .num_dapm_routes = ARRAY_SIZE(hx4700_audio_map), +}; + +static struct gpio hx4700_audio_gpios[] = { + { GPIO107_HX4700_SPK_nSD, GPIOF_OUT_INIT_HIGH, "SPK_POWER" }, + { GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" }, +}; + +static int __devinit hx4700_audio_probe(struct platform_device *pdev) +{ + int ret; + + if (!machine_is_h4700()) + return -ENODEV; + + ret = gpio_request_array(hx4700_audio_gpios, + ARRAY_SIZE(hx4700_audio_gpios)); + if (ret) + return ret; + + snd_soc_card_hx4700.dev = &pdev->dev; + ret = snd_soc_register_card(&snd_soc_card_hx4700); + if (ret) + return ret; + + return 0; +} + +static int __devexit hx4700_audio_remove(struct platform_device *pdev) +{ + snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); + snd_soc_unregister_card(&snd_soc_card_hx4700); + + gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); + gpio_set_value(GPIO107_HX4700_SPK_nSD, 0); + + gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios)); + return 0; +} + +static struct platform_driver hx4700_audio_driver = { + .driver = { + .name = "hx4700-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = hx4700_audio_probe, + .remove = __devexit_p(hx4700_audio_remove), +}; + +static int __init hx4700_modinit(void) +{ + return platform_driver_register(&hx4700_audio_driver); +} +module_init(hx4700_modinit); + +static void __exit hx4700_modexit(void) +{ + platform_driver_unregister(&hx4700_audio_driver); +} + +module_exit(hx4700_modexit); + +MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:hx4700-audio"); diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 2afabaf59491..1a591f1ebfbd 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = { .hw_params = raumfeld_cs4270_hw_params, }; -static int raumfeld_line_suspend(struct snd_soc_card *card) +static int raumfeld_analog_suspend(struct snd_soc_card *card) { raumfeld_enable_audio(false); return 0; } -static int raumfeld_line_resume(struct snd_soc_card *card) +static int raumfeld_analog_resume(struct snd_soc_card *card) { raumfeld_enable_audio(true); return 0; @@ -225,32 +225,53 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .hw_params = raumfeld_ak4104_hw_params, }; -static struct snd_soc_dai_link raumfeld_dai[] = { +#define DAI_LINK_CS4270 \ +{ \ + .name = "CS4270", \ + .stream_name = "CS4270", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "cs4270-hifi", \ + .codec_name = "cs4270-codec.0-0048", \ + .ops = &raumfeld_cs4270_ops, \ +} + +#define DAI_LINK_AK4104 \ +{ \ + .name = "ak4104", \ + .stream_name = "Playback", \ + .cpu_dai_name = "pxa-ssp-dai.1", \ + .codec_dai_name = "ak4104-hifi", \ + .platform_name = "pxa-pcm-audio", \ + .ops = &raumfeld_ak4104_ops, \ + .codec_name = "spi0.0", \ +} + +static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { - .name = "ak4104", - .stream_name = "Playback", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "ak4104-hifi", - .platform_name = "pxa-pcm-audio", - .ops = &raumfeld_ak4104_ops, - .codec_name = "ak4104-codec.0", -}, + DAI_LINK_CS4270, + DAI_LINK_AK4104, +}; + +static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { - .name = "CS4270", - .stream_name = "CS4270", - .cpu_dai_name = "pxa-ssp-dai.0", - .platform_name = "pxa-pcm-audio", - .codec_dai_name = "cs4270-hifi", - .codec_name = "cs4270-codec.0-0048", - .ops = &raumfeld_cs4270_ops, -},}; - -static struct snd_soc_card snd_soc_raumfeld = { - .name = "Raumfeld", - .dai_link = raumfeld_dai, - .suspend_post = raumfeld_line_suspend, - .resume_pre = raumfeld_line_resume, - .num_links = ARRAY_SIZE(raumfeld_dai), + DAI_LINK_CS4270, +}; + +static struct snd_soc_card snd_soc_raumfeld_connector = { + .name = "Raumfeld Connector", + .dai_link = snd_soc_raumfeld_connector_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, +}; + +static struct snd_soc_card snd_soc_raumfeld_speaker = { + .name = "Raumfeld Speaker", + .dai_link = snd_soc_raumfeld_speaker_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, }; static struct platform_device *raumfeld_audio_device; @@ -271,22 +292,25 @@ static int __init raumfeld_audio_init(void) set_max9485_clk(MAX9485_MCLK_FREQ_122880); - /* Register LINE and SPDIF */ + /* Register analog device */ raumfeld_audio_device = platform_device_alloc("soc-audio", 0); if (!raumfeld_audio_device) return -ENOMEM; - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld); - ret = platform_device_add(raumfeld_audio_device); - - /* no S/PDIF on Speakers */ if (machine_is_raumfeld_speaker()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_speaker); + + if (machine_is_raumfeld_connector()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_connector); + + ret = platform_device_add(raumfeld_audio_device); + if (ret < 0) return ret; raumfeld_enable_audio(true); - - return ret; + return 0; } static void __exit raumfeld_audio_exit(void) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d1d4059be04e..d8ce34c83d8b 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,25 +20,15 @@ #include <trace/events/asoc.h> -#if defined(CONFIG_SPI_MASTER) -static int do_spi_write(void *control_data, const void *msg, - int len) +#ifdef CONFIG_SPI_MASTER +static int do_spi_write(void *control, const char *data, int len) { - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; - - if (len <= 0) - return 0; - - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = msg; - t.len = len; + struct spi_device *spi = control; + int ret; - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); + ret = spi_write(spi, data, len); + if (ret < 0) + return ret; return len; } @@ -101,28 +91,12 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; - - data[0] = (reg << 4) | ((value >> 8) & 0x000f); - data[1] = value & 0x00ff; - - return do_hw_write(codec, reg, value, data, 2); -} - -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_4_12_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[2]; + u16 data; - msg[0] = data[1]; - msg[1] = data[0]; + data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - return do_spi_write(control_data, msg, len); + return do_hw_write(codec, reg, value, &data, 2); } -#else -#define snd_soc_4_12_spi_write NULL -#endif static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) @@ -140,21 +114,6 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, &data, 2); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_7_9_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[2]; - - msg[0] = data[0]; - msg[1] = data[1]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_7_9_spi_write NULL -#endif - static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -173,21 +132,6 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, return do_hw_read(codec, reg); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_8_8_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[2]; - - msg[0] = data[0]; - msg[1] = data[1]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_8_8_spi_write NULL -#endif - static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -206,22 +150,6 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, return do_hw_read(codec, reg); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_8_16_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[3]; - - msg[0] = data[0]; - msg[1] = data[1]; - msg[2] = data[2]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_8_16_spi_write NULL -#endif - #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int do_i2c_read(struct snd_soc_codec *codec, void *reg, int reglen, @@ -318,27 +246,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, memcpy(data, &rval, sizeof(rval)); data[2] = value; - reg &= 0xff; return do_hw_write(codec, reg, value, data, 3); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_16_8_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[3]; - - msg[0] = data[0]; - msg[1] = data[1]; - msg[2] = data[2]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_16_8_spi_write NULL -#endif - #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, unsigned int r) @@ -373,23 +284,6 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, data, sizeof(data)); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_16_16_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[4]; - - msg[0] = data[0]; - msg[1] = data[1]; - msg[2] = data[2]; - msg[3] = data[3]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_16_16_spi_write NULL -#endif - /* Primitive bulk write support for soc-cache. The data pointed to by * `data' needs to already be in the form the hardware expects * including any leading register specific data. Any data written @@ -419,7 +313,7 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r #endif #if defined(CONFIG_SPI_MASTER) case SND_SOC_SPI: - ret = do_spi_write(codec->control_data, data, len); + ret = spi_write(codec->control_data, data, len); break; #endif default: @@ -438,43 +332,36 @@ static struct { int addr_bits; int data_bits; int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - int (*spi_write)(void *, const char *, int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, .write = snd_soc_4_12_write, .read = snd_soc_4_12_read, - .spi_write = snd_soc_4_12_spi_write, }, { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, - .spi_write = snd_soc_7_9_spi_write, }, { .addr_bits = 8, .data_bits = 8, .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, .i2c_read = snd_soc_8_8_read_i2c, - .spi_write = snd_soc_8_8_spi_write, }, { .addr_bits = 8, .data_bits = 16, .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, .i2c_read = snd_soc_8_16_read_i2c, - .spi_write = snd_soc_8_16_spi_write, }, { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .read = snd_soc_16_8_read, .i2c_read = snd_soc_16_8_read_i2c, - .spi_write = snd_soc_16_8_spi_write, }, { .addr_bits = 16, .data_bits = 16, .write = snd_soc_16_16_write, .read = snd_soc_16_16_read, .i2c_read = snd_soc_16_16_read_i2c, - .spi_write = snd_soc_16_16_spi_write, }, }; @@ -535,8 +422,9 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, break; case SND_SOC_SPI: - if (io_types[i].spi_write) - codec->hw_write = io_types[i].spi_write; +#ifdef CONFIG_SPI_MASTER + codec->hw_write = do_spi_write; +#endif codec->control_data = container_of(codec->dev, struct spi_device, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c261eeb835b4..13a40fc78d32 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -242,7 +242,7 @@ static ssize_t codec_reg_write_file(struct file *file, const char __user *user_buf, size_t count, loff_t *ppos) { char buf[32]; - int buf_size; + size_t buf_size; char *start = buf; unsigned long reg, value; int step = 1; @@ -1307,10 +1307,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) { - - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - rtd->cpu_dai = cpu_dai; goto find_codec; } @@ -1623,11 +1619,15 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the cpu_dai */ if (!cpu_dai->probed) { + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; + if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n", cpu_dai->name); + module_put(cpu_dai->dev->driver->owner); return ret; } } @@ -1927,9 +1927,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), - "%s", card->name); - snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->name); + snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), + "%s", card->long_name ? card->long_name : card->name); + snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), + "%s", card->driver_name ? card->driver_name : card->name); if (card->late_probe) { ret = card->late_probe(card); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 456617e63789..776e6f418306 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -325,6 +325,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, } static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *kcontrolw, const struct snd_kcontrol_new *kcontrol_new, struct snd_kcontrol **kcontrol) { @@ -334,6 +335,8 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, *kcontrol = NULL; list_for_each_entry(w, &dapm->card->widgets, list) { + if (w == kcontrolw || w->dapm != kcontrolw->dapm) + continue; for (i = 0; i < w->num_kcontrols; i++) { if (&w->kcontrol_news[i] == kcontrol_new) { if (w->kcontrols) @@ -468,7 +471,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0], + shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0], &kcontrol); if (kcontrol) { wlist = kcontrol->private_data; @@ -1110,7 +1113,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) - if (d->n_widgets) + if (d->n_widgets || d->codec == NULL) d->dev_power = 0; /* Check which widgets we need to power and store them in diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 4f5e2c90b020..6b817e20548c 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -114,7 +114,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) { } |